Merge "Change drm/crypto service names to "default"" into oc-dev
diff --git a/camera/ndk/Android.bp b/camera/ndk/Android.bp
index c5fc646..ade0d72 100644
--- a/camera/ndk/Android.bp
+++ b/camera/ndk/Android.bp
@@ -17,7 +17,7 @@
// frameworks/av/include.
ndk_library {
- name: "libcamera2ndk.ndk",
+ name: "libcamera2ndk",
symbol_file: "libcamera2ndk.map.txt",
first_version: "24",
unversioned_until: "current",
diff --git a/include/media/omx/1.0/WOmxNode.h b/include/media/omx/1.0/WOmxNode.h
index 1d575e7..eebc8c6 100644
--- a/include/media/omx/1.0/WOmxNode.h
+++ b/include/media/omx/1.0/WOmxNode.h
@@ -102,9 +102,6 @@
const char *parameter_name,
OMX_INDEXTYPE *index) override;
status_t dispatchMessage(const omx_message &msg) override;
-
- // TODO: this is temporary, will be removed when quirks move to OMX side.
- status_t setQuirks(OMX_U32 quirks) override;
};
struct TWOmxNode : public IOmxNode {
@@ -153,7 +150,6 @@
hidl_string const& parameterName,
getExtensionIndex_cb _hidl_cb) override;
Return<Status> dispatchMessage(Message const& msg) override;
- Return<void> setQuirks(uint32_t quirks) override;
};
} // namespace utils
diff --git a/include/media/vndk/xmlparser/1.0/MediaCodecsXmlParser.h b/include/media/vndk/xmlparser/1.0/MediaCodecsXmlParser.h
new file mode 100644
index 0000000..b324cd8
--- /dev/null
+++ b/include/media/vndk/xmlparser/1.0/MediaCodecsXmlParser.h
@@ -0,0 +1,135 @@
+/*
+ * Copyright 2017, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_CODECS_XML_PARSER_H_
+
+#define MEDIA_CODECS_XML_PARSER_H_
+
+#include <map>
+#include <vector>
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/AString.h>
+
+#include <sys/types.h>
+#include <utils/Errors.h>
+#include <utils/Vector.h>
+#include <utils/StrongPointer.h>
+
+namespace android {
+
+struct AMessage;
+
+// Quirk still supported, even though deprecated
+enum Quirks {
+ kRequiresAllocateBufferOnInputPorts = 1,
+ kRequiresAllocateBufferOnOutputPorts = 2,
+
+ kQuirksMask = kRequiresAllocateBufferOnInputPorts
+ | kRequiresAllocateBufferOnOutputPorts,
+};
+
+// Lightweight struct for querying components.
+struct TypeInfo {
+ AString mName;
+ std::map<AString, AString> mStringFeatures;
+ std::map<AString, bool> mBoolFeatures;
+ std::map<AString, AString> mDetails;
+};
+
+struct ProfileLevel {
+ uint32_t mProfile;
+ uint32_t mLevel;
+};
+
+struct CodecInfo {
+ std::vector<TypeInfo> mTypes;
+ std::vector<ProfileLevel> mProfileLevels;
+ std::vector<uint32_t> mColorFormats;
+ uint32_t mFlags;
+ bool mIsEncoder;
+};
+
+class MediaCodecsXmlParser {
+public:
+ MediaCodecsXmlParser();
+ ~MediaCodecsXmlParser();
+
+ void getGlobalSettings(std::map<AString, AString> *settings) const;
+
+ status_t getCodecInfo(const char *name, CodecInfo *info) const;
+
+ status_t getQuirks(const char *name, std::vector<AString> *quirks) const;
+
+private:
+ enum Section {
+ SECTION_TOPLEVEL,
+ SECTION_SETTINGS,
+ SECTION_DECODERS,
+ SECTION_DECODER,
+ SECTION_DECODER_TYPE,
+ SECTION_ENCODERS,
+ SECTION_ENCODER,
+ SECTION_ENCODER_TYPE,
+ SECTION_INCLUDE,
+ };
+
+ status_t mInitCheck;
+ Section mCurrentSection;
+ bool mUpdate;
+ Vector<Section> mPastSections;
+ int32_t mDepth;
+ AString mHrefBase;
+
+ std::map<AString, AString> mGlobalSettings;
+
+ // name -> CodecInfo
+ std::map<AString, CodecInfo> mCodecInfos;
+ std::map<AString, std::vector<AString>> mQuirks;
+ AString mCurrentName;
+ std::vector<TypeInfo>::iterator mCurrentType;
+
+ status_t initCheck() const;
+ void parseTopLevelXMLFile(const char *path, bool ignore_errors = false);
+
+ void parseXMLFile(const char *path);
+
+ static void StartElementHandlerWrapper(
+ void *me, const char *name, const char **attrs);
+
+ static void EndElementHandlerWrapper(void *me, const char *name);
+
+ void startElementHandler(const char *name, const char **attrs);
+ void endElementHandler(const char *name);
+
+ status_t includeXMLFile(const char **attrs);
+ status_t addSettingFromAttributes(const char **attrs);
+ status_t addMediaCodecFromAttributes(bool encoder, const char **attrs);
+ void addMediaCodec(bool encoder, const char *name, const char *type = NULL);
+
+ status_t addQuirk(const char **attrs);
+ status_t addTypeFromAttributes(const char **attrs, bool encoder);
+ status_t addLimit(const char **attrs);
+ status_t addFeature(const char **attrs);
+ void addType(const char *name);
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaCodecsXmlParser);
+};
+
+} // namespace android
+
+#endif // MEDIA_CODECS_XML_PARSER_H_
+
diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
index 63fa16b..afd1189 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -6,6 +6,7 @@
main_audioserver.cpp
LOCAL_SHARED_LIBRARIES := \
+ libaaudioservice \
libaudioflinger \
libaudiopolicyservice \
libbinder \
@@ -18,6 +19,7 @@
libutils \
libhwbinder
+# TODO oboeservice is the old folder name for aaudioservice. It will be changed.
LOCAL_C_INCLUDES := \
frameworks/av/services/audioflinger \
frameworks/av/services/audiopolicy \
@@ -26,8 +28,12 @@
frameworks/av/services/audiopolicy/engine/interface \
frameworks/av/services/audiopolicy/service \
frameworks/av/services/medialog \
+ frameworks/av/services/oboeservice \
frameworks/av/services/radio \
frameworks/av/services/soundtrigger \
+ frameworks/av/media/libaaudio/include \
+ frameworks/av/media/libaaudio/src \
+ frameworks/av/media/libaaudio/src/binding \
$(call include-path-for, audio-utils) \
external/sonic \
diff --git a/media/audioserver/main_audioserver.cpp b/media/audioserver/main_audioserver.cpp
index bcd0342..ee02d23 100644
--- a/media/audioserver/main_audioserver.cpp
+++ b/media/audioserver/main_audioserver.cpp
@@ -34,6 +34,7 @@
// from LOCAL_C_INCLUDES
#include "AudioFlinger.h"
#include "AudioPolicyService.h"
+#include "AAudioService.h"
#include "MediaLogService.h"
#include "RadioService.h"
#include "SoundTriggerHwService.h"
@@ -131,6 +132,7 @@
ALOGI("ServiceManager: %p", sm.get());
AudioFlinger::instantiate();
AudioPolicyService::instantiate();
+ AAudioService::instantiate();
RadioService::instantiate();
SoundTriggerHwService::instantiate();
ProcessState::self()->startThreadPool();
diff --git a/media/libaaudio/Android.bp b/media/libaaudio/Android.bp
index e41d62b..f539ba9 100644
--- a/media/libaaudio/Android.bp
+++ b/media/libaaudio/Android.bp
@@ -21,7 +21,7 @@
}
ndk_library {
- name: "libaaudio.ndk",
+ name: "libaaudio",
symbol_file: "libaaudio.map.txt",
first_version: "26",
unversioned_until: "current",
diff --git a/media/libaaudio/examples/input_monitor/Android.mk b/media/libaaudio/examples/input_monitor/Android.mk
new file mode 100644
index 0000000..b56328b
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/Android.mk
@@ -0,0 +1,6 @@
+# include $(call all-subdir-makefiles)
+
+# Just include static/ for now.
+LOCAL_PATH := $(call my-dir)
+#include $(LOCAL_PATH)/jni/Android.mk
+include $(LOCAL_PATH)/static/Android.mk
diff --git a/media/libaaudio/examples/input_monitor/README.md b/media/libaaudio/examples/input_monitor/README.md
new file mode 100644
index 0000000..3e54ef0
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/README.md
@@ -0,0 +1 @@
+Monitor input level and print value.
diff --git a/media/libaaudio/examples/input_monitor/jni/Android.mk b/media/libaaudio/examples/input_monitor/jni/Android.mk
new file mode 100644
index 0000000..51a5a85
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/jni/Android.mk
@@ -0,0 +1,35 @@
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+LOCAL_MODULE_TAGS := tests
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/media/liboboe/include
+
+LOCAL_SRC_FILES:= frameworks/av/media/liboboe/src/write_sine.cpp
+LOCAL_SHARED_LIBRARIES := libaudioutils libmedia libtinyalsa \
+ libbinder libcutils libutils
+LOCAL_STATIC_LIBRARIES := libsndfile
+LOCAL_MODULE := write_sine_ndk
+LOCAL_SHARED_LIBRARIES += liboboe_prebuilt
+include $(BUILD_EXECUTABLE)
+
+include $(CLEAR_VARS)
+LOCAL_MODULE_TAGS := tests
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/media/liboboe/include
+
+LOCAL_SRC_FILES:= frameworks/av/media/liboboe/src/write_sine_threaded.cpp
+LOCAL_SHARED_LIBRARIES := libaudioutils libmedia libtinyalsa \
+ libbinder libcutils libutils
+LOCAL_STATIC_LIBRARIES := libsndfile
+LOCAL_MODULE := write_sine_threaded_ndk
+LOCAL_SHARED_LIBRARIES += liboboe_prebuilt
+include $(BUILD_EXECUTABLE)
+
+include $(CLEAR_VARS)
+LOCAL_MODULE := liboboe_prebuilt
+LOCAL_SRC_FILES := liboboe.so
+LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
+include $(PREBUILT_SHARED_LIBRARY)
diff --git a/media/libaaudio/examples/input_monitor/jni/Application.mk b/media/libaaudio/examples/input_monitor/jni/Application.mk
new file mode 100644
index 0000000..e74475c
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/jni/Application.mk
@@ -0,0 +1,3 @@
+# TODO remove then when we support other architectures
+APP_ABI := arm64-v8a
+APP_CPPFLAGS += -std=c++11
diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
new file mode 100644
index 0000000..545496f
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
@@ -0,0 +1,194 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Record input using AAudio and display the peak amplitudes.
+
+#include <new>
+#include <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+#include <aaudio/AAudioDefinitions.h>
+#include <aaudio/AAudio.h>
+
+#define SAMPLE_RATE 48000
+#define NUM_SECONDS 10
+#define NANOS_PER_MICROSECOND ((int64_t)1000)
+#define NANOS_PER_MILLISECOND (NANOS_PER_MICROSECOND * 1000)
+#define NANOS_PER_SECOND (NANOS_PER_MILLISECOND * 1000)
+
+#define DECAY_FACTOR 0.999
+#define MIN_FRAMES_TO_READ 48 /* arbitrary, 1 msec at 48000 Hz */
+
+static const char *getSharingModeText(aaudio_sharing_mode_t mode) {
+ const char *modeText = "unknown";
+ switch (mode) {
+ case AAUDIO_SHARING_MODE_EXCLUSIVE:
+ modeText = "EXCLUSIVE";
+ break;
+ case AAUDIO_SHARING_MODE_SHARED:
+ modeText = "SHARED";
+ break;
+ default:
+ break;
+ }
+ return modeText;
+}
+
+int main(int argc, char **argv)
+{
+ (void)argc; // unused
+
+ aaudio_result_t result;
+
+ int actualSamplesPerFrame;
+ int actualSampleRate;
+ const aaudio_audio_format_t requestedDataFormat = AAUDIO_FORMAT_PCM_I16;
+ aaudio_audio_format_t actualDataFormat;
+
+ const aaudio_sharing_mode_t requestedSharingMode = AAUDIO_SHARING_MODE_SHARED;
+ aaudio_sharing_mode_t actualSharingMode;
+
+ AAudioStreamBuilder *aaudioBuilder = nullptr;
+ AAudioStream *aaudioStream = nullptr;
+ aaudio_stream_state_t state;
+ int32_t framesPerBurst = 0;
+ int32_t framesPerRead = 0;
+ int32_t framesToRecord = 0;
+ int32_t framesLeft = 0;
+ int32_t xRunCount = 0;
+ int16_t *data = nullptr;
+ float peakLevel = 0.0;
+ int loopCounter = 0;
+
+ // Make printf print immediately so that debug info is not stuck
+ // in a buffer if we hang or crash.
+ setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
+
+ printf("%s - Monitor input level using AAudio\n", argv[0]);
+
+ // Use an AAudioStreamBuilder to contain requested parameters.
+ result = AAudio_createStreamBuilder(&aaudioBuilder);
+ if (result != AAUDIO_OK) {
+ goto finish;
+ }
+
+ // Request stream properties.
+ AAudioStreamBuilder_setDirection(aaudioBuilder, AAUDIO_DIRECTION_INPUT);
+ AAudioStreamBuilder_setFormat(aaudioBuilder, requestedDataFormat);
+ AAudioStreamBuilder_setSharingMode(aaudioBuilder, requestedSharingMode);
+
+ // Create an AAudioStream using the Builder.
+ result = AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream);
+ if (result != AAUDIO_OK) {
+ goto finish;
+ }
+
+ actualSamplesPerFrame = AAudioStream_getSamplesPerFrame(aaudioStream);
+ printf("SamplesPerFrame = %d\n", actualSamplesPerFrame);
+ actualSampleRate = AAudioStream_getSampleRate(aaudioStream);
+ printf("SamplesPerFrame = %d\n", actualSampleRate);
+
+ actualSharingMode = AAudioStream_getSharingMode(aaudioStream);
+ printf("SharingMode: requested = %s, actual = %s\n",
+ getSharingModeText(requestedSharingMode),
+ getSharingModeText(actualSharingMode));
+
+ // This is the number of frames that are written in one chunk by a DMA controller
+ // or a DSP.
+ framesPerBurst = AAudioStream_getFramesPerBurst(aaudioStream);
+ printf("DataFormat: framesPerBurst = %d\n",framesPerBurst);
+
+ // Some DMA might use very short bursts of 16 frames. We don't need to read such small
+ // buffers. But it helps to use a multiple of the burst size for predictable scheduling.
+ framesPerRead = framesPerBurst;
+ while (framesPerRead < MIN_FRAMES_TO_READ) {
+ framesPerRead *= 2;
+ }
+ printf("DataFormat: framesPerRead = %d\n",framesPerRead);
+
+ actualDataFormat = AAudioStream_getFormat(aaudioStream);
+ printf("DataFormat: requested = %d, actual = %d\n", requestedDataFormat, actualDataFormat);
+ // TODO handle other data formats
+ assert(actualDataFormat == AAUDIO_FORMAT_PCM_I16);
+
+ // Allocate a buffer for the audio data.
+ data = new(std::nothrow) int16_t[framesPerRead * actualSamplesPerFrame];
+ if (data == nullptr) {
+ fprintf(stderr, "ERROR - could not allocate data buffer\n");
+ result = AAUDIO_ERROR_NO_MEMORY;
+ goto finish;
+ }
+
+ // Start the stream.
+ printf("call AAudioStream_requestStart()\n");
+ result = AAudioStream_requestStart(aaudioStream);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - AAudioStream_requestStart() returned %d\n", result);
+ goto finish;
+ }
+
+ state = AAudioStream_getState(aaudioStream);
+ printf("after start, state = %s\n", AAudio_convertStreamStateToText(state));
+
+ // Play for a while.
+ framesToRecord = actualSampleRate * NUM_SECONDS;
+ framesLeft = framesToRecord;
+ while (framesLeft > 0) {
+ // Read audio data from the stream.
+ int64_t timeoutNanos = 100 * NANOS_PER_MILLISECOND;
+ int minFrames = (framesToRecord < framesPerRead) ? framesToRecord : framesPerRead;
+ int actual = AAudioStream_read(aaudioStream, data, minFrames, timeoutNanos);
+ if (actual < 0) {
+ fprintf(stderr, "ERROR - AAudioStream_read() returned %zd\n", actual);
+ goto finish;
+ } else if (actual == 0) {
+ fprintf(stderr, "WARNING - AAudioStream_read() returned %zd\n", actual);
+ goto finish;
+ }
+ framesLeft -= actual;
+
+ // Peak follower.
+ for (int frameIndex = 0; frameIndex < actual; frameIndex++) {
+ float sample = data[frameIndex * actualSamplesPerFrame] * (1.0/32768);
+ peakLevel *= DECAY_FACTOR;
+ if (sample > peakLevel) {
+ peakLevel = sample;
+ }
+ }
+
+ // Display level as stars, eg. "******".
+ if ((loopCounter++ % 10) == 0) {
+ printf("%5.3f ", peakLevel);
+ int numStars = (int)(peakLevel * 50);
+ for (int i = 0; i < numStars; i++) {
+ printf("*");
+ }
+ printf("\n");
+ }
+ }
+
+ xRunCount = AAudioStream_getXRunCount(aaudioStream);
+ printf("AAudioStream_getXRunCount %d\n", xRunCount);
+
+finish:
+ delete[] data;
+ AAudioStream_close(aaudioStream);
+ AAudioStreamBuilder_delete(aaudioBuilder);
+ printf("exiting - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
+ return (result != AAUDIO_OK) ? EXIT_FAILURE : EXIT_SUCCESS;
+}
+
diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
new file mode 100644
index 0000000..8d40d94
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
@@ -0,0 +1,284 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Record input using AAudio and display the peak amplitudes.
+
+#include <assert.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <math.h>
+#include <time.h>
+#include <aaudio/AAudioDefinitions.h>
+#include <aaudio/AAudio.h>
+
+#define NUM_SECONDS 10
+#define NANOS_PER_MICROSECOND ((int64_t)1000)
+#define NANOS_PER_MILLISECOND (NANOS_PER_MICROSECOND * 1000)
+#define NANOS_PER_SECOND (NANOS_PER_MILLISECOND * 1000)
+
+//#define SHARING_MODE AAUDIO_SHARING_MODE_EXCLUSIVE
+#define SHARING_MODE AAUDIO_SHARING_MODE_SHARED
+
+/**
+ * Simple wrapper for AAudio that opens a default stream and then calls
+ * a callback function to fill the output buffers.
+ */
+class SimpleAAudioPlayer {
+public:
+ SimpleAAudioPlayer() {}
+ ~SimpleAAudioPlayer() {
+ close();
+ };
+
+ /**
+ * Call this before calling open().
+ * @param requestedSharingMode
+ */
+ void setSharingMode(aaudio_sharing_mode_t requestedSharingMode) {
+ mRequestedSharingMode = requestedSharingMode;
+ }
+
+ /**
+ * Also known as "sample rate"
+ * Only call this after open() has been called.
+ */
+ int32_t getFramesPerSecond() {
+ if (mStream == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ return AAudioStream_getSampleRate(mStream);;
+ }
+
+ /**
+ * Only call this after open() has been called.
+ */
+ int32_t getSamplesPerFrame() {
+ if (mStream == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ return AAudioStream_getSamplesPerFrame(mStream);;
+ }
+
+ /**
+ * Open a stream
+ */
+ aaudio_result_t open(AAudioStream_dataCallback proc, void *userContext) {
+ aaudio_result_t result = AAUDIO_OK;
+
+ // Use an AAudioStreamBuilder to contain requested parameters.
+ result = AAudio_createStreamBuilder(&mBuilder);
+ if (result != AAUDIO_OK) return result;
+
+ AAudioStreamBuilder_setDirection(mBuilder, AAUDIO_DIRECTION_INPUT);
+ AAudioStreamBuilder_setSharingMode(mBuilder, mRequestedSharingMode);
+ AAudioStreamBuilder_setDataCallback(mBuilder, proc, userContext);
+ AAudioStreamBuilder_setFormat(mBuilder, AAUDIO_FORMAT_PCM_I16);
+
+ // Open an AAudioStream using the Builder.
+ result = AAudioStreamBuilder_openStream(mBuilder, &mStream);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - AAudioStreamBuilder_openStream() returned %d %s\n",
+ result, AAudio_convertResultToText(result));
+ goto finish1;
+ }
+
+ printf("AAudioStream_getFramesPerBurst() = %d\n",
+ AAudioStream_getFramesPerBurst(mStream));
+ printf("AAudioStream_getBufferSizeInFrames() = %d\n",
+ AAudioStream_getBufferSizeInFrames(mStream));
+ printf("AAudioStream_getBufferCapacityInFrames() = %d\n",
+ AAudioStream_getBufferCapacityInFrames(mStream));
+ return result;
+
+ finish1:
+ AAudioStreamBuilder_delete(mBuilder);
+ mBuilder = nullptr;
+ return result;
+ }
+
+ aaudio_result_t close() {
+ if (mStream != nullptr) {
+ printf("call AAudioStream_close(%p)\n", mStream); fflush(stdout);
+ AAudioStream_close(mStream);
+ mStream = nullptr;
+ AAudioStreamBuilder_delete(mBuilder);
+ mBuilder = nullptr;
+ }
+ return AAUDIO_OK;
+ }
+
+ // Write zero data to fill up the buffer and prevent underruns.
+ // Assume format is PCM_I16. TODO use floats.
+ aaudio_result_t prime() {
+ int32_t samplesPerFrame = AAudioStream_getSamplesPerFrame(mStream);
+ const int numFrames = 32; // arbitrary
+ int16_t zeros[numFrames * samplesPerFrame];
+ memset(zeros, 0, sizeof(zeros));
+ aaudio_result_t result = numFrames;
+ while (result == numFrames) {
+ result = AAudioStream_write(mStream, zeros, numFrames, 0);
+ }
+ return result;
+ }
+
+ // Start the stream. AAudio will start calling your callback function.
+ aaudio_result_t start() {
+ aaudio_result_t result = AAudioStream_requestStart(mStream);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - AAudioStream_requestStart() returned %d %s\n",
+ result, AAudio_convertResultToText(result));
+ }
+ return result;
+ }
+
+ // Stop the stream. AAudio will stop calling your callback function.
+ aaudio_result_t stop() {
+ aaudio_result_t result = AAudioStream_requestStop(mStream);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - AAudioStream_requestStop() returned %d %s\n",
+ result, AAudio_convertResultToText(result));
+ }
+ int32_t xRunCount = AAudioStream_getXRunCount(mStream);
+ printf("AAudioStream_getXRunCount %d\n", xRunCount);
+ return result;
+ }
+
+private:
+ AAudioStreamBuilder *mBuilder = nullptr;
+ AAudioStream *mStream = nullptr;
+ aaudio_sharing_mode_t mRequestedSharingMode = SHARING_MODE;
+};
+
+// Application data that gets passed to the callback.
+typedef struct PeakTrackerData {
+ float peakLevel;
+} PeakTrackerData_t;
+
+#define DECAY_FACTOR 0.999
+
+// Callback function that fills the audio output buffer.
+aaudio_data_callback_result_t MyDataCallbackProc(
+ AAudioStream *stream,
+ void *userData,
+ void *audioData,
+ int32_t numFrames
+ ) {
+
+ PeakTrackerData_t *data = (PeakTrackerData_t *) userData;
+ // printf("MyCallbackProc(): frameCount = %d\n", numFrames);
+ int32_t samplesPerFrame = AAudioStream_getSamplesPerFrame(stream);
+ float sample;
+ // This code assume mono or stereo.
+ switch (AAudioStream_getFormat(stream)) {
+ case AAUDIO_FORMAT_PCM_I16: {
+ int16_t *audioBuffer = (int16_t *) audioData;
+ // Peak follower
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ sample = audioBuffer[frameIndex * samplesPerFrame] * (1.0/32768);
+ data->peakLevel *= DECAY_FACTOR;
+ if (sample > data->peakLevel) {
+ data->peakLevel = sample;
+ }
+ }
+ }
+ break;
+ case AAUDIO_FORMAT_PCM_FLOAT: {
+ float *audioBuffer = (float *) audioData;
+ // Peak follower
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ sample = audioBuffer[frameIndex * samplesPerFrame];
+ data->peakLevel *= DECAY_FACTOR;
+ if (sample > data->peakLevel) {
+ data->peakLevel = sample;
+ }
+ }
+ }
+ break;
+ default:
+ return AAUDIO_CALLBACK_RESULT_STOP;
+ }
+
+ return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+void displayPeakLevel(float peakLevel) {
+ printf("%5.3f ", peakLevel);
+ const int maxStars = 50; // arbitrary, fits on one line
+ int numStars = (int) (peakLevel * maxStars);
+ for (int i = 0; i < numStars; i++) {
+ printf("*");
+ }
+ printf("\n");
+}
+
+int main(int argc, char **argv)
+{
+ (void)argc; // unused
+ SimpleAAudioPlayer player;
+ PeakTrackerData_t myData = {0.0};
+ aaudio_result_t result;
+ const int displayRateHz = 20; // arbitrary
+ const int loopsNeeded = NUM_SECONDS * displayRateHz;
+
+ // Make printf print immediately so that debug info is not stuck
+ // in a buffer if we hang or crash.
+ setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
+ printf("%s - Display audio input using an AAudio callback\n", argv[0]);
+
+ player.setSharingMode(SHARING_MODE);
+
+ result = player.open(MyDataCallbackProc, &myData);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - player.open() returned %d\n", result);
+ goto error;
+ }
+ printf("player.getFramesPerSecond() = %d\n", player.getFramesPerSecond());
+ printf("player.getSamplesPerFrame() = %d\n", player.getSamplesPerFrame());
+
+ result = player.start();
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - player.start() returned %d\n", result);
+ goto error;
+ }
+
+ printf("Sleep for %d seconds while audio plays in a callback thread.\n", NUM_SECONDS);
+ for (int i = 0; i < loopsNeeded; i++)
+ {
+ const struct timespec request = { .tv_sec = 0,
+ .tv_nsec = NANOS_PER_SECOND / displayRateHz };
+ (void) clock_nanosleep(CLOCK_MONOTONIC, 0 /*flags*/, &request, NULL /*remain*/);
+ displayPeakLevel(myData.peakLevel);
+ }
+ printf("Woke up now.\n");
+
+ result = player.stop();
+ if (result != AAUDIO_OK) {
+ goto error;
+ }
+ result = player.close();
+ if (result != AAUDIO_OK) {
+ goto error;
+ }
+
+ printf("SUCCESS\n");
+ return EXIT_SUCCESS;
+error:
+ player.close();
+ printf("exiting - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
+ return EXIT_FAILURE;
+}
+
diff --git a/media/libaaudio/examples/input_monitor/static/Android.mk b/media/libaaudio/examples/input_monitor/static/Android.mk
new file mode 100644
index 0000000..e83f179
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/static/Android.mk
@@ -0,0 +1,35 @@
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+LOCAL_MODULE_TAGS := examples
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/media/libaaudio/include
+
+# TODO reorganize folders to avoid using ../
+LOCAL_SRC_FILES:= ../src/input_monitor.cpp
+
+LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
+ libbinder libcutils libutils \
+ libaudioclient liblog libtinyalsa
+LOCAL_STATIC_LIBRARIES := libaaudio
+
+LOCAL_MODULE := input_monitor
+include $(BUILD_EXECUTABLE)
+
+
+include $(CLEAR_VARS)
+LOCAL_MODULE_TAGS := tests
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/media/libaaudio/include
+
+LOCAL_SRC_FILES:= ../src/input_monitor_callback.cpp
+
+LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
+ libbinder libcutils libutils \
+ libaudioclient liblog
+LOCAL_STATIC_LIBRARIES := libaaudio
+
+LOCAL_MODULE := input_monitor_callback
+include $(BUILD_EXECUTABLE)
diff --git a/media/libaaudio/examples/input_monitor/static/README.md b/media/libaaudio/examples/input_monitor/static/README.md
new file mode 100644
index 0000000..6e26d7b
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/static/README.md
@@ -0,0 +1,2 @@
+Makefile for building simple command line examples.
+They link with AAudio as a static library.
diff --git a/media/libaaudio/examples/write_sine/Android.mk b/media/libaaudio/examples/write_sine/Android.mk
index b56328b..5053e7d 100644
--- a/media/libaaudio/examples/write_sine/Android.mk
+++ b/media/libaaudio/examples/write_sine/Android.mk
@@ -1,6 +1 @@
-# include $(call all-subdir-makefiles)
-
-# Just include static/ for now.
-LOCAL_PATH := $(call my-dir)
-#include $(LOCAL_PATH)/jni/Android.mk
-include $(LOCAL_PATH)/static/Android.mk
+include $(call all-subdir-makefiles)
diff --git a/media/libaaudio/examples/write_sine/jni/Android.mk b/media/libaaudio/examples/write_sine/jni/Android.mk
index 51a5a85..8cd0f03 100644
--- a/media/libaaudio/examples/write_sine/jni/Android.mk
+++ b/media/libaaudio/examples/write_sine/jni/Android.mk
@@ -4,32 +4,27 @@
LOCAL_MODULE_TAGS := tests
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-utils) \
- frameworks/av/media/liboboe/include
+ frameworks/av/media/libaaudio/include
-LOCAL_SRC_FILES:= frameworks/av/media/liboboe/src/write_sine.cpp
-LOCAL_SHARED_LIBRARIES := libaudioutils libmedia libtinyalsa \
- libbinder libcutils libutils
-LOCAL_STATIC_LIBRARIES := libsndfile
+# NDK recommends using this kind of relative path instead of an absolute path.
+LOCAL_SRC_FILES:= ../src/write_sine.cpp
+LOCAL_SHARED_LIBRARIES := libaaudio
LOCAL_MODULE := write_sine_ndk
-LOCAL_SHARED_LIBRARIES += liboboe_prebuilt
include $(BUILD_EXECUTABLE)
include $(CLEAR_VARS)
LOCAL_MODULE_TAGS := tests
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-utils) \
- frameworks/av/media/liboboe/include
+ frameworks/av/media/libaaudio/include
-LOCAL_SRC_FILES:= frameworks/av/media/liboboe/src/write_sine_threaded.cpp
-LOCAL_SHARED_LIBRARIES := libaudioutils libmedia libtinyalsa \
- libbinder libcutils libutils
-LOCAL_STATIC_LIBRARIES := libsndfile
+LOCAL_SRC_FILES:= ../src/write_sine_threaded.cpp
+LOCAL_SHARED_LIBRARIES := libaaudio
LOCAL_MODULE := write_sine_threaded_ndk
-LOCAL_SHARED_LIBRARIES += liboboe_prebuilt
include $(BUILD_EXECUTABLE)
include $(CLEAR_VARS)
-LOCAL_MODULE := liboboe_prebuilt
-LOCAL_SRC_FILES := liboboe.so
+LOCAL_MODULE := libaaudio_prebuilt
+LOCAL_SRC_FILES := libaaudio.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_SHARED_LIBRARY)
diff --git a/media/libaaudio/examples/write_sine/src/SineGenerator.h b/media/libaaudio/examples/write_sine/src/SineGenerator.h
index ade7527..64b772d 100644
--- a/media/libaaudio/examples/write_sine/src/SineGenerator.h
+++ b/media/libaaudio/examples/write_sine/src/SineGenerator.h
@@ -79,7 +79,7 @@
}
}
- double mAmplitude = 0.01;
+ double mAmplitude = 0.05; // unitless scaler
double mPhase = 0.0;
double mPhaseIncrement = 440 * M_PI * 2 / 48000;
double mFrameRate = 48000;
diff --git a/media/libaaudio/examples/write_sine/src/write_sine.cpp b/media/libaaudio/examples/write_sine/src/write_sine.cpp
index 80b6252..d8e5ec1 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine.cpp
@@ -19,7 +19,6 @@
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
-#include <aaudio/AAudioDefinitions.h>
#include <aaudio/AAudio.h>
#include "SineGenerator.h"
@@ -44,6 +43,7 @@
return modeText;
}
+// TODO move to a common utility library
static int64_t getNanoseconds(clockid_t clockId = CLOCK_MONOTONIC) {
struct timespec time;
int result = clock_gettime(clockId, &time);
@@ -74,6 +74,8 @@
AAudioStream *aaudioStream = nullptr;
aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNINITIALIZED;
int32_t framesPerBurst = 0;
+ int32_t framesPerWrite = 0;
+ int32_t bufferCapacity = 0;
int32_t framesToPlay = 0;
int32_t framesLeft = 0;
int32_t xRunCount = 0;
@@ -100,7 +102,6 @@
AAudioStreamBuilder_setFormat(aaudioBuilder, requestedDataFormat);
AAudioStreamBuilder_setSharingMode(aaudioBuilder, requestedSharingMode);
-
// Create an AAudioStream using the Builder.
result = AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream);
if (result != AAUDIO_OK) {
@@ -129,21 +130,25 @@
// This is the number of frames that are read in one chunk by a DMA controller
// or a DSP or a mixer.
framesPerBurst = AAudioStream_getFramesPerBurst(aaudioStream);
- printf("DataFormat: original framesPerBurst = %d\n",framesPerBurst);
+ printf("DataFormat: framesPerBurst = %d\n",framesPerBurst);
+ bufferCapacity = AAudioStream_getBufferCapacityInFrames(aaudioStream);
+ printf("DataFormat: bufferCapacity = %d, remainder = %d\n",
+ bufferCapacity, bufferCapacity % framesPerBurst);
// Some DMA might use very short bursts of 16 frames. We don't need to write such small
// buffers. But it helps to use a multiple of the burst size for predictable scheduling.
- while (framesPerBurst < 48) {
- framesPerBurst *= 2;
+ framesPerWrite = framesPerBurst;
+ while (framesPerWrite < 48) {
+ framesPerWrite *= 2;
}
- printf("DataFormat: final framesPerBurst = %d\n",framesPerBurst);
+ printf("DataFormat: framesPerWrite = %d\n",framesPerWrite);
actualDataFormat = AAudioStream_getFormat(aaudioStream);
printf("DataFormat: requested = %d, actual = %d\n", requestedDataFormat, actualDataFormat);
// TODO handle other data formats
// Allocate a buffer for the audio data.
- data = new int16_t[framesPerBurst * actualSamplesPerFrame];
+ data = new int16_t[framesPerWrite * actualSamplesPerFrame];
if (data == nullptr) {
fprintf(stderr, "ERROR - could not allocate data buffer\n");
result = AAUDIO_ERROR_NO_MEMORY;
@@ -166,14 +171,14 @@
framesLeft = framesToPlay;
while (framesLeft > 0) {
// Render sine waves to left and right channels.
- sineOsc1.render(&data[0], actualSamplesPerFrame, framesPerBurst);
+ sineOsc1.render(&data[0], actualSamplesPerFrame, framesPerWrite);
if (actualSamplesPerFrame > 1) {
- sineOsc2.render(&data[1], actualSamplesPerFrame, framesPerBurst);
+ sineOsc2.render(&data[1], actualSamplesPerFrame, framesPerWrite);
}
// Write audio data to the stream.
int64_t timeoutNanos = 100 * NANOS_PER_MILLISECOND;
- int minFrames = (framesToPlay < framesPerBurst) ? framesToPlay : framesPerBurst;
+ int minFrames = (framesToPlay < framesPerWrite) ? framesToPlay : framesPerWrite;
int actual = AAudioStream_write(aaudioStream, data, minFrames, timeoutNanos);
if (actual < 0) {
fprintf(stderr, "ERROR - AAudioStream_write() returned %zd\n", actual);
diff --git a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
new file mode 100644
index 0000000..9414236
--- /dev/null
+++ b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
@@ -0,0 +1,320 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Play sine waves using an AAudio callback.
+
+#include <assert.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <sched.h>
+#include <stdio.h>
+#include <math.h>
+#include <time.h>
+#include <aaudio/AAudio.h>
+#include "SineGenerator.h"
+
+#define NUM_SECONDS 5
+
+//#define SHARING_MODE AAUDIO_SHARING_MODE_EXCLUSIVE
+#define SHARING_MODE AAUDIO_SHARING_MODE_SHARED
+
+#define CALLBACK_SIZE_FRAMES 128
+
+// TODO refactor common code into a single SimpleAAudio class
+/**
+ * Simple wrapper for AAudio that opens a default stream and then calls
+ * a callback function to fill the output buffers.
+ */
+class SimpleAAudioPlayer {
+public:
+ SimpleAAudioPlayer() {}
+ ~SimpleAAudioPlayer() {
+ close();
+ };
+
+ /**
+ * Call this before calling open().
+ * @param requestedSharingMode
+ */
+ void setSharingMode(aaudio_sharing_mode_t requestedSharingMode) {
+ mRequestedSharingMode = requestedSharingMode;
+ }
+
+ /**
+ * Also known as "sample rate"
+ * Only call this after open() has been called.
+ */
+ int32_t getFramesPerSecond() {
+ if (mStream == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ return AAudioStream_getSampleRate(mStream);;
+ }
+
+ /**
+ * Only call this after open() has been called.
+ */
+ int32_t getSamplesPerFrame() {
+ if (mStream == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ return AAudioStream_getSamplesPerFrame(mStream);;
+ }
+
+ /**
+ * Open a stream
+ */
+ aaudio_result_t open(AAudioStream_dataCallback dataProc, void *userContext) {
+ aaudio_result_t result = AAUDIO_OK;
+
+ // Use an AAudioStreamBuilder to contain requested parameters.
+ result = AAudio_createStreamBuilder(&mBuilder);
+ if (result != AAUDIO_OK) return result;
+
+ AAudioStreamBuilder_setSharingMode(mBuilder, mRequestedSharingMode);
+ AAudioStreamBuilder_setDataCallback(mBuilder, dataProc, userContext);
+ AAudioStreamBuilder_setFormat(mBuilder, AAUDIO_FORMAT_PCM_FLOAT);
+ AAudioStreamBuilder_setFramesPerDataCallback(mBuilder, CALLBACK_SIZE_FRAMES);
+ // AAudioStreamBuilder_setBufferCapacityInFrames(mBuilder, CALLBACK_SIZE_FRAMES * 4);
+
+ // Open an AAudioStream using the Builder.
+ result = AAudioStreamBuilder_openStream(mBuilder, &mStream);
+ if (result != AAUDIO_OK) goto finish1;
+
+ printf("AAudioStream_getFramesPerBurst() = %d\n",
+ AAudioStream_getFramesPerBurst(mStream));
+ printf("AAudioStream_getBufferSizeInFrames() = %d\n",
+ AAudioStream_getBufferSizeInFrames(mStream));
+ printf("AAudioStream_getBufferCapacityInFrames() = %d\n",
+ AAudioStream_getBufferCapacityInFrames(mStream));
+ return result;
+
+ finish1:
+ AAudioStreamBuilder_delete(mBuilder);
+ mBuilder = nullptr;
+ return result;
+ }
+
+ aaudio_result_t close() {
+ if (mStream != nullptr) {
+ printf("call AAudioStream_close(%p)\n", mStream); fflush(stdout);
+ AAudioStream_close(mStream);
+ mStream = nullptr;
+ AAudioStreamBuilder_delete(mBuilder);
+ mBuilder = nullptr;
+ }
+ return AAUDIO_OK;
+ }
+
+ // Write zero data to fill up the buffer and prevent underruns.
+ aaudio_result_t prime() {
+ int32_t samplesPerFrame = AAudioStream_getSamplesPerFrame(mStream);
+ const int numFrames = 32;
+ float zeros[numFrames * samplesPerFrame];
+ memset(zeros, 0, sizeof(zeros));
+ aaudio_result_t result = numFrames;
+ while (result == numFrames) {
+ result = AAudioStream_write(mStream, zeros, numFrames, 0);
+ }
+ return result;
+ }
+
+ // Start the stream. AAudio will start calling your callback function.
+ aaudio_result_t start() {
+ aaudio_result_t result = AAudioStream_requestStart(mStream);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - AAudioStream_requestStart() returned %d %s\n",
+ result, AAudio_convertResultToText(result));
+ }
+ return result;
+ }
+
+ // Stop the stream. AAudio will stop calling your callback function.
+ aaudio_result_t stop() {
+ aaudio_result_t result = AAudioStream_requestStop(mStream);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - AAudioStream_requestStop() returned %d %s\n",
+ result, AAudio_convertResultToText(result));
+ }
+ int32_t xRunCount = AAudioStream_getXRunCount(mStream);
+ printf("AAudioStream_getXRunCount %d\n", xRunCount);
+ return result;
+ }
+
+ AAudioStream *getStream() const {
+ return mStream;
+ }
+
+private:
+ AAudioStreamBuilder *mBuilder = nullptr;
+ AAudioStream *mStream = nullptr;
+ aaudio_sharing_mode_t mRequestedSharingMode = SHARING_MODE;
+};
+
+// Application data that gets passed to the callback.
+#define MAX_FRAME_COUNT_RECORDS 256
+typedef struct SineThreadedData_s {
+ SineGenerator sineOsc1;
+ SineGenerator sineOsc2;
+ // Remove these variables used for testing.
+ int32_t numFrameCounts;
+ int32_t frameCounts[MAX_FRAME_COUNT_RECORDS];
+ int scheduler;
+ bool schedulerChecked;
+} SineThreadedData_t;
+
+// Callback function that fills the audio output buffer.
+aaudio_data_callback_result_t MyDataCallbackProc(
+ AAudioStream *stream,
+ void *userData,
+ void *audioData,
+ int32_t numFrames
+ ) {
+
+ SineThreadedData_t *sineData = (SineThreadedData_t *) userData;
+
+ if (sineData->numFrameCounts < MAX_FRAME_COUNT_RECORDS) {
+ sineData->frameCounts[sineData->numFrameCounts++] = numFrames;
+ }
+
+ if (!sineData->schedulerChecked) {
+ sineData->scheduler = sched_getscheduler(gettid());
+ sineData->schedulerChecked = true;
+ }
+
+ int32_t samplesPerFrame = AAudioStream_getSamplesPerFrame(stream);
+ // This code only plays on the first one or two channels.
+ // TODO Support arbitrary number of channels.
+ switch (AAudioStream_getFormat(stream)) {
+ case AAUDIO_FORMAT_PCM_I16: {
+ int16_t *audioBuffer = (int16_t *) audioData;
+ // Render sine waves as shorts to first channel.
+ sineData->sineOsc1.render(&audioBuffer[0], samplesPerFrame, numFrames);
+ // Render sine waves to second channel if there is one.
+ if (samplesPerFrame > 1) {
+ sineData->sineOsc2.render(&audioBuffer[1], samplesPerFrame, numFrames);
+ }
+ }
+ break;
+ case AAUDIO_FORMAT_PCM_FLOAT: {
+ float *audioBuffer = (float *) audioData;
+ // Render sine waves as floats to first channel.
+ sineData->sineOsc1.render(&audioBuffer[0], samplesPerFrame, numFrames);
+ // Render sine waves to second channel if there is one.
+ if (samplesPerFrame > 1) {
+ sineData->sineOsc2.render(&audioBuffer[1], samplesPerFrame, numFrames);
+ }
+ }
+ break;
+ default:
+ return AAUDIO_CALLBACK_RESULT_STOP;
+ }
+
+ return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+int main(int argc, char **argv)
+{
+ (void)argc; // unused
+ SimpleAAudioPlayer player;
+ SineThreadedData_t myData;
+ aaudio_result_t result;
+
+ // Make printf print immediately so that debug info is not stuck
+ // in a buffer if we hang or crash.
+ setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
+ printf("%s - Play a sine sweep using an AAudio callback\n", argv[0]);
+
+ player.setSharingMode(SHARING_MODE);
+
+ myData.numFrameCounts = 0;
+ myData.schedulerChecked = false;
+
+ result = player.open(MyDataCallbackProc, &myData);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - player.open() returned %d\n", result);
+ goto error;
+ }
+ printf("player.getFramesPerSecond() = %d\n", player.getFramesPerSecond());
+ printf("player.getSamplesPerFrame() = %d\n", player.getSamplesPerFrame());
+ myData.sineOsc1.setup(440.0, 48000);
+ myData.sineOsc1.setSweep(300.0, 600.0, 5.0);
+ myData.sineOsc2.setup(660.0, 48000);
+ myData.sineOsc2.setSweep(350.0, 900.0, 7.0);
+
+#if 0
+ result = player.prime(); // FIXME crashes AudioTrack.cpp
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - player.prime() returned %d\n", result);
+ goto error;
+ }
+#endif
+
+ result = player.start();
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - player.start() returned %d\n", result);
+ goto error;
+ }
+
+ printf("Sleep for %d seconds while audio plays in a callback thread.\n", NUM_SECONDS);
+ for (int second = 0; second < NUM_SECONDS; second++)
+ {
+ const struct timespec request = { .tv_sec = 1, .tv_nsec = 0 };
+ (void) clock_nanosleep(CLOCK_MONOTONIC, 0 /*flags*/, &request, NULL /*remain*/);
+
+ aaudio_stream_state_t state;
+ result = AAudioStream_waitForStateChange(player.getStream(),
+ AAUDIO_STREAM_STATE_CLOSED,
+ &state,
+ 0);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - AAudioStream_waitForStateChange() returned %d\n", result);
+ goto error;
+ }
+ if (state != AAUDIO_STREAM_STATE_STARTING && state != AAUDIO_STREAM_STATE_STARTED) {
+ printf("Stream state is %d %s!\n", state, AAudio_convertStreamStateToText(state));
+ break;
+ }
+ }
+ printf("Woke up now.\n");
+
+ result = player.stop();
+ if (result != AAUDIO_OK) {
+ goto error;
+ }
+ result = player.close();
+ if (result != AAUDIO_OK) {
+ goto error;
+ }
+
+ // Report data gathered in the callback.
+ for (int i = 0; i < myData.numFrameCounts; i++) {
+ printf("numFrames[%4d] = %4d\n", i, myData.frameCounts[i]);
+ }
+ if (myData.schedulerChecked) {
+ printf("scheduler = 0x%08x, SCHED_FIFO = 0x%08X\n",
+ myData.scheduler,
+ SCHED_FIFO);
+ }
+
+ printf("SUCCESS\n");
+ return EXIT_SUCCESS;
+error:
+ player.close();
+ printf("exiting - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
+ return EXIT_FAILURE;
+}
+
diff --git a/media/libaaudio/examples/write_sine/src/write_sine_threaded.cpp b/media/libaaudio/examples/write_sine/src/write_sine_threaded.cpp
index 40e5016..9bc5886 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine_threaded.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine_threaded.cpp
@@ -16,24 +16,24 @@
// Play sine waves using an AAudio background thread.
-#include <assert.h>
+//#include <assert.h>
+#include <atomic>
#include <unistd.h>
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include <time.h>
-#include <aaudio/AAudioDefinitions.h>
#include <aaudio/AAudio.h>
#include "SineGenerator.h"
-#define NUM_SECONDS 10
+#define NUM_SECONDS 5
#define NANOS_PER_MICROSECOND ((int64_t)1000)
#define NANOS_PER_MILLISECOND (NANOS_PER_MICROSECOND * 1000)
#define MILLIS_PER_SECOND 1000
#define NANOS_PER_SECOND (NANOS_PER_MILLISECOND * MILLIS_PER_SECOND)
-//#define SHARING_MODE AAUDIO_SHARING_MODE_EXCLUSIVE
-#define SHARING_MODE AAUDIO_SHARING_MODE_SHARED
+#define SHARING_MODE AAUDIO_SHARING_MODE_EXCLUSIVE
+//#define SHARING_MODE AAUDIO_SHARING_MODE_SHARED
// Prototype for a callback.
typedef int audio_callback_proc_t(float *outputBuffer,
@@ -42,6 +42,16 @@
static void *SimpleAAudioPlayerThreadProc(void *arg);
+// TODO merge into common code
+static int64_t getNanoseconds(clockid_t clockId = CLOCK_MONOTONIC) {
+ struct timespec time;
+ int result = clock_gettime(clockId, &time);
+ if (result < 0) {
+ return -errno; // TODO standardize return value
+ }
+ return (time.tv_sec * NANOS_PER_SECOND) + time.tv_nsec;
+}
+
/**
* Simple wrapper for AAudio that opens a default stream and then calls
* a callback function to fill the output buffers.
@@ -49,7 +59,7 @@
class SimpleAAudioPlayer {
public:
SimpleAAudioPlayer() {}
- virtual ~SimpleAAudioPlayer() {
+ ~SimpleAAudioPlayer() {
close();
};
@@ -80,21 +90,25 @@
if (result != AAUDIO_OK) return result;
AAudioStreamBuilder_setSharingMode(mBuilder, mRequestedSharingMode);
+ AAudioStreamBuilder_setSampleRate(mBuilder, 48000);
// Open an AAudioStream using the Builder.
result = AAudioStreamBuilder_openStream(mBuilder, &mStream);
- if (result != AAUDIO_OK) goto finish1;
+ if (result != AAUDIO_OK) goto error;
+
+ printf("Requested sharing mode = %d\n", mRequestedSharingMode);
+ printf("Actual sharing mode = %d\n", AAudioStream_getSharingMode(mStream));
// Check to see what kind of stream we actually got.
mFramesPerSecond = AAudioStream_getSampleRate(mStream);
- printf("open() mFramesPerSecond = %d\n", mFramesPerSecond);
+ printf("Actual framesPerSecond = %d\n", mFramesPerSecond);
mSamplesPerFrame = AAudioStream_getSamplesPerFrame(mStream);
- printf("open() mSamplesPerFrame = %d\n", mSamplesPerFrame);
+ printf("Actual samplesPerFrame = %d\n", mSamplesPerFrame);
{
int32_t bufferCapacity = AAudioStream_getBufferCapacityInFrames(mStream);
- printf("open() got bufferCapacity = %d\n", bufferCapacity);
+ printf("Actual bufferCapacity = %d\n", bufferCapacity);
}
// This is the number of frames that are read in one chunk by a DMA controller
@@ -105,9 +119,10 @@
while (mFramesPerBurst < 48) {
mFramesPerBurst *= 2;
}
- printf("DataFormat: final framesPerBurst = %d\n",mFramesPerBurst);
+ printf("Actual framesPerBurst = %d\n",mFramesPerBurst);
mDataFormat = AAudioStream_getFormat(mStream);
+ printf("Actual dataFormat = %d\n", mDataFormat);
// Allocate a buffer for the audio data.
mOutputBuffer = new float[mFramesPerBurst * mSamplesPerFrame];
@@ -118,6 +133,7 @@
// If needed allocate a buffer for converting float to int16_t.
if (mDataFormat == AAUDIO_FORMAT_PCM_I16) {
+ printf("Allocate data conversion buffer for float=>pcm16\n");
mConversionBuffer = new int16_t[mFramesPerBurst * mSamplesPerFrame];
if (mConversionBuffer == nullptr) {
fprintf(stderr, "ERROR - could not allocate conversion buffer\n");
@@ -126,7 +142,7 @@
}
return result;
- finish1:
+ error:
AAudioStreamBuilder_delete(mBuilder);
mBuilder = nullptr;
return result;
@@ -150,7 +166,7 @@
// Start a thread that will call the callback proc.
aaudio_result_t start() {
- mEnabled = true;
+ mEnabled.store(true);
int64_t nanosPerBurst = mFramesPerBurst * NANOS_PER_SECOND
/ mFramesPerSecond;
return AAudioStream_createThread(mStream, nanosPerBurst,
@@ -160,56 +176,106 @@
// Tell the thread to stop.
aaudio_result_t stop() {
- mEnabled = false;
+ mEnabled.store(false);
return AAudioStream_joinThread(mStream, nullptr, 2 * NANOS_PER_SECOND);
}
- aaudio_result_t callbackLoop() {
- int32_t framesWritten = 0;
- int32_t xRunCount = 0;
- aaudio_result_t result = AAUDIO_OK;
+ bool isEnabled() const {
+ return mEnabled.load();
+ }
- result = AAudioStream_requestStart(mStream);
- if (result != AAUDIO_OK) {
- fprintf(stderr, "ERROR - AAudioStream_requestStart() returned %d\n", result);
- return result;
- }
+ aaudio_result_t callbackLoop() {
+ aaudio_result_t result = 0;
+ int64_t framesWritten = 0;
+ int32_t xRunCount = 0;
+ bool started = false;
+ int64_t framesInBuffer =
+ AAudioStream_getFramesWritten(mStream) -
+ AAudioStream_getFramesRead(mStream);
+ int64_t framesAvailable =
+ AAudioStream_getBufferSizeInFrames(mStream) - framesInBuffer;
+
+ int64_t startTime = 0;
+ int64_t startPosition = 0;
+ int32_t loopCount = 0;
// Give up after several burst periods have passed.
const int burstsPerTimeout = 8;
- int64_t nanosPerTimeout =
- burstsPerTimeout * mFramesPerBurst * NANOS_PER_SECOND
- / mFramesPerSecond;
+ int64_t nanosPerTimeout = 0;
+ int64_t runningNanosPerTimeout = 500 * NANOS_PER_MILLISECOND;
- while (mEnabled && result >= 0) {
+ while (isEnabled() && result >= 0) {
// Call application's callback function to fill the buffer.
if (mCallbackProc(mOutputBuffer, mFramesPerBurst, mUserContext)) {
- mEnabled = false;
+ mEnabled.store(false);
}
+
// if needed, convert from float to int16_t PCM
+ //printf("app callbackLoop writing %d frames, state = %s\n", mFramesPerBurst,
+ // AAudio_convertStreamStateToText(AAudioStream_getState(mStream)));
if (mConversionBuffer != nullptr) {
int32_t numSamples = mFramesPerBurst * mSamplesPerFrame;
for (int i = 0; i < numSamples; i++) {
mConversionBuffer[i] = (int16_t)(32767.0 * mOutputBuffer[i]);
}
// Write the application data to stream.
- result = AAudioStream_write(mStream, mConversionBuffer, mFramesPerBurst, nanosPerTimeout);
+ result = AAudioStream_write(mStream, mConversionBuffer,
+ mFramesPerBurst, nanosPerTimeout);
} else {
// Write the application data to stream.
- result = AAudioStream_write(mStream, mOutputBuffer, mFramesPerBurst, nanosPerTimeout);
+ result = AAudioStream_write(mStream, mOutputBuffer,
+ mFramesPerBurst, nanosPerTimeout);
}
- framesWritten += result;
+
if (result < 0) {
- fprintf(stderr, "ERROR - AAudioStream_write() returned %zd\n", result);
+ fprintf(stderr, "ERROR - AAudioStream_write() returned %d %s\n", result,
+ AAudio_convertResultToText(result));
+ break;
+ } else if (started && result != mFramesPerBurst) {
+ fprintf(stderr, "ERROR - AAudioStream_write() timed out! %d\n", result);
+ break;
+ } else {
+ framesWritten += result;
+ }
+
+ if (startTime > 0 && ((loopCount & 0x01FF) == 0)) {
+ double elapsedFrames = (double)(framesWritten - startPosition);
+ int64_t elapsedTime = getNanoseconds() - startTime;
+ double measuredRate = elapsedFrames * NANOS_PER_SECOND / elapsedTime;
+ printf("app callbackLoop write() measured rate %f\n", measuredRate);
+ }
+ loopCount++;
+
+ if (!started && framesWritten >= framesAvailable) {
+ // Start buffer if fully primed.{
+ result = AAudioStream_requestStart(mStream);
+ printf("app callbackLoop requestStart returned %d\n", result);
+ if (result != AAUDIO_OK) {
+ fprintf(stderr, "ERROR - AAudioStream_requestStart() returned %d %s\n", result,
+ AAudio_convertResultToText(result));
+ mEnabled.store(false);
+ return result;
+ }
+ started = true;
+ nanosPerTimeout = runningNanosPerTimeout;
+ startPosition = framesWritten;
+ startTime = getNanoseconds();
+ }
+
+ {
+ int32_t tempXRunCount = AAudioStream_getXRunCount(mStream);
+ if (tempXRunCount != xRunCount) {
+ xRunCount = tempXRunCount;
+ printf("AAudioStream_getXRunCount returns %d at frame %d\n",
+ xRunCount, (int) framesWritten);
+ }
}
}
- xRunCount = AAudioStream_getXRunCount(mStream);
- printf("AAudioStream_getXRunCount %d\n", xRunCount);
-
result = AAudioStream_requestStop(mStream);
if (result != AAUDIO_OK) {
- fprintf(stderr, "ERROR - AAudioStream_requestStart() returned %d\n", result);
+ fprintf(stderr, "ERROR - AAudioStream_requestStop() returned %d %s\n", result,
+ AAudio_convertResultToText(result));
return result;
}
@@ -230,7 +296,7 @@
int32_t mFramesPerBurst = 0;
aaudio_audio_format_t mDataFormat = AAUDIO_FORMAT_PCM_I16;
- volatile bool mEnabled = false; // used to request that callback exit its loop
+ std::atomic<bool> mEnabled; // used to request that callback exit its loop
};
static void *SimpleAAudioPlayerThreadProc(void *arg) {
@@ -289,19 +355,21 @@
}
printf("Sleep for %d seconds while audio plays in a background thread.\n", NUM_SECONDS);
- {
+ for (int i = 0; i < NUM_SECONDS && player.isEnabled(); i++) {
// FIXME sleep is not an NDK API
// sleep(NUM_SECONDS);
- const struct timespec request = { .tv_sec = NUM_SECONDS, .tv_nsec = 0 };
+ const struct timespec request = { .tv_sec = 1, .tv_nsec = 0 };
(void) clock_nanosleep(CLOCK_MONOTONIC, 0 /*flags*/, &request, NULL /*remain*/);
}
- printf("Woke up now.\n");
+ printf("Woke up now!\n");
result = player.stop();
if (result != AAUDIO_OK) {
fprintf(stderr, "ERROR - player.stop() returned %d\n", result);
goto error;
}
+
+ printf("Player stopped.\n");
result = player.close();
if (result != AAUDIO_OK) {
fprintf(stderr, "ERROR - player.close() returned %d\n", result);
diff --git a/media/libaaudio/examples/write_sine/static/Android.mk b/media/libaaudio/examples/write_sine/static/Android.mk
index 139b70a..c02b91c 100644
--- a/media/libaaudio/examples/write_sine/static/Android.mk
+++ b/media/libaaudio/examples/write_sine/static/Android.mk
@@ -6,7 +6,7 @@
$(call include-path-for, audio-utils) \
frameworks/av/media/libaaudio/include
-# TODO reorganize folders to avoid using ../
+# NDK recommends using this kind of relative path instead of an absolute path.
LOCAL_SRC_FILES:= ../src/write_sine.cpp
LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
@@ -17,6 +17,8 @@
LOCAL_MODULE := write_sine
include $(BUILD_EXECUTABLE)
+
+
include $(CLEAR_VARS)
LOCAL_MODULE_TAGS := tests
LOCAL_C_INCLUDES := \
@@ -27,8 +29,26 @@
LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
libbinder libcutils libutils \
- libaudioclient liblog libtinyalsa
+ libaudioclient liblog
LOCAL_STATIC_LIBRARIES := libaaudio
LOCAL_MODULE := write_sine_threaded
include $(BUILD_EXECUTABLE)
+
+
+
+include $(CLEAR_VARS)
+LOCAL_MODULE_TAGS := tests
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/media/libaaudio/include
+
+LOCAL_SRC_FILES:= ../src/write_sine_callback.cpp
+
+LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
+ libbinder libcutils libutils \
+ libaudioclient liblog
+LOCAL_STATIC_LIBRARIES := libaaudio
+
+LOCAL_MODULE := write_sine_callback
+include $(BUILD_EXECUTABLE)
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index 551dcc9..25ad5f8 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -89,7 +89,8 @@
* Request an audio device identified device using an ID.
* On Android, for example, the ID could be obtained from the Java AudioManager.
*
- * By default, the primary device will be used.
+ * The default, if you do not call this function, is AAUDIO_DEVICE_UNSPECIFIED,
+ * in which case the primary device will be used.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param deviceId device identifier or AAUDIO_DEVICE_UNSPECIFIED
@@ -98,52 +99,71 @@
int32_t deviceId);
/**
- * Request a sample rate in Hz.
+ * Request a sample rate in Hertz.
+ *
* The stream may be opened with a different sample rate.
* So the application should query for the actual rate after the stream is opened.
*
* Technically, this should be called the "frame rate" or "frames per second",
* because it refers to the number of complete frames transferred per second.
- * But it is traditionally called "sample rate". Se we use that term.
+ * But it is traditionally called "sample rate". So we use that term.
*
- * Default is AAUDIO_UNSPECIFIED.
-
+ * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param sampleRate frames per second. Common rates include 44100 and 48000 Hz.
*/
AAUDIO_API void AAudioStreamBuilder_setSampleRate(AAudioStreamBuilder* builder,
int32_t sampleRate);
/**
* Request a number of samples per frame.
+ *
* The stream may be opened with a different value.
* So the application should query for the actual value after the stream is opened.
*
- * Default is AAUDIO_UNSPECIFIED.
+ * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
*
* Note, this quantity is sometimes referred to as "channel count".
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param samplesPerFrame Number of samples in one frame, ie. numChannels.
*/
AAUDIO_API void AAudioStreamBuilder_setSamplesPerFrame(AAudioStreamBuilder* builder,
int32_t samplesPerFrame);
/**
* Request a sample data format, for example AAUDIO_FORMAT_PCM_I16.
- * The application should query for the actual format after the stream is opened.
+ *
+ * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
+ *
+ * The stream may be opened with a different value.
+ * So the application should query for the actual value after the stream is opened.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param format Most common formats are AAUDIO_FORMAT_PCM_FLOAT and AAUDIO_FORMAT_PCM_I16.
*/
AAUDIO_API void AAudioStreamBuilder_setFormat(AAudioStreamBuilder* builder,
aaudio_audio_format_t format);
/**
* Request a mode for sharing the device.
+ *
+ * The default, if you do not call this function, is AAUDIO_SHARING_MODE_SHARED.
+ *
* The requested sharing mode may not be available.
- * So the application should query for the actual mode after the stream is opened.
+ * The application can query for the actual mode after the stream is opened.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param sharingMode AAUDIO_SHARING_MODE_LEGACY or AAUDIO_SHARING_MODE_EXCLUSIVE
+ * @param sharingMode AAUDIO_SHARING_MODE_SHARED or AAUDIO_SHARING_MODE_EXCLUSIVE
*/
AAUDIO_API void AAudioStreamBuilder_setSharingMode(AAudioStreamBuilder* builder,
aaudio_sharing_mode_t sharingMode);
/**
- * Request the direction for a stream. The default is AAUDIO_DIRECTION_OUTPUT.
+ * Request the direction for a stream.
+ *
+ * The default, if you do not call this function, is AAUDIO_DIRECTION_OUTPUT.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param direction AAUDIO_DIRECTION_OUTPUT or AAUDIO_DIRECTION_INPUT
@@ -152,16 +172,162 @@
aaudio_direction_t direction);
/**
- * Set the requested maximum buffer capacity in frames.
+ * Set the requested buffer capacity in frames.
* The final AAudioStream capacity may differ, but will probably be at least this big.
*
- * Default is AAUDIO_UNSPECIFIED.
+ * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param frames the desired buffer capacity in frames or AAUDIO_UNSPECIFIED
+ * @param numFrames the desired buffer capacity in frames or AAUDIO_UNSPECIFIED
*/
AAUDIO_API void AAudioStreamBuilder_setBufferCapacityInFrames(AAudioStreamBuilder* builder,
- int32_t frames);
+ int32_t numFrames);
+/**
+ * Return one of these values from the data callback function.
+ */
+enum {
+
+ /**
+ * Continue calling the callback.
+ */
+ AAUDIO_CALLBACK_RESULT_CONTINUE = 0,
+
+ /**
+ * Stop calling the callback.
+ *
+ * The application will still need to call AAudioStream_requestPause()
+ * or AAudioStream_requestStop().
+ */
+ AAUDIO_CALLBACK_RESULT_STOP,
+
+};
+typedef int32_t aaudio_data_callback_result_t;
+
+/**
+ * Prototype for the data function that is passed to AAudioStreamBuilder_setDataCallback().
+ *
+ * For an output stream, this function should render and write numFrames of data
+ * in the streams current data format to the audioData buffer.
+ *
+ * For an input stream, this function should read and process numFrames of data
+ * from the audioData buffer.
+ *
+ * Note that this callback function should be considered a "real-time" function.
+ * It must not do anything that could cause an unbounded delay because that can cause the
+ * audio to glitch or pop.
+ *
+ * These are things the function should NOT do:
+ * <ul>
+ * <li>allocate memory using, for example, malloc() or new</li>
+ * <li>any file operations such as opening, closing, reading or writing</li>
+ * <li>any network operations such as streaming</li>
+ * <li>use any mutexes or other synchronization primitives</li>
+ * <li>sleep</li>
+ * </ul>
+ *
+ * If you need to move data, eg. MIDI commands, in or out of the callback function then
+ * we recommend the use of non-blocking techniques such as an atomic FIFO.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @param userData the same address that was passed to AAudioStreamBuilder_setCallback()
+ * @param audioData a pointer to the audio data
+ * @param numFrames the number of frames to be processed
+ * @return AAUDIO_CALLBACK_RESULT_*
+ */
+typedef aaudio_data_callback_result_t (*AAudioStream_dataCallback)(
+ AAudioStream *stream,
+ void *userData,
+ void *audioData,
+ int32_t numFrames);
+
+/**
+ * Request that AAudio call this functions when the stream is running.
+ *
+ * Note that when using this callback, the audio data will be passed in or out
+ * of the function as an argument.
+ * So you cannot call AAudioStream_write() or AAudioStream_read() on the same stream
+ * that has an active data callback.
+ *
+ * The callback function will start being called after AAudioStream_requestStart() is called.
+ * It will stop being called after AAudioStream_requestPause() or
+ * AAudioStream_requestStop() is called.
+ *
+ * This callback function will be called on a real-time thread owned by AAudio. See
+ * {@link aaudio_data_callback_proc_t} for more information.
+ *
+ * Note that the AAudio callbacks will never be called simultaneously from multiple threads.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param callback pointer to a function that will process audio data.
+ * @param userData pointer to an application data structure that will be passed
+ * to the callback functions.
+ */
+AAUDIO_API void AAudioStreamBuilder_setDataCallback(AAudioStreamBuilder* builder,
+ AAudioStream_dataCallback callback,
+ void *userData);
+
+/**
+ * Set the requested data callback buffer size in frames.
+ * See {@link AAudioStream_dataCallback}.
+ *
+ * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
+ *
+ * For the lowest possible latency, do not call this function. AAudio will then
+ * call the dataProc callback function with whatever size is optimal.
+ * That size may vary from one callback to another.
+ *
+ * Only use this function if the application requires a specific number of frames for processing.
+ * The application might, for example, be using an FFT that requires
+ * a specific power-of-two sized buffer.
+ *
+ * AAudio may need to add additional buffering in order to adapt between the internal
+ * buffer size and the requested buffer size.
+ *
+ * If you do call this function then the requested size should be less than
+ * half the buffer capacity, to allow double buffering.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param numFrames the desired buffer size in frames or AAUDIO_UNSPECIFIED
+ */
+AAUDIO_API void AAudioStreamBuilder_setFramesPerDataCallback(AAudioStreamBuilder* builder,
+ int32_t numFrames);
+
+/**
+ * Prototype for the callback function that is passed to
+ * AAudioStreamBuilder_setErrorCallback().
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @param userData the same address that was passed to AAudioStreamBuilder_setErrorCallback()
+ * @param error an AAUDIO_ERROR_* value.
+ */
+typedef void (*AAudioStream_errorCallback)(
+ AAudioStream *stream,
+ void *userData,
+ aaudio_result_t error);
+
+/**
+ * Request that AAudio call this functions if any error occurs on a callback thread.
+ *
+ * It will be called, for example, if a headset or a USB device is unplugged causing the stream's
+ * device to be unavailable.
+ * In response, this function could signal or launch another thread to reopen a
+ * stream on another device. Do not reopen the stream in this callback.
+ *
+ * This will not be called because of actions by the application, such as stopping
+ * or closing a stream.
+ *
+ * Another possible cause of error would be a timeout or an unanticipated internal error.
+ *
+ * Note that the AAudio callbacks will never be called simultaneously from multiple threads.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param callback pointer to a function that will be called if an error occurs.
+ * @param userData pointer to an application data structure that will be passed
+ * to the callback functions.
+ */
+AAUDIO_API void AAudioStreamBuilder_setErrorCallback(AAudioStreamBuilder* builder,
+ AAudioStream_errorCallback callback,
+ void *userData);
/**
* Open a stream based on the options in the StreamBuilder.
@@ -333,9 +499,14 @@
// High priority audio threads
// ============================================================
+/**
+ * @deprecated Use AudioStreamBuilder_setCallback()
+ */
typedef void *(*aaudio_audio_thread_proc_t)(void *);
/**
+ * @deprecated Use AudioStreamBuilder_setCallback()
+ *
* Create a thread associated with a stream. The thread has special properties for
* low latency audio performance. This thread can be used to implement a callback API.
*
@@ -360,6 +531,8 @@
void *arg);
/**
+ * @deprecated Use AudioStreamBuilder_setCallback()
+ *
* Wait until the thread exits or an error occurs.
*
* @param stream A stream created using AAudioStreamBuilder_openStream().
@@ -388,11 +561,11 @@
* Call AAudioStream_getBufferSizeInFrames() to see what the actual final size is.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @param requestedFrames requested number of frames that can be filled without blocking
+ * @param numFrames requested number of frames that can be filled without blocking
* @return actual buffer size in frames or a negative error
*/
AAUDIO_API aaudio_result_t AAudioStream_setBufferSizeInFrames(AAudioStream* stream,
- int32_t requestedFrames);
+ int32_t numFrames);
/**
* Query the maximum number of frames that can be filled without blocking.
@@ -421,11 +594,32 @@
* Query maximum buffer capacity in frames.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return the buffer capacity in frames
+ * @return buffer capacity in frames
*/
AAUDIO_API int32_t AAudioStream_getBufferCapacityInFrames(AAudioStream* stream);
/**
+ * Query the size of the buffer that will be passed to the dataProc callback
+ * in the numFrames parameter.
+ *
+ * This call can be used if the application needs to know the value of numFrames before
+ * the stream is started. This is not normally necessary.
+ *
+ * If a specific size was requested by calling AAudioStreamBuilder_setCallbackSizeInFrames()
+ * then this will be the same size.
+ *
+ * If AAudioStreamBuilder_setCallbackSizeInFrames() was not called then this will
+ * return the size chosen by AAudio, or AAUDIO_UNSPECIFIED.
+ *
+ * AAUDIO_UNSPECIFIED indicates that the callback buffer size for this stream
+ * may vary from one dataProc callback to the next.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return callback buffer size in frames or AAUDIO_UNSPECIFIED
+ */
+AAUDIO_API int32_t AAudioStream_getFramesPerDataCallback(AAudioStream* stream);
+
+/**
* An XRun is an Underrun or an Overrun.
* During playing, an underrun will occur if the stream is not written in time
* and the system runs out of valid data.
diff --git a/media/libaaudio/include/aaudio/AAudioDefinitions.h b/media/libaaudio/include/aaudio/AAudioDefinitions.h
index fbd284c..57e3dbd 100644
--- a/media/libaaudio/include/aaudio/AAudioDefinitions.h
+++ b/media/libaaudio/include/aaudio/AAudioDefinitions.h
@@ -39,7 +39,7 @@
* and would accept whatever it was given.
*/
#define AAUDIO_UNSPECIFIED 0
-#define AAUDIO_DEVICE_UNSPECIFIED ((int32_t) -1)
+#define AAUDIO_DEVICE_UNSPECIFIED 0
enum {
AAUDIO_DIRECTION_OUTPUT,
@@ -82,9 +82,10 @@
AAUDIO_ERROR_NULL,
AAUDIO_ERROR_TIMEOUT,
AAUDIO_ERROR_WOULD_BLOCK,
- AAUDIO_ERROR_INVALID_ORDER,
+ AAUDIO_ERROR_INVALID_FORMAT,
AAUDIO_ERROR_OUT_OF_RANGE,
- AAUDIO_ERROR_NO_SERVICE
+ AAUDIO_ERROR_NO_SERVICE,
+ AAUDIO_ERROR_INVALID_RATE
};
typedef int32_t aaudio_result_t;
@@ -103,9 +104,11 @@
AAUDIO_STREAM_STATE_STOPPED,
AAUDIO_STREAM_STATE_CLOSING,
AAUDIO_STREAM_STATE_CLOSED,
+ AAUDIO_STREAM_STATE_DISCONNECTED
};
typedef int32_t aaudio_stream_state_t;
+
enum {
/**
* This will be the only stream using a particular source or sink.
diff --git a/media/libaaudio/libaaudio.map.txt b/media/libaaudio/libaaudio.map.txt
index a9e9109..f22fdfe 100644
--- a/media/libaaudio/libaaudio.map.txt
+++ b/media/libaaudio/libaaudio.map.txt
@@ -4,6 +4,9 @@
AAudio_convertStreamStateToText;
AAudio_createStreamBuilder;
AAudioStreamBuilder_setDeviceId;
+ AAudioStreamBuilder_setDataCallback;
+ AAudioStreamBuilder_setErrorCallback;
+ AAudioStreamBuilder_setFramesPerDataCallback;
AAudioStreamBuilder_setSampleRate;
AAudioStreamBuilder_setSamplesPerFrame;
AAudioStreamBuilder_setFormat;
@@ -25,6 +28,7 @@
AAudioStream_joinThread;
AAudioStream_setBufferSizeInFrames;
AAudioStream_getBufferSizeInFrames;
+ AAudioStream_getFramesPerDataCallback;
AAudioStream_getFramesPerBurst;
AAudioStream_getBufferCapacityInFrames;
AAudioStream_getXRunCount;
diff --git a/media/libaaudio/src/Android.mk b/media/libaaudio/src/Android.mk
index a016b49..b5bb75f 100644
--- a/media/libaaudio/src/Android.mk
+++ b/media/libaaudio/src/Android.mk
@@ -26,26 +26,32 @@
$(LOCAL_PATH)/legacy \
$(LOCAL_PATH)/utility
+# If you add a file here then also add it below in the SHARED target
LOCAL_SRC_FILES = \
core/AudioStream.cpp \
core/AudioStreamBuilder.cpp \
core/AAudioAudio.cpp \
+ legacy/AudioStreamLegacy.cpp \
legacy/AudioStreamRecord.cpp \
legacy/AudioStreamTrack.cpp \
utility/HandleTracker.cpp \
utility/AAudioUtilities.cpp \
+ utility/FixedBlockAdapter.cpp \
+ utility/FixedBlockReader.cpp \
+ utility/FixedBlockWriter.cpp \
fifo/FifoBuffer.cpp \
fifo/FifoControllerBase.cpp \
client/AudioEndpoint.cpp \
client/AudioStreamInternal.cpp \
client/IsochronousClockModel.cpp \
- binding/SharedMemoryParcelable.cpp \
- binding/SharedRegionParcelable.cpp \
- binding/RingBufferParcelable.cpp \
binding/AudioEndpointParcelable.cpp \
+ binding/AAudioBinderClient.cpp \
binding/AAudioStreamRequest.cpp \
binding/AAudioStreamConfiguration.cpp \
- binding/IAAudioService.cpp
+ binding/IAAudioService.cpp \
+ binding/RingBufferParcelable.cpp \
+ binding/SharedMemoryParcelable.cpp \
+ binding/SharedRegionParcelable.cpp
LOCAL_CFLAGS += -Wno-unused-parameter -Wall -Werror
@@ -79,22 +85,27 @@
LOCAL_SRC_FILES = core/AudioStream.cpp \
core/AudioStreamBuilder.cpp \
core/AAudioAudio.cpp \
+ legacy/AudioStreamLegacy.cpp \
legacy/AudioStreamRecord.cpp \
legacy/AudioStreamTrack.cpp \
utility/HandleTracker.cpp \
utility/AAudioUtilities.cpp \
+ utility/FixedBlockAdapter.cpp \
+ utility/FixedBlockReader.cpp \
+ utility/FixedBlockWriter.cpp \
fifo/FifoBuffer.cpp \
fifo/FifoControllerBase.cpp \
client/AudioEndpoint.cpp \
client/AudioStreamInternal.cpp \
client/IsochronousClockModel.cpp \
- binding/SharedMemoryParcelable.cpp \
- binding/SharedRegionParcelable.cpp \
- binding/RingBufferParcelable.cpp \
binding/AudioEndpointParcelable.cpp \
+ binding/AAudioBinderClient.cpp \
binding/AAudioStreamRequest.cpp \
binding/AAudioStreamConfiguration.cpp \
- binding/IAAudioService.cpp
+ binding/IAAudioService.cpp \
+ binding/RingBufferParcelable.cpp \
+ binding/SharedMemoryParcelable.cpp \
+ binding/SharedRegionParcelable.cpp
LOCAL_CFLAGS += -Wno-unused-parameter -Wall -Werror
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.cpp b/media/libaaudio/src/binding/AAudioBinderClient.cpp
new file mode 100644
index 0000000..8315c40
--- /dev/null
+++ b/media/libaaudio/src/binding/AAudioBinderClient.cpp
@@ -0,0 +1,167 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#define LOG_TAG "AAudio"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <binder/IServiceManager.h>
+#include <utils/Mutex.h>
+#include <utils/RefBase.h>
+
+#include <aaudio/AAudio.h>
+
+#include "AudioEndpointParcelable.h"
+#include "binding/AAudioStreamRequest.h"
+#include "binding/AAudioStreamConfiguration.h"
+#include "binding/IAAudioService.h"
+#include "binding/AAudioServiceMessage.h"
+
+#include "AAudioBinderClient.h"
+#include "AAudioServiceInterface.h"
+
+using android::String16;
+using android::IServiceManager;
+using android::defaultServiceManager;
+using android::interface_cast;
+using android::IAAudioService;
+using android::Mutex;
+using android::sp;
+
+using namespace aaudio;
+
+static android::Mutex gServiceLock;
+static sp<IAAudioService> gAAudioService;
+
+// TODO Share code with other service clients.
+// Helper function to get access to the "AAudioService" service.
+// This code was modeled after frameworks/av/media/libaudioclient/AudioSystem.cpp
+static const sp<IAAudioService> getAAudioService() {
+ sp<IBinder> binder;
+ Mutex::Autolock _l(gServiceLock);
+ if (gAAudioService == 0) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ // Try several times to get the service.
+ int retries = 4;
+ do {
+ binder = sm->getService(String16(AAUDIO_SERVICE_NAME)); // This will wait a while.
+ if (binder != 0) {
+ break;
+ }
+ } while (retries-- > 0);
+
+ if (binder != 0) {
+ // TODO Add linkToDeath() like in frameworks/av/media/libaudioclient/AudioSystem.cpp
+ // TODO Create a DeathRecipient that disconnects all active streams.
+ gAAudioService = interface_cast<IAAudioService>(binder);
+ } else {
+ ALOGE("AudioStreamInternal could not get %s", AAUDIO_SERVICE_NAME);
+ }
+ }
+ return gAAudioService;
+}
+
+
+AAudioBinderClient::AAudioBinderClient()
+ : AAudioServiceInterface() {}
+
+AAudioBinderClient::~AAudioBinderClient() {}
+
+/**
+* @param request info needed to create the stream
+* @param configuration contains information about the created stream
+* @return handle to the stream or a negative error
+*/
+aaudio_handle_t AAudioBinderClient::openStream(const AAudioStreamRequest &request,
+ AAudioStreamConfiguration &configurationOutput) {
+
+ const sp<IAAudioService> &service = getAAudioService();
+ if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
+ return service->openStream(request, configurationOutput);
+}
+
+aaudio_result_t AAudioBinderClient::closeStream(aaudio_handle_t streamHandle) {
+
+ const sp<IAAudioService> &service = getAAudioService();
+ if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
+ return service->closeStream(streamHandle);
+}
+
+/* Get an immutable description of the in-memory queues
+* used to communicate with the underlying HAL or Service.
+*/
+aaudio_result_t AAudioBinderClient::getStreamDescription(aaudio_handle_t streamHandle,
+ AudioEndpointParcelable &parcelable) {
+
+ const sp<IAAudioService> &service = getAAudioService();
+ if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
+ return service->getStreamDescription(streamHandle, parcelable);
+}
+
+/**
+* Start the flow of data.
+*/
+aaudio_result_t AAudioBinderClient::startStream(aaudio_handle_t streamHandle) {
+ const sp<IAAudioService> &service = getAAudioService();
+ if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
+ return service->startStream(streamHandle);
+}
+
+/**
+* Stop the flow of data such that start() can resume without loss of data.
+*/
+aaudio_result_t AAudioBinderClient::pauseStream(aaudio_handle_t streamHandle) {
+ const sp<IAAudioService> &service = getAAudioService();
+ if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
+ return service->startStream(streamHandle);
+}
+
+/**
+* Discard any data held by the underlying HAL or Service.
+*/
+aaudio_result_t AAudioBinderClient::flushStream(aaudio_handle_t streamHandle) {
+ const sp<IAAudioService> &service = getAAudioService();
+ if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
+ return service->startStream(streamHandle);
+}
+
+/**
+* Manage the specified thread as a low latency audio thread.
+*/
+aaudio_result_t AAudioBinderClient::registerAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId,
+ int64_t periodNanoseconds) {
+ const sp<IAAudioService> &service = getAAudioService();
+ if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
+ return service->registerAudioThread(streamHandle,
+ clientProcessId,
+ clientThreadId,
+ periodNanoseconds);
+}
+
+aaudio_result_t AAudioBinderClient::unregisterAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId) {
+ const sp<IAAudioService> &service = getAAudioService();
+ if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
+ return service->unregisterAudioThread(streamHandle,
+ clientProcessId,
+ clientThreadId);
+}
+
+
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.h b/media/libaaudio/src/binding/AAudioBinderClient.h
new file mode 100644
index 0000000..5613d5b
--- /dev/null
+++ b/media/libaaudio/src/binding/AAudioBinderClient.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_AAUDIO_BINDER_CLIENT_H
+#define AAUDIO_AAUDIO_BINDER_CLIENT_H
+
+#include <aaudio/AAudioDefinitions.h>
+#include "AAudioServiceDefinitions.h"
+#include "AAudioServiceInterface.h"
+#include "binding/AAudioStreamRequest.h"
+#include "binding/AAudioStreamConfiguration.h"
+#include "binding/AudioEndpointParcelable.h"
+
+/**
+ * Implements the AAudioServiceInterface by talking to the actual service through Binder.
+ */
+
+namespace aaudio {
+
+class AAudioBinderClient : public AAudioServiceInterface {
+
+public:
+
+ AAudioBinderClient();
+
+ virtual ~AAudioBinderClient();
+
+ /**
+ * @param request info needed to create the stream
+ * @param configuration contains resulting information about the created stream
+ * @return handle to the stream or a negative error
+ */
+ aaudio_handle_t openStream(const AAudioStreamRequest &request,
+ AAudioStreamConfiguration &configurationOutput) override;
+
+ aaudio_result_t closeStream(aaudio_handle_t streamHandle) override;
+
+ /* Get an immutable description of the in-memory queues
+ * used to communicate with the underlying HAL or Service.
+ */
+ aaudio_result_t getStreamDescription(aaudio_handle_t streamHandle,
+ AudioEndpointParcelable &parcelable) override;
+
+ /**
+ * Start the flow of data.
+ * This is asynchronous. When complete, the service will send a STARTED event.
+ */
+ aaudio_result_t startStream(aaudio_handle_t streamHandle) override;
+
+ /**
+ * Stop the flow of data such that start() can resume without loss of data.
+ * This is asynchronous. When complete, the service will send a PAUSED event.
+ */
+ aaudio_result_t pauseStream(aaudio_handle_t streamHandle) override;
+
+ /**
+ * Discard any data held by the underlying HAL or Service.
+ * This is asynchronous. When complete, the service will send a FLUSHED event.
+ */
+ aaudio_result_t flushStream(aaudio_handle_t streamHandle) override;
+
+ /**
+ * Manage the specified thread as a low latency audio thread.
+ * TODO Consider passing this information as part of the startStream() call.
+ */
+ aaudio_result_t registerAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId,
+ int64_t periodNanoseconds) override;
+
+ aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId) override;
+};
+
+
+} /* namespace aaudio */
+
+#endif //AAUDIO_AAUDIO_BINDER_CLIENT_H
diff --git a/media/libaaudio/src/binding/AAudioServiceDefinitions.h b/media/libaaudio/src/binding/AAudioServiceDefinitions.h
index b58d170..0d5bae5 100644
--- a/media/libaaudio/src/binding/AAudioServiceDefinitions.h
+++ b/media/libaaudio/src/binding/AAudioServiceDefinitions.h
@@ -48,25 +48,6 @@
#define AAUDIO_HANDLE_INVALID ((aaudio_handle_t) -1)
-enum aaudio_commands_t {
- OPEN_STREAM = IBinder::FIRST_CALL_TRANSACTION,
- CLOSE_STREAM,
- GET_STREAM_DESCRIPTION,
- START_STREAM,
- PAUSE_STREAM,
- FLUSH_STREAM,
- REGISTER_AUDIO_THREAD,
- UNREGISTER_AUDIO_THREAD
-};
-
-// TODO Expand this to include all the open parameters.
-typedef struct AAudioServiceStreamInfo_s {
- int32_t deviceId;
- int32_t samplesPerFrame; // number of channels
- int32_t sampleRate;
- aaudio_audio_format_t audioFormat;
-} AAudioServiceStreamInfo;
-
// This must be a fixed width so it can be in shared memory.
enum RingbufferFlags : uint32_t {
NONE = 0,
diff --git a/media/libaaudio/src/binding/AAudioServiceInterface.h b/media/libaaudio/src/binding/AAudioServiceInterface.h
new file mode 100644
index 0000000..62fd894
--- /dev/null
+++ b/media/libaaudio/src/binding/AAudioServiceInterface.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_BINDING_AAUDIO_SERVICE_INTERFACE_H
+#define AAUDIO_BINDING_AAUDIO_SERVICE_INTERFACE_H
+
+#include "binding/AAudioServiceDefinitions.h"
+#include "binding/AAudioStreamRequest.h"
+#include "binding/AAudioStreamConfiguration.h"
+#include "binding/AudioEndpointParcelable.h"
+
+/**
+ * This has the same methods as IAAudioService but without the Binder features.
+ *
+ * It allows us to abstract the Binder interface and use an AudioStreamInternal
+ * both in the client and in the service.
+ */
+namespace aaudio {
+
+class AAudioServiceInterface {
+public:
+
+ AAudioServiceInterface() {};
+ virtual ~AAudioServiceInterface() = default;
+
+ /**
+ * @param request info needed to create the stream
+ * @param configuration contains information about the created stream
+ * @return handle to the stream or a negative error
+ */
+ virtual aaudio_handle_t openStream(const AAudioStreamRequest &request,
+ AAudioStreamConfiguration &configuration) = 0;
+
+ virtual aaudio_result_t closeStream(aaudio_handle_t streamHandle) = 0;
+
+ /* Get an immutable description of the in-memory queues
+ * used to communicate with the underlying HAL or Service.
+ */
+ virtual aaudio_result_t getStreamDescription(aaudio_handle_t streamHandle,
+ AudioEndpointParcelable &parcelable) = 0;
+
+ /**
+ * Start the flow of data.
+ */
+ virtual aaudio_result_t startStream(aaudio_handle_t streamHandle) = 0;
+
+ /**
+ * Stop the flow of data such that start() can resume without loss of data.
+ */
+ virtual aaudio_result_t pauseStream(aaudio_handle_t streamHandle) = 0;
+
+ /**
+ * Discard any data held by the underlying HAL or Service.
+ */
+ virtual aaudio_result_t flushStream(aaudio_handle_t streamHandle) = 0;
+
+ /**
+ * Manage the specified thread as a low latency audio thread.
+ */
+ virtual aaudio_result_t registerAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId,
+ int64_t periodNanoseconds) = 0;
+
+ virtual aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId) = 0;
+};
+
+} /* namespace aaudio */
+
+#endif //AAUDIO_BINDING_AAUDIO_SERVICE_INTERFACE_H
diff --git a/media/libaaudio/src/binding/AAudioServiceMessage.h b/media/libaaudio/src/binding/AAudioServiceMessage.h
index cc77d59..b74b6c2 100644
--- a/media/libaaudio/src/binding/AAudioServiceMessage.h
+++ b/media/libaaudio/src/binding/AAudioServiceMessage.h
@@ -25,10 +25,11 @@
// TODO move this to an "include" folder for the service.
+// Used to send information about the HAL to the client.
struct AAudioMessageTimestamp {
- int64_t position;
- int64_t deviceOffset; // add to client position to get device position
- int64_t timestamp;
+ int64_t position; // number of frames transferred so far
+ int64_t deviceOffset; // add to client position to get device position
+ int64_t timestamp; // time when that position was reached
};
typedef enum aaudio_service_event_e : uint32_t {
@@ -36,13 +37,14 @@
AAUDIO_SERVICE_EVENT_PAUSED,
AAUDIO_SERVICE_EVENT_FLUSHED,
AAUDIO_SERVICE_EVENT_CLOSED,
- AAUDIO_SERVICE_EVENT_DISCONNECTED
+ AAUDIO_SERVICE_EVENT_DISCONNECTED,
+ AAUDIO_SERVICE_EVENT_VOLUME
} aaudio_service_event_t;
struct AAudioMessageEvent {
aaudio_service_event_t event;
- int32_t data1;
- int64_t data2;
+ double dataDouble;
+ int64_t dataLong;
};
typedef struct AAudioServiceMessage_s {
@@ -54,12 +56,11 @@
code what;
union {
- AAudioMessageTimestamp timestamp;
- AAudioMessageEvent event;
+ AAudioMessageTimestamp timestamp; // what == TIMESTAMP
+ AAudioMessageEvent event; // what == EVENT
};
} AAudioServiceMessage;
-
} /* namespace aaudio */
#endif //AAUDIO_AAUDIO_SERVICE_MESSAGE_H
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
index fe3a59f..ba41a3b 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
@@ -35,26 +35,50 @@
AAudioStreamConfiguration::~AAudioStreamConfiguration() {}
status_t AAudioStreamConfiguration::writeToParcel(Parcel* parcel) const {
- parcel->writeInt32(mDeviceId);
- parcel->writeInt32(mSampleRate);
- parcel->writeInt32(mSamplesPerFrame);
- parcel->writeInt32((int32_t) mAudioFormat);
- parcel->writeInt32(mBufferCapacity);
- return NO_ERROR; // TODO check for errors above
+ status_t status;
+ status = parcel->writeInt32(mDeviceId);
+ if (status != NO_ERROR) goto error;
+ status = parcel->writeInt32(mSampleRate);
+ if (status != NO_ERROR) goto error;
+ status = parcel->writeInt32(mSamplesPerFrame);
+ if (status != NO_ERROR) goto error;
+ status = parcel->writeInt32((int32_t) mSharingMode);
+ ALOGD("AAudioStreamConfiguration.writeToParcel(): mSharingMode = %d", mSharingMode);
+ if (status != NO_ERROR) goto error;
+ status = parcel->writeInt32((int32_t) mAudioFormat);
+ if (status != NO_ERROR) goto error;
+ status = parcel->writeInt32(mBufferCapacity);
+ if (status != NO_ERROR) goto error;
+ return NO_ERROR;
+error:
+ ALOGE("AAudioStreamConfiguration.writeToParcel(): write failed = %d", status);
+ return status;
}
status_t AAudioStreamConfiguration::readFromParcel(const Parcel* parcel) {
int32_t temp;
- parcel->readInt32(&mDeviceId);
- parcel->readInt32(&mSampleRate);
- parcel->readInt32(&mSamplesPerFrame);
- parcel->readInt32(&temp);
+ status_t status = parcel->readInt32(&mDeviceId);
+ if (status != NO_ERROR) goto error;
+ status = parcel->readInt32(&mSampleRate);
+ if (status != NO_ERROR) goto error;
+ status = parcel->readInt32(&mSamplesPerFrame);
+ if (status != NO_ERROR) goto error;
+ status = parcel->readInt32(&temp);
+ if (status != NO_ERROR) goto error;
+ mSharingMode = (aaudio_sharing_mode_t) temp;
+ ALOGD("AAudioStreamConfiguration.readFromParcel(): mSharingMode = %d", mSharingMode);
+ status = parcel->readInt32(&temp);
+ if (status != NO_ERROR) goto error;
mAudioFormat = (aaudio_audio_format_t) temp;
- parcel->readInt32(&mBufferCapacity);
- return NO_ERROR; // TODO check for errors above
+ status = parcel->readInt32(&mBufferCapacity);
+ if (status != NO_ERROR) goto error;
+ return NO_ERROR;
+error:
+ ALOGE("AAudioStreamConfiguration.readFromParcel(): read failed = %d", status);
+ return status;
}
-aaudio_result_t AAudioStreamConfiguration::validate() {
+aaudio_result_t AAudioStreamConfiguration::validate() const {
// Validate results of the open.
if (mSampleRate < 0 || mSampleRate >= 8 * 48000) { // TODO review limits
ALOGE("AAudioStreamConfiguration.validate(): invalid sampleRate = %d", mSampleRate);
@@ -84,9 +108,11 @@
return AAUDIO_OK;
}
-void AAudioStreamConfiguration::dump() {
- ALOGD("AAudioStreamConfiguration mSampleRate = %d -----", mSampleRate);
+void AAudioStreamConfiguration::dump() const {
+ ALOGD("AAudioStreamConfiguration mDeviceId = %d", mDeviceId);
+ ALOGD("AAudioStreamConfiguration mSampleRate = %d", mSampleRate);
ALOGD("AAudioStreamConfiguration mSamplesPerFrame = %d", mSamplesPerFrame);
+ ALOGD("AAudioStreamConfiguration mSharingMode = %d", (int)mSharingMode);
ALOGD("AAudioStreamConfiguration mAudioFormat = %d", (int)mAudioFormat);
ALOGD("AAudioStreamConfiguration mBufferCapacity = %d", mBufferCapacity);
}
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.h b/media/libaaudio/src/binding/AAudioStreamConfiguration.h
index 57b1c59..b68d8b2 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.h
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.h
@@ -66,6 +66,14 @@
mAudioFormat = audioFormat;
}
+ aaudio_sharing_mode_t getSharingMode() const {
+ return mSharingMode;
+ }
+
+ void setSharingMode(aaudio_sharing_mode_t sharingMode) {
+ mSharingMode = sharingMode;
+ }
+
int32_t getBufferCapacity() const {
return mBufferCapacity;
}
@@ -78,14 +86,15 @@
virtual status_t readFromParcel(const Parcel* parcel) override;
- aaudio_result_t validate();
+ aaudio_result_t validate() const;
- void dump();
+ void dump() const;
-protected:
+private:
int32_t mDeviceId = AAUDIO_DEVICE_UNSPECIFIED;
int32_t mSampleRate = AAUDIO_UNSPECIFIED;
int32_t mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+ aaudio_sharing_mode_t mSharingMode = AAUDIO_SHARING_MODE_SHARED;
aaudio_audio_format_t mAudioFormat = AAUDIO_FORMAT_UNSPECIFIED;
int32_t mBufferCapacity = AAUDIO_UNSPECIFIED;
};
diff --git a/media/libaaudio/src/binding/AAudioStreamRequest.cpp b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
index 5202b73..b8a0429 100644
--- a/media/libaaudio/src/binding/AAudioStreamRequest.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+#define LOG_TAG "AAudio"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
#include <stdint.h>
#include <sys/mman.h>
@@ -39,28 +43,48 @@
AAudioStreamRequest::~AAudioStreamRequest() {}
status_t AAudioStreamRequest::writeToParcel(Parcel* parcel) const {
- parcel->writeInt32((int32_t) mUserId);
- parcel->writeInt32((int32_t) mProcessId);
- mConfiguration.writeToParcel(parcel);
- return NO_ERROR; // TODO check for errors above
+ status_t status = parcel->writeInt32((int32_t) mUserId);
+ if (status != NO_ERROR) goto error;
+ status = parcel->writeInt32((int32_t) mProcessId);
+ if (status != NO_ERROR) goto error;
+ status = parcel->writeInt32((int32_t) mDirection);
+ if (status != NO_ERROR) goto error;
+ status = mConfiguration.writeToParcel(parcel);
+ if (status != NO_ERROR) goto error;
+ return NO_ERROR;
+
+error:
+ ALOGE("AAudioStreamRequest.writeToParcel(): write failed = %d", status);
+ return status;
}
status_t AAudioStreamRequest::readFromParcel(const Parcel* parcel) {
int32_t temp;
- parcel->readInt32(&temp);
+ status_t status = parcel->readInt32(&temp);
+ if (status != NO_ERROR) goto error;
mUserId = (uid_t) temp;
- parcel->readInt32(&temp);
+ status = parcel->readInt32(&temp);
+ if (status != NO_ERROR) goto error;
mProcessId = (pid_t) temp;
- mConfiguration.readFromParcel(parcel);
- return NO_ERROR; // TODO check for errors above
+ status = parcel->readInt32(&temp);
+ if (status != NO_ERROR) goto error;
+ mDirection = (aaudio_direction_t) temp;
+ status = mConfiguration.readFromParcel(parcel);
+ if (status != NO_ERROR) goto error;
+ return NO_ERROR;
+
+error:
+ ALOGE("AAudioStreamRequest.readFromParcel(): read failed = %d", status);
+ return status;
}
-aaudio_result_t AAudioStreamRequest::validate() {
+aaudio_result_t AAudioStreamRequest::validate() const {
return mConfiguration.validate();
}
-void AAudioStreamRequest::dump() {
- ALOGD("AAudioStreamRequest mUserId = %d -----", mUserId);
+void AAudioStreamRequest::dump() const {
+ ALOGD("AAudioStreamRequest mUserId = %d", mUserId);
ALOGD("AAudioStreamRequest mProcessId = %d", mProcessId);
+ ALOGD("AAudioStreamRequest mDirection = %d", mDirection);
mConfiguration.dump();
}
diff --git a/media/libaaudio/src/binding/AAudioStreamRequest.h b/media/libaaudio/src/binding/AAudioStreamRequest.h
index 0fd28ba..6546562 100644
--- a/media/libaaudio/src/binding/AAudioStreamRequest.h
+++ b/media/libaaudio/src/binding/AAudioStreamRequest.h
@@ -52,6 +52,18 @@
mProcessId = processId;
}
+ aaudio_direction_t getDirection() const {
+ return mDirection;
+ }
+
+ void setDirection(aaudio_direction_t direction) {
+ mDirection = direction;
+ }
+
+ const AAudioStreamConfiguration &getConstantConfiguration() const {
+ return mConfiguration;
+ }
+
AAudioStreamConfiguration &getConfiguration() {
return mConfiguration;
}
@@ -60,14 +72,15 @@
virtual status_t readFromParcel(const Parcel* parcel) override;
- aaudio_result_t validate();
+ aaudio_result_t validate() const;
- void dump();
+ void dump() const;
protected:
AAudioStreamConfiguration mConfiguration;
- uid_t mUserId;
- pid_t mProcessId;
+ uid_t mUserId;
+ pid_t mProcessId;
+ aaudio_direction_t mDirection;
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
index f40ee02..ee92ee3 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
@@ -14,11 +14,15 @@
* limitations under the License.
*/
+#define LOG_TAG "AAudio"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
#include <stdint.h>
-#include <sys/mman.h>
#include <binder/Parcel.h>
#include <binder/Parcelable.h>
+#include <utility/AAudioUtilities.h>
#include "binding/AAudioServiceDefinitions.h"
#include "binding/RingBufferParcelable.h"
@@ -82,13 +86,27 @@
}
aaudio_result_t AudioEndpointParcelable::resolve(EndpointDescriptor *descriptor) {
- // TODO error check
- mUpMessageQueueParcelable.resolve(mSharedMemories, &descriptor->upMessageQueueDescriptor);
- mDownMessageQueueParcelable.resolve(mSharedMemories,
+ aaudio_result_t result = mUpMessageQueueParcelable.resolve(mSharedMemories,
+ &descriptor->upMessageQueueDescriptor);
+ if (result != AAUDIO_OK) return result;
+ result = mDownMessageQueueParcelable.resolve(mSharedMemories,
&descriptor->downMessageQueueDescriptor);
- mUpDataQueueParcelable.resolve(mSharedMemories, &descriptor->upDataQueueDescriptor);
- mDownDataQueueParcelable.resolve(mSharedMemories, &descriptor->downDataQueueDescriptor);
- return AAUDIO_OK;
+ if (result != AAUDIO_OK) return result;
+
+ result = mUpDataQueueParcelable.resolve(mSharedMemories, &descriptor->upDataQueueDescriptor);
+ if (result != AAUDIO_OK) return result;
+ result = mDownDataQueueParcelable.resolve(mSharedMemories,
+ &descriptor->downDataQueueDescriptor);
+ return result;
+}
+
+aaudio_result_t AudioEndpointParcelable::close() {
+ int err = 0;
+ for (int i = 0; i < mNumSharedMemories; i++) {
+ int lastErr = mSharedMemories[i].close();
+ if (lastErr < 0) err = lastErr;
+ }
+ return AAudioConvert_androidToAAudioResult(err);
}
aaudio_result_t AudioEndpointParcelable::validate() {
@@ -100,6 +118,7 @@
for (int i = 0; i < mNumSharedMemories; i++) {
result = mSharedMemories[i].validate();
if (result != AAUDIO_OK) {
+ ALOGE("AudioEndpointParcelable invalid mSharedMemories[%d] = %d", i, result);
return result;
}
}
diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.h b/media/libaaudio/src/binding/AudioEndpointParcelable.h
index d4646d0..4a1cb72 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.h
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.h
@@ -57,6 +57,8 @@
aaudio_result_t validate();
+ aaudio_result_t close();
+
void dump();
public: // TODO add getters
diff --git a/media/libaaudio/src/binding/IAAudioService.cpp b/media/libaaudio/src/binding/IAAudioService.cpp
index c21033e..20cbbc8 100644
--- a/media/libaaudio/src/binding/IAAudioService.cpp
+++ b/media/libaaudio/src/binding/IAAudioService.cpp
@@ -40,11 +40,13 @@
{
}
- virtual aaudio_handle_t openStream(aaudio::AAudioStreamRequest &request,
- aaudio::AAudioStreamConfiguration &configuration) override {
+ virtual aaudio_handle_t openStream(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) override {
Parcel data, reply;
// send command
data.writeInterfaceToken(IAAudioService::getInterfaceDescriptor());
+ ALOGE("BpAAudioService::client openStream request dump --------------------");
+ request.dump();
request.writeToParcel(&data);
status_t err = remote()->transact(OPEN_STREAM, data, &reply);
if (err != NO_ERROR) {
@@ -53,7 +55,12 @@
// parse reply
aaudio_handle_t stream;
reply.readInt32(&stream);
- configuration.readFromParcel(&reply);
+ err = configurationOutput.readFromParcel(&reply);
+ if (err != NO_ERROR) {
+ ALOGE("BpAAudioService::client openStream readFromParcel failed %d", err);
+ closeStream(stream);
+ return AAudioConvert_androidToAAudioResult(err);
+ }
return stream;
}
@@ -80,16 +87,30 @@
data.writeInt32(streamHandle);
status_t err = remote()->transact(GET_STREAM_DESCRIPTION, data, &reply);
if (err != NO_ERROR) {
+ ALOGE("BpAAudioService::client transact(GET_STREAM_DESCRIPTION) returns %d", err);
return AAudioConvert_androidToAAudioResult(err);
}
// parse reply
- parcelable.readFromParcel(&reply);
- parcelable.dump();
- aaudio_result_t result = parcelable.validate();
- if (result != AAUDIO_OK) {
+ aaudio_result_t result;
+ err = reply.readInt32(&result);
+ if (err != NO_ERROR) {
+ ALOGE("BpAAudioService::client transact(GET_STREAM_DESCRIPTION) readInt %d", err);
+ return AAudioConvert_androidToAAudioResult(err);
+ } else if (result != AAUDIO_OK) {
+ ALOGE("BpAAudioService::client GET_STREAM_DESCRIPTION passed result %d", result);
return result;
}
- reply.readInt32(&result);
+ err = parcelable.readFromParcel(&reply);;
+ if (err != NO_ERROR) {
+ ALOGE("BpAAudioService::client transact(GET_STREAM_DESCRIPTION) read endpoint %d", err);
+ return AAudioConvert_androidToAAudioResult(err);
+ }
+ //parcelable.dump();
+ result = parcelable.validate();
+ if (result != AAUDIO_OK) {
+ ALOGE("BpAAudioService::client GET_STREAM_DESCRIPTION validation fails %d", result);
+ return result;
+ }
return result;
}
@@ -139,13 +160,16 @@
return res;
}
- virtual aaudio_result_t registerAudioThread(aaudio_handle_t streamHandle, pid_t clientThreadId,
+ virtual aaudio_result_t registerAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId,
int64_t periodNanoseconds)
override {
Parcel data, reply;
// send command
data.writeInterfaceToken(IAAudioService::getInterfaceDescriptor());
data.writeInt32(streamHandle);
+ data.writeInt32((int32_t) clientProcessId);
data.writeInt32((int32_t) clientThreadId);
data.writeInt64(periodNanoseconds);
status_t err = remote()->transact(REGISTER_AUDIO_THREAD, data, &reply);
@@ -158,12 +182,15 @@
return res;
}
- virtual aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle, pid_t clientThreadId)
+ virtual aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId)
override {
Parcel data, reply;
// send command
data.writeInterfaceToken(IAAudioService::getInterfaceDescriptor());
data.writeInt32(streamHandle);
+ data.writeInt32((int32_t) clientProcessId);
data.writeInt32((int32_t) clientThreadId);
status_t err = remote()->transact(UNREGISTER_AUDIO_THREAD, data, &reply);
if (err != NO_ERROR) {
@@ -178,7 +205,7 @@
};
// Implement an interface to the service.
-// This is here so that you don't have to link with liboboe static library.
+// This is here so that you don't have to link with libaaudio static library.
IMPLEMENT_META_INTERFACE(AAudioService, "IAAudioService");
// The order of parameters in the Parcels must match with code in BpAAudioService
@@ -189,6 +216,7 @@
aaudio::AAudioStreamRequest request;
aaudio::AAudioStreamConfiguration configuration;
pid_t pid;
+ pid_t tid;
int64_t nanoseconds;
aaudio_result_t result;
ALOGV("BnAAudioService::onTransact(%i) %i", code, flags);
@@ -197,8 +225,12 @@
switch(code) {
case OPEN_STREAM: {
request.readFromParcel(&data);
+
+ ALOGD("BnAAudioService::client openStream request dump --------------------");
+ request.dump();
+
stream = openStream(request, configuration);
- ALOGD("BnAAudioService::onTransact OPEN_STREAM server handle = 0x%08X", stream);
+ ALOGV("BnAAudioService::onTransact OPEN_STREAM server handle = 0x%08X", stream);
reply->writeInt32(stream);
configuration.writeToParcel(reply);
return NO_ERROR;
@@ -206,7 +238,7 @@
case CLOSE_STREAM: {
data.readInt32(&stream);
- ALOGD("BnAAudioService::onTransact CLOSE_STREAM 0x%08X", stream);
+ ALOGV("BnAAudioService::onTransact CLOSE_STREAM 0x%08X", stream);
result = closeStream(stream);
reply->writeInt32(result);
return NO_ERROR;
@@ -214,26 +246,28 @@
case GET_STREAM_DESCRIPTION: {
data.readInt32(&stream);
- ALOGD("BnAAudioService::onTransact GET_STREAM_DESCRIPTION 0x%08X", stream);
+ ALOGI("BnAAudioService::onTransact GET_STREAM_DESCRIPTION 0x%08X", stream);
aaudio::AudioEndpointParcelable parcelable;
result = getStreamDescription(stream, parcelable);
+ ALOGI("BnAAudioService::onTransact getStreamDescription() returns %d", result);
if (result != AAUDIO_OK) {
return AAudioConvert_aaudioToAndroidStatus(result);
}
- parcelable.dump();
result = parcelable.validate();
if (result != AAUDIO_OK) {
+ ALOGE("BnAAudioService::onTransact getStreamDescription() returns %d", result);
+ parcelable.dump();
return AAudioConvert_aaudioToAndroidStatus(result);
}
- parcelable.writeToParcel(reply);
reply->writeInt32(result);
+ parcelable.writeToParcel(reply);
return NO_ERROR;
} break;
case START_STREAM: {
data.readInt32(&stream);
result = startStream(stream);
- ALOGD("BnAAudioService::onTransact START_STREAM 0x%08X, result = %d",
+ ALOGV("BnAAudioService::onTransact START_STREAM 0x%08X, result = %d",
stream, result);
reply->writeInt32(result);
return NO_ERROR;
@@ -242,7 +276,7 @@
case PAUSE_STREAM: {
data.readInt32(&stream);
result = pauseStream(stream);
- ALOGD("BnAAudioService::onTransact PAUSE_STREAM 0x%08X, result = %d",
+ ALOGV("BnAAudioService::onTransact PAUSE_STREAM 0x%08X, result = %d",
stream, result);
reply->writeInt32(result);
return NO_ERROR;
@@ -251,7 +285,7 @@
case FLUSH_STREAM: {
data.readInt32(&stream);
result = flushStream(stream);
- ALOGD("BnAAudioService::onTransact FLUSH_STREAM 0x%08X, result = %d",
+ ALOGV("BnAAudioService::onTransact FLUSH_STREAM 0x%08X, result = %d",
stream, result);
reply->writeInt32(result);
return NO_ERROR;
@@ -260,9 +294,10 @@
case REGISTER_AUDIO_THREAD: {
data.readInt32(&stream);
data.readInt32(&pid);
+ data.readInt32(&tid);
data.readInt64(&nanoseconds);
- result = registerAudioThread(stream, pid, nanoseconds);
- ALOGD("BnAAudioService::onTransact REGISTER_AUDIO_THREAD 0x%08X, result = %d",
+ result = registerAudioThread(stream, pid, tid, nanoseconds);
+ ALOGV("BnAAudioService::onTransact REGISTER_AUDIO_THREAD 0x%08X, result = %d",
stream, result);
reply->writeInt32(result);
return NO_ERROR;
@@ -271,8 +306,9 @@
case UNREGISTER_AUDIO_THREAD: {
data.readInt32(&stream);
data.readInt32(&pid);
- result = unregisterAudioThread(stream, pid);
- ALOGD("BnAAudioService::onTransact UNREGISTER_AUDIO_THREAD 0x%08X, result = %d",
+ data.readInt32(&tid);
+ result = unregisterAudioThread(stream, pid, tid);
+ ALOGV("BnAAudioService::onTransact UNREGISTER_AUDIO_THREAD 0x%08X, result = %d",
stream, result);
reply->writeInt32(result);
return NO_ERROR;
diff --git a/media/libaaudio/src/binding/IAAudioService.h b/media/libaaudio/src/binding/IAAudioService.h
index 53c3b45..ab7fd1b 100644
--- a/media/libaaudio/src/binding/IAAudioService.h
+++ b/media/libaaudio/src/binding/IAAudioService.h
@@ -28,9 +28,12 @@
#include "binding/AudioEndpointParcelable.h"
#include "binding/AAudioStreamRequest.h"
#include "binding/AAudioStreamConfiguration.h"
+#include "utility/HandleTracker.h"
namespace android {
+#define AAUDIO_SERVICE_NAME "media.aaudio"
+
// Interface (our AIDL) - Shared by server and client
class IAAudioService : public IInterface {
public:
@@ -42,8 +45,8 @@
* @param configuration contains information about the created stream
* @return handle to the stream or a negative error
*/
- virtual aaudio::aaudio_handle_t openStream(aaudio::AAudioStreamRequest &request,
- aaudio::AAudioStreamConfiguration &configuration) = 0;
+ virtual aaudio_handle_t openStream(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) = 0;
virtual aaudio_result_t closeStream(aaudio::aaudio_handle_t streamHandle) = 0;
@@ -55,26 +58,33 @@
/**
* Start the flow of data.
+ * This is asynchronous. When complete, the service will send a STARTED event.
*/
virtual aaudio_result_t startStream(aaudio::aaudio_handle_t streamHandle) = 0;
/**
* Stop the flow of data such that start() can resume without loss of data.
+ * This is asynchronous. When complete, the service will send a PAUSED event.
*/
virtual aaudio_result_t pauseStream(aaudio::aaudio_handle_t streamHandle) = 0;
/**
* Discard any data held by the underlying HAL or Service.
+ * This is asynchronous. When complete, the service will send a FLUSHED event.
*/
virtual aaudio_result_t flushStream(aaudio::aaudio_handle_t streamHandle) = 0;
/**
* Manage the specified thread as a low latency audio thread.
+ * TODO Consider passing this information as part of the startStream() call.
*/
- virtual aaudio_result_t registerAudioThread(aaudio::aaudio_handle_t streamHandle, pid_t clientThreadId,
+ virtual aaudio_result_t registerAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
+ pid_t clientThreadId,
int64_t periodNanoseconds) = 0;
- virtual aaudio_result_t unregisterAudioThread(aaudio::aaudio_handle_t streamHandle,
+ virtual aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
pid_t clientThreadId) = 0;
};
diff --git a/media/libaaudio/src/binding/RingBufferParcelable.cpp b/media/libaaudio/src/binding/RingBufferParcelable.cpp
index 3a92929..05451f9 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.cpp
+++ b/media/libaaudio/src/binding/RingBufferParcelable.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+#define LOG_TAG "AAudio"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
#include <stdint.h>
#include <binder/Parcelable.h>
diff --git a/media/libaaudio/src/binding/RingBufferParcelable.h b/media/libaaudio/src/binding/RingBufferParcelable.h
index 3f82c79..5fc5d00 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.h
+++ b/media/libaaudio/src/binding/RingBufferParcelable.h
@@ -55,6 +55,8 @@
void setCapacityInFrames(int32_t capacityInFrames);
+ bool isFileDescriptorSafe(SharedMemoryParcelable *memoryParcels);
+
/**
* The read and write must be symmetric.
*/
diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
index 1102dec..cfb820f 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
@@ -14,12 +14,18 @@
* limitations under the License.
*/
+#define LOG_TAG "AAudio"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
#include <stdint.h>
+#include <stdio.h>
#include <sys/mman.h>
#include <aaudio/AAudioDefinitions.h>
#include <binder/Parcelable.h>
+#include <utility/AAudioUtilities.h>
#include "binding/SharedMemoryParcelable.h"
@@ -36,28 +42,55 @@
void SharedMemoryParcelable::setup(int fd, int32_t sizeInBytes) {
mFd = fd;
mSizeInBytes = sizeInBytes;
+
}
status_t SharedMemoryParcelable::writeToParcel(Parcel* parcel) const {
- parcel->writeInt32(mSizeInBytes);
+ status_t status = parcel->writeInt32(mSizeInBytes);
+ if (status != NO_ERROR) return status;
if (mSizeInBytes > 0) {
- parcel->writeDupFileDescriptor(mFd);
+ status = parcel->writeDupFileDescriptor(mFd);
+ ALOGE_IF(status != NO_ERROR, "SharedMemoryParcelable writeDupFileDescriptor failed : %d", status);
}
- return NO_ERROR; // TODO check for errors above
+ return status;
}
status_t SharedMemoryParcelable::readFromParcel(const Parcel* parcel) {
- parcel->readInt32(&mSizeInBytes);
- if (mSizeInBytes > 0) {
- mFd = dup(parcel->readFileDescriptor());
+ status_t status = parcel->readInt32(&mSizeInBytes);
+ if (status != NO_ERROR) {
+ return status;
}
- return NO_ERROR; // TODO check for errors above
+ if (mSizeInBytes > 0) {
+// FIXME mFd = dup(parcel->readFileDescriptor());
+ // Why is the ALSA resource not getting freed?!
+ mFd = fcntl(parcel->readFileDescriptor(), F_DUPFD_CLOEXEC, 0);
+ if (mFd == -1) {
+ status = -errno;
+ ALOGE("SharedMemoryParcelable readFileDescriptor fcntl() failed : %d", status);
+ }
+ }
+ return status;
}
-// TODO Add code to unmmap()
+aaudio_result_t SharedMemoryParcelable::close() {
+ if (mResolvedAddress != nullptr) {
+ int err = munmap(mResolvedAddress, mSizeInBytes);
+ if (err < 0) {
+ ALOGE("SharedMemoryParcelable::close() munmap() failed %d", err);
+ return AAudioConvert_androidToAAudioResult(err);
+ }
+ mResolvedAddress = nullptr;
+ }
+ if (mFd != -1) {
+ ::close(mFd);
+ mFd = -1;
+ }
+ return AAUDIO_OK;
+}
aaudio_result_t SharedMemoryParcelable::resolve(int32_t offsetInBytes, int32_t sizeInBytes,
void **regionAddressPtr) {
+
if (offsetInBytes < 0) {
ALOGE("SharedMemoryParcelable illegal offsetInBytes = %d", offsetInBytes);
return AAUDIO_ERROR_OUT_OF_RANGE;
@@ -68,6 +101,11 @@
return AAUDIO_ERROR_OUT_OF_RANGE;
}
if (mResolvedAddress == nullptr) {
+ /* TODO remove
+ int fd = fcntl(mFd, F_DUPFD_CLOEXEC, 0);
+ ALOGE_IF(fd==-1, "cannot dup fd=%d, size=%zd, (%s)",
+ mFd, mSizeInBytes, strerror(errno));
+ */
mResolvedAddress = (uint8_t *) mmap(0, mSizeInBytes, PROT_READ|PROT_WRITE,
MAP_SHARED, mFd, 0);
if (mResolvedAddress == nullptr) {
@@ -76,8 +114,8 @@
}
}
*regionAddressPtr = mResolvedAddress + offsetInBytes;
- ALOGD("SharedMemoryParcelable mResolvedAddress = %p", mResolvedAddress);
- ALOGD("SharedMemoryParcelable offset by %d, *regionAddressPtr = %p",
+ ALOGV("SharedMemoryParcelable mResolvedAddress = %p", mResolvedAddress);
+ ALOGV("SharedMemoryParcelable offset by %d, *regionAddressPtr = %p",
offsetInBytes, *regionAddressPtr);
return AAUDIO_OK;
}
diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.h b/media/libaaudio/src/binding/SharedMemoryParcelable.h
index 7e0bf1a..22e16f0 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.h
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.h
@@ -49,8 +49,14 @@
virtual status_t readFromParcel(const Parcel* parcel) override;
+ // mmap() shared memory
aaudio_result_t resolve(int32_t offsetInBytes, int32_t sizeInBytes, void **regionAddressPtr);
+ // munmap() any mapped memory
+ aaudio_result_t close();
+
+ bool isFileDescriptorSafe();
+
int32_t getSizeInBytes();
aaudio_result_t validate();
diff --git a/media/libaaudio/src/binding/SharedRegionParcelable.cpp b/media/libaaudio/src/binding/SharedRegionParcelable.cpp
index 8ca0023..8e57832 100644
--- a/media/libaaudio/src/binding/SharedRegionParcelable.cpp
+++ b/media/libaaudio/src/binding/SharedRegionParcelable.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+#define LOG_TAG "AAudio"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
#include <stdint.h>
#include <sys/mman.h>
diff --git a/media/libaaudio/src/binding/SharedRegionParcelable.h b/media/libaaudio/src/binding/SharedRegionParcelable.h
index d6c2281..5fb2a4c 100644
--- a/media/libaaudio/src/binding/SharedRegionParcelable.h
+++ b/media/libaaudio/src/binding/SharedRegionParcelable.h
@@ -45,6 +45,8 @@
aaudio_result_t resolve(SharedMemoryParcelable *memoryParcels, void **regionAddressPtr);
+ bool isFileDescriptorSafe(SharedMemoryParcelable *memoryParcels);
+
aaudio_result_t validate();
void dump();
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 47c4774..fe049b2 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -19,7 +19,7 @@
#include <utils/Log.h>
#include <cassert>
-#include <aaudio/AAudioDefinitions.h>
+#include <aaudio/AAudio.h>
#include "AudioEndpointParcelable.h"
#include "AudioEndpoint.h"
@@ -39,11 +39,26 @@
{
}
-static void AudioEndpoint_validateQueueDescriptor(const char *type,
+static aaudio_result_t AudioEndpoint_validateQueueDescriptor(const char *type,
const RingBufferDescriptor *descriptor) {
- assert(descriptor->capacityInFrames > 0);
- assert(descriptor->bytesPerFrame > 1);
- assert(descriptor->dataAddress != nullptr);
+ if (descriptor == nullptr) {
+ ALOGE("AudioEndpoint_validateQueueDescriptor() NULL descriptor");
+ return AAUDIO_ERROR_NULL;
+ }
+ if (descriptor->capacityInFrames <= 0) {
+ ALOGE("AudioEndpoint_validateQueueDescriptor() bad capacityInFrames = %d",
+ descriptor->capacityInFrames);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+ }
+ if (descriptor->bytesPerFrame <= 1) {
+ ALOGE("AudioEndpoint_validateQueueDescriptor() bad bytesPerFrame = %d",
+ descriptor->bytesPerFrame);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+ }
+ if (descriptor->dataAddress == nullptr) {
+ ALOGE("AudioEndpoint_validateQueueDescriptor() NULL dataAddress");
+ return AAUDIO_ERROR_NULL;
+ }
ALOGD("AudioEndpoint_validateQueueDescriptor %s, dataAddress at %p ====================",
type,
descriptor->dataAddress);
@@ -52,11 +67,12 @@
descriptor->writeCounterAddress);
// Try to READ from the data area.
+ // This code will crash if the mmap failed.
uint8_t value = descriptor->dataAddress[0];
ALOGD("AudioEndpoint_validateQueueDescriptor() dataAddress[0] = %d, then try to write",
(int) value);
// Try to WRITE to the data area.
- descriptor->dataAddress[0] = value;
+ descriptor->dataAddress[0] = value * 3;
ALOGD("AudioEndpoint_validateQueueDescriptor() wrote successfully");
if (descriptor->readCounterAddress) {
@@ -73,17 +89,28 @@
*descriptor->writeCounterAddress = counter;
ALOGD("AudioEndpoint_validateQueueDescriptor() wrote writeCounterAddress successfully");
}
+ return AAUDIO_OK;
}
-void AudioEndpoint_validateDescriptor(const EndpointDescriptor *pEndpointDescriptor) {
- AudioEndpoint_validateQueueDescriptor("msg", &pEndpointDescriptor->upMessageQueueDescriptor);
- AudioEndpoint_validateQueueDescriptor("data", &pEndpointDescriptor->downDataQueueDescriptor);
+aaudio_result_t AudioEndpoint_validateDescriptor(const EndpointDescriptor *pEndpointDescriptor) {
+ aaudio_result_t result = AudioEndpoint_validateQueueDescriptor("messages",
+ &pEndpointDescriptor->upMessageQueueDescriptor);
+ if (result == AAUDIO_OK) {
+ result = AudioEndpoint_validateQueueDescriptor("data",
+ &pEndpointDescriptor->downDataQueueDescriptor);
+ }
+ return result;
}
aaudio_result_t AudioEndpoint::configure(const EndpointDescriptor *pEndpointDescriptor)
{
- aaudio_result_t result = AAUDIO_OK;
- AudioEndpoint_validateDescriptor(pEndpointDescriptor); // FIXME remove after debugging
+ // TODO maybe remove after debugging
+ aaudio_result_t result = AudioEndpoint_validateDescriptor(pEndpointDescriptor);
+ if (result != AAUDIO_OK) {
+ ALOGD("AudioEndpoint_validateQueueDescriptor returned %d %s",
+ result, AAudio_convertResultToText(result));
+ return result;
+ }
const RingBufferDescriptor *descriptor = &pEndpointDescriptor->upMessageQueueDescriptor;
assert(descriptor->bytesPerFrame == sizeof(AAudioServiceMessage));
@@ -125,6 +152,7 @@
int64_t *writeCounterAddress = (descriptor->writeCounterAddress == nullptr)
? &mDataWriteCounter
: descriptor->writeCounterAddress;
+
mDownDataQueue = new FifoBuffer(
descriptor->bytesPerFrame,
descriptor->capacityInFrames,
@@ -144,9 +172,19 @@
aaudio_result_t AudioEndpoint::writeDataNow(const void *buffer, int32_t numFrames)
{
+ // TODO Make it easier for the AAudioStreamInternal to scale floats and write shorts
+ // TODO Similar to block adapter write through technique. Add a DataConverter.
return mDownDataQueue->write(buffer, numFrames);
}
+void AudioEndpoint::getEmptyRoomAvailable(WrappingBuffer *wrappingBuffer) {
+ mDownDataQueue->getEmptyRoomAvailable(wrappingBuffer);
+}
+
+void AudioEndpoint::advanceWriteIndex(int32_t deltaFrames) {
+ mDownDataQueue->getFifoControllerBase()->advanceWriteIndex(deltaFrames);
+}
+
void AudioEndpoint::setDownDataReadCounter(fifo_counter_t framesRead)
{
mDownDataQueue->setReadCounter(framesRead);
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index caee488..a24a705 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -19,13 +19,13 @@
#include <aaudio/AAudio.h>
-#include "AAudioServiceMessage.h"
-#include "AudioEndpointParcelable.h"
+#include "binding/AAudioServiceMessage.h"
+#include "binding/AudioEndpointParcelable.h"
#include "fifo/FifoBuffer.h"
namespace aaudio {
-#define ENDPOINT_DATA_QUEUE_SIZE_MIN 64
+#define ENDPOINT_DATA_QUEUE_SIZE_MIN 48
/**
* A sink for audio.
@@ -54,15 +54,19 @@
*/
aaudio_result_t writeDataNow(const void *buffer, int32_t numFrames);
+ void getEmptyRoomAvailable(android::WrappingBuffer *wrappingBuffer);
+
+ void advanceWriteIndex(int32_t deltaFrames);
+
/**
* Set the read index in the downData queue.
* This is needed if the reader is not updating the index itself.
*/
- void setDownDataReadCounter(fifo_counter_t framesRead);
- fifo_counter_t getDownDataReadCounter();
+ void setDownDataReadCounter(android::fifo_counter_t framesRead);
+ android::fifo_counter_t getDownDataReadCounter();
- void setDownDataWriteCounter(fifo_counter_t framesWritten);
- fifo_counter_t getDownDataWriteCounter();
+ void setDownDataWriteCounter(android::fifo_counter_t framesWritten);
+ android::fifo_counter_t getDownDataWriteCounter();
/**
* The result is not valid until after configure() is called.
@@ -80,11 +84,11 @@
int32_t getFullFramesAvailable();
private:
- FifoBuffer * mUpCommandQueue;
- FifoBuffer * mDownDataQueue;
- bool mOutputFreeRunning;
- fifo_counter_t mDataReadCounter; // only used if free-running
- fifo_counter_t mDataWriteCounter; // only used if free-running
+ android::FifoBuffer *mUpCommandQueue;
+ android::FifoBuffer *mDownDataQueue;
+ bool mOutputFreeRunning;
+ android::fifo_counter_t mDataReadCounter; // only used if free-running
+ android::fifo_counter_t mDataWriteCounter; // only used if free-running
};
} // namespace aaudio
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 54f4870..7304205 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -22,9 +22,9 @@
#include <assert.h>
#include <binder/IServiceManager.h>
-#include <utils/Mutex.h>
#include <aaudio/AAudio.h>
+#include <utils/String16.h>
#include "AudioClock.h"
#include "AudioEndpointParcelable.h"
@@ -32,6 +32,7 @@
#include "binding/AAudioStreamConfiguration.h"
#include "binding/IAAudioService.h"
#include "binding/AAudioServiceMessage.h"
+#include "fifo/FifoBuffer.h"
#include "core/AudioStreamBuilder.h"
#include "AudioStreamInternal.h"
@@ -43,47 +44,25 @@
using android::defaultServiceManager;
using android::interface_cast;
using android::Mutex;
+using android::WrappingBuffer;
using namespace aaudio;
-static android::Mutex gServiceLock;
-static sp<IAAudioService> gAAudioService;
+#define MIN_TIMEOUT_NANOS (1000 * AAUDIO_NANOS_PER_MILLISECOND)
-#define AAUDIO_SERVICE_NAME "AAudioService"
+// Wait at least this many times longer than the operation should take.
+#define MIN_TIMEOUT_OPERATIONS 4
-// Helper function to get access to the "AAudioService" service.
-// This code was modeled after frameworks/av/media/libaudioclient/AudioSystem.cpp
-static const sp<IAAudioService> getAAudioService() {
- sp<IBinder> binder;
- Mutex::Autolock _l(gServiceLock);
- if (gAAudioService == 0) {
- sp<IServiceManager> sm = defaultServiceManager();
- // Try several times to get the service.
- int retries = 4;
- do {
- binder = sm->getService(String16(AAUDIO_SERVICE_NAME)); // This will wait a while.
- if (binder != 0) {
- break;
- }
- } while (retries-- > 0);
+#define ALOG_CONDITION (mInService == false)
- if (binder != 0) {
- // TODO Add linkToDeath() like in frameworks/av/media/libaudioclient/AudioSystem.cpp
- // TODO Create a DeathRecipient that disconnects all active streams.
- gAAudioService = interface_cast<IAAudioService>(binder);
- } else {
- ALOGE("AudioStreamInternal could not get %s", AAUDIO_SERVICE_NAME);
- }
- }
- return gAAudioService;
-}
-
-AudioStreamInternal::AudioStreamInternal()
+AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService)
: AudioStream()
, mClockModel()
, mAudioEndpoint()
, mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
, mFramesPerBurst(16)
+ , mServiceInterface(serviceInterface)
+ , mInService(inService)
{
}
@@ -92,9 +71,6 @@
aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
- const sp<IAAudioService>& service = getAAudioService();
- if (service == 0) return AAUDIO_ERROR_NO_SERVICE;
-
aaudio_result_t result = AAUDIO_OK;
AAudioStreamRequest request;
AAudioStreamConfiguration configuration;
@@ -104,22 +80,31 @@
return result;
}
+ // We have to do volume scaling. So we prefer FLOAT format.
+ if (getFormat() == AAUDIO_UNSPECIFIED) {
+ setFormat(AAUDIO_FORMAT_PCM_FLOAT);
+ }
+
// Build the request to send to the server.
request.setUserId(getuid());
request.setProcessId(getpid());
+ request.setDirection(getDirection());
+
request.getConfiguration().setDeviceId(getDeviceId());
request.getConfiguration().setSampleRate(getSampleRate());
request.getConfiguration().setSamplesPerFrame(getSamplesPerFrame());
request.getConfiguration().setAudioFormat(getFormat());
+ aaudio_sharing_mode_t sharingMode = getSharingMode();
+ ALOGE("AudioStreamInternal.open(): sharingMode %d", sharingMode);
+ request.getConfiguration().setSharingMode(sharingMode);
request.getConfiguration().setBufferCapacity(builder.getBufferCapacity());
- request.dump();
- mServiceStreamHandle = service->openStream(request, configuration);
- ALOGD("AudioStreamInternal.open(): openStream returned mServiceStreamHandle = 0x%08X",
+ mServiceStreamHandle = mServiceInterface.openStream(request, configuration);
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal.open(): openStream returned mServiceStreamHandle = 0x%08X",
(unsigned int)mServiceStreamHandle);
if (mServiceStreamHandle < 0) {
result = mServiceStreamHandle;
- ALOGE("AudioStreamInternal.open(): acquireRealtimeStream aaudio_result_t = 0x%08X", result);
+ ALOGE("AudioStreamInternal.open(): openStream() returned %d", result);
} else {
result = configuration.validate();
if (result != AAUDIO_OK) {
@@ -129,17 +114,27 @@
// Save results of the open.
setSampleRate(configuration.getSampleRate());
setSamplesPerFrame(configuration.getSamplesPerFrame());
- setFormat(configuration.getAudioFormat());
+ setDeviceId(configuration.getDeviceId());
- aaudio::AudioEndpointParcelable parcelable;
- result = service->getStreamDescription(mServiceStreamHandle, parcelable);
+ // Save device format so we can do format conversion and volume scaling together.
+ mDeviceFormat = configuration.getAudioFormat();
+
+ result = mServiceInterface.getStreamDescription(mServiceStreamHandle, mEndPointParcelable);
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal.open(): getStreamDescriptor(0x%08X) returns %d",
+ mServiceStreamHandle, result);
if (result != AAUDIO_OK) {
ALOGE("AudioStreamInternal.open(): getStreamDescriptor returns %d", result);
- service->closeStream(mServiceStreamHandle);
+ mServiceInterface.closeStream(mServiceStreamHandle);
return result;
}
+
// resolve parcelable into a descriptor
- parcelable.resolve(&mEndpointDescriptor);
+ result = mEndPointParcelable.resolve(&mEndpointDescriptor);
+ if (result != AAUDIO_OK) {
+ ALOGE("AudioStreamInternal.open(): resolve() returns %d", result);
+ mServiceInterface.closeStream(mServiceStreamHandle);
+ return result;
+ }
// Configure endpoint based on descriptor.
mAudioEndpoint.configure(&mEndpointDescriptor);
@@ -151,67 +146,186 @@
mClockModel.setSampleRate(getSampleRate());
mClockModel.setFramesPerBurst(mFramesPerBurst);
+ if (getDataCallbackProc()) {
+ mCallbackFrames = builder.getFramesPerDataCallback();
+ if (mCallbackFrames > getBufferCapacity() / 2) {
+ ALOGE("AudioStreamInternal.open(): framesPerCallback too large");
+ mServiceInterface.closeStream(mServiceStreamHandle);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+
+ } else if (mCallbackFrames < 0) {
+ ALOGE("AudioStreamInternal.open(): framesPerCallback negative");
+ mServiceInterface.closeStream(mServiceStreamHandle);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+
+ }
+ if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
+ mCallbackFrames = mFramesPerBurst;
+ }
+
+ int32_t bytesPerFrame = getSamplesPerFrame()
+ * AAudioConvert_formatToSizeInBytes(getFormat());
+ int32_t callbackBufferSize = mCallbackFrames * bytesPerFrame;
+ mCallbackBuffer = new uint8_t[callbackBufferSize];
+ }
+
setState(AAUDIO_STREAM_STATE_OPEN);
}
return result;
}
aaudio_result_t AudioStreamInternal::close() {
- ALOGD("AudioStreamInternal.close(): mServiceStreamHandle = 0x%08X", mServiceStreamHandle);
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal.close(): mServiceStreamHandle = 0x%08X", mServiceStreamHandle);
if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) {
aaudio_handle_t serviceStreamHandle = mServiceStreamHandle;
mServiceStreamHandle = AAUDIO_HANDLE_INVALID;
- const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
- aaudioService->closeStream(serviceStreamHandle);
- return AAUDIO_OK;
+
+ mServiceInterface.closeStream(serviceStreamHandle);
+ delete[] mCallbackBuffer;
+ return mEndPointParcelable.close();
} else {
return AAUDIO_ERROR_INVALID_HANDLE;
}
}
+
+// Render audio in the application callback and then write the data to the stream.
+void *AudioStreamInternal::callbackLoop() {
+ aaudio_result_t result = AAUDIO_OK;
+ aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
+ AAudioStream_dataCallback appCallback = getDataCallbackProc();
+ if (appCallback == nullptr) return NULL;
+
+ // result might be a frame count
+ while (mCallbackEnabled.load() && isPlaying() && (result >= 0)) {
+ // Call application using the AAudio callback interface.
+ callbackResult = (*appCallback)(
+ (AAudioStream *) this,
+ getDataCallbackUserData(),
+ mCallbackBuffer,
+ mCallbackFrames);
+
+ if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
+ // Write audio data to stream.
+ int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);
+
+ // This is a BLOCKING WRITE!
+ result = write(mCallbackBuffer, mCallbackFrames, timeoutNanos);
+ if ((result != mCallbackFrames)) {
+ ALOGE("AudioStreamInternal(): callbackLoop: write() returned %d", result);
+ if (result >= 0) {
+ // Only wrote some of the frames requested. Must have timed out.
+ result = AAUDIO_ERROR_TIMEOUT;
+ }
+ if (getErrorCallbackProc() != nullptr) {
+ (*getErrorCallbackProc())(
+ (AAudioStream *) this,
+ getErrorCallbackUserData(),
+ result);
+ }
+ break;
+ }
+ } else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
+ ALOGD("AudioStreamInternal(): callback returned AAUDIO_CALLBACK_RESULT_STOP");
+ break;
+ }
+ }
+
+ ALOGD("AudioStreamInternal(): callbackLoop() exiting, result = %d, isPlaying() = %d",
+ result, (int) isPlaying());
+ return NULL; // TODO review
+}
+
+static void *aaudio_callback_thread_proc(void *context)
+{
+ AudioStreamInternal *stream = (AudioStreamInternal *)context;
+ //LOGD("AudioStreamInternal(): oboe_callback_thread, stream = %p", stream);
+ if (stream != NULL) {
+ return stream->callbackLoop();
+ } else {
+ return NULL;
+ }
+}
+
aaudio_result_t AudioStreamInternal::requestStart()
{
int64_t startTime;
- ALOGD("AudioStreamInternal(): start()");
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal(): start()");
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
return AAUDIO_ERROR_INVALID_STATE;
}
- const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
+
startTime = AudioClock::getNanoseconds();
mClockModel.start(startTime);
processTimestamp(0, startTime);
setState(AAUDIO_STREAM_STATE_STARTING);
- return aaudioService->startStream(mServiceStreamHandle);
+ aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle);;
+
+ if (result == AAUDIO_OK && getDataCallbackProc() != nullptr) {
+ // Launch the callback loop thread.
+ int64_t periodNanos = mCallbackFrames
+ * AAUDIO_NANOS_PER_SECOND
+ / getSampleRate();
+ mCallbackEnabled.store(true);
+ result = createThread(periodNanos, aaudio_callback_thread_proc, this);
+ }
+ return result;
}
-aaudio_result_t AudioStreamInternal::requestPause()
+int64_t AudioStreamInternal::calculateReasonableTimeout(int32_t framesPerOperation) {
+
+ // Wait for at least a second or some number of callbacks to join the thread.
+ int64_t timeoutNanoseconds = (MIN_TIMEOUT_OPERATIONS * framesPerOperation * AAUDIO_NANOS_PER_SECOND)
+ / getSampleRate();
+ if (timeoutNanoseconds < MIN_TIMEOUT_NANOS) { // arbitrary number of seconds
+ timeoutNanoseconds = MIN_TIMEOUT_NANOS;
+ }
+ return timeoutNanoseconds;
+}
+
+aaudio_result_t AudioStreamInternal::stopCallback()
+{
+ if (isDataCallbackActive()) {
+ mCallbackEnabled.store(false);
+ return joinThread(NULL, calculateReasonableTimeout(mCallbackFrames));
+ } else {
+ return AAUDIO_OK;
+ }
+}
+
+aaudio_result_t AudioStreamInternal::requestPauseInternal()
{
ALOGD("AudioStreamInternal(): pause()");
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
return AAUDIO_ERROR_INVALID_STATE;
}
- const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
+
mClockModel.stop(AudioClock::getNanoseconds());
setState(AAUDIO_STREAM_STATE_PAUSING);
- return aaudioService->pauseStream(mServiceStreamHandle);
+ return mServiceInterface.startStream(mServiceStreamHandle);
+}
+
+aaudio_result_t AudioStreamInternal::requestPause()
+{
+ aaudio_result_t result = stopCallback();
+ if (result != AAUDIO_OK) {
+ return result;
+ }
+ return requestPauseInternal();
}
aaudio_result_t AudioStreamInternal::requestFlush() {
- ALOGD("AudioStreamInternal(): flush()");
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal(): flush()");
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
return AAUDIO_ERROR_INVALID_STATE;
}
- const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
-setState(AAUDIO_STREAM_STATE_FLUSHING);
- return aaudioService->flushStream(mServiceStreamHandle);
+
+ setState(AAUDIO_STREAM_STATE_FLUSHING);
+ return mServiceInterface.flushStream(mServiceStreamHandle);
}
void AudioStreamInternal::onFlushFromServer() {
- ALOGD("AudioStreamInternal(): onFlushFromServer()");
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal(): onFlushFromServer()");
int64_t readCounter = mAudioEndpoint.getDownDataReadCounter();
int64_t writeCounter = mAudioEndpoint.getDownDataWriteCounter();
// Bump offset so caller does not see the retrograde motion in getFramesRead().
@@ -239,39 +353,38 @@
}
aaudio_result_t AudioStreamInternal::registerThread() {
- ALOGD("AudioStreamInternal(): registerThread()");
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal(): registerThread()");
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
return AAUDIO_ERROR_INVALID_STATE;
}
- const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
- return aaudioService->registerAudioThread(mServiceStreamHandle,
- gettid(),
- getPeriodNanoseconds());
+ return mServiceInterface.registerAudioThread(mServiceStreamHandle,
+ getpid(),
+ gettid(),
+ getPeriodNanoseconds());
}
aaudio_result_t AudioStreamInternal::unregisterThread() {
- ALOGD("AudioStreamInternal(): unregisterThread()");
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal(): unregisterThread()");
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
return AAUDIO_ERROR_INVALID_STATE;
}
- const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
- return aaudioService->unregisterAudioThread(mServiceStreamHandle, gettid());
+ return mServiceInterface.unregisterAudioThread(mServiceStreamHandle, getpid(), gettid());
}
-// TODO use aaudio_clockid_t all the way down to AudioClock
aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds) {
-// TODO implement using real HAL
+ // TODO implement using real HAL
int64_t time = AudioClock::getNanoseconds();
*framePosition = mClockModel.convertTimeToPosition(time);
*timeNanoseconds = time + (10 * AAUDIO_NANOS_PER_MILLISECOND); // Fake hardware delay
return AAUDIO_OK;
}
-aaudio_result_t AudioStreamInternal::updateState() {
+aaudio_result_t AudioStreamInternal::updateStateWhileWaiting() {
+ if (isDataCallbackActive()) {
+ return AAUDIO_OK; // state is getting updated by the callback thread read/write call
+ }
return processCommands();
}
@@ -281,16 +394,16 @@
static int64_t oldTime = 0;
int64_t framePosition = command.timestamp.position;
int64_t nanoTime = command.timestamp.timestamp;
- ALOGD("AudioStreamInternal() timestamp says framePosition = %08lld at nanoTime %llu",
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal() timestamp says framePosition = %08lld at nanoTime %llu",
(long long) framePosition,
(long long) nanoTime);
int64_t nanosDelta = nanoTime - oldTime;
if (nanosDelta > 0 && oldTime > 0) {
int64_t framesDelta = framePosition - oldPosition;
int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta;
- ALOGD("AudioStreamInternal() - framesDelta = %08lld", (long long) framesDelta);
- ALOGD("AudioStreamInternal() - nanosDelta = %08lld", (long long) nanosDelta);
- ALOGD("AudioStreamInternal() - measured rate = %llu", (unsigned long long) rate);
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal() - framesDelta = %08lld", (long long) framesDelta);
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal() - nanosDelta = %08lld", (long long) nanosDelta);
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal() - measured rate = %llu", (unsigned long long) rate);
}
oldPosition = framePosition;
oldTime = nanoTime;
@@ -309,29 +422,34 @@
aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *message) {
aaudio_result_t result = AAUDIO_OK;
- ALOGD("processCommands() got event %d", message->event.event);
+ ALOGD_IF(ALOG_CONDITION, "processCommands() got event %d", message->event.event);
switch (message->event.event) {
case AAUDIO_SERVICE_EVENT_STARTED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STARTED");
+ ALOGD_IF(ALOG_CONDITION, "processCommands() got AAUDIO_SERVICE_EVENT_STARTED");
setState(AAUDIO_STREAM_STATE_STARTED);
break;
case AAUDIO_SERVICE_EVENT_PAUSED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_PAUSED");
+ ALOGD_IF(ALOG_CONDITION, "processCommands() got AAUDIO_SERVICE_EVENT_PAUSED");
setState(AAUDIO_STREAM_STATE_PAUSED);
break;
case AAUDIO_SERVICE_EVENT_FLUSHED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_FLUSHED");
+ ALOGD_IF(ALOG_CONDITION, "processCommands() got AAUDIO_SERVICE_EVENT_FLUSHED");
setState(AAUDIO_STREAM_STATE_FLUSHED);
onFlushFromServer();
break;
case AAUDIO_SERVICE_EVENT_CLOSED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_CLOSED");
+ ALOGD_IF(ALOG_CONDITION, "processCommands() got AAUDIO_SERVICE_EVENT_CLOSED");
setState(AAUDIO_STREAM_STATE_CLOSED);
break;
case AAUDIO_SERVICE_EVENT_DISCONNECTED:
result = AAUDIO_ERROR_DISCONNECTED;
+ setState(AAUDIO_STREAM_STATE_DISCONNECTED);
ALOGW("WARNING - processCommands() AAUDIO_SERVICE_EVENT_DISCONNECTED");
break;
+ case AAUDIO_SERVICE_EVENT_VOLUME:
+ mVolume = message->event.dataDouble;
+ ALOGD_IF(ALOG_CONDITION, "processCommands() AAUDIO_SERVICE_EVENT_VOLUME %f", mVolume);
+ break;
default:
ALOGW("WARNING - processCommands() Unrecognized event = %d",
(int) message->event.event);
@@ -345,6 +463,7 @@
aaudio_result_t result = AAUDIO_OK;
while (result == AAUDIO_OK) {
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::processCommands() - looping, %d", result);
AAudioServiceMessage message;
if (mAudioEndpoint.readUpCommand(&message) != 1) {
break; // no command this time, no problem
@@ -373,21 +492,26 @@
int64_t timeoutNanoseconds)
{
aaudio_result_t result = AAUDIO_OK;
+ int32_t loopCount = 0;
uint8_t* source = (uint8_t*)buffer;
int64_t currentTimeNanos = AudioClock::getNanoseconds();
int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
int32_t framesLeft = numFrames;
-// ALOGD("AudioStreamInternal::write(%p, %d) at time %08llu , mState = %d ------------------",
-// buffer, numFrames, (unsigned long long) currentTimeNanos, mState);
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::write(%p, %d) at time %08llu , mState = %s",
+ // buffer, numFrames, (unsigned long long) currentTimeNanos,
+ // AAudio_convertStreamStateToText(getState()));
// Write until all the data has been written or until a timeout occurs.
while (framesLeft > 0) {
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::write() loop: framesLeft = %d, loopCount = %d =====",
+ // framesLeft, loopCount++);
// The call to writeNow() will not block. It will just write as much as it can.
int64_t wakeTimeNanos = 0;
aaudio_result_t framesWritten = writeNow(source, framesLeft,
currentTimeNanos, &wakeTimeNanos);
-// ALOGD("AudioStreamInternal::write() writeNow() framesLeft = %d --> framesWritten = %d", framesLeft, framesWritten);
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::write() loop: framesWritten = %d", framesWritten);
if (framesWritten < 0) {
+ ALOGE("AudioStreamInternal::write() loop: writeNow returned %d", framesWritten);
result = framesWritten;
break;
}
@@ -398,18 +522,19 @@
if (timeoutNanoseconds == 0) {
break; // don't block
} else if (framesLeft > 0) {
- //ALOGD("AudioStreamInternal:: original wakeTimeNanos %lld", (long long) wakeTimeNanos);
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal:: original wakeTimeNanos %lld", (long long) wakeTimeNanos);
// clip the wake time to something reasonable
if (wakeTimeNanos < currentTimeNanos) {
wakeTimeNanos = currentTimeNanos;
}
if (wakeTimeNanos > deadlineNanos) {
// If we time out, just return the framesWritten so far.
- ALOGE("AudioStreamInternal::write(): timed out after %lld nanos", (long long) timeoutNanoseconds);
+ ALOGE("AudioStreamInternal::write(): timed out after %lld nanos",
+ (long long) timeoutNanoseconds);
break;
}
- //ALOGD("AudioStreamInternal:: sleep until %lld, dur = %lld", (long long) wakeTimeNanos,
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal:: sleep until %lld, dur = %lld", (long long) wakeTimeNanos,
// (long long) (wakeTimeNanos - currentTimeNanos));
AudioClock::sleepForNanos(wakeTimeNanos - currentTimeNanos);
currentTimeNanos = AudioClock::getNanoseconds();
@@ -417,43 +542,52 @@
}
// return error or framesWritten
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::write() result = %d, framesLeft = %d, #%d",
+ // result, framesLeft, loopCount);
+ (void) loopCount;
return (result < 0) ? result : numFrames - framesLeft;
}
// Write as much data as we can without blocking.
aaudio_result_t AudioStreamInternal::writeNow(const void *buffer, int32_t numFrames,
int64_t currentNanoTime, int64_t *wakeTimePtr) {
+
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNow(%p) - enter", buffer);
{
aaudio_result_t result = processCommands();
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNow() - processCommands() returned %d", result);
if (result != AAUDIO_OK) {
return result;
}
}
if (mAudioEndpoint.isOutputFreeRunning()) {
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNow() - update read counter");
// Update data queue based on the timing model.
int64_t estimatedReadCounter = mClockModel.convertTimeToPosition(currentNanoTime);
mAudioEndpoint.setDownDataReadCounter(estimatedReadCounter);
- // If the read index passed the write index then consider it an underrun.
- if (mAudioEndpoint.getFullFramesAvailable() < 0) {
- mXRunCount++;
- }
}
// TODO else query from endpoint cuz set by actual reader, maybe
- // Write some data to the buffer.
- int32_t framesWritten = mAudioEndpoint.writeDataNow(buffer, numFrames);
- if (framesWritten > 0) {
- incrementFramesWritten(framesWritten);
+ // If the read index passed the write index then consider it an underrun.
+ if (mAudioEndpoint.getFullFramesAvailable() < 0) {
+ mXRunCount++;
}
- //ALOGD("AudioStreamInternal::writeNow() - tried to write %d frames, wrote %d",
+
+ // Write some data to the buffer.
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNow() - writeNowWithConversion(%d)", numFrames);
+ int32_t framesWritten = writeNowWithConversion(buffer, numFrames);
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNow() - tried to write %d frames, wrote %d",
// numFrames, framesWritten);
// Calculate an ideal time to wake up.
if (wakeTimePtr != nullptr && framesWritten >= 0) {
// By default wake up a few milliseconds from now. // TODO review
- int64_t wakeTime = currentNanoTime + (2 * AAUDIO_NANOS_PER_MILLISECOND);
- switch (getState()) {
+ int64_t wakeTime = currentNanoTime + (1 * AAUDIO_NANOS_PER_MILLISECOND);
+ aaudio_stream_state_t state = getState();
+ //ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNow() - wakeTime based on %s",
+ // AAudio_convertStreamStateToText(state));
+ switch (state) {
case AAUDIO_STREAM_STATE_OPEN:
case AAUDIO_STREAM_STATE_STARTING:
if (framesWritten != 0) {
@@ -478,50 +612,68 @@
*wakeTimePtr = wakeTime;
}
-// ALOGD("AudioStreamInternal::writeNow finished: now = %llu, read# = %llu, wrote# = %llu",
+// ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNow finished: now = %llu, read# = %llu, wrote# = %llu",
// (unsigned long long)currentNanoTime,
// (unsigned long long)mAudioEndpoint.getDownDataReadCounter(),
// (unsigned long long)mAudioEndpoint.getDownDataWriteCounter());
return framesWritten;
}
-aaudio_result_t AudioStreamInternal::waitForStateChange(aaudio_stream_state_t currentState,
- aaudio_stream_state_t *nextState,
- int64_t timeoutNanoseconds)
-{
- aaudio_result_t result = processCommands();
-// ALOGD("AudioStreamInternal::waitForStateChange() - processCommands() returned %d", result);
- if (result != AAUDIO_OK) {
- return result;
- }
- // TODO replace this polling with a timed sleep on a futex on the message queue
- int32_t durationNanos = 5 * AAUDIO_NANOS_PER_MILLISECOND;
- aaudio_stream_state_t state = getState();
-// ALOGD("AudioStreamInternal::waitForStateChange() - state = %d", state);
- while (state == currentState && timeoutNanoseconds > 0) {
- // TODO use futex from service message queue
- if (durationNanos > timeoutNanoseconds) {
- durationNanos = timeoutNanoseconds;
+// TODO this function needs a major cleanup.
+aaudio_result_t AudioStreamInternal::writeNowWithConversion(const void *buffer,
+ int32_t numFrames) {
+ // ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNowWithConversion(%p, %d)", buffer, numFrames);
+ WrappingBuffer wrappingBuffer;
+ mAudioEndpoint.getEmptyRoomAvailable(&wrappingBuffer);
+ uint8_t *source = (uint8_t *) buffer;
+ int32_t framesLeft = numFrames;
+
+ mAudioEndpoint.getEmptyRoomAvailable(&wrappingBuffer);
+
+ // Read data in one or two parts.
+ int partIndex = 0;
+ while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
+ int32_t framesToWrite = framesLeft;
+ int32_t framesAvailable = wrappingBuffer.numFrames[partIndex];
+ if (framesAvailable > 0) {
+ if (framesToWrite > framesAvailable) {
+ framesToWrite = framesAvailable;
+ }
+ int32_t numBytes = getBytesPerFrame();
+ // TODO handle volume scaling
+ if (getFormat() == mDeviceFormat) {
+ // Copy straight through.
+ memcpy(wrappingBuffer.data[partIndex], source, numBytes);
+ } else if (getFormat() == AAUDIO_FORMAT_PCM_FLOAT
+ && mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
+ // Data conversion.
+ AAudioConvert_floatToPcm16(
+ (const float *) source,
+ framesToWrite * getSamplesPerFrame(),
+ (int16_t *) wrappingBuffer.data[partIndex]);
+ } else {
+ // TODO handle more conversions
+ ALOGE("AudioStreamInternal::writeNowWithConversion() unsupported formats: %d, %d",
+ getFormat(), mDeviceFormat);
+ return AAUDIO_ERROR_UNEXPECTED_VALUE;
+ }
+
+ source += numBytes;
+ framesLeft -= framesToWrite;
}
- AudioClock::sleepForNanos(durationNanos);
- timeoutNanoseconds -= durationNanos;
-
- result = processCommands();
- if (result != AAUDIO_OK) {
- return result;
- }
-
- state = getState();
-// ALOGD("AudioStreamInternal::waitForStateChange() - state = %d", state);
+ partIndex++;
}
- if (nextState != nullptr) {
- *nextState = state;
+ int32_t framesWritten = numFrames - framesLeft;
+ mAudioEndpoint.advanceWriteIndex(framesWritten);
+
+ if (framesWritten > 0) {
+ incrementFramesWritten(framesWritten);
}
- return (state == currentState) ? AAUDIO_ERROR_TIMEOUT : AAUDIO_OK;
+ // ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNowWithConversion() returns %d", framesWritten);
+ return framesWritten;
}
-
void AudioStreamInternal::processTimestamp(uint64_t position, int64_t time) {
mClockModel.processTimestamp( position, time);
}
@@ -562,7 +714,7 @@
} else {
mLastFramesRead = framesRead;
}
- ALOGD("AudioStreamInternal::getFramesRead() returns %lld", (long long)framesRead);
+ ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::getFramesRead() returns %lld", (long long)framesRead);
return framesRead;
}
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 6f3a7ac..1aa3b0f 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -26,6 +26,8 @@
#include "client/AudioEndpoint.h"
#include "core/AudioStream.h"
+#include "binding/AAudioServiceInterface.h"
+
using android::sp;
using android::IAAudioService;
@@ -35,61 +37,66 @@
class AudioStreamInternal : public AudioStream {
public:
- AudioStreamInternal();
+ AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService = false);
virtual ~AudioStreamInternal();
// =========== Begin ABSTRACT methods ===========================
- virtual aaudio_result_t requestStart() override;
+ aaudio_result_t requestStart() override;
- virtual aaudio_result_t requestPause() override;
+ aaudio_result_t requestPause() override;
- virtual aaudio_result_t requestFlush() override;
+ aaudio_result_t requestFlush() override;
- virtual aaudio_result_t requestStop() override;
+ aaudio_result_t requestStop() override;
// TODO use aaudio_clockid_t all the way down to AudioClock
- virtual aaudio_result_t getTimestamp(clockid_t clockId,
+ aaudio_result_t getTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds) override;
- virtual aaudio_result_t updateState() override;
+
+ virtual aaudio_result_t updateStateWhileWaiting() override;
+
// =========== End ABSTRACT methods ===========================
- virtual aaudio_result_t open(const AudioStreamBuilder &builder) override;
+ aaudio_result_t open(const AudioStreamBuilder &builder) override;
- virtual aaudio_result_t close() override;
+ aaudio_result_t close() override;
- virtual aaudio_result_t write(const void *buffer,
+ aaudio_result_t write(const void *buffer,
int32_t numFrames,
int64_t timeoutNanoseconds) override;
- virtual aaudio_result_t waitForStateChange(aaudio_stream_state_t currentState,
- aaudio_stream_state_t *nextState,
- int64_t timeoutNanoseconds) override;
+ aaudio_result_t setBufferSize(int32_t requestedFrames) override;
- virtual aaudio_result_t setBufferSize(int32_t requestedFrames) override;
+ int32_t getBufferSize() const override;
- virtual int32_t getBufferSize() const override;
+ int32_t getBufferCapacity() const override;
- virtual int32_t getBufferCapacity() const override;
+ int32_t getFramesPerBurst() const override;
- virtual int32_t getFramesPerBurst() const override;
+ int64_t getFramesRead() override;
- virtual int64_t getFramesRead() override;
-
- virtual int32_t getXRunCount() const override {
+ int32_t getXRunCount() const override {
return mXRunCount;
}
- virtual aaudio_result_t registerThread() override;
+ aaudio_result_t registerThread() override;
- virtual aaudio_result_t unregisterThread() override;
+ aaudio_result_t unregisterThread() override;
+
+ // Called internally from 'C'
+ void *callbackLoop();
protected:
aaudio_result_t processCommands();
+ aaudio_result_t requestPauseInternal();
+
+ aaudio_result_t stopCallback();
+
/**
* Low level write that will not block. It will just write as much as it can.
*
@@ -97,10 +104,10 @@
*
* @return the number of frames written or a negative error code.
*/
- virtual aaudio_result_t writeNow(const void *buffer,
- int32_t numFrames,
- int64_t currentTimeNanos,
- int64_t *wakeTimePtr);
+ aaudio_result_t writeNow(const void *buffer,
+ int32_t numFrames,
+ int64_t currentTimeNanos,
+ int64_t *wakeTimePtr);
void onFlushFromServer();
@@ -108,18 +115,45 @@
aaudio_result_t onTimestampFromServer(AAudioServiceMessage *message);
-private:
- IsochronousClockModel mClockModel;
- AudioEndpoint mAudioEndpoint;
- aaudio_handle_t mServiceStreamHandle;
- EndpointDescriptor mEndpointDescriptor;
- // Offset from underlying frame position.
- int64_t mFramesOffsetFromService = 0;
- int64_t mLastFramesRead = 0;
- int32_t mFramesPerBurst;
- int32_t mXRunCount = 0;
+ // Calculate timeout for an operation involving framesPerOperation.
+ int64_t calculateReasonableTimeout(int32_t framesPerOperation);
+private:
+ /*
+ * Asynchronous write with data conversion.
+ * @param buffer
+ * @param numFrames
+ * @return fdrames written or negative error
+ */
+ aaudio_result_t writeNowWithConversion(const void *buffer,
+ int32_t numFrames);
void processTimestamp(uint64_t position, int64_t time);
+
+ // Adjust timing model based on timestamp from service.
+
+ IsochronousClockModel mClockModel; // timing model for chasing the HAL
+ AudioEndpoint mAudioEndpoint; // sink for writes
+ aaudio_handle_t mServiceStreamHandle; // opaque handle returned from service
+
+ AudioEndpointParcelable mEndPointParcelable; // description of the buffers filled by service
+ EndpointDescriptor mEndpointDescriptor; // buffer description with resolved addresses
+
+ aaudio_audio_format_t mDeviceFormat = AAUDIO_FORMAT_UNSPECIFIED;
+
+ uint8_t *mCallbackBuffer = nullptr;
+ int32_t mCallbackFrames = 0;
+
+ // Offset from underlying frame position.
+ int64_t mFramesOffsetFromService = 0; // offset for timestamps
+ int64_t mLastFramesRead = 0; // used to prevent retrograde motion
+ int32_t mFramesPerBurst; // frames per HAL transfer
+ int32_t mXRunCount = 0; // how many underrun events?
+ float mVolume = 1.0; // volume that the server told us to use
+
+ AAudioServiceInterface &mServiceInterface; // abstract interface to the service
+
+ // The service uses this for SHARED mode.
+ bool mInService = false; // Are running in the client or the service?
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/client/IsochronousClockModel.cpp b/media/libaaudio/src/client/IsochronousClockModel.cpp
index 4c8aabc..c278c8b 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.cpp
+++ b/media/libaaudio/src/client/IsochronousClockModel.cpp
@@ -19,7 +19,6 @@
#include <utils/Log.h>
#include <stdint.h>
-#include <aaudio/AAudioDefinitions.h>
#include "utility/AudioClock.h"
#include "IsochronousClockModel.h"
diff --git a/media/libaaudio/src/client/IsochronousClockModel.h b/media/libaaudio/src/client/IsochronousClockModel.h
index 524c286..205c341 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.h
+++ b/media/libaaudio/src/client/IsochronousClockModel.h
@@ -14,11 +14,10 @@
* limitations under the License.
*/
-#ifndef AAUDIO_ISOCHRONOUSCLOCKMODEL_H
-#define AAUDIO_ISOCHRONOUSCLOCKMODEL_H
+#ifndef AAUDIO_ISOCHRONOUS_CLOCK_MODEL_H
+#define AAUDIO_ISOCHRONOUS_CLOCK_MODEL_H
#include <stdint.h>
-#include <aaudio/AAudio.h>
namespace aaudio {
@@ -107,4 +106,4 @@
} /* namespace aaudio */
-#endif //AAUDIO_ISOCHRONOUSCLOCKMODEL_H
+#endif //AAUDIO_ISOCHRONOUS_CLOCK_MODEL_H
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 52bad70..f687e7d 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -49,10 +49,13 @@
AAUDIO_API const char * AAudio_convertResultToText(aaudio_result_t returnCode) {
switch (returnCode) {
AAUDIO_CASE_ENUM(AAUDIO_OK);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_DISCONNECTED);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_ILLEGAL_ARGUMENT);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_INCOMPATIBLE);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_INTERNAL);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_STATE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNEXPECTED_STATE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNEXPECTED_VALUE);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_HANDLE);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_QUERY);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNIMPLEMENTED);
@@ -62,9 +65,10 @@
AAUDIO_CASE_ENUM(AAUDIO_ERROR_NULL);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_TIMEOUT);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_WOULD_BLOCK);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_ORDER);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_FORMAT);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_OUT_OF_RANGE);
AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_SERVICE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_RATE);
}
return "Unrecognized AAudio error.";
}
@@ -82,6 +86,7 @@
AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHED);
AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPING);
AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_DISCONNECTED);
AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSING);
AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSED);
}
@@ -102,7 +107,6 @@
AAUDIO_API aaudio_result_t AAudio_createStreamBuilder(AAudioStreamBuilder** builder)
{
- ALOGD("AAudio_createStreamBuilder(): check sHandleTracker.isInitialized ()");
AudioStreamBuilder *audioStreamBuilder = new AudioStreamBuilder();
if (audioStreamBuilder == nullptr) {
return AAUDIO_ERROR_NO_MEMORY;
@@ -114,53 +118,79 @@
AAUDIO_API void AAudioStreamBuilder_setDeviceId(AAudioStreamBuilder* builder,
int32_t deviceId)
{
- AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);;
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
streamBuilder->setDeviceId(deviceId);
}
AAUDIO_API void AAudioStreamBuilder_setSampleRate(AAudioStreamBuilder* builder,
int32_t sampleRate)
{
- AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);;
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
streamBuilder->setSampleRate(sampleRate);
}
AAUDIO_API void AAudioStreamBuilder_setSamplesPerFrame(AAudioStreamBuilder* builder,
int32_t samplesPerFrame)
{
- AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);;
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
streamBuilder->setSamplesPerFrame(samplesPerFrame);
}
AAUDIO_API void AAudioStreamBuilder_setDirection(AAudioStreamBuilder* builder,
aaudio_direction_t direction)
{
- AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);;
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
streamBuilder->setDirection(direction);
}
-
AAUDIO_API void AAudioStreamBuilder_setFormat(AAudioStreamBuilder* builder,
aaudio_audio_format_t format)
{
- AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);;
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
streamBuilder->setFormat(format);
}
AAUDIO_API void AAudioStreamBuilder_setSharingMode(AAudioStreamBuilder* builder,
aaudio_sharing_mode_t sharingMode)
{
- AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);;
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
streamBuilder->setSharingMode(sharingMode);
}
AAUDIO_API void AAudioStreamBuilder_setBufferCapacityInFrames(AAudioStreamBuilder* builder,
int32_t frames)
{
- AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);;
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
streamBuilder->setBufferCapacity(frames);
}
+AAUDIO_API void AAudioStreamBuilder_setDataCallback(AAudioStreamBuilder* builder,
+ AAudioStream_dataCallback callback,
+ void *userData)
+{
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+ ALOGD("AAudioStreamBuilder_setCallback(): userData = %p", userData);
+ streamBuilder->setDataCallbackProc(callback);
+ streamBuilder->setDataCallbackUserData(userData);
+}
+AAUDIO_API void AAudioStreamBuilder_setErrorCallback(AAudioStreamBuilder* builder,
+ AAudioStream_errorCallback callback,
+ void *userData)
+{
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+ ALOGD("AAudioStreamBuilder_setCallback(): userData = %p", userData);
+ streamBuilder->setErrorCallbackProc(callback);
+ streamBuilder->setErrorCallbackUserData(userData);
+}
+
+AAUDIO_API void AAudioStreamBuilder_setFramesPerDataCallback(AAudioStreamBuilder* builder,
+ int32_t frames)
+{
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+ ALOGD("%s: frames = %d", __func__, frames);
+ streamBuilder->setFramesPerDataCallback(frames);
+}
+
static aaudio_result_t AAudioInternal_openStream(AudioStreamBuilder *streamBuilder,
AAudioStream** streamPtr)
{
@@ -276,6 +306,13 @@
if (buffer == nullptr) {
return AAUDIO_ERROR_NULL;
}
+
+ // Don't allow writes when playing with a callback.
+ if (audioStream->getDataCallbackProc() != nullptr && audioStream->isPlaying()) {
+ ALOGE("Cannot write to a callback stream when running.");
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+
if (numFrames < 0) {
return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
} else if (numFrames == 0) {
@@ -297,6 +334,9 @@
aaudio_audio_thread_proc_t threadProc, void *arg)
{
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+ if (audioStream->getDataCallbackProc() != nullptr) {
+ return AAUDIO_ERROR_INCOMPATIBLE;
+ }
return audioStream->createThread(periodNanoseconds, threadProc, arg);
}
@@ -361,6 +401,12 @@
return audioStream->getFramesPerBurst();
}
+AAUDIO_API int32_t AAudioStream_getFramesPerDataCallback(AAudioStream* stream)
+{
+ AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+ return audioStream->getFramesPerDataCallback();
+}
+
AAUDIO_API int32_t AAudioStream_getBufferCapacityInFrames(AAudioStream* stream)
{
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index b054d94..7c0b5ae 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -28,7 +28,9 @@
using namespace aaudio;
-AudioStream::AudioStream() {
+AudioStream::AudioStream()
+ : mCallbackEnabled(false)
+{
// mThread is a pthread_t of unknown size so we need memset.
memset(&mThread, 0, sizeof(mThread));
setPeriodNanoseconds(0);
@@ -36,13 +38,31 @@
aaudio_result_t AudioStream::open(const AudioStreamBuilder& builder)
{
- // TODO validate parameters.
+
// Copy parameters from the Builder because the Builder may be deleted after this call.
mSamplesPerFrame = builder.getSamplesPerFrame();
mSampleRate = builder.getSampleRate();
mDeviceId = builder.getDeviceId();
mFormat = builder.getFormat();
+ mDirection = builder.getDirection();
mSharingMode = builder.getSharingMode();
+
+ // callbacks
+ mFramesPerDataCallback = builder.getFramesPerDataCallback();
+ mDataCallbackProc = builder.getDataCallbackProc();
+ mErrorCallbackProc = builder.getErrorCallbackProc();
+ mDataCallbackUserData = builder.getDataCallbackUserData();
+
+ // TODO validate more parameters.
+ if (mErrorCallbackProc != nullptr && mDataCallbackProc == nullptr) {
+ ALOGE("AudioStream::open(): disconnect callback cannot be used without a data callback.");
+ return AAUDIO_ERROR_UNEXPECTED_VALUE;
+ }
+ if (mDirection != AAUDIO_DIRECTION_INPUT && mDirection != AAUDIO_DIRECTION_OUTPUT) {
+ ALOGE("AudioStream::open(): illegal direction %d", mDirection);
+ return AAUDIO_ERROR_UNEXPECTED_VALUE;
+ }
+
return AAUDIO_OK;
}
@@ -75,8 +95,13 @@
aaudio_stream_state_t *nextState,
int64_t timeoutNanoseconds)
{
+ aaudio_result_t result = updateStateWhileWaiting();
+ if (result != AAUDIO_OK) {
+ return result;
+ }
+
// TODO replace this when similar functionality added to AudioTrack.cpp
- int64_t durationNanos = 20 * AAUDIO_NANOS_PER_MILLISECOND;
+ int64_t durationNanos = 20 * AAUDIO_NANOS_PER_MILLISECOND; // arbitrary
aaudio_stream_state_t state = getState();
while (state == currentState && timeoutNanoseconds > 0) {
if (durationNanos > timeoutNanoseconds) {
@@ -85,7 +110,7 @@
AudioClock::sleepForNanos(durationNanos);
timeoutNanoseconds -= durationNanos;
- aaudio_result_t result = updateState();
+ aaudio_result_t result = updateStateWhileWaiting();
if (result != AAUDIO_OK) {
return result;
}
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 6ac8554..da71906 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -18,12 +18,12 @@
#define AAUDIO_AUDIOSTREAM_H
#include <atomic>
+#include <mutex>
#include <stdint.h>
-#include <aaudio/AAudioDefinitions.h>
#include <aaudio/AAudio.h>
-#include "AAudioUtilities.h"
-#include "MonotonicCounter.h"
+#include "utility/AAudioUtilities.h"
+#include "utility/MonotonicCounter.h"
namespace aaudio {
@@ -55,14 +55,18 @@
int64_t *timeNanoseconds) = 0;
- virtual aaudio_result_t updateState() = 0;
+ /**
+ * Update state while in the middle of waitForStateChange()
+ * @return
+ */
+ virtual aaudio_result_t updateStateWhileWaiting() = 0;
// =========== End ABSTRACT methods ===========================
virtual aaudio_result_t waitForStateChange(aaudio_stream_state_t currentState,
- aaudio_stream_state_t *nextState,
- int64_t timeoutNanoseconds);
+ aaudio_stream_state_t *nextState,
+ int64_t timeoutNanoseconds);
/**
* Open the stream using the parameters in the builder.
@@ -152,10 +156,16 @@
return mDirection;
}
+ /**
+ * This is only valid after setSamplesPerFrame() and setFormat() have been called.
+ */
int32_t getBytesPerFrame() const {
return mSamplesPerFrame * getBytesPerSample();
}
+ /**
+ * This is only valid after setFormat() has been called.
+ */
int32_t getBytesPerSample() const {
return AAudioConvert_formatToSizeInBytes(mFormat);
}
@@ -168,6 +178,27 @@
return mFramesRead.get();
}
+ AAudioStream_dataCallback getDataCallbackProc() const {
+ return mDataCallbackProc;
+ }
+ AAudioStream_errorCallback getErrorCallbackProc() const {
+ return mErrorCallbackProc;
+ }
+
+ void *getDataCallbackUserData() const {
+ return mDataCallbackUserData;
+ }
+ void *getErrorCallbackUserData() const {
+ return mErrorCallbackUserData;
+ }
+
+ int32_t getFramesPerDataCallback() const {
+ return mFramesPerDataCallback;
+ }
+
+ bool isDataCallbackActive() {
+ return (mDataCallbackProc != nullptr) && isPlaying();
+ }
// ============== I/O ===========================
// A Stream will only implement read() or write() depending on its direction.
@@ -235,7 +266,13 @@
mState = state;
}
+ void setDeviceId(int32_t deviceId) {
+ mDeviceId = deviceId;
+ }
+ std::mutex mStreamMutex;
+
+ std::atomic<bool> mCallbackEnabled;
protected:
MonotonicCounter mFramesWritten;
@@ -259,6 +296,15 @@
aaudio_direction_t mDirection = AAUDIO_DIRECTION_OUTPUT;
aaudio_stream_state_t mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
+ // callback ----------------------------------
+
+ AAudioStream_dataCallback mDataCallbackProc = nullptr; // external callback functions
+ void *mDataCallbackUserData = nullptr;
+ int32_t mFramesPerDataCallback = AAUDIO_UNSPECIFIED; // frames
+
+ AAudioStream_errorCallback mErrorCallbackProc = nullptr;
+ void *mErrorCallbackUserData = nullptr;
+
// background thread ----------------------------------
bool mHasThread = false;
pthread_t mThread; // initialized in constructor
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 5a54e62..c0b59bb 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -24,12 +24,18 @@
#include <aaudio/AAudioDefinitions.h>
#include <aaudio/AAudio.h>
+#include "binding/AAudioBinderClient.h"
#include "client/AudioStreamInternal.h"
#include "core/AudioStream.h"
#include "core/AudioStreamBuilder.h"
#include "legacy/AudioStreamRecord.h"
#include "legacy/AudioStreamTrack.h"
+// Enable a mixer in AAudio service that will mix stream to an ALSA MMAP buffer.
+#define MMAP_SHARED_ENABLED 0
+// Enable AAUDIO_SHARING_MODE_EXCLUSIVE that uses an ALSA MMAP buffer.
+#define MMAP_EXCLUSIVE_ENABLED 1
+
using namespace aaudio;
/*
@@ -43,8 +49,11 @@
aaudio_result_t AudioStreamBuilder::build(AudioStream** streamPtr) {
AudioStream* audioStream = nullptr;
+ AAudioBinderClient *aaudioClient = nullptr;
const aaudio_sharing_mode_t sharingMode = getSharingMode();
+ ALOGD("AudioStreamBuilder.build() sharingMode = %d", sharingMode);
switch (getDirection()) {
+
case AAUDIO_DIRECTION_INPUT:
switch (sharingMode) {
case AAUDIO_SHARING_MODE_SHARED:
@@ -56,26 +65,37 @@
break;
}
break;
+
case AAUDIO_DIRECTION_OUTPUT:
switch (sharingMode) {
case AAUDIO_SHARING_MODE_SHARED:
+#if MMAP_SHARED_ENABLED
+ aaudioClient = new AAudioBinderClient();
+ audioStream = new(std::nothrow) AudioStreamInternal(*aaudioClient, false);
+#else
audioStream = new(std::nothrow) AudioStreamTrack();
+#endif
break;
+#if MMAP_EXCLUSIVE_ENABLED
case AAUDIO_SHARING_MODE_EXCLUSIVE:
- audioStream = new(std::nothrow) AudioStreamInternal();
+ aaudioClient = new AAudioBinderClient();
+ audioStream = new(std::nothrow) AudioStreamInternal(*aaudioClient, false);
break;
+#endif
default:
ALOGE("AudioStreamBuilder(): bad sharing mode = %d", sharingMode);
return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
break;
}
break;
+
default:
ALOGE("AudioStreamBuilder(): bad direction = %d", getDirection());
return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
break;
}
if (audioStream == nullptr) {
+ delete aaudioClient;
return AAUDIO_ERROR_NO_MEMORY;
}
ALOGD("AudioStreamBuilder(): created audioStream = %p", audioStream);
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.h b/media/libaaudio/src/core/AudioStreamBuilder.h
index 7b5f35c..93ca7f5 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.h
+++ b/media/libaaudio/src/core/AudioStreamBuilder.h
@@ -14,8 +14,8 @@
* limitations under the License.
*/
-#ifndef AAUDIO_AUDIOSTREAMBUILDER_H
-#define AAUDIO_AUDIOSTREAMBUILDER_H
+#ifndef AAUDIO_AUDIO_STREAM_BUILDER_H
+#define AAUDIO_AUDIO_STREAM_BUILDER_H
#include <stdint.h>
@@ -101,6 +101,52 @@
return this;
}
+ AAudioStream_dataCallback getDataCallbackProc() const {
+ return mDataCallbackProc;
+ }
+
+ AudioStreamBuilder* setDataCallbackProc(AAudioStream_dataCallback proc) {
+ mDataCallbackProc = proc;
+ return this;
+ }
+
+
+ void *getDataCallbackUserData() const {
+ return mDataCallbackUserData;
+ }
+
+ AudioStreamBuilder* setDataCallbackUserData(void *userData) {
+ mDataCallbackUserData = userData;
+ return this;
+ }
+
+ AAudioStream_errorCallback getErrorCallbackProc() const {
+ return mErrorCallbackProc;
+ }
+
+ AudioStreamBuilder* setErrorCallbackProc(AAudioStream_errorCallback proc) {
+ mErrorCallbackProc = proc;
+ return this;
+ }
+
+ AudioStreamBuilder* setErrorCallbackUserData(void *userData) {
+ mErrorCallbackUserData = userData;
+ return this;
+ }
+
+ void *getErrorCallbackUserData() const {
+ return mErrorCallbackUserData;
+ }
+
+ int32_t getFramesPerDataCallback() const {
+ return mFramesPerDataCallback;
+ }
+
+ AudioStreamBuilder* setFramesPerDataCallback(int32_t sizeInFrames) {
+ mFramesPerDataCallback = sizeInFrames;
+ return this;
+ }
+
aaudio_result_t build(AudioStream **streamPtr);
private:
@@ -111,8 +157,15 @@
aaudio_audio_format_t mFormat = AAUDIO_FORMAT_UNSPECIFIED;
aaudio_direction_t mDirection = AAUDIO_DIRECTION_OUTPUT;
int32_t mBufferCapacity = AAUDIO_UNSPECIFIED;
+
+ AAudioStream_dataCallback mDataCallbackProc = nullptr; // external callback functions
+ void *mDataCallbackUserData = nullptr;
+ int32_t mFramesPerDataCallback = AAUDIO_UNSPECIFIED; // frames
+
+ AAudioStream_errorCallback mErrorCallbackProc = nullptr;
+ void *mErrorCallbackUserData = nullptr;
};
} /* namespace aaudio */
-#endif /* AAUDIO_AUDIOSTREAMBUILDER_H */
+#endif //AAUDIO_AUDIO_STREAM_BUILDER_H
diff --git a/media/libaaudio/src/fifo/FifoBuffer.cpp b/media/libaaudio/src/fifo/FifoBuffer.cpp
index c5489f1..857780c 100644
--- a/media/libaaudio/src/fifo/FifoBuffer.cpp
+++ b/media/libaaudio/src/fifo/FifoBuffer.cpp
@@ -17,6 +17,7 @@
#include <cstring>
#include <unistd.h>
+
#define LOG_TAG "FifoBuffer"
//#define LOG_NDEBUG 0
#include <utils/Log.h>
@@ -26,6 +27,8 @@
#include "FifoControllerIndirect.h"
#include "FifoBuffer.h"
+using namespace android; // TODO just import names needed
+
FifoBuffer::FifoBuffer(int32_t bytesPerFrame, fifo_frames_t capacityInFrames)
: mFrameCapacity(capacityInFrames)
, mBytesPerFrame(bytesPerFrame)
@@ -79,80 +82,102 @@
return frames * mBytesPerFrame;
}
-fifo_frames_t FifoBuffer::read(void *buffer, fifo_frames_t numFrames) {
- size_t numBytes;
- fifo_frames_t framesAvailable = mFifo->getFullFramesAvailable();
- fifo_frames_t framesToRead = numFrames;
- // Is there enough data in the FIFO
- if (framesToRead > framesAvailable) {
- framesToRead = framesAvailable;
- }
- if (framesToRead == 0) {
- return 0;
- }
+void FifoBuffer::fillWrappingBuffer(WrappingBuffer *wrappingBuffer,
+ int32_t framesAvailable,
+ int32_t startIndex) {
+ wrappingBuffer->data[1] = nullptr;
+ wrappingBuffer->numFrames[1] = 0;
+ if (framesAvailable > 0) {
- fifo_frames_t readIndex = mFifo->getReadIndex();
- uint8_t *destination = (uint8_t *) buffer;
- uint8_t *source = &mStorage[convertFramesToBytes(readIndex)];
- if ((readIndex + framesToRead) > mFrameCapacity) {
- // read in two parts, first part here
- fifo_frames_t frames1 = mFrameCapacity - readIndex;
- int32_t numBytes = convertFramesToBytes(frames1);
- memcpy(destination, source, numBytes);
- destination += numBytes;
- // read second part
- source = &mStorage[0];
- fifo_frames_t frames2 = framesToRead - frames1;
- numBytes = convertFramesToBytes(frames2);
- memcpy(destination, source, numBytes);
+ uint8_t *source = &mStorage[convertFramesToBytes(startIndex)];
+ // Does the available data cross the end of the FIFO?
+ if ((startIndex + framesAvailable) > mFrameCapacity) {
+ wrappingBuffer->data[0] = source;
+ wrappingBuffer->numFrames[0] = mFrameCapacity - startIndex;
+ wrappingBuffer->data[1] = &mStorage[0];
+ wrappingBuffer->numFrames[1] = mFrameCapacity - startIndex;
+
+ } else {
+ wrappingBuffer->data[0] = source;
+ wrappingBuffer->numFrames[0] = framesAvailable;
+ }
} else {
- // just read in one shot
- numBytes = convertFramesToBytes(framesToRead);
- memcpy(destination, source, numBytes);
+ wrappingBuffer->data[0] = nullptr;
+ wrappingBuffer->numFrames[0] = 0;
}
- mFifo->advanceReadIndex(framesToRead);
- return framesToRead;
}
-fifo_frames_t FifoBuffer::write(const void *buffer, fifo_frames_t framesToWrite) {
+void FifoBuffer::getFullDataAvailable(WrappingBuffer *wrappingBuffer) {
+ fifo_frames_t framesAvailable = mFifo->getFullFramesAvailable();
+ fifo_frames_t startIndex = mFifo->getReadIndex();
+ fillWrappingBuffer(wrappingBuffer, framesAvailable, startIndex);
+}
+
+void FifoBuffer::getEmptyRoomAvailable(WrappingBuffer *wrappingBuffer) {
fifo_frames_t framesAvailable = mFifo->getEmptyFramesAvailable();
-// ALOGD("FifoBuffer::write() framesToWrite = %d, framesAvailable = %d",
-// framesToWrite, framesAvailable);
- if (framesToWrite > framesAvailable) {
- framesToWrite = framesAvailable;
- }
- if (framesToWrite <= 0) {
- return 0;
- }
+ fifo_frames_t startIndex = mFifo->getWriteIndex();
+ fillWrappingBuffer(wrappingBuffer, framesAvailable, startIndex);
+}
- size_t numBytes;
- fifo_frames_t writeIndex = mFifo->getWriteIndex();
- int byteIndex = convertFramesToBytes(writeIndex);
- const uint8_t *source = (const uint8_t *) buffer;
- uint8_t *destination = &mStorage[byteIndex];
- if ((writeIndex + framesToWrite) > mFrameCapacity) {
- // write in two parts, first part here
- fifo_frames_t frames1 = mFrameCapacity - writeIndex;
- numBytes = convertFramesToBytes(frames1);
- memcpy(destination, source, numBytes);
-// ALOGD("FifoBuffer::write(%p to %p, numBytes = %d", source, destination, numBytes);
- // read second part
- source += convertFramesToBytes(frames1);
- destination = &mStorage[0];
- fifo_frames_t framesLeft = framesToWrite - frames1;
- numBytes = convertFramesToBytes(framesLeft);
-// ALOGD("FifoBuffer::write(%p to %p, numBytes = %d", source, destination, numBytes);
- memcpy(destination, source, numBytes);
- } else {
- // just write in one shot
- numBytes = convertFramesToBytes(framesToWrite);
-// ALOGD("FifoBuffer::write(%p to %p, numBytes = %d", source, destination, numBytes);
- memcpy(destination, source, numBytes);
- }
- mFifo->advanceWriteIndex(framesToWrite);
+fifo_frames_t FifoBuffer::read(void *buffer, fifo_frames_t numFrames) {
+ WrappingBuffer wrappingBuffer;
+ uint8_t *destination = (uint8_t *) buffer;
+ fifo_frames_t framesLeft = numFrames;
- return framesToWrite;
+ getFullDataAvailable(&wrappingBuffer);
+
+ // Read data in one or two parts.
+ int partIndex = 0;
+ while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
+ fifo_frames_t framesToRead = framesLeft;
+ fifo_frames_t framesAvailable = wrappingBuffer.numFrames[partIndex];
+ //ALOGD("FifoProcessor::read() framesAvailable = %d, partIndex = %d",
+ // framesAvailable, partIndex);
+ if (framesAvailable > 0) {
+ if (framesToRead > framesAvailable) {
+ framesToRead = framesAvailable;
+ }
+ int32_t numBytes = convertFramesToBytes(framesToRead);
+ memcpy(destination, wrappingBuffer.data[partIndex], numBytes);
+
+ destination += numBytes;
+ framesLeft -= framesToRead;
+ }
+ partIndex++;
+ }
+ fifo_frames_t framesRead = numFrames - framesLeft;
+ mFifo->advanceReadIndex(framesRead);
+ return framesRead;
+}
+
+fifo_frames_t FifoBuffer::write(const void *buffer, fifo_frames_t numFrames) {
+ WrappingBuffer wrappingBuffer;
+ uint8_t *source = (uint8_t *) buffer;
+ fifo_frames_t framesLeft = numFrames;
+
+ getEmptyRoomAvailable(&wrappingBuffer);
+
+ // Read data in one or two parts.
+ int partIndex = 0;
+ while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
+ fifo_frames_t framesToWrite = framesLeft;
+ fifo_frames_t framesAvailable = wrappingBuffer.numFrames[partIndex];
+ if (framesAvailable > 0) {
+ if (framesToWrite > framesAvailable) {
+ framesToWrite = framesAvailable;
+ }
+ int32_t numBytes = convertFramesToBytes(framesToWrite);
+ memcpy(wrappingBuffer.data[partIndex], source, numBytes);
+
+ source += numBytes;
+ framesLeft -= framesToWrite;
+ }
+ partIndex++;
+ }
+ fifo_frames_t framesWritten = numFrames - framesLeft;
+ mFifo->advanceWriteIndex(framesWritten);
+ return framesWritten;
}
fifo_frames_t FifoBuffer::readNow(void *buffer, fifo_frames_t numFrames) {
diff --git a/media/libaaudio/src/fifo/FifoBuffer.h b/media/libaaudio/src/fifo/FifoBuffer.h
index faa9ae2..2b262a1 100644
--- a/media/libaaudio/src/fifo/FifoBuffer.h
+++ b/media/libaaudio/src/fifo/FifoBuffer.h
@@ -21,15 +21,29 @@
#include "FifoControllerBase.h"
+namespace android {
+
+/**
+ * Structure that represents a region in a circular buffer that might be at the
+ * end of the array and split in two.
+ */
+struct WrappingBuffer {
+ enum {
+ SIZE = 2
+ };
+ void *data[SIZE];
+ int32_t numFrames[SIZE];
+};
+
class FifoBuffer {
public:
FifoBuffer(int32_t bytesPerFrame, fifo_frames_t capacityInFrames);
- FifoBuffer(int32_t bytesPerFrame,
- fifo_frames_t capacityInFrames,
- fifo_counter_t * readCounterAddress,
- fifo_counter_t * writeCounterAddress,
- void * dataStorageAddress);
+ FifoBuffer(int32_t bytesPerFrame,
+ fifo_frames_t capacityInFrames,
+ fifo_counter_t *readCounterAddress,
+ fifo_counter_t *writeCounterAddress,
+ void *dataStorageAddress);
~FifoBuffer();
@@ -40,10 +54,33 @@
fifo_frames_t write(const void *source, fifo_frames_t framesToWrite);
fifo_frames_t getThreshold();
+
void setThreshold(fifo_frames_t threshold);
fifo_frames_t getBufferCapacityInFrames();
+ /**
+ * Return pointer to available full frames in data1 and set size in numFrames1.
+ * if the data is split across the end of the FIFO then set data2 and numFrames2.
+ * Other wise set them to null
+ * @param wrappingBuffer
+ */
+ void getFullDataAvailable(WrappingBuffer *wrappingBuffer);
+
+ /**
+ * Return pointer to available empty frames in data1 and set size in numFrames1.
+ * if the room is split across the end of the FIFO then set data2 and numFrames2.
+ * Other wise set them to null
+ * @param wrappingBuffer
+ */
+ void getEmptyRoomAvailable(WrappingBuffer *wrappingBuffer);
+
+ /**
+ * Copy data from the FIFO into the buffer.
+ * @param buffer
+ * @param numFrames
+ * @return
+ */
fifo_frames_t readNow(void *buffer, fifo_frames_t numFrames);
int64_t getNextReadTime(int32_t frameRate);
@@ -73,15 +110,21 @@
}
private:
+
+ void fillWrappingBuffer(WrappingBuffer *wrappingBuffer,
+ int32_t framesAvailable, int32_t startIndex);
+
const fifo_frames_t mFrameCapacity;
- const int32_t mBytesPerFrame;
- uint8_t * mStorage;
- bool mStorageOwned; // did this object allocate the storage?
+ const int32_t mBytesPerFrame;
+ uint8_t *mStorage;
+ bool mStorageOwned; // did this object allocate the storage?
FifoControllerBase *mFifo;
- fifo_counter_t mFramesReadCount;
- fifo_counter_t mFramesUnderrunCount;
- int32_t mUnderrunCount; // need? just use frames
- int32_t mLastReadSize;
+ fifo_counter_t mFramesReadCount;
+ fifo_counter_t mFramesUnderrunCount;
+ int32_t mUnderrunCount; // need? just use frames
+ int32_t mLastReadSize;
};
+} // android
+
#endif //FIFO_FIFO_BUFFER_H
diff --git a/media/libaaudio/src/fifo/FifoController.h b/media/libaaudio/src/fifo/FifoController.h
index 7434634..79d98a1 100644
--- a/media/libaaudio/src/fifo/FifoController.h
+++ b/media/libaaudio/src/fifo/FifoController.h
@@ -22,6 +22,8 @@
#include "FifoControllerBase.h"
+namespace android {
+
/**
* A FIFO with counters contained in the class.
*/
@@ -55,5 +57,6 @@
std::atomic<fifo_counter_t> mWriteCounter;
};
+} // android
#endif //FIFO_FIFO_CONTROLLER_H
diff --git a/media/libaaudio/src/fifo/FifoControllerBase.cpp b/media/libaaudio/src/fifo/FifoControllerBase.cpp
index 33a253e..14a2be1 100644
--- a/media/libaaudio/src/fifo/FifoControllerBase.cpp
+++ b/media/libaaudio/src/fifo/FifoControllerBase.cpp
@@ -21,6 +21,8 @@
#include <stdint.h>
#include "FifoControllerBase.h"
+using namespace android; // TODO just import names needed
+
FifoControllerBase::FifoControllerBase(fifo_frames_t capacity, fifo_frames_t threshold)
: mCapacity(capacity)
, mThreshold(threshold)
diff --git a/media/libaaudio/src/fifo/FifoControllerBase.h b/media/libaaudio/src/fifo/FifoControllerBase.h
index c543519..64af777 100644
--- a/media/libaaudio/src/fifo/FifoControllerBase.h
+++ b/media/libaaudio/src/fifo/FifoControllerBase.h
@@ -19,6 +19,8 @@
#include <stdint.h>
+namespace android {
+
typedef int64_t fifo_counter_t;
typedef int32_t fifo_frames_t;
@@ -118,4 +120,6 @@
fifo_frames_t mThreshold;
};
+} // android
+
#endif // FIFO_FIFO_CONTROLLER_BASE_H
diff --git a/media/libaaudio/src/fifo/FifoControllerIndirect.h b/media/libaaudio/src/fifo/FifoControllerIndirect.h
index 1aaf9ea..5832d9c 100644
--- a/media/libaaudio/src/fifo/FifoControllerIndirect.h
+++ b/media/libaaudio/src/fifo/FifoControllerIndirect.h
@@ -22,6 +22,8 @@
#include "FifoControllerBase.h"
+namespace android {
+
/**
* A FifoControllerBase with counters external to the class.
*
@@ -66,4 +68,6 @@
std::atomic<fifo_counter_t> * mWriteCounterAddress;
};
+} // android
+
#endif //FIFO_FIFO_CONTROLLER_INDIRECT_H
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
new file mode 100644
index 0000000..baa24c9
--- /dev/null
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -0,0 +1,110 @@
+/*
+ * Copyright 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioStreamLegacy"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <stdint.h>
+#include <utils/String16.h>
+#include <media/AudioTrack.h>
+#include <aaudio/AAudio.h>
+
+#include "core/AudioStream.h"
+#include "legacy/AudioStreamLegacy.h"
+
+using namespace android;
+using namespace aaudio;
+
+AudioStreamLegacy::AudioStreamLegacy()
+ : AudioStream() {
+}
+
+AudioStreamLegacy::~AudioStreamLegacy() {
+}
+
+// Called from AudioTrack.cpp or AudioRecord.cpp
+static void AudioStreamLegacy_callback(int event, void* userData, void *info) {
+ AudioStreamLegacy *streamLegacy = (AudioStreamLegacy *) userData;
+ streamLegacy->processCallback(event, info);
+}
+
+aaudio_legacy_callback_t AudioStreamLegacy::getLegacyCallback() {
+ return AudioStreamLegacy_callback;
+}
+
+// Implement FixedBlockProcessor
+int32_t AudioStreamLegacy::onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) {
+ int32_t frameCount = numBytes / getBytesPerFrame();
+ // Call using the AAudio callback interface.
+ AAudioStream_dataCallback appCallback = getDataCallbackProc();
+ return (*appCallback)(
+ (AAudioStream *) this,
+ getDataCallbackUserData(),
+ buffer,
+ frameCount);
+}
+
+void AudioStreamLegacy::processCallbackCommon(aaudio_callback_operation_t opcode, void *info) {
+ aaudio_data_callback_result_t callbackResult;
+ switch (opcode) {
+ case AAUDIO_CALLBACK_OPERATION_PROCESS_DATA: {
+ // Note that this code assumes an AudioTrack::Buffer is the same as AudioRecord::Buffer
+ // TODO define our own AudioBuffer and pass it from the subclasses.
+ AudioTrack::Buffer *audioBuffer = static_cast<AudioTrack::Buffer *>(info);
+ if (audioBuffer->frameCount == 0) return;
+
+ // If the caller specified an exact size then use a block size adapter.
+ if (mBlockAdapter != nullptr) {
+ int32_t byteCount = audioBuffer->frameCount * getBytesPerFrame();
+ callbackResult = mBlockAdapter->processVariableBlock((uint8_t *) audioBuffer->raw,
+ byteCount);
+ } else {
+ // Call using the AAudio callback interface.
+ callbackResult = (*getDataCallbackProc())(
+ (AAudioStream *) this,
+ getDataCallbackUserData(),
+ audioBuffer->raw,
+ audioBuffer->frameCount
+ );
+ }
+ if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
+ audioBuffer->size = audioBuffer->frameCount * getBytesPerFrame();
+ } else {
+ audioBuffer->size = 0;
+ }
+ }
+ break;
+
+ // Stream got rerouted so we disconnect.
+ case AAUDIO_CALLBACK_OPERATION_DISCONNECTED: {
+ ALOGD("AudioStreamAAudio(): callbackLoop() stream disconnected");
+ if (getErrorCallbackProc() != nullptr) {
+ (*getErrorCallbackProc())(
+ (AAudioStream *) this,
+ getErrorCallbackUserData(),
+ AAUDIO_OK
+ );
+ }
+ mCallbackEnabled.store(false);
+ }
+ break;
+
+ default:
+ break;
+ }
+}
+
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
new file mode 100644
index 0000000..c109ee7
--- /dev/null
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef LEGACY_AUDIO_STREAM_LEGACY_H
+#define LEGACY_AUDIO_STREAM_LEGACY_H
+
+
+#include <aaudio/AAudio.h>
+
+#include "AudioStream.h"
+#include "AAudioLegacy.h"
+#include "utility/FixedBlockAdapter.h"
+
+namespace aaudio {
+
+
+typedef void (*aaudio_legacy_callback_t)(int event, void* user, void *info);
+
+enum {
+ /**
+ * Request that the callback function should fill the data buffer of an output stream,
+ * or process the data of an input stream.
+ * The address parameter passed to the callback function will point to a data buffer.
+ * For an input stream, the data is read-only.
+ * The value1 parameter will be the number of frames.
+ * The value2 parameter is reserved and will be set to zero.
+ * The callback should return AAUDIO_CALLBACK_RESULT_CONTINUE or AAUDIO_CALLBACK_RESULT_STOP.
+ */
+ AAUDIO_CALLBACK_OPERATION_PROCESS_DATA,
+
+ /**
+ * Inform the callback function that the stream was disconnected.
+ * The address parameter passed to the callback function will be NULL.
+ * The value1 will be an error code or AAUDIO_OK.
+ * The value2 parameter is reserved and will be set to zero.
+ * The callback return value will be ignored.
+ */
+ AAUDIO_CALLBACK_OPERATION_DISCONNECTED,
+};
+typedef int32_t aaudio_callback_operation_t;
+
+
+class AudioStreamLegacy : public AudioStream, public FixedBlockProcessor {
+public:
+ AudioStreamLegacy();
+
+ virtual ~AudioStreamLegacy();
+
+ aaudio_legacy_callback_t getLegacyCallback();
+
+ // This is public so it can be called from the C callback function.
+ // This is called from the AudioTrack/AudioRecord client.
+ virtual void processCallback(int event, void *info) = 0;
+
+ void processCallbackCommon(aaudio_callback_operation_t opcode, void *info);
+
+ // Implement FixedBlockProcessor
+ int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) override;
+
+protected:
+ FixedBlockAdapter *mBlockAdapter = nullptr;
+ aaudio_wrapping_frames_t mPositionWhenStarting = 0;
+ int32_t mCallbackBufferSize = 0;
+};
+
+} /* namespace aaudio */
+
+#endif //LEGACY_AUDIO_STREAM_LEGACY_H
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index d380eb8..f0a6ceb 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -24,14 +24,16 @@
#include <aaudio/AAudio.h>
#include "AudioClock.h"
-#include "AudioStreamRecord.h"
-#include "utility/AAudioUtilities.h"
+#include "legacy/AudioStreamLegacy.h"
+#include "legacy/AudioStreamRecord.h"
+#include "utility/FixedBlockWriter.h"
using namespace android;
using namespace aaudio;
AudioStreamRecord::AudioStreamRecord()
- : AudioStream()
+ : AudioStreamLegacy()
+ , mFixedBlockWriter(*this)
{
}
@@ -58,7 +60,6 @@
? 2 : getSamplesPerFrame();
audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(samplesPerFrame);
- AudioRecord::callback_t callback = nullptr;
audio_input_flags_t flags = (audio_input_flags_t) AUDIO_INPUT_FLAG_NONE;
size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0
@@ -68,6 +69,17 @@
? AUDIO_FORMAT_PCM_FLOAT
: AAudioConvert_aaudioToAndroidDataFormat(getFormat());
+ // Setup the callback if there is one.
+ AudioRecord::callback_t callback = nullptr;
+ void *callbackData = nullptr;
+ AudioRecord::transfer_type streamTransferType = AudioRecord::transfer_type::TRANSFER_SYNC;
+ if (builder.getDataCallbackProc() != nullptr) {
+ streamTransferType = AudioRecord::transfer_type::TRANSFER_CALLBACK;
+ callback = getLegacyCallback();
+ callbackData = this;
+ }
+ mCallbackBufferSize = builder.getFramesPerDataCallback();
+
mAudioRecord = new AudioRecord(
AUDIO_SOURCE_DEFAULT,
getSampleRate(),
@@ -76,10 +88,10 @@
mOpPackageName, // const String16& opPackageName TODO does not compile
frameCount,
callback,
- nullptr, // void* user = nullptr,
+ callbackData,
0, // uint32_t notificationFrames = 0,
AUDIO_SESSION_ALLOCATE,
- AudioRecord::TRANSFER_DEFAULT,
+ streamTransferType,
flags
// int uid = -1,
// pid_t pid = -1,
@@ -99,6 +111,15 @@
setSamplesPerFrame(mAudioRecord->channelCount());
setFormat(AAudioConvert_androidToAAudioDataFormat(mAudioRecord->format()));
+ // We may need to pass the data through a block size adapter to guarantee constant size.
+ if (mCallbackBufferSize != AAUDIO_UNSPECIFIED) {
+ int callbackSizeBytes = getBytesPerFrame() * mCallbackBufferSize;
+ mFixedBlockWriter.open(callbackSizeBytes);
+ mBlockAdapter = &mFixedBlockWriter;
+ } else {
+ mBlockAdapter = nullptr;
+ }
+
setState(AAUDIO_STREAM_STATE_OPEN);
return AAUDIO_OK;
@@ -111,9 +132,29 @@
mAudioRecord.clear();
setState(AAUDIO_STREAM_STATE_CLOSED);
}
+ mFixedBlockWriter.close();
return AAUDIO_OK;
}
+void AudioStreamRecord::processCallback(int event, void *info) {
+
+ ALOGD("AudioStreamRecord::processCallback(), event %d", event);
+ switch (event) {
+ case AudioRecord::EVENT_MORE_DATA:
+ processCallbackCommon(AAUDIO_CALLBACK_OPERATION_PROCESS_DATA, info);
+ break;
+
+ // Stream got rerouted so we disconnect.
+ case AudioRecord::EVENT_NEW_IAUDIORECORD:
+ processCallbackCommon(AAUDIO_CALLBACK_OPERATION_DISCONNECTED, info);
+ break;
+
+ default:
+ break;
+ }
+ return;
+}
+
aaudio_result_t AudioStreamRecord::requestStart()
{
if (mAudioRecord.get() == nullptr) {
@@ -124,6 +165,7 @@
if (err != OK) {
return AAudioConvert_androidToAAudioResult(err);
}
+
err = mAudioRecord->start();
if (err != OK) {
return AAudioConvert_androidToAAudioResult(err);
@@ -151,7 +193,7 @@
return AAUDIO_OK;
}
-aaudio_result_t AudioStreamRecord::updateState()
+aaudio_result_t AudioStreamRecord::updateStateWhileWaiting()
{
aaudio_result_t result = AAUDIO_OK;
aaudio_wrapping_frames_t position;
@@ -222,7 +264,7 @@
int32_t AudioStreamRecord::getFramesPerBurst() const
{
- return 192; // TODO add query to AudioRecord.cpp
+ return static_cast<int32_t>(mAudioRecord->getNotificationPeriodInFrames());
}
aaudio_result_t AudioStreamRecord::getTimestamp(clockid_t clockId,
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index 4667f05..897a5b3 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -23,51 +23,58 @@
#include "AudioStreamBuilder.h"
#include "AudioStream.h"
#include "AAudioLegacy.h"
+#include "legacy/AudioStreamLegacy.h"
+#include "utility/FixedBlockWriter.h"
namespace aaudio {
/**
* Internal stream that uses the legacy AudioTrack path.
*/
-class AudioStreamRecord : public AudioStream {
+class AudioStreamRecord : public AudioStreamLegacy {
public:
AudioStreamRecord();
virtual ~AudioStreamRecord();
- virtual aaudio_result_t open(const AudioStreamBuilder & builder) override;
- virtual aaudio_result_t close() override;
+ aaudio_result_t open(const AudioStreamBuilder & builder) override;
+ aaudio_result_t close() override;
- virtual aaudio_result_t requestStart() override;
- virtual aaudio_result_t requestPause() override;
- virtual aaudio_result_t requestFlush() override;
- virtual aaudio_result_t requestStop() override;
+ aaudio_result_t requestStart() override;
+ aaudio_result_t requestPause() override;
+ aaudio_result_t requestFlush() override;
+ aaudio_result_t requestStop() override;
virtual aaudio_result_t getTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds) override;
- virtual aaudio_result_t read(void *buffer,
+ aaudio_result_t read(void *buffer,
int32_t numFrames,
int64_t timeoutNanoseconds) override;
- virtual aaudio_result_t setBufferSize(int32_t requestedFrames) override;
+ aaudio_result_t setBufferSize(int32_t requestedFrames) override;
- virtual int32_t getBufferSize() const override;
+ int32_t getBufferSize() const override;
- virtual int32_t getBufferCapacity() const override;
+ int32_t getBufferCapacity() const override;
- virtual int32_t getXRunCount() const override;
+ int32_t getXRunCount() const override;
- virtual int32_t getFramesPerBurst() const override;
+ int32_t getFramesPerBurst() const override;
- virtual aaudio_result_t updateState() override;
+ aaudio_result_t updateStateWhileWaiting() override;
+
+ // This is public so it can be called from the C callback function.
+ void processCallback(int event, void *info) override;
private:
android::sp<android::AudioRecord> mAudioRecord;
+ // adapts between variable sized blocks and fixed size blocks
+ FixedBlockWriter mFixedBlockWriter;
+
// TODO add 64-bit position reporting to AudioRecord and use it.
- aaudio_wrapping_frames_t mPositionWhenStarting = 0;
- android::String16 mOpPackageName;
+ android::String16 mOpPackageName;
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 8bb6aee..1bb9e53 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -20,20 +20,25 @@
#include <stdint.h>
#include <media/AudioTrack.h>
-#include <aaudio/AAudio.h>
+#include <aaudio/AAudio.h>
#include "utility/AudioClock.h"
-#include "AudioStreamTrack.h"
-#include "utility/AAudioUtilities.h"
+#include "legacy/AudioStreamLegacy.h"
+#include "legacy/AudioStreamTrack.h"
+#include "utility/FixedBlockReader.h"
using namespace android;
using namespace aaudio;
+// Arbitrary and somewhat generous number of bursts.
+#define DEFAULT_BURSTS_PER_BUFFER_CAPACITY 8
+
/*
* Create a stream that uses the AudioTrack.
*/
AudioStreamTrack::AudioStreamTrack()
- : AudioStream()
+ : AudioStreamLegacy()
+ , mFixedBlockReader(*this)
{
}
@@ -53,6 +58,8 @@
return result;
}
+ ALOGD("AudioStreamTrack::open = %p", this);
+
// Try to create an AudioTrack
// TODO Support UNSPECIFIED in AudioTrack. For now, use stereo if unspecified.
int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED)
@@ -61,16 +68,40 @@
ALOGD("AudioStreamTrack::open(), samplesPerFrame = %d, channelMask = 0x%08x",
samplesPerFrame, channelMask);
- AudioTrack::callback_t callback = nullptr;
// TODO add more performance options
audio_output_flags_t flags = (audio_output_flags_t) AUDIO_OUTPUT_FLAG_FAST;
- size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0
- : builder.getBufferCapacity();
+
+ int32_t frameCount = builder.getBufferCapacity();
+ ALOGD("AudioStreamTrack::open(), requested buffer capacity %d", frameCount);
+
+ int32_t notificationFrames = 0;
+
// TODO implement an unspecified AudioTrack format then use that.
- audio_format_t format = (getFormat() == AAUDIO_UNSPECIFIED)
+ audio_format_t format = (getFormat() == AAUDIO_FORMAT_UNSPECIFIED)
? AUDIO_FORMAT_PCM_FLOAT
: AAudioConvert_aaudioToAndroidDataFormat(getFormat());
+ // Setup the callback if there is one.
+ AudioTrack::callback_t callback = nullptr;
+ void *callbackData = nullptr;
+ // Note that TRANSFER_SYNC does not allow FAST track
+ AudioTrack::transfer_type streamTransferType = AudioTrack::transfer_type::TRANSFER_SYNC;
+ if (builder.getDataCallbackProc() != nullptr) {
+ streamTransferType = AudioTrack::transfer_type::TRANSFER_CALLBACK;
+ callback = getLegacyCallback();
+ callbackData = this;
+
+ notificationFrames = builder.getFramesPerDataCallback();
+ // If the total buffer size is unspecified then base the size on the burst size.
+ if (frameCount == AAUDIO_UNSPECIFIED) {
+ // Take advantage of a special trick that allows us to create a buffer
+ // that is some multiple of the burst size.
+ notificationFrames = 0 - DEFAULT_BURSTS_PER_BUFFER_CAPACITY;
+ }
+ }
+ mCallbackBufferSize = builder.getFramesPerDataCallback();
+
+ ALOGD("AudioStreamTrack::open(), notificationFrames = %d", notificationFrames);
mAudioTrack = new AudioTrack(
(audio_stream_type_t) AUDIO_STREAM_MUSIC,
getSampleRate(),
@@ -79,10 +110,10 @@
frameCount,
flags,
callback,
- nullptr, // user callback data
- 0, // notificationFrames
+ callbackData,
+ notificationFrames,
AUDIO_SESSION_ALLOCATE,
- AudioTrack::transfer_type::TRANSFER_SYNC // TODO - this does not allow FAST
+ streamTransferType
);
// Did we get a valid track?
@@ -97,9 +128,21 @@
// Get the actual values from the AudioTrack.
setSamplesPerFrame(mAudioTrack->channelCount());
setSampleRate(mAudioTrack->getSampleRate());
- setFormat(AAudioConvert_androidToAAudioDataFormat(mAudioTrack->format()));
+ aaudio_audio_format_t aaudioFormat =
+ AAudioConvert_androidToAAudioDataFormat(mAudioTrack->format());
+ setFormat(aaudioFormat);
+
+ // We may need to pass the data through a block size adapter to guarantee constant size.
+ if (mCallbackBufferSize != AAUDIO_UNSPECIFIED) {
+ int callbackSizeBytes = getBytesPerFrame() * mCallbackBufferSize;
+ mFixedBlockReader.open(callbackSizeBytes);
+ mBlockAdapter = &mFixedBlockReader;
+ } else {
+ mBlockAdapter = nullptr;
+ }
setState(AAUDIO_STREAM_STATE_OPEN);
+ setDeviceId(mAudioTrack->getRoutedDeviceId());
return AAUDIO_OK;
}
@@ -111,11 +154,32 @@
mAudioTrack.clear(); // TODO is this right?
setState(AAUDIO_STREAM_STATE_CLOSED);
}
+ mFixedBlockReader.close();
return AAUDIO_OK;
}
+void AudioStreamTrack::processCallback(int event, void *info) {
+
+ switch (event) {
+ case AudioTrack::EVENT_MORE_DATA:
+ processCallbackCommon(AAUDIO_CALLBACK_OPERATION_PROCESS_DATA, info);
+ break;
+
+ // Stream got rerouted so we disconnect.
+ case AudioTrack::EVENT_NEW_IAUDIOTRACK:
+ processCallbackCommon(AAUDIO_CALLBACK_OPERATION_DISCONNECTED, info);
+ break;
+
+ default:
+ break;
+ }
+ return;
+}
+
aaudio_result_t AudioStreamTrack::requestStart()
{
+ std::lock_guard<std::mutex> lock(mStreamMutex);
+
if (mAudioTrack.get() == nullptr) {
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -124,6 +188,7 @@
if (err != OK) {
return AAudioConvert_androidToAAudioResult(err);
}
+
err = mAudioTrack->start();
if (err != OK) {
return AAudioConvert_androidToAAudioResult(err);
@@ -135,11 +200,14 @@
aaudio_result_t AudioStreamTrack::requestPause()
{
+ std::lock_guard<std::mutex> lock(mStreamMutex);
+
if (mAudioTrack.get() == nullptr) {
return AAUDIO_ERROR_INVALID_STATE;
} else if (getState() != AAUDIO_STREAM_STATE_STARTING
&& getState() != AAUDIO_STREAM_STATE_STARTED) {
- ALOGE("requestPause(), called when state is %s", AAudio_convertStreamStateToText(getState()));
+ ALOGE("requestPause(), called when state is %s",
+ AAudio_convertStreamStateToText(getState()));
return AAUDIO_ERROR_INVALID_STATE;
}
setState(AAUDIO_STREAM_STATE_PAUSING);
@@ -152,6 +220,8 @@
}
aaudio_result_t AudioStreamTrack::requestFlush() {
+ std::lock_guard<std::mutex> lock(mStreamMutex);
+
if (mAudioTrack.get() == nullptr) {
return AAUDIO_ERROR_INVALID_STATE;
} else if (getState() != AAUDIO_STREAM_STATE_PAUSED) {
@@ -165,6 +235,8 @@
}
aaudio_result_t AudioStreamTrack::requestStop() {
+ std::lock_guard<std::mutex> lock(mStreamMutex);
+
if (mAudioTrack.get() == nullptr) {
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -175,7 +247,7 @@
return AAUDIO_OK;
}
-aaudio_result_t AudioStreamTrack::updateState()
+aaudio_result_t AudioStreamTrack::updateStateWhileWaiting()
{
status_t err;
aaudio_wrapping_frames_t position;
@@ -272,7 +344,7 @@
int32_t AudioStreamTrack::getFramesPerBurst() const
{
- return 192; // TODO add query to AudioTrack.cpp
+ return static_cast<int32_t>(mAudioTrack->getNotificationPeriodInFrames());
}
int64_t AudioStreamTrack::getFramesRead() {
@@ -303,7 +375,7 @@
}
// TODO Merge common code into AudioStreamLegacy after rebasing.
int timebase;
- switch(clockId) {
+ switch (clockId) {
case CLOCK_BOOTTIME:
timebase = ExtendedTimestamp::TIMEBASE_BOOTTIME;
break;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.h b/media/libaaudio/src/legacy/AudioStreamTrack.h
index 7a53022..29f5d15 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.h
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.h
@@ -17,54 +17,63 @@
#ifndef LEGACY_AUDIO_STREAM_TRACK_H
#define LEGACY_AUDIO_STREAM_TRACK_H
+#include <math.h>
#include <media/AudioTrack.h>
#include <aaudio/AAudio.h>
#include "AudioStreamBuilder.h"
#include "AudioStream.h"
-#include "AAudioLegacy.h"
+#include "legacy/AAudioLegacy.h"
+#include "legacy/AudioStreamLegacy.h"
+#include "utility/FixedBlockReader.h"
namespace aaudio {
-
/**
* Internal stream that uses the legacy AudioTrack path.
*/
-class AudioStreamTrack : public AudioStream {
+class AudioStreamTrack : public AudioStreamLegacy {
public:
AudioStreamTrack();
virtual ~AudioStreamTrack();
- virtual aaudio_result_t open(const AudioStreamBuilder & builder) override;
- virtual aaudio_result_t close() override;
+ aaudio_result_t open(const AudioStreamBuilder & builder) override;
+ aaudio_result_t close() override;
- virtual aaudio_result_t requestStart() override;
- virtual aaudio_result_t requestPause() override;
- virtual aaudio_result_t requestFlush() override;
- virtual aaudio_result_t requestStop() override;
+ aaudio_result_t requestStart() override;
+ aaudio_result_t requestPause() override;
+ aaudio_result_t requestFlush() override;
+ aaudio_result_t requestStop() override;
- virtual aaudio_result_t getTimestamp(clockid_t clockId,
+ aaudio_result_t getTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds) override;
- virtual aaudio_result_t write(const void *buffer,
+ aaudio_result_t write(const void *buffer,
int32_t numFrames,
int64_t timeoutNanoseconds) override;
- virtual aaudio_result_t setBufferSize(int32_t requestedFrames) override;
- virtual int32_t getBufferSize() const override;
- virtual int32_t getBufferCapacity() const override;
- virtual int32_t getFramesPerBurst()const override;
- virtual int32_t getXRunCount() const override;
+ aaudio_result_t setBufferSize(int32_t requestedFrames) override;
+ int32_t getBufferSize() const override;
+ int32_t getBufferCapacity() const override;
+ int32_t getFramesPerBurst()const override;
+ int32_t getXRunCount() const override;
- virtual int64_t getFramesRead() override;
+ int64_t getFramesRead() override;
- virtual aaudio_result_t updateState() override;
+ aaudio_result_t updateStateWhileWaiting() override;
+
+ // This is public so it can be called from the C callback function.
+ void processCallback(int event, void *info) override;
private:
+
android::sp<android::AudioTrack> mAudioTrack;
+ // adapts between variable sized blocks and fixed size blocks
+ FixedBlockReader mFixedBlockReader;
+
// TODO add 64-bit position reporting to AudioRecord and use it.
aaudio_wrapping_frames_t mPositionWhenStarting = 0;
aaudio_wrapping_frames_t mPositionWhenPausing = 0;
diff --git a/media/libaaudio/src/utility/FixedBlockAdapter.cpp b/media/libaaudio/src/utility/FixedBlockAdapter.cpp
new file mode 100644
index 0000000..f4666af
--- /dev/null
+++ b/media/libaaudio/src/utility/FixedBlockAdapter.cpp
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdint.h>
+
+#include "FixedBlockAdapter.h"
+
+FixedBlockAdapter::~FixedBlockAdapter() {
+ close();
+}
+
+int32_t FixedBlockAdapter::open(int32_t bytesPerFixedBlock)
+{
+ mSize = bytesPerFixedBlock;
+ mStorage = new uint8_t[bytesPerFixedBlock]; // TODO use std::nothrow
+ mPosition = 0;
+ return 0;
+}
+
+int32_t FixedBlockAdapter::close()
+{
+ delete[] mStorage;
+ mStorage = nullptr;
+ mSize = 0;
+ mPosition = 0;
+ return 0;
+}
diff --git a/media/libaaudio/src/utility/FixedBlockAdapter.h b/media/libaaudio/src/utility/FixedBlockAdapter.h
new file mode 100644
index 0000000..7008b25
--- /dev/null
+++ b/media/libaaudio/src/utility/FixedBlockAdapter.h
@@ -0,0 +1,71 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_FIXED_BLOCK_ADAPTER_H
+#define AAUDIO_FIXED_BLOCK_ADAPTER_H
+
+#include <stdio.h>
+
+/**
+ * Interface for a class that needs fixed-size blocks.
+ */
+class FixedBlockProcessor {
+public:
+ virtual ~FixedBlockProcessor() = default;
+ virtual int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) = 0;
+};
+
+/**
+ * Base class for a variable-to-fixed-size block adapter.
+ */
+class FixedBlockAdapter
+{
+public:
+ FixedBlockAdapter(FixedBlockProcessor &fixedBlockProcessor)
+ : mFixedBlockProcessor(fixedBlockProcessor) {}
+
+ virtual ~FixedBlockAdapter();
+
+ /**
+ * Allocate internal resources needed for buffering data.
+ */
+ virtual int32_t open(int32_t bytesPerFixedBlock);
+
+ /**
+ * Note that if the fixed-sized blocks must be aligned, then the variable-sized blocks
+ * must have the same alignment.
+ * For example, if the fixed-size blocks must be a multiple of 8, then the variable-sized
+ * blocks must also be a multiple of 8.
+ *
+ * @param buffer
+ * @param numBytes
+ * @return zero if OK or a non-zero code
+ */
+ virtual int32_t processVariableBlock(uint8_t *buffer, int32_t numBytes) = 0;
+
+ /**
+ * Free internal resources.
+ */
+ int32_t close();
+
+protected:
+ FixedBlockProcessor &mFixedBlockProcessor;
+ uint8_t *mStorage = nullptr; // Store data here while assembling buffers.
+ int32_t mSize = 0; // Size in bytes of the fixed size buffer.
+ int32_t mPosition = 0; // Offset of the last byte read or written.
+};
+
+#endif /* AAUDIO_FIXED_BLOCK_ADAPTER_H */
diff --git a/media/libaaudio/src/utility/FixedBlockReader.cpp b/media/libaaudio/src/utility/FixedBlockReader.cpp
new file mode 100644
index 0000000..21ea70e
--- /dev/null
+++ b/media/libaaudio/src/utility/FixedBlockReader.cpp
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdint.h>
+#include <memory.h>
+
+#include "FixedBlockAdapter.h"
+
+#include "FixedBlockReader.h"
+
+
+FixedBlockReader::FixedBlockReader(FixedBlockProcessor &fixedBlockProcessor)
+ : FixedBlockAdapter(fixedBlockProcessor) {
+ mPosition = mSize;
+}
+
+int32_t FixedBlockReader::open(int32_t bytesPerFixedBlock) {
+ int32_t result = FixedBlockAdapter::open(bytesPerFixedBlock);
+ mPosition = mSize; // Indicate no data in storage.
+ return result;
+}
+
+int32_t FixedBlockReader::readFromStorage(uint8_t *buffer, int32_t numBytes) {
+ int32_t bytesToRead = numBytes;
+ int32_t dataAvailable = mSize - mPosition;
+ if (bytesToRead > dataAvailable) {
+ bytesToRead = dataAvailable;
+ }
+ memcpy(buffer, mStorage + mPosition, bytesToRead);
+ mPosition += bytesToRead;
+ return bytesToRead;
+}
+
+int32_t FixedBlockReader::processVariableBlock(uint8_t *buffer, int32_t numBytes) {
+ int32_t result = 0;
+ int32_t bytesLeft = numBytes;
+ while(bytesLeft > 0 && result == 0) {
+ if (mPosition < mSize) {
+ // Use up bytes currently in storage.
+ int32_t bytesRead = readFromStorage(buffer, bytesLeft);
+ buffer += bytesRead;
+ bytesLeft -= bytesRead;
+ } else if (bytesLeft >= mSize) {
+ // Read through if enough for a complete block.
+ result = mFixedBlockProcessor.onProcessFixedBlock(buffer, mSize);
+ buffer += mSize;
+ bytesLeft -= mSize;
+ } else {
+ // Just need a partial block so we have to use storage.
+ result = mFixedBlockProcessor.onProcessFixedBlock(mStorage, mSize);
+ mPosition = 0;
+ }
+ }
+ return result;
+}
+
diff --git a/media/libaaudio/src/utility/FixedBlockReader.h b/media/libaaudio/src/utility/FixedBlockReader.h
new file mode 100644
index 0000000..128dd52
--- /dev/null
+++ b/media/libaaudio/src/utility/FixedBlockReader.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_FIXED_BLOCK_READER_H
+#define AAUDIO_FIXED_BLOCK_READER_H
+
+#include <stdint.h>
+
+#include "FixedBlockAdapter.h"
+
+/**
+ * Read from a fixed-size block to a variable sized block.
+ *
+ * This can be used to convert a pull data flow from fixed sized buffers to variable sized buffers.
+ * An example would be an audio output callback that reads from the app.
+ */
+class FixedBlockReader : public FixedBlockAdapter
+{
+public:
+ FixedBlockReader(FixedBlockProcessor &fixedBlockProcessor);
+
+ virtual ~FixedBlockReader() = default;
+
+ int32_t open(int32_t bytesPerFixedBlock) override;
+
+ int32_t readFromStorage(uint8_t *buffer, int32_t numBytes);
+
+ /**
+ * Read into a variable sized block.
+ */
+ int32_t processVariableBlock(uint8_t *buffer, int32_t numBytes) override;
+};
+
+
+#endif /* AAUDIO_FIXED_BLOCK_READER_H */
diff --git a/media/libaaudio/src/utility/FixedBlockWriter.cpp b/media/libaaudio/src/utility/FixedBlockWriter.cpp
new file mode 100644
index 0000000..2ce8046
--- /dev/null
+++ b/media/libaaudio/src/utility/FixedBlockWriter.cpp
@@ -0,0 +1,67 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdint.h>
+#include <memory.h>
+
+#include "FixedBlockAdapter.h"
+#include "FixedBlockWriter.h"
+
+FixedBlockWriter::FixedBlockWriter(FixedBlockProcessor &fixedBlockProcessor)
+ : FixedBlockAdapter(fixedBlockProcessor) {}
+
+
+int32_t FixedBlockWriter::writeToStorage(uint8_t *buffer, int32_t numBytes) {
+ int32_t bytesToStore = numBytes;
+ int32_t roomAvailable = mSize - mPosition;
+ if (bytesToStore > roomAvailable) {
+ bytesToStore = roomAvailable;
+ }
+ memcpy(mStorage + mPosition, buffer, bytesToStore);
+ mPosition += bytesToStore;
+ return bytesToStore;
+}
+
+int32_t FixedBlockWriter::processVariableBlock(uint8_t *buffer, int32_t numBytes) {
+ int32_t result = 0;
+ int32_t bytesLeft = numBytes;
+
+ // If we already have data in storage then add to it.
+ if (mPosition > 0) {
+ int32_t bytesWritten = writeToStorage(buffer, bytesLeft);
+ buffer += bytesWritten;
+ bytesLeft -= bytesWritten;
+ // If storage full then flush it out
+ if (mPosition == mSize) {
+ result = mFixedBlockProcessor.onProcessFixedBlock(mStorage, mSize);
+ mPosition = 0;
+ }
+ }
+
+ // Write through if enough for a complete block.
+ while(bytesLeft > mSize && result == 0) {
+ result = mFixedBlockProcessor.onProcessFixedBlock(buffer, mSize);
+ buffer += mSize;
+ bytesLeft -= mSize;
+ }
+
+ // Save any remaining partial block for next time.
+ if (bytesLeft > 0) {
+ writeToStorage(buffer, bytesLeft);
+ }
+
+ return result;
+}
diff --git a/media/libaaudio/src/utility/FixedBlockWriter.h b/media/libaaudio/src/utility/FixedBlockWriter.h
new file mode 100644
index 0000000..f1d917c
--- /dev/null
+++ b/media/libaaudio/src/utility/FixedBlockWriter.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_FIXED_BLOCK_WRITER_H
+#define AAUDIO_FIXED_BLOCK_WRITER_H
+
+#include <stdint.h>
+
+#include "FixedBlockAdapter.h"
+
+/**
+ * This can be used to convert a push data flow from variable sized buffers to fixed sized buffers.
+ * An example would be an audio input callback.
+ */
+class FixedBlockWriter : public FixedBlockAdapter
+{
+public:
+ FixedBlockWriter(FixedBlockProcessor &fixedBlockProcessor);
+
+ virtual ~FixedBlockWriter() = default;
+
+ int32_t writeToStorage(uint8_t *buffer, int32_t numBytes);
+
+ /**
+ * Write from a variable sized block.
+ */
+ int32_t processVariableBlock(uint8_t *buffer, int32_t numBytes) override;
+};
+
+#endif /* AAUDIO_FIXED_BLOCK_WRITER_H */
diff --git a/media/libaaudio/tests/Android.mk b/media/libaaudio/tests/Android.mk
index 7899cf5..06c9364 100644
--- a/media/libaaudio/tests/Android.mk
+++ b/media/libaaudio/tests/Android.mk
@@ -4,8 +4,7 @@
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-utils) \
frameworks/av/media/libaaudio/include \
- frameworks/av/media/libaaudio/src/core \
- frameworks/av/media/libaaudio/src/utility
+ frameworks/av/media/libaaudio/src
LOCAL_SRC_FILES:= test_handle_tracker.cpp
LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
libcutils liblog libmedia libutils
@@ -17,13 +16,22 @@
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-utils) \
frameworks/av/media/libaaudio/include \
- frameworks/av/media/libaaudio/src \
- frameworks/av/media/libaaudio/src/core \
- frameworks/av/media/libaaudio/src/fifo \
- frameworks/av/media/libaaudio/src/utility
+ frameworks/av/media/libaaudio/src
LOCAL_SRC_FILES:= test_marshalling.cpp
LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
libcutils liblog libmedia libutils
LOCAL_STATIC_LIBRARIES := libaaudio
-LOCAL_MODULE := test_marshalling
+LOCAL_MODULE := test_aaudio_marshalling
+include $(BUILD_NATIVE_TEST)
+
+include $(CLEAR_VARS)
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/media/libaaudio/include \
+ frameworks/av/media/libaaudio/src
+LOCAL_SRC_FILES:= test_block_adapter.cpp
+LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
+ libcutils liblog libmedia libutils
+LOCAL_STATIC_LIBRARIES := libaaudio
+LOCAL_MODULE := test_block_adapter
include $(BUILD_NATIVE_TEST)
diff --git a/media/libaaudio/tests/test_block_adapter.cpp b/media/libaaudio/tests/test_block_adapter.cpp
new file mode 100644
index 0000000..a22abb9
--- /dev/null
+++ b/media/libaaudio/tests/test_block_adapter.cpp
@@ -0,0 +1,151 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <iostream>
+
+#include <gtest/gtest.h>
+
+#include "utility/FixedBlockAdapter.h"
+#include "utility/FixedBlockWriter.h"
+#include "utility/FixedBlockReader.h"
+
+#define FIXED_BLOCK_SIZE 47
+#define TEST_BUFFER_SIZE 103
+
+// Pass varying sized blocks.
+// Frames contain a sequential index, which are easily checked.
+class TestBlockAdapter {
+public:
+ TestBlockAdapter()
+ : mTestIndex(0), mLastIndex(0) {
+ }
+
+ ~TestBlockAdapter() = default;
+
+ void fillSequence(int32_t *indexBuffer, int32_t frameCount) {
+ ASSERT_LE(frameCount, TEST_BUFFER_SIZE);
+ for (int i = 0; i < frameCount; i++) {
+ indexBuffer[i] = mLastIndex++;
+ }
+ }
+
+ int checkSequence(const int32_t *indexBuffer, int32_t frameCount) {
+ // This is equivalent to calling an output callback.
+ for (int i = 0; i < frameCount; i++) {
+ int32_t expected = mTestIndex++;
+ int32_t actual = indexBuffer[i];
+ EXPECT_EQ(expected, actual);
+ if (actual != expected) {
+ return -1;
+ }
+ }
+ return 0;
+ }
+
+ int32_t mTestBuffer[TEST_BUFFER_SIZE];
+ int32_t mTestIndex;
+ int32_t mLastIndex;
+};
+
+class TestBlockWriter : public TestBlockAdapter, FixedBlockProcessor {
+public:
+ TestBlockWriter()
+ : mFixedBlockWriter(*this) {
+ mFixedBlockWriter.open(sizeof(int32_t) * FIXED_BLOCK_SIZE);
+ }
+
+ ~TestBlockWriter() {
+ mFixedBlockWriter.close();
+ }
+
+ int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) override {
+ int32_t frameCount = numBytes / sizeof(int32_t);
+ return checkSequence((int32_t *) buffer, frameCount);
+ }
+
+ // Simulate audio input from a variable sized callback.
+ int32_t testInputWrite(int32_t variableCount) {
+ fillSequence(mTestBuffer, variableCount);
+ int32_t sizeBytes = variableCount * sizeof(int32_t);
+ return mFixedBlockWriter.processVariableBlock((uint8_t *) mTestBuffer, sizeBytes);
+ }
+
+private:
+ FixedBlockWriter mFixedBlockWriter;
+};
+
+class TestBlockReader : public TestBlockAdapter, FixedBlockProcessor {
+public:
+ TestBlockReader()
+ : mFixedBlockReader(*this) {
+ mFixedBlockReader.open(sizeof(int32_t) * FIXED_BLOCK_SIZE);
+ }
+
+ ~TestBlockReader() {
+ mFixedBlockReader.close();
+ }
+
+ int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) override {
+ int32_t frameCount = numBytes / sizeof(int32_t);
+ fillSequence((int32_t *) buffer, frameCount);
+ return 0;
+ }
+
+ // Simulate audio output from a variable sized callback.
+ int32_t testOutputRead(int32_t variableCount) {
+ int32_t sizeBytes = variableCount * sizeof(int32_t);
+ int32_t result = mFixedBlockReader.processVariableBlock((uint8_t *) mTestBuffer, sizeBytes);
+ if (result >= 0) {
+ result = checkSequence((int32_t *)mTestBuffer, variableCount);
+ }
+ return result;
+ }
+
+private:
+ FixedBlockReader mFixedBlockReader;
+};
+
+
+TEST(test_block_adapter, block_adapter_write) {
+ TestBlockWriter tester;
+ int result = 0;
+ const int numLoops = 1000;
+
+ for (int i = 0; i<numLoops && result == 0; i++) {
+ long r = random();
+ int32_t size = (r % TEST_BUFFER_SIZE);
+ ASSERT_LE(size, TEST_BUFFER_SIZE);
+ ASSERT_GE(size, 0);
+ result = tester.testInputWrite(size);
+ }
+ ASSERT_EQ(0, result);
+}
+
+TEST(test_block_adapter, block_adapter_read) {
+ TestBlockReader tester;
+ int result = 0;
+ const int numLoops = 1000;
+
+ for (int i = 0; i < numLoops && result == 0; i++) {
+ long r = random();
+ int32_t size = (r % TEST_BUFFER_SIZE);
+ ASSERT_LE(size, TEST_BUFFER_SIZE);
+ ASSERT_GE(size, 0);
+ result = tester.testOutputRead(size);
+ }
+ ASSERT_EQ(0, result);
+};
+
diff --git a/media/libaaudio/tests/test_handle_tracker.cpp b/media/libaaudio/tests/test_handle_tracker.cpp
index e51c39c..e1cb676 100644
--- a/media/libaaudio/tests/test_handle_tracker.cpp
+++ b/media/libaaudio/tests/test_handle_tracker.cpp
@@ -22,7 +22,7 @@
#include <gtest/gtest.h>
#include <aaudio/AAudioDefinitions.h>
-#include "HandleTracker.h"
+#include "utility/HandleTracker.h"
// Test adding one address.
TEST(test_handle_tracker, aaudio_handle_tracker) {
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 6c7cdde..5c54bb2 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -645,10 +645,10 @@
mAwaitBoost = false;
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
if (flags & AUDIO_INPUT_FLAG_FAST) {
- ALOGI("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu", frameCount);
+ ALOGI("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu -> %zu", frameCount, temp);
mAwaitBoost = true;
} else {
- ALOGW("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
+ ALOGW("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", frameCount, temp);
mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST |
AUDIO_INPUT_FLAG_RAW));
continue; // retry
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index d590cb7..3a0ce5e 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -1479,12 +1479,13 @@
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
- ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount);
+ ALOGI("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu", frameCount, temp);
if (!mThreadCanCallJava) {
mAwaitBoost = true;
}
} else {
- ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
+ ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", frameCount,
+ temp);
}
}
mFlags = flags;
diff --git a/media/libaudioclient/AudioTrackShared.cpp b/media/libaudioclient/AudioTrackShared.cpp
index 846f8b8..2ce6c63 100644
--- a/media/libaudioclient/AudioTrackShared.cpp
+++ b/media/libaudioclient/AudioTrackShared.cpp
@@ -696,7 +696,8 @@
ssize_t filled = rear - front;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
+ ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); shutting down",
+ filled, mFrameCount);
mIsShutdown = true;
}
if (mIsShutdown) {
@@ -820,7 +821,8 @@
ssize_t filled = rear - cblk->u.mStreaming.mFront;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
+ ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); shutting down",
+ filled, mFrameCount);
mIsShutdown = true;
return 0;
}
diff --git a/media/libaudioclient/include/AudioRecord.h b/media/libaudioclient/include/AudioRecord.h
index 1c8746f..1b034b5 100644
--- a/media/libaudioclient/include/AudioRecord.h
+++ b/media/libaudioclient/include/AudioRecord.h
@@ -243,6 +243,13 @@
size_t frameSize() const { return mFrameSize; }
audio_source_t inputSource() const { return mAttributes.source; }
+ /*
+ * Return the period of the notification callback in frames.
+ * This value is set when the AudioRecord is constructed.
+ * It can be modified if the AudioRecord is rerouted.
+ */
+ uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; }
+
/* After it's created the track is not active. Call start() to
* make it active. If set, the callback will start being called.
* If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until
diff --git a/media/libaudioclient/include/AudioTrack.h b/media/libaudioclient/include/AudioTrack.h
index 0358363..16eb225 100644
--- a/media/libaudioclient/include/AudioTrack.h
+++ b/media/libaudioclient/include/AudioTrack.h
@@ -348,7 +348,12 @@
uint32_t channelCount() const { return mChannelCount; }
size_t frameCount() const { return mFrameCount; }
- // TODO consider notificationFrames() if needed
+ /*
+ * Return the period of the notification callback in frames.
+ * This value is set when the AudioTrack is constructed.
+ * It can be modified if the AudioTrack is rerouted.
+ */
+ uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; }
/* Return effective size of audio buffer that an application writes to
* or a negative error if the track is uninitialized.
diff --git a/media/libmedia/IOMX.cpp b/media/libmedia/IOMX.cpp
index 223ca6b..43130eb 100644
--- a/media/libmedia/IOMX.cpp
+++ b/media/libmedia/IOMX.cpp
@@ -549,11 +549,6 @@
virtual status_t dispatchMessage(const omx_message &msg) {
return mBase->dispatchMessage(msg);
}
-
- // TODO: this is temporary, will be removed when quirks move to OMX side
- virtual status_t setQuirks(OMX_U32 quirks) {
- return mBase->setQuirks(quirks);
- }
};
IMPLEMENT_META_INTERFACE(OMX, "android.hardware.IOMX");
@@ -975,17 +970,6 @@
return NO_ERROR;
}
- case SET_QUIRKS:
- {
- CHECK_OMX_INTERFACE(IOMXNode, data, reply);
-
- OMX_U32 quirks = data.readInt32();
-
- reply->writeInt32(setQuirks(quirks));
-
- return NO_ERROR;
- }
-
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libmedia/include/IOMX.h b/media/libmedia/include/IOMX.h
index 62067c7..9a0ada1 100644
--- a/media/libmedia/include/IOMX.h
+++ b/media/libmedia/include/IOMX.h
@@ -170,9 +170,6 @@
OMX_INDEXTYPE *index) = 0;
virtual status_t dispatchMessage(const omx_message &msg) = 0;
-
- // TODO: this is temporary, will be removed when quirks move to OMX side
- virtual status_t setQuirks(OMX_U32 quirks) = 0;
};
struct omx_message {
diff --git a/media/libmedia/omx/1.0/WOmxNode.cpp b/media/libmedia/omx/1.0/WOmxNode.cpp
index 6c92b52..194378c 100644
--- a/media/libmedia/omx/1.0/WOmxNode.cpp
+++ b/media/libmedia/omx/1.0/WOmxNode.cpp
@@ -242,11 +242,6 @@
return status;
}
-// TODO: this is temporary, will be removed when quirks move to OMX side.
-status_t LWOmxNode::setQuirks(OMX_U32 quirks) {
- return toStatusT(mBase->setQuirks(static_cast<uint32_t>(quirks)));;
-}
-
// TWOmxNode
TWOmxNode::TWOmxNode(sp<IOMXNode> const& base) : mBase(base) {
}
@@ -424,11 +419,6 @@
return toStatus(mBase->dispatchMessage(lMsg));
}
-Return<void> TWOmxNode::setQuirks(uint32_t quirks) {
- mBase->setQuirks(static_cast<OMX_U32>(quirks));
- return Void();
-}
-
} // namespace utils
} // namespace V1_0
} // namespace omx
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index d9a5c26..d048777 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -1159,7 +1159,8 @@
}
if (mSource != nullptr) {
if (audio) {
- if (mVideoDecoderError || mSource->getFormat(false /* audio */) == NULL) {
+ if (mVideoDecoderError || mSource->getFormat(false /* audio */) == NULL
+ || mSurface == NULL) {
// When both audio and video have error, or this stream has only audio
// which has error, notify client of error.
notifyListener(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err);
@@ -1169,7 +1170,8 @@
}
mAudioDecoderError = true;
} else {
- if (mAudioDecoderError || mSource->getFormat(true /* audio */) == NULL) {
+ if (mAudioDecoderError || mSource->getFormat(true /* audio */) == NULL
+ || mAudioSink == NULL) {
// When both audio and video have error, or this stream has only video
// which has error, notify client of error.
notifyListener(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err);
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 72645ab..63b9571 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -6307,7 +6307,6 @@
AString mime;
AString componentName;
- uint32_t quirks = 0;
int32_t encoder = false;
if (msg->findString("componentName", &componentName)) {
sp<IMediaCodecList> list = MediaCodecList::getInstance();
@@ -6335,7 +6334,6 @@
for (size_t matchIndex = 0; matchIndex < matchingCodecs.size();
++matchIndex) {
componentName = matchingCodecs[matchIndex];
- quirks = MediaCodecList::getQuirksFor(componentName.c_str());
pid_t tid = gettid();
int prevPriority = androidGetThreadPriority(tid);
@@ -6392,7 +6390,6 @@
mCodec->mFlags |= kFlagPushBlankBuffersToNativeWindowOnShutdown;
}
- omxNode->setQuirks(quirks);
mCodec->mOMX = omx;
mCodec->mOMXNode = omxNode;
mCodec->mCallback->onComponentAllocated(mCodec->mComponentName.c_str());
diff --git a/media/libstagefright/ACodecBufferChannel.cpp b/media/libstagefright/ACodecBufferChannel.cpp
index 40ac986..0d9696f 100644
--- a/media/libstagefright/ACodecBufferChannel.cpp
+++ b/media/libstagefright/ACodecBufferChannel.cpp
@@ -300,8 +300,10 @@
});
size_t destinationBufferSize = maxSize;
size_t heapSize = totalSize + destinationBufferSize;
- mDealer = makeMemoryDealer(heapSize);
- mDecryptDestination = mDealer->allocate(destinationBufferSize);
+ if (heapSize > 0) {
+ mDealer = makeMemoryDealer(heapSize);
+ mDecryptDestination = mDealer->allocate(destinationBufferSize);
+ }
}
std::vector<const BufferInfo> inputBuffers;
for (const BufferAndId &elem : array) {
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index bdc37a5..61b8f9d 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -121,6 +121,7 @@
android.hidl.allocator@1.0 \
android.hidl.memory@1.0 \
android.hardware.media.omx@1.0 \
+ libstagefright_xmlparser@1.0 \
LOCAL_EXPORT_SHARED_LIBRARY_HEADERS := libmedia
diff --git a/media/libstagefright/include/OMX.h b/media/libstagefright/include/OMX.h
index 5b22a2f..4af3d39 100644
--- a/media/libstagefright/include/OMX.h
+++ b/media/libstagefright/include/OMX.h
@@ -20,7 +20,7 @@
#include <media/IOMX.h>
#include <utils/threads.h>
#include <utils/KeyedVector.h>
-
+#include <media/vndk/xmlparser/1.0/MediaCodecsXmlParser.h>
#include "OmxNodeOwner.h"
namespace android {
@@ -54,6 +54,7 @@
private:
Mutex mLock;
OMXMaster *mMaster;
+ MediaCodecsXmlParser mParser;
KeyedVector<wp<IBinder>, sp<OMXNodeInstance> > mLiveNodes;
diff --git a/media/libstagefright/omx/1.0/Omx.cpp b/media/libstagefright/omx/1.0/Omx.cpp
index e5b89da..365ea5d 100644
--- a/media/libstagefright/omx/1.0/Omx.cpp
+++ b/media/libstagefright/omx/1.0/Omx.cpp
@@ -43,7 +43,9 @@
constexpr size_t kMaxNodeInstances = (1 << 16);
-Omx::Omx() : mMaster(new OMXMaster()) {
+Omx::Omx() :
+ mMaster(new OMXMaster()),
+ mParser() {
}
Omx::~Omx() {
@@ -111,6 +113,19 @@
return Void();
}
instance->setHandle(handle);
+ std::vector<AString> quirkVector;
+ if (mParser.getQuirks(name.c_str(), &quirkVector) == OK) {
+ uint32_t quirks = 0;
+ for (const AString quirk : quirkVector) {
+ if (quirk == "requires-allocate-on-input-ports") {
+ quirks |= kRequiresAllocateBufferOnInputPorts;
+ }
+ if (quirk == "requires-allocate-on-output-ports") {
+ quirks |= kRequiresAllocateBufferOnOutputPorts;
+ }
+ }
+ instance->setQuirks(quirks);
+ }
mLiveNodes.add(observer.get(), instance);
observer->linkToDeath(this, 0);
diff --git a/media/libstagefright/omx/1.0/Omx.h b/media/libstagefright/omx/1.0/Omx.h
index 001e8cb..23784aa 100644
--- a/media/libstagefright/omx/1.0/Omx.h
+++ b/media/libstagefright/omx/1.0/Omx.h
@@ -23,6 +23,7 @@
#include "../../include/OMXNodeInstance.h"
#include <android/hardware/media/omx/1.0/IOmx.h>
+#include <media/vndk/xmlparser/1.0/MediaCodecsXmlParser.h>
namespace android {
@@ -76,6 +77,7 @@
Mutex mLock;
KeyedVector<wp<IBase>, sp<OMXNodeInstance> > mLiveNodes;
KeyedVector<OMXNodeInstance*, wp<IBase> > mNode2Observer;
+ MediaCodecsXmlParser mParser;
};
extern "C" IOmx* HIDL_FETCH_IOmx(const char* name);
diff --git a/media/libstagefright/omx/1.0/WOmxNode.cpp b/media/libstagefright/omx/1.0/WOmxNode.cpp
index ea9fb35..1a61007 100644
--- a/media/libstagefright/omx/1.0/WOmxNode.cpp
+++ b/media/libstagefright/omx/1.0/WOmxNode.cpp
@@ -245,11 +245,6 @@
return status;
}
-// TODO: this is temporary, will be removed when quirks move to OMX side.
-status_t LWOmxNode::setQuirks(OMX_U32 quirks) {
- return toStatusT(mBase->setQuirks(static_cast<uint32_t>(quirks)));;
-}
-
// TWOmxNode
TWOmxNode::TWOmxNode(sp<IOMXNode> const& base) : mBase(base) {
}
@@ -427,11 +422,6 @@
return toStatus(mBase->dispatchMessage(lMsg));
}
-Return<void> TWOmxNode::setQuirks(uint32_t quirks) {
- mBase->setQuirks(static_cast<OMX_U32>(quirks));
- return Void();
-}
-
} // namespace implementation
} // namespace V1_0
} // namespace omx
diff --git a/media/libstagefright/omx/1.0/WOmxNode.h b/media/libstagefright/omx/1.0/WOmxNode.h
index 8ca3e67..d715374 100644
--- a/media/libstagefright/omx/1.0/WOmxNode.h
+++ b/media/libstagefright/omx/1.0/WOmxNode.h
@@ -103,9 +103,6 @@
const char *parameter_name,
OMX_INDEXTYPE *index) override;
status_t dispatchMessage(const omx_message &msg) override;
-
- // TODO: this is temporary, will be removed when quirks move to OMX side.
- status_t setQuirks(OMX_U32 quirks) override;
};
struct TWOmxNode : public IOmxNode {
@@ -154,7 +151,6 @@
hidl_string const& parameterName,
getExtensionIndex_cb _hidl_cb) override;
Return<Status> dispatchMessage(Message const& msg) override;
- Return<void> setQuirks(uint32_t quirks) override;
};
} // namespace implementation
diff --git a/media/libstagefright/omx/Android.mk b/media/libstagefright/omx/Android.mk
index b1508dc..90333ef 100644
--- a/media/libstagefright/omx/Android.mk
+++ b/media/libstagefright/omx/Android.mk
@@ -41,6 +41,7 @@
libdl \
libhidlbase \
libhidlmemory \
+ libstagefright_xmlparser@1.0 \
android.hidl.base@1.0 \
android.hidl.memory@1.0 \
android.hardware.media@1.0 \
diff --git a/media/libstagefright/omx/OMX.cpp b/media/libstagefright/omx/OMX.cpp
index bf1418f..8c1141d 100644
--- a/media/libstagefright/omx/OMX.cpp
+++ b/media/libstagefright/omx/OMX.cpp
@@ -37,8 +37,7 @@
// node ids are created by concatenating the pid with a 16-bit counter
static size_t kMaxNodeInstances = (1 << 16);
-OMX::OMX()
- : mMaster(new OMXMaster) {
+OMX::OMX() : mMaster(new OMXMaster), mParser() {
}
OMX::~OMX() {
@@ -119,6 +118,19 @@
return StatusFromOMXError(err);
}
instance->setHandle(handle);
+ std::vector<AString> quirkVector;
+ if (mParser.getQuirks(name, &quirkVector) == OK) {
+ uint32_t quirks = 0;
+ for (const AString quirk : quirkVector) {
+ if (quirk == "requires-allocate-on-input-ports") {
+ quirks |= kRequiresAllocateBufferOnInputPorts;
+ }
+ if (quirk == "requires-allocate-on-output-ports") {
+ quirks |= kRequiresAllocateBufferOnOutputPorts;
+ }
+ }
+ instance->setQuirks(quirks);
+ }
mLiveNodes.add(IInterface::asBinder(observer), instance);
IInterface::asBinder(observer)->linkToDeath(this);
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index 807c2ea..d0f64ca 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -41,21 +41,13 @@
#include <utils/misc.h>
#include <utils/NativeHandle.h>
#include <media/OMXBuffer.h>
+#include <media/vndk/xmlparser/1.0/MediaCodecsXmlParser.h>
#include <hidlmemory/mapping.h>
static const OMX_U32 kPortIndexInput = 0;
static const OMX_U32 kPortIndexOutput = 1;
-// Quirk still supported, even though deprecated
-enum Quirks {
- kRequiresAllocateBufferOnInputPorts = 1,
- kRequiresAllocateBufferOnOutputPorts = 2,
-
- kQuirksMask = kRequiresAllocateBufferOnInputPorts
- | kRequiresAllocateBufferOnOutputPorts,
-};
-
#define CLOGW(fmt, ...) ALOGW("[%p:%s] " fmt, mHandle, mName, ##__VA_ARGS__)
#define CLOG_ERROR_IF(cond, fn, err, fmt, ...) \
@@ -1033,6 +1025,11 @@
}
Mutex::Autolock autoLock(mLock);
+ if (!mSailed) {
+ ALOGE("b/35467458");
+ android_errorWriteLog(0x534e4554, "35467458");
+ return BAD_VALUE;
+ }
switch (omxBuffer.mBufferType) {
case OMXBuffer::kBufferTypePreset:
@@ -1470,6 +1467,11 @@
Mutex::Autolock autoLock(mLock);
+ if (!mSailed) {
+ ALOGE("b/35467458");
+ android_errorWriteLog(0x534e4554, "35467458");
+ return BAD_VALUE;
+ }
BufferMeta *buffer_meta = new BufferMeta(portIndex);
OMX_BUFFERHEADERTYPE *header;
diff --git a/media/ndk/Android.bp b/media/ndk/Android.bp
index e4e3d8f..824872f 100644
--- a/media/ndk/Android.bp
+++ b/media/ndk/Android.bp
@@ -17,7 +17,7 @@
// frameworks/av/include.
ndk_library {
- name: "libmediandk.ndk",
+ name: "libmediandk",
symbol_file: "libmediandk.map.txt",
first_version: "21",
unversioned_until: "current",
diff --git a/media/vndk/Android.bp b/media/vndk/Android.bp
new file mode 100644
index 0000000..a233d6c
--- /dev/null
+++ b/media/vndk/Android.bp
@@ -0,0 +1,4 @@
+subdirs = [
+ "*",
+]
+
diff --git a/media/vndk/xmlparser/1.0/Android.bp b/media/vndk/xmlparser/1.0/Android.bp
new file mode 100644
index 0000000..c48703c
--- /dev/null
+++ b/media/vndk/xmlparser/1.0/Android.bp
@@ -0,0 +1,37 @@
+cc_library_shared {
+
+ name: "libstagefright_xmlparser@1.0",
+
+ srcs: [
+ "MediaCodecsXmlParser.cpp",
+ ],
+
+ include_dirs: [
+ "frameworks/av/media/libstagefright",
+ "frameworks/av/include",
+ ],
+
+ shared_libs: [
+ "libexpat",
+ "libutils",
+ "liblog",
+ "libcutils",
+ "libstagefright_foundation",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ clang: true,
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+
+}
+
diff --git a/media/vndk/xmlparser/1.0/MediaCodecsXmlParser.cpp b/media/vndk/xmlparser/1.0/MediaCodecsXmlParser.cpp
new file mode 100644
index 0000000..84e5514
--- /dev/null
+++ b/media/vndk/xmlparser/1.0/MediaCodecsXmlParser.cpp
@@ -0,0 +1,862 @@
+/*
+ * Copyright 2017, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaCodecsXmlParser"
+#include <utils/Log.h>
+
+#include <media/vndk/xmlparser/1.0/MediaCodecsXmlParser.h>
+
+#include <media/MediaCodecInfo.h>
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/AUtils.h>
+#include <media/stagefright/MediaErrors.h>
+
+#include <sys/stat.h>
+
+#include <expat.h>
+#include <string>
+
+#define MEDIA_CODECS_CONFIG_FILE_PATH_MAX_LENGTH 256
+
+namespace android {
+
+namespace { // Local variables and functions
+
+const char *kProfilingResults =
+ "/data/misc/media/media_codecs_profiling_results.xml";
+
+// Treblized media codec list will be located in /odm/etc or /vendor/etc.
+const char *kConfigLocationList[] =
+ {"/odm/etc", "/vendor/etc", "/etc"};
+constexpr int kConfigLocationListSize =
+ (sizeof(kConfigLocationList) / sizeof(kConfigLocationList[0]));
+
+bool findMediaCodecListFileFullPath(
+ const char *file_name, std::string *out_path) {
+ for (int i = 0; i < kConfigLocationListSize; i++) {
+ *out_path = std::string(kConfigLocationList[i]) + "/" + file_name;
+ struct stat file_stat;
+ if (stat(out_path->c_str(), &file_stat) == 0 &&
+ S_ISREG(file_stat.st_mode)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+// Find TypeInfo by name.
+std::vector<TypeInfo>::iterator findTypeInfo(
+ CodecInfo &codecInfo, const AString &typeName) {
+ return std::find_if(
+ codecInfo.mTypes.begin(), codecInfo.mTypes.end(),
+ [typeName](const auto &typeInfo) {
+ return typeInfo.mName == typeName;
+ });
+}
+
+// Convert a string into a boolean value.
+bool ParseBoolean(const char *s) {
+ if (!strcasecmp(s, "true") || !strcasecmp(s, "yes") || !strcasecmp(s, "y")) {
+ return true;
+ }
+ char *end;
+ unsigned long res = strtoul(s, &end, 10);
+ return *s != '\0' && *end == '\0' && res > 0;
+}
+
+} // unnamed namespace
+
+MediaCodecsXmlParser::MediaCodecsXmlParser() :
+ mInitCheck(NO_INIT),
+ mUpdate(false) {
+ std::string config_file_path;
+ if (findMediaCodecListFileFullPath(
+ "media_codecs.xml", &config_file_path)) {
+ parseTopLevelXMLFile(config_file_path.c_str(), false);
+ } else {
+ mInitCheck = NAME_NOT_FOUND;
+ }
+ if (findMediaCodecListFileFullPath(
+ "media_codecs_performance.xml", &config_file_path)) {
+ parseTopLevelXMLFile(config_file_path.c_str(), true);
+ }
+ parseTopLevelXMLFile(kProfilingResults, true);
+}
+
+void MediaCodecsXmlParser::parseTopLevelXMLFile(
+ const char *codecs_xml, bool ignore_errors) {
+ // get href_base
+ const char *href_base_end = strrchr(codecs_xml, '/');
+ if (href_base_end != NULL) {
+ mHrefBase = AString(codecs_xml, href_base_end - codecs_xml + 1);
+ }
+
+ mInitCheck = OK; // keeping this here for safety
+ mCurrentSection = SECTION_TOPLEVEL;
+ mDepth = 0;
+
+ parseXMLFile(codecs_xml);
+
+ if (mInitCheck != OK) {
+ if (ignore_errors) {
+ mInitCheck = OK;
+ return;
+ }
+ mCodecInfos.clear();
+ return;
+ }
+}
+
+MediaCodecsXmlParser::~MediaCodecsXmlParser() {
+}
+
+status_t MediaCodecsXmlParser::initCheck() const {
+ return mInitCheck;
+}
+
+void MediaCodecsXmlParser::parseXMLFile(const char *path) {
+ FILE *file = fopen(path, "r");
+
+ if (file == NULL) {
+ ALOGW("unable to open media codecs configuration xml file: %s", path);
+ mInitCheck = NAME_NOT_FOUND;
+ return;
+ }
+
+ ALOGV("Start parsing %s", path);
+ XML_Parser parser = ::XML_ParserCreate(NULL);
+ CHECK(parser != NULL);
+
+ ::XML_SetUserData(parser, this);
+ ::XML_SetElementHandler(
+ parser, StartElementHandlerWrapper, EndElementHandlerWrapper);
+
+ const int BUFF_SIZE = 512;
+ while (mInitCheck == OK) {
+ void *buff = ::XML_GetBuffer(parser, BUFF_SIZE);
+ if (buff == NULL) {
+ ALOGE("failed in call to XML_GetBuffer()");
+ mInitCheck = UNKNOWN_ERROR;
+ break;
+ }
+
+ int bytes_read = ::fread(buff, 1, BUFF_SIZE, file);
+ if (bytes_read < 0) {
+ ALOGE("failed in call to read");
+ mInitCheck = ERROR_IO;
+ break;
+ }
+
+ XML_Status status = ::XML_ParseBuffer(parser, bytes_read, bytes_read == 0);
+ if (status != XML_STATUS_OK) {
+ ALOGE("malformed (%s)", ::XML_ErrorString(::XML_GetErrorCode(parser)));
+ mInitCheck = ERROR_MALFORMED;
+ break;
+ }
+
+ if (bytes_read == 0) {
+ break;
+ }
+ }
+
+ ::XML_ParserFree(parser);
+
+ fclose(file);
+ file = NULL;
+}
+
+// static
+void MediaCodecsXmlParser::StartElementHandlerWrapper(
+ void *me, const char *name, const char **attrs) {
+ static_cast<MediaCodecsXmlParser *>(me)->startElementHandler(name, attrs);
+}
+
+// static
+void MediaCodecsXmlParser::EndElementHandlerWrapper(void *me, const char *name) {
+ static_cast<MediaCodecsXmlParser *>(me)->endElementHandler(name);
+}
+
+status_t MediaCodecsXmlParser::includeXMLFile(const char **attrs) {
+ const char *href = NULL;
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (!strcmp(attrs[i], "href")) {
+ if (attrs[i + 1] == NULL) {
+ return -EINVAL;
+ }
+ href = attrs[i + 1];
+ ++i;
+ } else {
+ ALOGE("includeXMLFile: unrecognized attribute: %s", attrs[i]);
+ return -EINVAL;
+ }
+ ++i;
+ }
+
+ // For security reasons and for simplicity, file names can only contain
+ // [a-zA-Z0-9_.] and must start with media_codecs_ and end with .xml
+ for (i = 0; href[i] != '\0'; i++) {
+ if (href[i] == '.' || href[i] == '_' ||
+ (href[i] >= '0' && href[i] <= '9') ||
+ (href[i] >= 'A' && href[i] <= 'Z') ||
+ (href[i] >= 'a' && href[i] <= 'z')) {
+ continue;
+ }
+ ALOGE("invalid include file name: %s", href);
+ return -EINVAL;
+ }
+
+ AString filename = href;
+ if (!filename.startsWith("media_codecs_") ||
+ !filename.endsWith(".xml")) {
+ ALOGE("invalid include file name: %s", href);
+ return -EINVAL;
+ }
+ filename.insert(mHrefBase, 0);
+
+ parseXMLFile(filename.c_str());
+ return mInitCheck;
+}
+
+void MediaCodecsXmlParser::startElementHandler(
+ const char *name, const char **attrs) {
+ if (mInitCheck != OK) {
+ return;
+ }
+
+ bool inType = true;
+
+ if (!strcmp(name, "Include")) {
+ mInitCheck = includeXMLFile(attrs);
+ if (mInitCheck == OK) {
+ mPastSections.push(mCurrentSection);
+ mCurrentSection = SECTION_INCLUDE;
+ }
+ ++mDepth;
+ return;
+ }
+
+ switch (mCurrentSection) {
+ case SECTION_TOPLEVEL:
+ {
+ if (!strcmp(name, "Decoders")) {
+ mCurrentSection = SECTION_DECODERS;
+ } else if (!strcmp(name, "Encoders")) {
+ mCurrentSection = SECTION_ENCODERS;
+ } else if (!strcmp(name, "Settings")) {
+ mCurrentSection = SECTION_SETTINGS;
+ }
+ break;
+ }
+
+ case SECTION_SETTINGS:
+ {
+ if (!strcmp(name, "Setting")) {
+ mInitCheck = addSettingFromAttributes(attrs);
+ }
+ break;
+ }
+
+ case SECTION_DECODERS:
+ {
+ if (!strcmp(name, "MediaCodec")) {
+ mInitCheck =
+ addMediaCodecFromAttributes(false /* encoder */, attrs);
+
+ mCurrentSection = SECTION_DECODER;
+ }
+ break;
+ }
+
+ case SECTION_ENCODERS:
+ {
+ if (!strcmp(name, "MediaCodec")) {
+ mInitCheck =
+ addMediaCodecFromAttributes(true /* encoder */, attrs);
+
+ mCurrentSection = SECTION_ENCODER;
+ }
+ break;
+ }
+
+ case SECTION_DECODER:
+ case SECTION_ENCODER:
+ {
+ if (!strcmp(name, "Quirk")) {
+ mInitCheck = addQuirk(attrs);
+ } else if (!strcmp(name, "Type")) {
+ mInitCheck = addTypeFromAttributes(attrs, (mCurrentSection == SECTION_ENCODER));
+ mCurrentSection =
+ (mCurrentSection == SECTION_DECODER
+ ? SECTION_DECODER_TYPE : SECTION_ENCODER_TYPE);
+ }
+ }
+ inType = false;
+ // fall through
+
+ case SECTION_DECODER_TYPE:
+ case SECTION_ENCODER_TYPE:
+ {
+ // ignore limits and features specified outside of type
+ bool outside = !inType && mCurrentType == mCodecInfos[mCurrentName].mTypes.end();
+ if (outside && (!strcmp(name, "Limit") || !strcmp(name, "Feature"))) {
+ ALOGW("ignoring %s specified outside of a Type", name);
+ } else if (!strcmp(name, "Limit")) {
+ mInitCheck = addLimit(attrs);
+ } else if (!strcmp(name, "Feature")) {
+ mInitCheck = addFeature(attrs);
+ }
+ break;
+ }
+
+ default:
+ break;
+ }
+
+ ++mDepth;
+}
+
+void MediaCodecsXmlParser::endElementHandler(const char *name) {
+ if (mInitCheck != OK) {
+ return;
+ }
+
+ switch (mCurrentSection) {
+ case SECTION_SETTINGS:
+ {
+ if (!strcmp(name, "Settings")) {
+ mCurrentSection = SECTION_TOPLEVEL;
+ }
+ break;
+ }
+
+ case SECTION_DECODERS:
+ {
+ if (!strcmp(name, "Decoders")) {
+ mCurrentSection = SECTION_TOPLEVEL;
+ }
+ break;
+ }
+
+ case SECTION_ENCODERS:
+ {
+ if (!strcmp(name, "Encoders")) {
+ mCurrentSection = SECTION_TOPLEVEL;
+ }
+ break;
+ }
+
+ case SECTION_DECODER_TYPE:
+ case SECTION_ENCODER_TYPE:
+ {
+ if (!strcmp(name, "Type")) {
+ mCurrentSection =
+ (mCurrentSection == SECTION_DECODER_TYPE
+ ? SECTION_DECODER : SECTION_ENCODER);
+
+ mCurrentType = mCodecInfos[mCurrentName].mTypes.end();
+ }
+ break;
+ }
+
+ case SECTION_DECODER:
+ {
+ if (!strcmp(name, "MediaCodec")) {
+ mCurrentSection = SECTION_DECODERS;
+ mCurrentName.clear();
+ }
+ break;
+ }
+
+ case SECTION_ENCODER:
+ {
+ if (!strcmp(name, "MediaCodec")) {
+ mCurrentSection = SECTION_ENCODERS;
+ mCurrentName.clear();
+ }
+ break;
+ }
+
+ case SECTION_INCLUDE:
+ {
+ if (!strcmp(name, "Include") && mPastSections.size() > 0) {
+ mCurrentSection = mPastSections.top();
+ mPastSections.pop();
+ }
+ break;
+ }
+
+ default:
+ break;
+ }
+
+ --mDepth;
+}
+
+status_t MediaCodecsXmlParser::addSettingFromAttributes(const char **attrs) {
+ const char *name = NULL;
+ const char *value = NULL;
+ const char *update = NULL;
+
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (!strcmp(attrs[i], "name")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addSettingFromAttributes: name is null");
+ return -EINVAL;
+ }
+ name = attrs[i + 1];
+ ++i;
+ } else if (!strcmp(attrs[i], "value")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addSettingFromAttributes: value is null");
+ return -EINVAL;
+ }
+ value = attrs[i + 1];
+ ++i;
+ } else if (!strcmp(attrs[i], "update")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addSettingFromAttributes: update is null");
+ return -EINVAL;
+ }
+ update = attrs[i + 1];
+ ++i;
+ } else {
+ ALOGE("addSettingFromAttributes: unrecognized attribute: %s", attrs[i]);
+ return -EINVAL;
+ }
+
+ ++i;
+ }
+
+ if (name == NULL || value == NULL) {
+ ALOGE("addSettingFromAttributes: name or value unspecified");
+ return -EINVAL;
+ }
+
+ mUpdate = (update != NULL) && ParseBoolean(update);
+ if (mUpdate != (mGlobalSettings.count(name) > 0)) {
+ ALOGE("addSettingFromAttributes: updating non-existing setting");
+ return -EINVAL;
+ }
+ mGlobalSettings[name] = value;
+
+ return OK;
+}
+
+status_t MediaCodecsXmlParser::addMediaCodecFromAttributes(
+ bool encoder, const char **attrs) {
+ const char *name = NULL;
+ const char *type = NULL;
+ const char *update = NULL;
+
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (!strcmp(attrs[i], "name")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addMediaCodecFromAttributes: name is null");
+ return -EINVAL;
+ }
+ name = attrs[i + 1];
+ ++i;
+ } else if (!strcmp(attrs[i], "type")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addMediaCodecFromAttributes: type is null");
+ return -EINVAL;
+ }
+ type = attrs[i + 1];
+ ++i;
+ } else if (!strcmp(attrs[i], "update")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addMediaCodecFromAttributes: update is null");
+ return -EINVAL;
+ }
+ update = attrs[i + 1];
+ ++i;
+ } else {
+ ALOGE("addMediaCodecFromAttributes: unrecognized attribute: %s", attrs[i]);
+ return -EINVAL;
+ }
+
+ ++i;
+ }
+
+ if (name == NULL) {
+ ALOGE("addMediaCodecFromAttributes: name not found");
+ return -EINVAL;
+ }
+
+ mUpdate = (update != NULL) && ParseBoolean(update);
+ if (mUpdate != (mCodecInfos.count(name) > 0)) {
+ ALOGE("addMediaCodecFromAttributes: updating non-existing codec or vice versa");
+ return -EINVAL;
+ }
+
+ CodecInfo *info = &mCodecInfos[name];
+ if (mUpdate) {
+ // existing codec
+ mCurrentName = name;
+ mCurrentType = info->mTypes.begin();
+ if (type != NULL) {
+ // existing type
+ mCurrentType = findTypeInfo(*info, type);
+ if (mCurrentType == info->mTypes.end()) {
+ ALOGE("addMediaCodecFromAttributes: updating non-existing type");
+ return -EINVAL;
+ }
+ }
+ } else {
+ // new codec
+ mCurrentName = name;
+ mQuirks[name].clear();
+ info->mTypes.clear();
+ info->mTypes.emplace_back();
+ mCurrentType = --info->mTypes.end();
+ mCurrentType->mName = type;
+ info->mIsEncoder = encoder;
+ }
+
+ return OK;
+}
+
+status_t MediaCodecsXmlParser::addQuirk(const char **attrs) {
+ const char *name = NULL;
+
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (!strcmp(attrs[i], "name")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addQuirk: name is null");
+ return -EINVAL;
+ }
+ name = attrs[i + 1];
+ ++i;
+ } else {
+ ALOGE("addQuirk: unrecognized attribute: %s", attrs[i]);
+ return -EINVAL;
+ }
+
+ ++i;
+ }
+
+ if (name == NULL) {
+ ALOGE("addQuirk: name not found");
+ return -EINVAL;
+ }
+
+ mQuirks[mCurrentName].emplace_back(name);
+ return OK;
+}
+
+status_t MediaCodecsXmlParser::addTypeFromAttributes(const char **attrs, bool encoder) {
+ const char *name = NULL;
+ const char *update = NULL;
+
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (!strcmp(attrs[i], "name")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addTypeFromAttributes: name is null");
+ return -EINVAL;
+ }
+ name = attrs[i + 1];
+ ++i;
+ } else if (!strcmp(attrs[i], "update")) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addTypeFromAttributes: update is null");
+ return -EINVAL;
+ }
+ update = attrs[i + 1];
+ ++i;
+ } else {
+ ALOGE("addTypeFromAttributes: unrecognized attribute: %s", attrs[i]);
+ return -EINVAL;
+ }
+
+ ++i;
+ }
+
+ if (name == NULL) {
+ return -EINVAL;
+ }
+
+ CodecInfo *info = &mCodecInfos[mCurrentName];
+ info->mIsEncoder = encoder;
+ mCurrentType = findTypeInfo(*info, name);
+ if (!mUpdate) {
+ if (mCurrentType != info->mTypes.end()) {
+ ALOGE("addTypeFromAttributes: re-defining existing type without update");
+ return -EINVAL;
+ }
+ info->mTypes.emplace_back();
+ mCurrentType = --info->mTypes.end();
+ } else if (mCurrentType == info->mTypes.end()) {
+ ALOGE("addTypeFromAttributes: updating non-existing type");
+ return -EINVAL;
+ }
+
+ return OK;
+}
+
+static status_t limitFoundMissingAttr(const AString &name, const char *attr, bool found = true) {
+ ALOGE("limit '%s' with %s'%s' attribute", name.c_str(),
+ (found ? "" : "no "), attr);
+ return -EINVAL;
+}
+
+static status_t limitError(const AString &name, const char *msg) {
+ ALOGE("limit '%s' %s", name.c_str(), msg);
+ return -EINVAL;
+}
+
+static status_t limitInvalidAttr(const AString &name, const char *attr, const AString &value) {
+ ALOGE("limit '%s' with invalid '%s' attribute (%s)", name.c_str(),
+ attr, value.c_str());
+ return -EINVAL;
+}
+
+status_t MediaCodecsXmlParser::addLimit(const char **attrs) {
+ sp<AMessage> msg = new AMessage();
+
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addLimit: limit is not given");
+ return -EINVAL;
+ }
+
+ // attributes with values
+ if (!strcmp(attrs[i], "name")
+ || !strcmp(attrs[i], "default")
+ || !strcmp(attrs[i], "in")
+ || !strcmp(attrs[i], "max")
+ || !strcmp(attrs[i], "min")
+ || !strcmp(attrs[i], "range")
+ || !strcmp(attrs[i], "ranges")
+ || !strcmp(attrs[i], "scale")
+ || !strcmp(attrs[i], "value")) {
+ msg->setString(attrs[i], attrs[i + 1]);
+ ++i;
+ } else {
+ ALOGE("addLimit: unrecognized limit: %s", attrs[i]);
+ return -EINVAL;
+ }
+ ++i;
+ }
+
+ AString name;
+ if (!msg->findString("name", &name)) {
+ ALOGE("limit with no 'name' attribute");
+ return -EINVAL;
+ }
+
+ // size, blocks, bitrate, frame-rate, blocks-per-second, aspect-ratio,
+ // measured-frame-rate, measured-blocks-per-second: range
+ // quality: range + default + [scale]
+ // complexity: range + default
+ bool found;
+ if (mCurrentType == mCodecInfos[mCurrentName].mTypes.end()) {
+ ALOGW("ignoring null type");
+ return OK;
+ }
+
+ if (name == "aspect-ratio" || name == "bitrate" || name == "block-count"
+ || name == "blocks-per-second" || name == "complexity"
+ || name == "frame-rate" || name == "quality" || name == "size"
+ || name == "measured-blocks-per-second" || name.startsWith("measured-frame-rate-")) {
+ AString min, max;
+ if (msg->findString("min", &min) && msg->findString("max", &max)) {
+ min.append("-");
+ min.append(max);
+ if (msg->contains("range") || msg->contains("value")) {
+ return limitError(name, "has 'min' and 'max' as well as 'range' or "
+ "'value' attributes");
+ }
+ msg->setString("range", min);
+ } else if (msg->contains("min") || msg->contains("max")) {
+ return limitError(name, "has only 'min' or 'max' attribute");
+ } else if (msg->findString("value", &max)) {
+ min = max;
+ min.append("-");
+ min.append(max);
+ if (msg->contains("range")) {
+ return limitError(name, "has both 'range' and 'value' attributes");
+ }
+ msg->setString("range", min);
+ }
+
+ AString range, scale = "linear", def, in_;
+ if (!msg->findString("range", &range)) {
+ return limitError(name, "with no 'range', 'value' or 'min'/'max' attributes");
+ }
+
+ if ((name == "quality" || name == "complexity") ^
+ (found = msg->findString("default", &def))) {
+ return limitFoundMissingAttr(name, "default", found);
+ }
+ if (name != "quality" && msg->findString("scale", &scale)) {
+ return limitFoundMissingAttr(name, "scale");
+ }
+ if ((name == "aspect-ratio") ^ (found = msg->findString("in", &in_))) {
+ return limitFoundMissingAttr(name, "in", found);
+ }
+
+ if (name == "aspect-ratio") {
+ if (!(in_ == "pixels") && !(in_ == "blocks")) {
+ return limitInvalidAttr(name, "in", in_);
+ }
+ in_.erase(5, 1); // (pixel|block)-aspect-ratio
+ in_.append("-");
+ in_.append(name);
+ name = in_;
+ }
+ if (name == "quality") {
+ mCurrentType->mDetails["quality-scale"] = scale;
+ }
+ if (name == "quality" || name == "complexity") {
+ AString tag = name;
+ tag.append("-default");
+ mCurrentType->mDetails[tag] = def;
+ }
+ AString tag = name;
+ tag.append("-range");
+ mCurrentType->mDetails[tag] = range;
+ } else {
+ AString max, value, ranges;
+ if (msg->contains("default")) {
+ return limitFoundMissingAttr(name, "default");
+ } else if (msg->contains("in")) {
+ return limitFoundMissingAttr(name, "in");
+ } else if ((name == "channel-count" || name == "concurrent-instances") ^
+ (found = msg->findString("max", &max))) {
+ return limitFoundMissingAttr(name, "max", found);
+ } else if (msg->contains("min")) {
+ return limitFoundMissingAttr(name, "min");
+ } else if (msg->contains("range")) {
+ return limitFoundMissingAttr(name, "range");
+ } else if ((name == "sample-rate") ^
+ (found = msg->findString("ranges", &ranges))) {
+ return limitFoundMissingAttr(name, "ranges", found);
+ } else if (msg->contains("scale")) {
+ return limitFoundMissingAttr(name, "scale");
+ } else if ((name == "alignment" || name == "block-size") ^
+ (found = msg->findString("value", &value))) {
+ return limitFoundMissingAttr(name, "value", found);
+ }
+
+ if (max.size()) {
+ AString tag = "max-";
+ tag.append(name);
+ mCurrentType->mDetails[tag] = max;
+ } else if (value.size()) {
+ mCurrentType->mDetails[name] = value;
+ } else if (ranges.size()) {
+ AString tag = name;
+ tag.append("-ranges");
+ mCurrentType->mDetails[tag] = ranges;
+ } else {
+ ALOGW("Ignoring unrecognized limit '%s'", name.c_str());
+ }
+ }
+
+ return OK;
+}
+
+status_t MediaCodecsXmlParser::addFeature(const char **attrs) {
+ size_t i = 0;
+ const char *name = NULL;
+ int32_t optional = -1;
+ int32_t required = -1;
+ const char *value = NULL;
+
+ while (attrs[i] != NULL) {
+ if (attrs[i + 1] == NULL) {
+ ALOGE("addFeature: feature is not given");
+ return -EINVAL;
+ }
+
+ // attributes with values
+ if (!strcmp(attrs[i], "name")) {
+ name = attrs[i + 1];
+ ++i;
+ } else if (!strcmp(attrs[i], "optional") || !strcmp(attrs[i], "required")) {
+ int value = (int)ParseBoolean(attrs[i + 1]);
+ if (!strcmp(attrs[i], "optional")) {
+ optional = value;
+ } else {
+ required = value;
+ }
+ ++i;
+ } else if (!strcmp(attrs[i], "value")) {
+ value = attrs[i + 1];
+ ++i;
+ } else {
+ ALOGE("addFeature: unrecognized attribute: %s", attrs[i]);
+ return -EINVAL;
+ }
+ ++i;
+ }
+ if (name == NULL) {
+ ALOGE("feature with no 'name' attribute");
+ return -EINVAL;
+ }
+
+ if (optional == required && optional != -1) {
+ ALOGE("feature '%s' is both/neither optional and required", name);
+ return -EINVAL;
+ }
+
+ if (mCurrentType == mCodecInfos[mCurrentName].mTypes.end()) {
+ ALOGW("ignoring null type");
+ return OK;
+ }
+ if (value != NULL) {
+ mCurrentType->mStringFeatures[name] = value;
+ } else {
+ mCurrentType->mBoolFeatures[name] = (required == 1) || (optional == 0);
+ }
+ return OK;
+}
+
+void MediaCodecsXmlParser::getGlobalSettings(
+ std::map<AString, AString> *settings) const {
+ settings->clear();
+ settings->insert(mGlobalSettings.begin(), mGlobalSettings.end());
+}
+
+status_t MediaCodecsXmlParser::getCodecInfo(const char *name, CodecInfo *info) const {
+ if (mCodecInfos.count(name) == 0) {
+ ALOGE("Codec not found with name '%s'", name);
+ return NAME_NOT_FOUND;
+ }
+ *info = mCodecInfos.at(name);
+ return OK;
+}
+
+status_t MediaCodecsXmlParser::getQuirks(const char *name, std::vector<AString> *quirks) const {
+ if (mQuirks.count(name) == 0) {
+ ALOGE("Codec not found with name '%s'", name);
+ return NAME_NOT_FOUND;
+ }
+ quirks->clear();
+ quirks->insert(quirks->end(), mQuirks.at(name).begin(), mQuirks.at(name).end());
+ return OK;
+}
+
+} // namespace android
diff --git a/media/vndk/xmlparser/Android.bp b/media/vndk/xmlparser/Android.bp
new file mode 100644
index 0000000..a233d6c
--- /dev/null
+++ b/media/vndk/xmlparser/Android.bp
@@ -0,0 +1,4 @@
+subdirs = [
+ "*",
+]
+
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 7eb179a..42e9c6b 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -117,6 +117,22 @@
Mutex gLock;
wp<AudioFlinger> gAudioFlinger;
+// Keep a strong reference to media.log service around forever.
+// The service is within our parent process so it can never die in a way that we could observe.
+// These two variables are const after initialization.
+static sp<IBinder> sMediaLogServiceAsBinder;
+static sp<IMediaLogService> sMediaLogService;
+
+static pthread_once_t sMediaLogOnce = PTHREAD_ONCE_INIT;
+
+static void sMediaLogInit()
+{
+ sMediaLogServiceAsBinder = defaultServiceManager()->getService(String16("media.log"));
+ if (sMediaLogServiceAsBinder != 0) {
+ sMediaLogService = interface_cast<IMediaLogService>(sMediaLogServiceAsBinder);
+ }
+}
+
// ----------------------------------------------------------------------------
std::string formatToString(audio_format_t format) {
@@ -154,6 +170,7 @@
if (doLog) {
mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
MemoryHeapBase::READ_ONLY);
+ (void) pthread_once(&sMediaLogOnce, sMediaLogInit);
}
// reset battery stats.
@@ -230,15 +247,11 @@
}
// Tell media.log service about any old writers that still need to be unregistered
- if (mLogMemoryDealer != 0) {
- sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
- if (binder != 0) {
- sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
- for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
- sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
- mUnregisteredWriters.pop();
- mediaLogService->unregisterWriter(iMemory);
- }
+ if (sMediaLogService != 0) {
+ for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
+ sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
+ mUnregisteredWriters.pop();
+ sMediaLogService->unregisterWriter(iMemory);
}
}
}
@@ -519,13 +532,10 @@
// append a copy of media.log here by forwarding fd to it, but don't attempt
// to lookup the service if it's not running, as it will block for a second
- if (mLogMemoryDealer != 0) {
- sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
- if (binder != 0) {
- dprintf(fd, "\nmedia.log:\n");
- Vector<String16> args;
- binder->dump(fd, args);
- }
+ if (sMediaLogServiceAsBinder != 0) {
+ dprintf(fd, "\nmedia.log:\n");
+ Vector<String16> args;
+ sMediaLogServiceAsBinder->dump(fd, args);
}
// check for optional arguments
@@ -570,16 +580,11 @@
sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
{
- // If there is no memory allocated for logs, return a dummy writer that does nothing
- if (mLogMemoryDealer == 0) {
+ // If there is no memory allocated for logs, return a dummy writer that does nothing.
+ // Similarly if we can't contact the media.log service, also return a dummy writer.
+ if (mLogMemoryDealer == 0 || sMediaLogService == 0) {
return new NBLog::Writer();
}
- sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
- // Similarly if we can't contact the media.log service, also return a dummy writer
- if (binder == 0) {
- return new NBLog::Writer();
- }
- sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
// If allocation fails, consult the vector of previously unregistered writers
// and garbage-collect one or more them until an allocation succeeds
@@ -590,7 +595,7 @@
// Pick the oldest stale writer to garbage-collect
sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
mUnregisteredWriters.removeAt(0);
- mediaLogService->unregisterWriter(iMemory);
+ sMediaLogService->unregisterWriter(iMemory);
// Now the media.log remote reference to IMemory is gone. When our last local
// reference to IMemory also drops to zero at end of this block,
// the IMemory destructor will deallocate the region from mLogMemoryDealer.
@@ -609,7 +614,7 @@
NBLog::Shared *sharedRawPtr = (NBLog::Shared *) shared->pointer();
new((void *) sharedRawPtr) NBLog::Shared(); // placement new here, but the corresponding
// explicit destructor not needed since it is POD
- mediaLogService->registerWriter(shared, size, name);
+ sMediaLogService->registerWriter(shared, size, name);
return new NBLog::Writer(shared, size);
}
@@ -1544,6 +1549,10 @@
}
bool AudioFlinger::MediaLogNotifier::threadLoop() {
+ // Should already have been checked, but just in case
+ if (sMediaLogService == 0) {
+ return false;
+ }
// Wait until there are pending requests
{
AutoMutex _l(mMutex);
@@ -1555,11 +1564,7 @@
mPendingRequests = false;
}
// Execute the actual MediaLogService binder call and ignore extra requests for a while
- sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
- if (binder != 0) {
- sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
- mediaLogService->requestMergeWakeup();
- }
+ sMediaLogService->requestMergeWakeup();
usleep(kPostTriggerSleepPeriod);
return true;
}
@@ -2044,7 +2049,7 @@
// the first primary output opened designates the primary hw device
if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
- ALOGI("Using module %d has the primary audio interface", module);
+ ALOGI("Using module %d as the primary audio interface", module);
mPrimaryHardwareDev = playbackThread->getOutput()->audioHwDev;
AutoMutex lock(mHardwareLock);
@@ -2133,7 +2138,7 @@
return BAD_VALUE;
}
mMmapThreads.removeItem(output);
- ALOGV("closing mmapThread %p", mmapThread.get());
+ ALOGD("closing mmapThread %p", mmapThread.get());
}
const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
ioDesc->mIoHandle = output;
@@ -2148,7 +2153,7 @@
closeOutputFinish(playbackThread);
}
} else if (mmapThread != 0) {
- ALOGV("mmapThread exit()");
+ ALOGD("mmapThread exit()");
mmapThread->exit();
AudioStreamOut *out = mmapThread->clearOutput();
ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
diff --git a/services/audioflinger/FastThread.cpp b/services/audioflinger/FastThread.cpp
index caf7905..cf9fce3 100644
--- a/services/audioflinger/FastThread.cpp
+++ b/services/audioflinger/FastThread.cpp
@@ -170,7 +170,7 @@
}
int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
if (!(policy == SCHED_FIFO || policy == SCHED_RR)) {
- ALOGE("did not receive expected priority boost");
+ ALOGE("did not receive expected priority boost on time");
}
// This may be overly conservative; there could be times that the normal mixer
// requests such a brief cold idle that it doesn't require resetting this flag.
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 3b1edec..3665875 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -447,6 +447,8 @@
return "RECORD";
case OFFLOAD:
return "OFFLOAD";
+ case MMAP:
+ return "MMAP";
default:
return "unknown";
}
@@ -533,7 +535,7 @@
{
status_t status = initCheck();
if (status == NO_ERROR) {
- ALOGI("AudioFlinger's thread %p ready to run", this);
+ ALOGI("AudioFlinger's thread %p tid=%d ready to run", this, getTid());
} else {
ALOGE("No working audio driver found.");
}
@@ -809,14 +811,15 @@
char buffer[SIZE];
String8 result;
+ dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input",
+ this, mThreadName, getTid(), type(), threadTypeToString(type()));
+
bool locked = AudioFlinger::dumpTryLock(mLock);
if (!locked) {
- dprintf(fd, "thread %p may be deadlocked\n", this);
+ dprintf(fd, " Thread may be deadlocked\n");
}
- dprintf(fd, " Thread name: %s\n", mThreadName);
dprintf(fd, " I/O handle: %d\n", mId);
- dprintf(fd, " TID: %d\n", getTid());
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
@@ -1778,8 +1781,6 @@
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
- dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type()));
-
dumpBase(fd, args);
dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
@@ -4714,34 +4715,42 @@
dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off");
- // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
- // while we are dumping it. It may be inconsistent, but it won't mutate!
- // This is a large object so we place it on the heap.
- // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
- const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState);
- copy->dump(fd);
- delete copy;
+ if (hasFastMixer()) {
+ dprintf(fd, " FastMixer thread %p tid=%d", mFastMixer.get(), mFastMixer->getTid());
+
+ // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
+ // while we are dumping it. It may be inconsistent, but it won't mutate!
+ // This is a large object so we place it on the heap.
+ // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
+ const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState);
+ copy->dump(fd);
+ delete copy;
#ifdef STATE_QUEUE_DUMP
- // Similar for state queue
- StateQueueObserverDump observerCopy = mStateQueueObserverDump;
- observerCopy.dump(fd);
- StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
- mutatorCopy.dump(fd);
+ // Similar for state queue
+ StateQueueObserverDump observerCopy = mStateQueueObserverDump;
+ observerCopy.dump(fd);
+ StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
+ mutatorCopy.dump(fd);
#endif
+#ifdef AUDIO_WATCHDOG
+ if (mAudioWatchdog != 0) {
+ // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
+ AudioWatchdogDump wdCopy = mAudioWatchdogDump;
+ wdCopy.dump(fd);
+ }
+#endif
+
+ } else {
+ dprintf(fd, " No FastMixer\n");
+ }
+
#ifdef TEE_SINK
// Write the tee output to a .wav file
dumpTee(fd, mTeeSource, mId);
#endif
-#ifdef AUDIO_WATCHDOG
- if (mAudioWatchdog != 0) {
- // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
- AudioWatchdogDump wdCopy = mAudioWatchdogDump;
- wdCopy.dump(fd);
- }
-#endif
}
uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
@@ -6791,7 +6800,7 @@
bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
ALOGV("RecordThread::stop");
AutoMutex _l(mLock);
- if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) {
+ if (mActiveTracks.indexOf(recordTrack) < 0 || recordTrack->mState == TrackBase::PAUSING) {
return false;
}
// note that threadLoop may still be processing the track at this point [without lock]
@@ -6805,7 +6814,7 @@
// FIXME incorrect usage of wait: no explicit predicate or loop
mStartStopCond.wait(mLock);
// if we have been restarted, recordTrack is in mActiveTracks here
- if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) {
+ if (exitPending() || mActiveTracks.indexOf(recordTrack) < 0) {
ALOGV("Record stopped OK");
return true;
}
@@ -6872,8 +6881,6 @@
void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
{
- dprintf(fd, "\nInput thread %p:\n", this);
-
dumpBase(fd, args);
AudioStreamIn *input = mInput;
@@ -7585,7 +7592,11 @@
if (mActiveTracks.size() == 0) {
// for the first track, reuse portId and session allocated when the stream was opened
- mHalStream->start();
+ ret = mHalStream->start();
+ if (ret != NO_ERROR) {
+ ALOGE("%s: error mHalStream->start() = %d for first track", __FUNCTION__, ret);
+ return ret;
+ }
portId = mPortId;
sessionId = mSessionId;
mStandby = false;
@@ -7640,6 +7651,7 @@
// abort if start is rejected by audio policy manager
if (ret != NO_ERROR) {
+ ALOGE("%s: error start rejected by AudioPolicyManager = %d", __FUNCTION__, ret);
if (mActiveTracks.size() != 0) {
if (isOutput()) {
AudioSystem::releaseOutput(mId, streamType(), sessionId);
@@ -7935,15 +7947,17 @@
if (isOutput() && mPrevOutDevice != mOutDevice) {
mPrevOutDevice = type;
sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
- if (mCallback != 0) {
- mCallback->onRoutingChanged(deviceId);
+ sp<MmapStreamCallback> callback = mCallback.promote();
+ if (callback != 0) {
+ callback->onRoutingChanged(deviceId);
}
}
if (!isOutput() && mPrevInDevice != mInDevice) {
mPrevInDevice = type;
sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
- if (mCallback != 0) {
- mCallback->onRoutingChanged(deviceId);
+ sp<MmapStreamCallback> callback = mCallback.promote();
+ if (callback != 0) {
+ callback->onRoutingChanged(deviceId);
}
}
return status;
@@ -8058,8 +8072,9 @@
void AudioFlinger::MmapThread::threadLoop_exit()
{
- if (mCallback != 0) {
- mCallback->onTearDown();
+ sp<MmapStreamCallback> callback = mCallback.promote();
+ if (callback != 0) {
+ callback->onTearDown();
}
}
@@ -8107,8 +8122,9 @@
{
for (const sp<MmapTrack> &track : mActiveTracks) {
if (track->isInvalid()) {
- if (mCallback != 0) {
- mCallback->onTearDown();
+ sp<MmapStreamCallback> callback = mCallback.promote();
+ if (callback != 0) {
+ callback->onTearDown();
}
break;
}
@@ -8124,8 +8140,6 @@
void AudioFlinger::MmapThread::dumpInternals(int fd, const Vector<String16>& args)
{
- dprintf(fd, "\nMmap thread %p:\n", this);
-
dumpBase(fd, args);
dprintf(fd, " Attributes: content type %d usage %d source %d\n",
@@ -8285,7 +8299,8 @@
mOutput->stream->setVolume(volume, volume);
- if (mCallback != 0) {
+ sp<MmapStreamCallback> callback = mCallback.promote();
+ if (callback != 0) {
int channelCount;
if (isOutput()) {
channelCount = audio_channel_count_from_out_mask(mChannelMask);
@@ -8296,7 +8311,7 @@
for (int i = 0; i < channelCount; i++) {
values.add(volume);
}
- mCallback->onVolumeChanged(mChannelMask, values);
+ callback->onVolumeChanged(mChannelMask, values);
}
}
}
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 8270e74..7469710 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -31,6 +31,7 @@
RECORD, // Thread class is RecordThread
OFFLOAD, // Thread class is OffloadThread
MMAP // control thread for MMAP stream
+ // If you add any values here, also update ThreadBase::threadTypeToString()
};
static const char *threadTypeToString(type_t type);
@@ -1527,7 +1528,7 @@
audio_session_t mSessionId;
audio_port_handle_t mPortId;
- sp<MmapStreamCallback> mCallback;
+ wp<MmapStreamCallback> mCallback;
sp<StreamHalInterface> mHalStream;
sp<DeviceHalInterface> mHalDevice;
AudioHwDevice* const mAudioHwDev;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
index dbdcca7..bea9f4f 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
@@ -91,8 +91,10 @@
AUDIO_CONFIG_BASE_INITIALIZER;
const audio_patch_handle_t patchHandle = (provider != NULL) ? provider->getPatchHandle() :
AUDIO_PATCH_HANDLE_NONE;
- mClientInterface->onRecordingConfigurationUpdate(event, mSession, mInputSource,
- &mConfig, &deviceConfig, patchHandle);
+ if (patchHandle != AUDIO_PATCH_HANDLE_NONE) {
+ mClientInterface->onRecordingConfigurationUpdate(event, mSession, mInputSource,
+ &mConfig, &deviceConfig, patchHandle);
+ }
}
return mActiveCount;
@@ -126,9 +128,11 @@
AUDIO_CONFIG_BASE_INITIALIZER;
const audio_patch_handle_t patchHandle = (provider != NULL) ? provider->getPatchHandle() :
AUDIO_PATCH_HANDLE_NONE;
- mClientInterface->onRecordingConfigurationUpdate(RECORD_CONFIG_EVENT_START,
- mSession, mInputSource,
- &mConfig, &deviceConfig, patchHandle);
+ if (patchHandle != AUDIO_PATCH_HANDLE_NONE) {
+ mClientInterface->onRecordingConfigurationUpdate(RECORD_CONFIG_EVENT_START,
+ mSession, mInputSource,
+ &mConfig, &deviceConfig, patchHandle);
+ }
}
}
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/PolicyConfigurableDomains.xml b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/PolicyConfigurableDomains.xml
index aa2af0f..b43f83b 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/PolicyConfigurableDomains.xml
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/PolicyConfigurableDomains.xml
@@ -3062,7 +3062,7 @@
<CompoundRule Type="All">
<SelectionCriterionRule SelectionCriterion="TelephonyMode" MatchesWhen="IsNot" Value="InCall"/>
<SelectionCriterionRule SelectionCriterion="TelephonyMode" MatchesWhen="IsNot" Value="InCommunication"/>
- <SelectionCriterionRule SelectionCriterion="ForceUseForMedia" MatchesWhen="Is" Value="ForceNoBtA2dp"/>
+ <SelectionCriterionRule SelectionCriterion="ForceUseForMedia" MatchesWhen="IsNot" Value="ForceNoBtA2dp"/>
<SelectionCriterionRule SelectionCriterion="AvailableOutputDevices" MatchesWhen="Includes" Value="BluetoothA2dp"/>
</CompoundRule>
</Configuration>
@@ -3070,7 +3070,7 @@
<CompoundRule Type="All">
<SelectionCriterionRule SelectionCriterion="TelephonyMode" MatchesWhen="IsNot" Value="InCall"/>
<SelectionCriterionRule SelectionCriterion="TelephonyMode" MatchesWhen="IsNot" Value="InCommunication"/>
- <SelectionCriterionRule SelectionCriterion="ForceUseForMedia" MatchesWhen="Is" Value="ForceNoBtA2dp"/>
+ <SelectionCriterionRule SelectionCriterion="ForceUseForMedia" MatchesWhen="IsNot" Value="ForceNoBtA2dp"/>
<SelectionCriterionRule SelectionCriterion="AvailableOutputDevices" MatchesWhen="Includes" Value="BluetoothA2dpHeadphones"/>
</CompoundRule>
</Configuration>
@@ -3078,7 +3078,7 @@
<CompoundRule Type="All">
<SelectionCriterionRule SelectionCriterion="TelephonyMode" MatchesWhen="IsNot" Value="InCall"/>
<SelectionCriterionRule SelectionCriterion="TelephonyMode" MatchesWhen="IsNot" Value="InCommunication"/>
- <SelectionCriterionRule SelectionCriterion="ForceUseForMedia" MatchesWhen="Is" Value="ForceNoBtA2dp"/>
+ <SelectionCriterionRule SelectionCriterion="ForceUseForMedia" MatchesWhen="IsNot" Value="ForceNoBtA2dp"/>
<SelectionCriterionRule SelectionCriterion="AvailableOutputDevices" MatchesWhen="Includes" Value="BluetoothA2dpSpeaker"/>
</CompoundRule>
</Configuration>
@@ -6472,7 +6472,7 @@
<SelectionCriterionRule SelectionCriterion="TelephonyMode" MatchesWhen="IsNot" Value="InCall"/>
<SelectionCriterionRule SelectionCriterion="TelephonyMode" MatchesWhen="IsNot" Value="InCommunication"/>
<SelectionCriterionRule SelectionCriterion="AvailableOutputDevices" MatchesWhen="Excludes" Value="UsbAccessory"/>
- <SelectionCriterionRule SelectionCriterion="ForceUseForCommunication" MatchesWhen="Is" Value="ForceSpeaker"/>
+ <SelectionCriterionRule SelectionCriterion="ForceUseForMedia" MatchesWhen="IsNot" Value="ForceSpeaker"/>
</CompoundRule>
</CompoundRule>
<SelectionCriterionRule SelectionCriterion="AvailableOutputDevices" MatchesWhen="Includes" Value="UsbDevice"/>
@@ -8416,6 +8416,7 @@
<ConfigurableElement Path="/Policy/policy/usages/assistance_navigation_guidance/applicable_strategy/strategy"/>
<ConfigurableElement Path="/Policy/policy/usages/assistance_sonification/applicable_strategy/strategy"/>
<ConfigurableElement Path="/Policy/policy/usages/game/applicable_strategy/strategy"/>
+ <ConfigurableElement Path="/Policy/policy/usages/assistant/applicable_strategy/strategy"/>
</ConfigurableElements>
<Settings>
<Configuration Name="Calibration">
@@ -8461,6 +8462,9 @@
<ConfigurableElement Path="/Policy/policy/usages/game/applicable_strategy/strategy">
<EnumParameter Name="strategy">media</EnumParameter>
</ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/usages/assistant/applicable_strategy/strategy">
+ <EnumParameter Name="strategy">media</EnumParameter>
+ </ConfigurableElement>
</Configuration>
</Settings>
</ConfigurableDomain>
@@ -8738,6 +8742,7 @@
<ConfigurableElement Path="/Policy/policy/input_sources/fm_tuner/applicable_input_device/mask/loopback"/>
<ConfigurableElement Path="/Policy/policy/input_sources/fm_tuner/applicable_input_device/mask/ip"/>
<ConfigurableElement Path="/Policy/policy/input_sources/fm_tuner/applicable_input_device/mask/bus"/>
+ <ConfigurableElement Path="/Policy/policy/input_sources/fm_tuner/applicable_input_device/mask/stub"/>
</ConfigurableElements>
<Settings>
<Configuration Name="Calibration">
@@ -9428,6 +9433,9 @@
<ConfigurableElement Path="/Policy/policy/input_sources/fm_tuner/applicable_input_device/mask/bus">
<BitParameter Name="bus">0</BitParameter>
</ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/fm_tuner/applicable_input_device/mask/stub">
+ <BitParameter Name="stub">0</BitParameter>
+ </ConfigurableElement>
</Configuration>
</Settings>
</ConfigurableDomain>
@@ -9758,7 +9766,7 @@
</Configuration>
</Settings>
</ConfigurableDomain>
- <ConfigurableDomain Name="DeviceForInputSource.VoiceRecognitionAndHotword" SequenceAware="false">
+ <ConfigurableDomain Name="DeviceForInputSource.VoiceRecognitionAndUnprocessedAndHotword" SequenceAware="false">
<Configurations>
<Configuration Name="ScoHeadset">
<CompoundRule Type="All">
@@ -9790,6 +9798,10 @@
<ConfigurableElement Path="/Policy/policy/input_sources/voice_recognition/applicable_input_device/mask/wired_headset"/>
<ConfigurableElement Path="/Policy/policy/input_sources/voice_recognition/applicable_input_device/mask/usb_device"/>
<ConfigurableElement Path="/Policy/policy/input_sources/voice_recognition/applicable_input_device/mask/builtin_mic"/>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/bluetooth_sco_headset"/>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/wired_headset"/>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/usb_device"/>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/builtin_mic"/>
<ConfigurableElement Path="/Policy/policy/input_sources/hotword/applicable_input_device/mask/bluetooth_sco_headset"/>
<ConfigurableElement Path="/Policy/policy/input_sources/hotword/applicable_input_device/mask/wired_headset"/>
<ConfigurableElement Path="/Policy/policy/input_sources/hotword/applicable_input_device/mask/usb_device"/>
@@ -9809,6 +9821,18 @@
<ConfigurableElement Path="/Policy/policy/input_sources/voice_recognition/applicable_input_device/mask/builtin_mic">
<BitParameter Name="builtin_mic">0</BitParameter>
</ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/bluetooth_sco_headset">
+ <BitParameter Name="bluetooth_sco_headset">1</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/wired_headset">
+ <BitParameter Name="wired_headset">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/usb_device">
+ <BitParameter Name="usb_device">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/builtin_mic">
+ <BitParameter Name="builtin_mic">0</BitParameter>
+ </ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/input_sources/hotword/applicable_input_device/mask/bluetooth_sco_headset">
<BitParameter Name="bluetooth_sco_headset">1</BitParameter>
</ConfigurableElement>
@@ -9835,6 +9859,18 @@
<ConfigurableElement Path="/Policy/policy/input_sources/voice_recognition/applicable_input_device/mask/builtin_mic">
<BitParameter Name="builtin_mic">0</BitParameter>
</ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/bluetooth_sco_headset">
+ <BitParameter Name="bluetooth_sco_headset">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/wired_headset">
+ <BitParameter Name="wired_headset">1</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/usb_device">
+ <BitParameter Name="usb_device">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/builtin_mic">
+ <BitParameter Name="builtin_mic">0</BitParameter>
+ </ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/input_sources/hotword/applicable_input_device/mask/bluetooth_sco_headset">
<BitParameter Name="bluetooth_sco_headset">0</BitParameter>
</ConfigurableElement>
@@ -9861,6 +9897,18 @@
<ConfigurableElement Path="/Policy/policy/input_sources/voice_recognition/applicable_input_device/mask/builtin_mic">
<BitParameter Name="builtin_mic">0</BitParameter>
</ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/bluetooth_sco_headset">
+ <BitParameter Name="bluetooth_sco_headset">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/wired_headset">
+ <BitParameter Name="wired_headset">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/usb_device">
+ <BitParameter Name="usb_device">1</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/builtin_mic">
+ <BitParameter Name="builtin_mic">0</BitParameter>
+ </ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/input_sources/hotword/applicable_input_device/mask/bluetooth_sco_headset">
<BitParameter Name="bluetooth_sco_headset">0</BitParameter>
</ConfigurableElement>
@@ -9887,6 +9935,18 @@
<ConfigurableElement Path="/Policy/policy/input_sources/voice_recognition/applicable_input_device/mask/builtin_mic">
<BitParameter Name="builtin_mic">1</BitParameter>
</ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/bluetooth_sco_headset">
+ <BitParameter Name="bluetooth_sco_headset">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/wired_headset">
+ <BitParameter Name="wired_headset">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/usb_device">
+ <BitParameter Name="usb_device">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/builtin_mic">
+ <BitParameter Name="builtin_mic">1</BitParameter>
+ </ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/input_sources/hotword/applicable_input_device/mask/bluetooth_sco_headset">
<BitParameter Name="bluetooth_sco_headset">0</BitParameter>
</ConfigurableElement>
@@ -9913,6 +9973,18 @@
<ConfigurableElement Path="/Policy/policy/input_sources/voice_recognition/applicable_input_device/mask/builtin_mic">
<BitParameter Name="builtin_mic">0</BitParameter>
</ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/bluetooth_sco_headset">
+ <BitParameter Name="bluetooth_sco_headset">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/wired_headset">
+ <BitParameter Name="wired_headset">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/usb_device">
+ <BitParameter Name="usb_device">0</BitParameter>
+ </ConfigurableElement>
+ <ConfigurableElement Path="/Policy/policy/input_sources/unprocessed/applicable_input_device/mask/builtin_mic">
+ <BitParameter Name="builtin_mic">0</BitParameter>
+ </ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/input_sources/hotword/applicable_input_device/mask/bluetooth_sco_headset">
<BitParameter Name="bluetooth_sco_headset">0</BitParameter>
</ConfigurableElement>
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_accessibility.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_accessibility.pfw
index ecd56b0..eb11980 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_accessibility.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_accessibility.pfw
@@ -375,7 +375,7 @@
TelephonyMode IsNot InCall
TelephonyMode IsNot InCommunication
AvailableOutputDevices Excludes UsbAccessory
- ForceUseForCommunication Is ForceSpeaker
+ ForceUseForMedia IsNot ForceSpeaker
AvailableOutputDevices Includes UsbDevice
component: /Policy/policy/strategies/accessibility/selected_output_devices/mask
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_sonification_respectful.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_sonification_respectful.pfw
index b30aa4c..cee7cd1 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_sonification_respectful.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_sonification_respectful.pfw
@@ -78,7 +78,7 @@
#
TelephonyMode IsNot InCall
TelephonyMode IsNot InCommunication
- ForceUseForMedia Is ForceNoBtA2dp
+ ForceUseForMedia IsNot ForceNoBtA2dp
AvailableOutputDevices Includes BluetoothA2dp
component: /Policy/policy/strategies/sonification_respectful/selected_output_devices/mask
@@ -105,7 +105,7 @@
#
TelephonyMode IsNot InCall
TelephonyMode IsNot InCommunication
- ForceUseForMedia Is ForceNoBtA2dp
+ ForceUseForMedia IsNot ForceNoBtA2dp
AvailableOutputDevices Includes BluetoothA2dpHeadphones
component: /Policy/policy/strategies/sonification_respectful/selected_output_devices/mask
@@ -132,7 +132,7 @@
#
TelephonyMode IsNot InCall
TelephonyMode IsNot InCommunication
- ForceUseForMedia Is ForceNoBtA2dp
+ ForceUseForMedia IsNot ForceNoBtA2dp
AvailableOutputDevices Includes BluetoothA2dpSpeaker
component: /Policy/policy/strategies/sonification_respectful/selected_output_devices/mask
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/strategy_for_usage.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/strategy_for_usage.pfw
index 3f5da13..b3115e7 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/strategy_for_usage.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/strategy_for_usage.pfw
@@ -16,6 +16,7 @@
/Policy/policy/usages/assistance_navigation_guidance/applicable_strategy/strategy = media
/Policy/policy/usages/assistance_sonification/applicable_strategy/strategy = media
/Policy/policy/usages/game/applicable_strategy/strategy = media
+ /Policy/policy/usages/assistant/applicable_strategy/strategy = media
domain: AssistanceAccessibility
conf: Sonification
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Structure/PolicySubsystem.xml b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Structure/PolicySubsystem.xml
index 71b2b62..ad9c356 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Structure/PolicySubsystem.xml
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Structure/PolicySubsystem.xml
@@ -68,7 +68,7 @@
<!--#################### USAGE BEGIN ####################-->
- <ComponentType Name="Usages" Description="associated to audio_stream_type_t definition,
+ <ComponentType Name="Usages" Description="associated to audio_usage_t definition,
identifier mapping must match the value of the enum">
<Component Name="unknown" Type="Usage" Mapping="Amend1:Unknown,Identifier:0"/>
<Component Name="media" Type="Usage" Mapping="Amend1:Media,Identifier:1"/>
@@ -97,6 +97,7 @@
<Component Name="game" Type="Usage" Mapping="Amend1:BluetoothSco,Identifier:14"/>
<Component Name="virtual_source" Type="Usage"
Mapping="Amend1:VirtualSource,Identifier:15"/>
+ <Component Name="assistant" Type="Usage" Mapping="Amend1:Assistant,Identifier:16"/>
</ComponentType>
<!--#################### USAGE END ####################-->
diff --git a/services/camera/libcameraservice/tests/CameraProviderManagerTest.cpp b/services/camera/libcameraservice/tests/CameraProviderManagerTest.cpp
index b18df5f..c1d6e85 100644
--- a/services/camera/libcameraservice/tests/CameraProviderManagerTest.cpp
+++ b/services/camera/libcameraservice/tests/CameraProviderManagerTest.cpp
@@ -352,12 +352,12 @@
metadataCopy.update(ANDROID_CONTROL_SCENE_MODE, &sceneMode, 1);
// However appending from same vendor tag provider should be fine
ASSERT_EQ(OK, metadata.append(secondMetadata));
- // Append from a metadata without vendor tag provider should not be supported
+ // Append from a metadata without vendor tag provider should be supported
CameraMetadata regularMetadata(10, 20);
uint8_t controlMode = ANDROID_CONTROL_MODE_AUTO;
regularMetadata.update(ANDROID_CONTROL_MODE, &controlMode, 1);
- ASSERT_NE(OK, secondMetadata.append(regularMetadata));
- ASSERT_EQ(1u, secondMetadata.entryCount());
+ ASSERT_EQ(OK, secondMetadata.append(regularMetadata));
+ ASSERT_EQ(2u, secondMetadata.entryCount());
ASSERT_EQ(2u, metadata.entryCount());
// Dump
diff --git a/services/oboeservice/AAudioEndpointManager.cpp b/services/oboeservice/AAudioEndpointManager.cpp
new file mode 100644
index 0000000..84fa227
--- /dev/null
+++ b/services/oboeservice/AAudioEndpointManager.cpp
@@ -0,0 +1,87 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <assert.h>
+#include <map>
+#include <mutex>
+#include <utils/Singleton.h>
+
+#include "AAudioEndpointManager.h"
+#include "AAudioServiceEndpoint.h"
+
+using namespace android;
+using namespace aaudio;
+
+ANDROID_SINGLETON_STATIC_INSTANCE(AAudioEndpointManager);
+
+AAudioEndpointManager::AAudioEndpointManager()
+ : Singleton<AAudioEndpointManager>() {
+}
+
+AAudioServiceEndpoint *AAudioEndpointManager::findEndpoint(AAudioService &audioService, int32_t deviceId,
+ aaudio_direction_t direction) {
+ AAudioServiceEndpoint *endpoint = nullptr;
+ std::lock_guard<std::mutex> lock(mLock);
+ switch (direction) {
+ case AAUDIO_DIRECTION_INPUT:
+ endpoint = mInputs[deviceId];
+ break;
+ case AAUDIO_DIRECTION_OUTPUT:
+ endpoint = mOutputs[deviceId];
+ break;
+ default:
+ assert(false); // There are only two possible directions.
+ break;
+ }
+
+ // If we can't find an existing one then open one.
+ ALOGD("AAudioEndpointManager::findEndpoint(), found %p", endpoint);
+ if (endpoint == nullptr) {
+ endpoint = new AAudioServiceEndpoint(audioService);
+ if (endpoint->open(deviceId, direction) != AAUDIO_OK) {
+ ALOGD("AAudioEndpointManager::findEndpoint(), open failed");
+ delete endpoint;
+ endpoint = nullptr;
+ } else {
+ switch(direction) {
+ case AAUDIO_DIRECTION_INPUT:
+ mInputs[deviceId] = endpoint;
+ break;
+ case AAUDIO_DIRECTION_OUTPUT:
+ mOutputs[deviceId] = endpoint;
+ break;
+ }
+ }
+ }
+ return endpoint;
+}
+
+// FIXME add reference counter for serviceEndpoints and removed on last use.
+
+void AAudioEndpointManager::removeEndpoint(AAudioServiceEndpoint *serviceEndpoint) {
+ aaudio_direction_t direction = serviceEndpoint->getDirection();
+ int32_t deviceId = serviceEndpoint->getDeviceId();
+
+ std::lock_guard<std::mutex> lock(mLock);
+ switch(direction) {
+ case AAUDIO_DIRECTION_INPUT:
+ mInputs.erase(deviceId);
+ break;
+ case AAUDIO_DIRECTION_OUTPUT:
+ mOutputs.erase(deviceId);
+ break;
+ }
+}
\ No newline at end of file
diff --git a/services/oboeservice/AAudioEndpointManager.h b/services/oboeservice/AAudioEndpointManager.h
new file mode 100644
index 0000000..48b27f0
--- /dev/null
+++ b/services/oboeservice/AAudioEndpointManager.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_AAUDIO_ENDPOINT_MANAGER_H
+#define AAUDIO_AAUDIO_ENDPOINT_MANAGER_H
+
+#include <map>
+#include <mutex>
+#include <utils/Singleton.h>
+
+#include "binding/AAudioServiceMessage.h"
+#include "AAudioServiceEndpoint.h"
+
+namespace aaudio {
+
+class AAudioEndpointManager : public android::Singleton<AAudioEndpointManager>{
+public:
+ AAudioEndpointManager();
+ ~AAudioEndpointManager() = default;
+
+ /**
+ * Find a service endpoint for the given deviceId and direction.
+ * If an endpoint does not already exist then it will try to create one.
+ *
+ * @param deviceId
+ * @param direction
+ * @return endpoint or nullptr
+ */
+ AAudioServiceEndpoint *findEndpoint(android::AAudioService &audioService,
+ int32_t deviceId,
+ aaudio_direction_t direction);
+
+ void removeEndpoint(AAudioServiceEndpoint *serviceEndpoint);
+
+private:
+
+ std::mutex mLock;
+
+ // We need separate inputs and outputs because they may both have device==0.
+ // TODO review
+ std::map<int32_t, AAudioServiceEndpoint *> mInputs;
+ std::map<int32_t, AAudioServiceEndpoint *> mOutputs;
+
+};
+
+} /* namespace aaudio */
+
+#endif //AAUDIO_AAUDIO_ENDPOINT_MANAGER_H
diff --git a/services/oboeservice/AAudioMixer.cpp b/services/oboeservice/AAudioMixer.cpp
new file mode 100644
index 0000000..70da339
--- /dev/null
+++ b/services/oboeservice/AAudioMixer.cpp
@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AAudioService"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <cstring>
+#include "AAudioMixer.h"
+
+using android::WrappingBuffer;
+using android::FifoBuffer;
+using android::fifo_frames_t;
+
+AAudioMixer::~AAudioMixer() {
+ delete[] mOutputBuffer;
+}
+
+void AAudioMixer::allocate(int32_t samplesPerFrame, int32_t framesPerBurst) {
+ mSamplesPerFrame = samplesPerFrame;
+ mFramesPerBurst = framesPerBurst;
+ int32_t samplesPerBuffer = samplesPerFrame * framesPerBurst;
+ mOutputBuffer = new float[samplesPerBuffer];
+ mBufferSizeInBytes = samplesPerBuffer * sizeof(float);
+}
+
+void AAudioMixer::clear() {
+ memset(mOutputBuffer, 0, mBufferSizeInBytes);
+}
+
+void AAudioMixer::mix(FifoBuffer *fifo, float volume) {
+ WrappingBuffer wrappingBuffer;
+ float *destination = mOutputBuffer;
+ fifo_frames_t framesLeft = mFramesPerBurst;
+
+ // Gather the data from the client. May be in two parts.
+ fifo->getFullDataAvailable(&wrappingBuffer);
+
+ // Mix data in one or two parts.
+ int partIndex = 0;
+ while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
+ fifo_frames_t framesToMix = framesLeft;
+ fifo_frames_t framesAvailable = wrappingBuffer.numFrames[partIndex];
+ if (framesAvailable > 0) {
+ if (framesToMix > framesAvailable) {
+ framesToMix = framesAvailable;
+ }
+ mixPart(destination, (float *)wrappingBuffer.data[partIndex], framesToMix, volume);
+
+ destination += framesToMix * mSamplesPerFrame;
+ framesLeft -= framesToMix;
+ }
+ partIndex++;
+ }
+ fifo->getFifoControllerBase()->advanceReadIndex(mFramesPerBurst - framesLeft);
+ if (framesLeft > 0) {
+ ALOGW("AAudioMixer::mix() UNDERFLOW by %d / %d frames ----- UNDERFLOW !!!!!!!!!!",
+ framesLeft, mFramesPerBurst);
+ }
+}
+
+void AAudioMixer::mixPart(float *destination, float *source, int32_t numFrames, float volume) {
+ int32_t numSamples = numFrames * mSamplesPerFrame;
+ // TODO maybe optimize using SIMD
+ for (int sampleIndex = 0; sampleIndex < numSamples; sampleIndex++) {
+ *destination++ += *source++ * volume;
+ }
+}
+
+float *AAudioMixer::getOutputBuffer() {
+ return mOutputBuffer;
+}
diff --git a/services/oboeservice/AAudioMixer.h b/services/oboeservice/AAudioMixer.h
new file mode 100644
index 0000000..2191183
--- /dev/null
+++ b/services/oboeservice/AAudioMixer.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_AAUDIO_MIXER_H
+#define AAUDIO_AAUDIO_MIXER_H
+
+#include <stdint.h>
+
+#include <aaudio/AAudio.h>
+#include <fifo/FifoBuffer.h>
+
+class AAudioMixer {
+public:
+ AAudioMixer() {}
+ ~AAudioMixer();
+
+ void allocate(int32_t samplesPerFrame, int32_t framesPerBurst);
+
+ void clear();
+
+ void mix(android::FifoBuffer *fifo, float volume);
+
+ void mixPart(float *destination, float *source, int32_t numFrames, float volume);
+
+ float *getOutputBuffer();
+
+private:
+ float *mOutputBuffer = nullptr;
+ int32_t mSamplesPerFrame = 0;
+ int32_t mFramesPerBurst = 0;
+ int32_t mBufferSizeInBytes = 0;
+};
+
+
+#endif //AAUDIO_AAUDIO_MIXER_H
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index 99b0b4d..e4fa1c5 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -18,28 +18,29 @@
//#define LOG_NDEBUG 0
#include <utils/Log.h>
-#include <time.h>
-#include <pthread.h>
+//#include <time.h>
+//#include <pthread.h>
#include <aaudio/AAudioDefinitions.h>
+#include <mediautils/SchedulingPolicyService.h>
+#include <utils/String16.h>
-#include "HandleTracker.h"
-#include "IAAudioService.h"
-#include "AAudioServiceDefinitions.h"
+#include "binding/AAudioServiceMessage.h"
#include "AAudioService.h"
-#include "AAudioServiceStreamFakeHal.h"
+#include "AAudioServiceStreamMMAP.h"
+#include "AAudioServiceStreamShared.h"
+#include "AAudioServiceStreamMMAP.h"
+#include "binding/IAAudioService.h"
+#include "utility/HandleTracker.h"
using namespace android;
using namespace aaudio;
typedef enum
{
- AAUDIO_HANDLE_TYPE_DUMMY1, // TODO remove DUMMYs
- AAUDIO_HANDLE_TYPE_DUMMY2, // make server handles different than client
- AAUDIO_HANDLE_TYPE_STREAM,
- AAUDIO_HANDLE_TYPE_COUNT
+ AAUDIO_HANDLE_TYPE_STREAM
} aaudio_service_handle_type_t;
-static_assert(AAUDIO_HANDLE_TYPE_COUNT <= HANDLE_TRACKER_MAX_TYPES, "Too many handle types.");
+static_assert(AAUDIO_HANDLE_TYPE_STREAM < HANDLE_TRACKER_MAX_TYPES, "Too many handle types.");
android::AAudioService::AAudioService()
: BnAAudioService() {
@@ -48,18 +49,50 @@
AAudioService::~AAudioService() {
}
-aaudio_handle_t AAudioService::openStream(aaudio::AAudioStreamRequest &request,
- aaudio::AAudioStreamConfiguration &configuration) {
- AAudioServiceStreamBase *serviceStream = new AAudioServiceStreamFakeHal();
- ALOGD("AAudioService::openStream(): created serviceStream = %p", serviceStream);
- aaudio_result_t result = serviceStream->open(request, configuration);
- if (result < 0) {
- ALOGE("AAudioService::openStream(): open returned %d", result);
+aaudio_handle_t AAudioService::openStream(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) {
+ aaudio_result_t result = AAUDIO_OK;
+ AAudioServiceStreamBase *serviceStream = nullptr;
+ const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
+ aaudio_sharing_mode_t sharingMode = configurationInput.getSharingMode();
+ ALOGE("AAudioService::openStream(): sharingMode = %d", sharingMode);
+
+ if (sharingMode != AAUDIO_SHARING_MODE_EXCLUSIVE && sharingMode != AAUDIO_SHARING_MODE_SHARED) {
+ ALOGE("AAudioService::openStream(): unrecognized sharing mode = %d", sharingMode);
+ return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ }
+
+ if (sharingMode == AAUDIO_SHARING_MODE_EXCLUSIVE) {
+ ALOGD("AAudioService::openStream(), sharingMode = AAUDIO_SHARING_MODE_EXCLUSIVE");
+ serviceStream = new AAudioServiceStreamMMAP();
+ result = serviceStream->open(request, configurationOutput);
+ if (result != AAUDIO_OK) {
+ // fall back to using a shared stream
+ ALOGD("AAudioService::openStream(), EXCLUSIVE mode failed");
+ delete serviceStream;
+ serviceStream = nullptr;
+ } else {
+ configurationOutput.setSharingMode(AAUDIO_SHARING_MODE_EXCLUSIVE);
+ }
+ }
+
+ // if SHARED requested or if EXCLUSIVE failed
+ if (serviceStream == nullptr) {
+ ALOGD("AAudioService::openStream(), sharingMode = AAUDIO_SHARING_MODE_SHARED");
+ serviceStream = new AAudioServiceStreamShared(*this);
+ result = serviceStream->open(request, configurationOutput);
+ configurationOutput.setSharingMode(AAUDIO_SHARING_MODE_SHARED);
+ }
+
+ if (result != AAUDIO_OK) {
+ delete serviceStream;
+ ALOGE("AAudioService::openStream(): failed, return %d", result);
return result;
} else {
aaudio_handle_t handle = mHandleTracker.put(AAUDIO_HANDLE_TYPE_STREAM, serviceStream);
ALOGD("AAudioService::openStream(): handle = 0x%08X", handle);
if (handle < 0) {
+ ALOGE("AAudioService::openStream(): handle table full");
delete serviceStream;
}
return handle;
@@ -72,7 +105,7 @@
streamHandle);
ALOGD("AAudioService.closeStream(0x%08X)", streamHandle);
if (serviceStream != nullptr) {
- ALOGD("AAudioService::closeStream(): deleting serviceStream = %p", serviceStream);
+ serviceStream->close();
delete serviceStream;
return AAUDIO_OK;
}
@@ -89,27 +122,32 @@
aaudio_handle_t streamHandle,
aaudio::AudioEndpointParcelable &parcelable) {
AAudioServiceStreamBase *serviceStream = convertHandleToServiceStream(streamHandle);
- ALOGD("AAudioService::getStreamDescription(), serviceStream = %p", serviceStream);
if (serviceStream == nullptr) {
+ ALOGE("AAudioService::getStreamDescription(), illegal stream handle = 0x%0x", streamHandle);
return AAUDIO_ERROR_INVALID_HANDLE;
}
- return serviceStream->getDescription(parcelable);
+ ALOGD("AAudioService::getStreamDescription(), handle = 0x%08x", streamHandle);
+ aaudio_result_t result = serviceStream->getDescription(parcelable);
+ ALOGD("AAudioService::getStreamDescription(), result = %d", result);
+ // parcelable.dump();
+ return result;
}
aaudio_result_t AAudioService::startStream(aaudio_handle_t streamHandle) {
AAudioServiceStreamBase *serviceStream = convertHandleToServiceStream(streamHandle);
- ALOGD("AAudioService::startStream(), serviceStream = %p", serviceStream);
if (serviceStream == nullptr) {
+ ALOGE("AAudioService::startStream(), illegal stream handle = 0x%0x", streamHandle);
return AAUDIO_ERROR_INVALID_HANDLE;
}
aaudio_result_t result = serviceStream->start();
+ ALOGD("AAudioService::startStream(), serviceStream->start() returned %d", result);
return result;
}
aaudio_result_t AAudioService::pauseStream(aaudio_handle_t streamHandle) {
AAudioServiceStreamBase *serviceStream = convertHandleToServiceStream(streamHandle);
- ALOGD("AAudioService::pauseStream(), serviceStream = %p", serviceStream);
if (serviceStream == nullptr) {
+ ALOGE("AAudioService::pauseStream(), illegal stream handle = 0x%0x", streamHandle);
return AAUDIO_ERROR_INVALID_HANDLE;
}
aaudio_result_t result = serviceStream->pause();
@@ -118,35 +156,33 @@
aaudio_result_t AAudioService::flushStream(aaudio_handle_t streamHandle) {
AAudioServiceStreamBase *serviceStream = convertHandleToServiceStream(streamHandle);
- ALOGD("AAudioService::flushStream(), serviceStream = %p", serviceStream);
if (serviceStream == nullptr) {
+ ALOGE("AAudioService::flushStream(), illegal stream handle = 0x%0x", streamHandle);
return AAUDIO_ERROR_INVALID_HANDLE;
}
return serviceStream->flush();
}
aaudio_result_t AAudioService::registerAudioThread(aaudio_handle_t streamHandle,
+ pid_t clientProcessId,
pid_t clientThreadId,
int64_t periodNanoseconds) {
AAudioServiceStreamBase *serviceStream = convertHandleToServiceStream(streamHandle);
ALOGD("AAudioService::registerAudioThread(), serviceStream = %p", serviceStream);
if (serviceStream == nullptr) {
- ALOGE("AAudioService::registerAudioThread(), serviceStream == nullptr");
+ ALOGE("AAudioService::registerAudioThread(), illegal stream handle = 0x%0x", streamHandle);
return AAUDIO_ERROR_INVALID_HANDLE;
}
if (serviceStream->getRegisteredThread() != AAudioServiceStreamBase::ILLEGAL_THREAD_ID) {
ALOGE("AAudioService::registerAudioThread(), thread already registered");
- return AAUDIO_ERROR_INVALID_ORDER;
+ return AAUDIO_ERROR_INVALID_STATE;
}
serviceStream->setRegisteredThread(clientThreadId);
- // Boost client thread to SCHED_FIFO
- struct sched_param sp;
- memset(&sp, 0, sizeof(sp));
- sp.sched_priority = 2; // TODO use 'requestPriority' function from frameworks/av/media/utils
- int err = sched_setscheduler(clientThreadId, SCHED_FIFO, &sp);
+ int err = android::requestPriority(clientProcessId, clientThreadId,
+ DEFAULT_AUDIO_PRIORITY, true /* isForApp */);
if (err != 0){
- ALOGE("AAudioService::sched_setscheduler() failed, errno = %d, priority = %d",
- errno, sp.sched_priority);
+ ALOGE("AAudioService::registerAudioThread() failed, errno = %d, priority = %d",
+ errno, DEFAULT_AUDIO_PRIORITY);
return AAUDIO_ERROR_INTERNAL;
} else {
return AAUDIO_OK;
@@ -154,11 +190,13 @@
}
aaudio_result_t AAudioService::unregisterAudioThread(aaudio_handle_t streamHandle,
- pid_t clientThreadId) {
+ pid_t clientProcessId,
+ pid_t clientThreadId) {
AAudioServiceStreamBase *serviceStream = convertHandleToServiceStream(streamHandle);
ALOGI("AAudioService::unregisterAudioThread(), serviceStream = %p", serviceStream);
if (serviceStream == nullptr) {
- ALOGE("AAudioService::unregisterAudioThread(), serviceStream == nullptr");
+ ALOGE("AAudioService::unregisterAudioThread(), illegal stream handle = 0x%0x",
+ streamHandle);
return AAUDIO_ERROR_INVALID_HANDLE;
}
if (serviceStream->getRegisteredThread() != clientThreadId) {
diff --git a/services/oboeservice/AAudioService.h b/services/oboeservice/AAudioService.h
index a520d7a..5a7a2b6 100644
--- a/services/oboeservice/AAudioService.h
+++ b/services/oboeservice/AAudioService.h
@@ -22,17 +22,19 @@
#include <binder/BinderService.h>
-#include <aaudio/AAudioDefinitions.h>
#include <aaudio/AAudio.h>
#include "utility/HandleTracker.h"
-#include "IAAudioService.h"
+#include "binding/IAAudioService.h"
+#include "binding/AAudioServiceInterface.h"
+
#include "AAudioServiceStreamBase.h"
namespace android {
class AAudioService :
public BinderService<AAudioService>,
- public BnAAudioService
+ public BnAAudioService,
+ public aaudio::AAudioServiceInterface
{
friend class BinderService<AAudioService>;
@@ -40,9 +42,9 @@
AAudioService();
virtual ~AAudioService();
- static const char* getServiceName() { return "media.audio_aaudio"; }
+ static const char* getServiceName() { return AAUDIO_SERVICE_NAME; }
- virtual aaudio_handle_t openStream(aaudio::AAudioStreamRequest &request,
+ virtual aaudio_handle_t openStream(const aaudio::AAudioStreamRequest &request,
aaudio::AAudioStreamConfiguration &configuration);
virtual aaudio_result_t closeStream(aaudio_handle_t streamHandle);
@@ -58,9 +60,11 @@
virtual aaudio_result_t flushStream(aaudio_handle_t streamHandle);
virtual aaudio_result_t registerAudioThread(aaudio_handle_t streamHandle,
- pid_t pid, int64_t periodNanoseconds) ;
+ pid_t pid, pid_t tid,
+ int64_t periodNanoseconds) ;
- virtual aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle, pid_t pid);
+ virtual aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
+ pid_t pid, pid_t tid);
private:
@@ -68,6 +72,9 @@
HandleTracker mHandleTracker;
+ enum constants {
+ DEFAULT_AUDIO_PRIORITY = 2
+ };
};
} /* namespace android */
diff --git a/services/oboeservice/AAudioServiceDefinitions.h b/services/oboeservice/AAudioServiceDefinitions.h
deleted file mode 100644
index f98acbf..0000000
--- a/services/oboeservice/AAudioServiceDefinitions.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AAUDIO_AAUDIO_SERVICE_DEFINITIONS_H
-#define AAUDIO_AAUDIO_SERVICE_DEFINITIONS_H
-
-#include <stdint.h>
-
-#include <aaudio/AAudio.h>
-
-#include "binding/RingBufferParcelable.h"
-
-namespace aaudio {
-
-// TODO move this an "include" folder for the service.
-
-struct AAudioMessageTimestamp {
- int64_t position;
- int64_t deviceOffset; // add to client position to get device position
- int64_t timestamp;
-};
-
-typedef enum aaudio_service_event_e : uint32_t {
- AAUDIO_SERVICE_EVENT_STARTED,
- AAUDIO_SERVICE_EVENT_PAUSED,
- AAUDIO_SERVICE_EVENT_FLUSHED,
- AAUDIO_SERVICE_EVENT_CLOSED,
- AAUDIO_SERVICE_EVENT_DISCONNECTED
-} aaudio_service_event_t;
-
-struct AAudioMessageEvent {
- aaudio_service_event_t event;
- int32_t data1;
- int64_t data2;
-};
-
-typedef struct AAudioServiceMessage_s {
- enum class code : uint32_t {
- NOTHING,
- TIMESTAMP,
- EVENT,
- };
-
- code what;
- union {
- AAudioMessageTimestamp timestamp;
- AAudioMessageEvent event;
- };
-} AAudioServiceMessage;
-
-} /* namespace aaudio */
-
-#endif //AAUDIO_AAUDIO_SERVICE_DEFINITIONS_H
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
new file mode 100644
index 0000000..80551c9
--- /dev/null
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -0,0 +1,188 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <algorithm>
+#include <mutex>
+#include <vector>
+
+#include "core/AudioStreamBuilder.h"
+#include "AAudioServiceEndpoint.h"
+#include "AAudioServiceStreamShared.h"
+
+using namespace android; // TODO just import names needed
+using namespace aaudio; // TODO just import names needed
+
+#define MIN_TIMEOUT_NANOS (1000 * AAUDIO_NANOS_PER_MILLISECOND)
+
+// Wait at least this many times longer than the operation should take.
+#define MIN_TIMEOUT_OPERATIONS 4
+
+// The mStreamInternal will use a service interface that does not go through Binder.
+AAudioServiceEndpoint::AAudioServiceEndpoint(AAudioService &audioService)
+ : mStreamInternal(audioService, true)
+ {
+}
+
+AAudioServiceEndpoint::~AAudioServiceEndpoint() {
+}
+
+// Set up an EXCLUSIVE MMAP stream that will be shared.
+aaudio_result_t AAudioServiceEndpoint::open(int32_t deviceId, aaudio_direction_t direction) {
+ AudioStreamBuilder builder;
+ builder.setSharingMode(AAUDIO_SHARING_MODE_EXCLUSIVE);
+ builder.setDeviceId(deviceId);
+ builder.setDirection(direction);
+ aaudio_result_t result = mStreamInternal.open(builder);
+ if (result == AAUDIO_OK) {
+ mMixer.allocate(mStreamInternal.getSamplesPerFrame(), mStreamInternal.getFramesPerBurst());
+ }
+ return result;
+}
+
+aaudio_result_t AAudioServiceEndpoint::close() {
+ return mStreamInternal.close();
+}
+
+// TODO, maybe use an interface to reduce exposure
+aaudio_result_t AAudioServiceEndpoint::registerStream(AAudioServiceStreamShared *sharedStream) {
+ ALOGD("AAudioServiceEndpoint::registerStream(%p)", sharedStream);
+ // TODO use real-time technique to avoid mutex, eg. atomic command FIFO
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ mRegisteredStreams.push_back(sharedStream);
+ return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceEndpoint::unregisterStream(AAudioServiceStreamShared *sharedStream) {
+ ALOGD("AAudioServiceEndpoint::unregisterStream(%p)", sharedStream);
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ mRegisteredStreams.erase(std::remove(mRegisteredStreams.begin(), mRegisteredStreams.end(), sharedStream),
+ mRegisteredStreams.end());
+ return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceEndpoint::startStream(AAudioServiceStreamShared *sharedStream) {
+ // TODO use real-time technique to avoid mutex, eg. atomic command FIFO
+ ALOGD("AAudioServiceEndpoint(): startStream() entering");
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ mRunningStreams.push_back(sharedStream);
+ if (mRunningStreams.size() == 1) {
+ startMixer_l();
+ }
+ return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceEndpoint::stopStream(AAudioServiceStreamShared *sharedStream) {
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ mRunningStreams.erase(std::remove(mRunningStreams.begin(), mRunningStreams.end(), sharedStream),
+ mRunningStreams.end());
+ if (mRunningStreams.size() == 0) {
+ stopMixer_l();
+ }
+ return AAUDIO_OK;
+}
+
+static void *aaudio_mixer_thread_proc(void *context) {
+ AAudioServiceEndpoint *stream = (AAudioServiceEndpoint *) context;
+ //LOGD("AudioStreamAAudio(): oboe_callback_thread, stream = %p", stream);
+ if (stream != NULL) {
+ return stream->callbackLoop();
+ } else {
+ return NULL;
+ }
+}
+
+// Render audio in the application callback and then write the data to the stream.
+void *AAudioServiceEndpoint::callbackLoop() {
+ aaudio_result_t result = AAUDIO_OK;
+
+ ALOGD("AAudioServiceEndpoint(): callbackLoop() entering");
+
+ result = mStreamInternal.requestStart();
+ ALOGD("AAudioServiceEndpoint(): callbackLoop() after requestStart() %d, isPlaying() = %d",
+ result, (int) mStreamInternal.isPlaying());
+
+ // result might be a frame count
+ while (mCallbackEnabled.load() && mStreamInternal.isPlaying() && (result >= 0)) {
+ // Mix data from each active stream.
+ {
+ mMixer.clear();
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ for(AAudioServiceStreamShared *sharedStream : mRunningStreams) {
+ FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
+ float volume = 0.5; // TODO get from system
+ mMixer.mix(fifo, volume);
+ }
+ }
+
+ // Write audio data to stream using a blocking write.
+ ALOGD("AAudioServiceEndpoint(): callbackLoop() write(%d)", getFramesPerBurst());
+ int64_t timeoutNanos = calculateReasonableTimeout(mStreamInternal.getFramesPerBurst());
+ result = mStreamInternal.write(mMixer.getOutputBuffer(), getFramesPerBurst(), timeoutNanos);
+ if (result == AAUDIO_ERROR_DISCONNECTED) {
+ disconnectRegisteredStreams();
+ break;
+ } else if (result != getFramesPerBurst()) {
+ ALOGW("AAudioServiceEndpoint(): callbackLoop() wrote %d / %d",
+ result, getFramesPerBurst());
+ break;
+ }
+ }
+
+ ALOGD("AAudioServiceEndpoint(): callbackLoop() exiting, result = %d, isPlaying() = %d",
+ result, (int) mStreamInternal.isPlaying());
+
+ result = mStreamInternal.requestStop();
+
+ return NULL; // TODO review
+}
+
+aaudio_result_t AAudioServiceEndpoint::startMixer_l() {
+ // Launch the callback loop thread.
+ int64_t periodNanos = mStreamInternal.getFramesPerBurst()
+ * AAUDIO_NANOS_PER_SECOND
+ / getSampleRate();
+ mCallbackEnabled.store(true);
+ return mStreamInternal.createThread(periodNanos, aaudio_mixer_thread_proc, this);
+}
+
+aaudio_result_t AAudioServiceEndpoint::stopMixer_l() {
+ mCallbackEnabled.store(false);
+ return mStreamInternal.joinThread(NULL, calculateReasonableTimeout(mStreamInternal.getFramesPerBurst()));
+}
+
+// TODO Call method in AudioStreamInternal when that callback CL is merged.
+int64_t AAudioServiceEndpoint::calculateReasonableTimeout(int32_t framesPerOperation) {
+
+ // Wait for at least a second or some number of callbacks to join the thread.
+ int64_t timeoutNanoseconds = (MIN_TIMEOUT_OPERATIONS * framesPerOperation * AAUDIO_NANOS_PER_SECOND)
+ / getSampleRate();
+ if (timeoutNanoseconds < MIN_TIMEOUT_NANOS) { // arbitrary number of seconds
+ timeoutNanoseconds = MIN_TIMEOUT_NANOS;
+ }
+ return timeoutNanoseconds;
+}
+
+void AAudioServiceEndpoint::disconnectRegisteredStreams() {
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ for(AAudioServiceStreamShared *sharedStream : mRunningStreams) {
+ sharedStream->onStop();
+ }
+ mRunningStreams.clear();
+ for(AAudioServiceStreamShared *sharedStream : mRegisteredStreams) {
+ sharedStream->onDisconnect();
+ }
+ mRegisteredStreams.clear();
+}
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
new file mode 100644
index 0000000..020d38a
--- /dev/null
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_SERVICE_ENDPOINT_H
+#define AAUDIO_SERVICE_ENDPOINT_H
+
+#include <atomic>
+#include <functional>
+#include <mutex>
+#include <vector>
+
+#include "client/AudioStreamInternal.h"
+#include "binding/AAudioServiceMessage.h"
+#include "AAudioServiceStreamShared.h"
+#include "AAudioServiceStreamMMAP.h"
+#include "AAudioMixer.h"
+#include "AAudioService.h"
+
+namespace aaudio {
+
+class AAudioServiceEndpoint {
+public:
+ explicit AAudioServiceEndpoint(android::AAudioService &audioService);
+ virtual ~AAudioServiceEndpoint();
+
+ aaudio_result_t open(int32_t deviceId, aaudio_direction_t direction);
+
+ int32_t getSampleRate() const { return mStreamInternal.getSampleRate(); }
+ int32_t getSamplesPerFrame() const { return mStreamInternal.getSamplesPerFrame(); }
+ int32_t getFramesPerBurst() const { return mStreamInternal.getFramesPerBurst(); }
+
+ aaudio_result_t registerStream(AAudioServiceStreamShared *sharedStream);
+ aaudio_result_t unregisterStream(AAudioServiceStreamShared *sharedStream);
+ aaudio_result_t startStream(AAudioServiceStreamShared *sharedStream);
+ aaudio_result_t stopStream(AAudioServiceStreamShared *sharedStream);
+ aaudio_result_t close();
+
+ int32_t getDeviceId() const { return mStreamInternal.getDeviceId(); }
+
+ aaudio_direction_t getDirection() const { return mStreamInternal.getDirection(); }
+
+ void disconnectRegisteredStreams();
+
+ void *callbackLoop();
+
+private:
+ aaudio_result_t startMixer_l();
+ aaudio_result_t stopMixer_l();
+
+ int64_t calculateReasonableTimeout(int32_t framesPerOperation);
+
+ AudioStreamInternal mStreamInternal;
+ AAudioMixer mMixer;
+ AAudioServiceStreamMMAP mStreamMMAP;
+
+ std::atomic<bool> mCallbackEnabled;
+
+ std::mutex mLockStreams;
+ std::vector<AAudioServiceStreamShared *> mRegisteredStreams;
+ std::vector<AAudioServiceStreamShared *> mRunningStreams;
+};
+
+} /* namespace aaudio */
+
+
+#endif //AAUDIO_SERVICE_ENDPOINT_H
diff --git a/services/oboeservice/AAudioServiceMain.cpp b/services/oboeservice/AAudioServiceMain.cpp
deleted file mode 100644
index aa89180..0000000
--- a/services/oboeservice/AAudioServiceMain.cpp
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "AAudioService"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-
-#include <stdint.h>
-#include <math.h>
-
-#include <utils/RefBase.h>
-#include <binder/TextOutput.h>
-
-#include <binder/IInterface.h>
-#include <binder/IBinder.h>
-#include <binder/ProcessState.h>
-#include <binder/IServiceManager.h>
-#include <binder/IPCThreadState.h>
-
-#include <cutils/ashmem.h>
-#include <sys/mman.h>
-
-#include "AAudioServiceDefinitions.h"
-#include "IAAudioService.h"
-#include "AAudioService.h"
-
-using namespace android;
-using namespace aaudio;
-
-/**
- * This is used to test the AAudioService as a standalone application.
- * It is not used when the AAudioService is integrated with AudioFlinger.
- */
-int main(int argc, char **argv) {
- printf("Test AAudioService %s\n", argv[1]);
- ALOGD("This is the AAudioService");
-
- defaultServiceManager()->addService(String16("AAudioService"), new AAudioService());
- android::ProcessState::self()->startThreadPool();
- printf("AAudioService service is now ready\n");
- IPCThreadState::self()->joinThreadPool();
- printf("AAudioService service thread joined\n");
-
- return 0;
-}
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index a7938dc..b15043d 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -18,43 +18,138 @@
//#define LOG_NDEBUG 0
#include <utils/Log.h>
-#include "IAAudioService.h"
-#include "AAudioServiceDefinitions.h"
-#include "AAudioServiceStreamBase.h"
-#include "AudioEndpointParcelable.h"
+#include <mutex>
-using namespace android;
-using namespace aaudio;
+#include "binding/IAAudioService.h"
+#include "binding/AAudioServiceMessage.h"
+#include "utility/AudioClock.h"
+
+#include "AAudioServiceStreamBase.h"
+#include "TimestampScheduler.h"
+
+using namespace android; // TODO just import names needed
+using namespace aaudio; // TODO just import names needed
/**
- * Construct the AudioCommandQueues and the AudioDataQueue
- * and fill in the endpoint parcelable.
+ * Base class for streams in the service.
+ * @return
*/
AAudioServiceStreamBase::AAudioServiceStreamBase()
: mUpMessageQueue(nullptr)
-{
- // TODO could fail so move out of constructor
- mUpMessageQueue = new SharedRingBuffer();
- mUpMessageQueue->allocate(sizeof(AAudioServiceMessage), QUEUE_UP_CAPACITY_COMMANDS);
+ , mAAudioThread() {
}
AAudioServiceStreamBase::~AAudioServiceStreamBase() {
- Mutex::Autolock _l(mLockUpMessageQueue);
- delete mUpMessageQueue;
+ close();
}
-void AAudioServiceStreamBase::sendServiceEvent(aaudio_service_event_t event,
- int32_t data1,
- int64_t data2) {
+aaudio_result_t AAudioServiceStreamBase::open(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) {
+ std::lock_guard<std::mutex> lock(mLockUpMessageQueue);
+ if (mUpMessageQueue != nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ } else {
+ mUpMessageQueue = new SharedRingBuffer();
+ return mUpMessageQueue->allocate(sizeof(AAudioServiceMessage), QUEUE_UP_CAPACITY_COMMANDS);
+ }
+}
- Mutex::Autolock _l(mLockUpMessageQueue);
+aaudio_result_t AAudioServiceStreamBase::close() {
+ std::lock_guard<std::mutex> lock(mLockUpMessageQueue);
+ delete mUpMessageQueue;
+ mUpMessageQueue = nullptr;
+ return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceStreamBase::start() {
+ sendServiceEvent(AAUDIO_SERVICE_EVENT_STARTED);
+ mState = AAUDIO_STREAM_STATE_STARTED;
+ mThreadEnabled.store(true);
+ return mAAudioThread.start(this);
+}
+
+aaudio_result_t AAudioServiceStreamBase::pause() {
+
+ sendCurrentTimestamp();
+ mThreadEnabled.store(false);
+ aaudio_result_t result = mAAudioThread.stop();
+ if (result != AAUDIO_OK) {
+ processError();
+ return result;
+ }
+ sendServiceEvent(AAUDIO_SERVICE_EVENT_PAUSED);
+ mState = AAUDIO_STREAM_STATE_PAUSED;
+ return result;
+}
+
+// implement Runnable
+void AAudioServiceStreamBase::run() {
+ ALOGD("AAudioServiceStreamMMAP::run() entering ----------------");
+ TimestampScheduler timestampScheduler;
+ timestampScheduler.setBurstPeriod(mFramesPerBurst, mSampleRate);
+ timestampScheduler.start(AudioClock::getNanoseconds());
+ int64_t nextTime = timestampScheduler.nextAbsoluteTime();
+ while(mThreadEnabled.load()) {
+ if (AudioClock::getNanoseconds() >= nextTime) {
+ aaudio_result_t result = sendCurrentTimestamp();
+ if (result != AAUDIO_OK) {
+ break;
+ }
+ nextTime = timestampScheduler.nextAbsoluteTime();
+ } else {
+ // Sleep until it is time to send the next timestamp.
+ AudioClock::sleepUntilNanoTime(nextTime);
+ }
+ }
+ ALOGD("AAudioServiceStreamMMAP::run() exiting ----------------");
+}
+
+void AAudioServiceStreamBase::processError() {
+ sendServiceEvent(AAUDIO_SERVICE_EVENT_DISCONNECTED);
+}
+
+aaudio_result_t AAudioServiceStreamBase::sendServiceEvent(aaudio_service_event_t event,
+ double dataDouble,
+ int64_t dataLong) {
AAudioServiceMessage command;
command.what = AAudioServiceMessage::code::EVENT;
command.event.event = event;
- command.event.data1 = data1;
- command.event.data2 = data2;
- mUpMessageQueue->getFifoBuffer()->write(&command, 1);
+ command.event.dataDouble = dataDouble;
+ command.event.dataLong = dataLong;
+ return writeUpMessageQueue(&command);
+}
+
+aaudio_result_t AAudioServiceStreamBase::writeUpMessageQueue(AAudioServiceMessage *command) {
+ std::lock_guard<std::mutex> lock(mLockUpMessageQueue);
+ int32_t count = mUpMessageQueue->getFifoBuffer()->write(command, 1);
+ if (count != 1) {
+ ALOGE("writeUpMessageQueue(): Queue full. Did client die?");
+ return AAUDIO_ERROR_WOULD_BLOCK;
+ } else {
+ return AAUDIO_OK;
+ }
+}
+
+aaudio_result_t AAudioServiceStreamBase::sendCurrentTimestamp() {
+ AAudioServiceMessage command;
+ aaudio_result_t result = getFreeRunningPosition(&command.timestamp.position,
+ &command.timestamp.timestamp);
+ if (result == AAUDIO_OK) {
+ command.what = AAudioServiceMessage::code::TIMESTAMP;
+ result = writeUpMessageQueue(&command);
+ }
+ return result;
}
+/**
+ * Get an immutable description of the in-memory queues
+ * used to communicate with the underlying HAL or Service.
+ */
+aaudio_result_t AAudioServiceStreamBase::getDescription(AudioEndpointParcelable &parcelable) {
+ // Gather information on the message queue.
+ mUpMessageQueue->fillParcelable(parcelable,
+ parcelable.mUpMessageQueueParcelable);
+ return getDownDataDescription(parcelable);
+}
\ No newline at end of file
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index 7a812f9..91eec35 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -17,13 +17,15 @@
#ifndef AAUDIO_AAUDIO_SERVICE_STREAM_BASE_H
#define AAUDIO_AAUDIO_SERVICE_STREAM_BASE_H
-#include <utils/Mutex.h>
+#include <mutex>
-#include "IAAudioService.h"
-#include "AAudioServiceDefinitions.h"
#include "fifo/FifoBuffer.h"
+#include "binding/IAAudioService.h"
+#include "binding/AudioEndpointParcelable.h"
+#include "binding/AAudioServiceMessage.h"
+#include "utility/AAudioUtilities.h"
+
#include "SharedRingBuffer.h"
-#include "AudioEndpointParcelable.h"
#include "AAudioThread.h"
namespace aaudio {
@@ -32,7 +34,11 @@
// This should be way more than we need.
#define QUEUE_UP_CAPACITY_COMMANDS (128)
-class AAudioServiceStreamBase {
+/**
+ * Base class for a stream in the AAudio service.
+ */
+class AAudioServiceStreamBase
+ : public Runnable {
public:
AAudioServiceStreamBase();
@@ -42,16 +48,14 @@
ILLEGAL_THREAD_ID = 0
};
- /**
- * Fill in a parcelable description of stream.
- */
- virtual aaudio_result_t getDescription(aaudio::AudioEndpointParcelable &parcelable) = 0;
-
+ // -------------------------------------------------------------------
/**
* Open the device.
*/
- virtual aaudio_result_t open(aaudio::AAudioStreamRequest &request,
- aaudio::AAudioStreamConfiguration &configuration) = 0;
+ virtual aaudio_result_t open(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) = 0;
+
+ virtual aaudio_result_t close();
/**
* Start the flow of data.
@@ -68,39 +72,69 @@
*/
virtual aaudio_result_t flush() = 0;
- virtual aaudio_result_t close() = 0;
+ // -------------------------------------------------------------------
- virtual void sendCurrentTimestamp() = 0;
+ /**
+ * Send a message to the client.
+ */
+ aaudio_result_t sendServiceEvent(aaudio_service_event_t event,
+ double dataDouble = 0.0,
+ int64_t dataLong = 0);
- int32_t getFramesPerBurst() {
- return mFramesPerBurst;
- }
+ /**
+ * Fill in a parcelable description of stream.
+ */
+ aaudio_result_t getDescription(AudioEndpointParcelable &parcelable);
- virtual void sendServiceEvent(aaudio_service_event_t event,
- int32_t data1 = 0,
- int64_t data2 = 0);
- virtual void setRegisteredThread(pid_t pid) {
+ void setRegisteredThread(pid_t pid) {
mRegisteredClientThread = pid;
}
- virtual pid_t getRegisteredThread() {
+ pid_t getRegisteredThread() const {
return mRegisteredClientThread;
}
+ int32_t getFramesPerBurst() const {
+ return mFramesPerBurst;
+ }
+
+ int32_t calculateBytesPerFrame() const {
+ return mSamplesPerFrame * AAudioConvert_formatToSizeInBytes(mAudioFormat);
+ }
+
+ void run() override; // to implement Runnable
+
+ void processError();
+
protected:
+ aaudio_result_t writeUpMessageQueue(AAudioServiceMessage *command);
+
+ aaudio_result_t sendCurrentTimestamp();
+
+ virtual aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) = 0;
+
+ virtual aaudio_result_t getDownDataDescription(AudioEndpointParcelable &parcelable) = 0;
+
+ aaudio_stream_state_t mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
pid_t mRegisteredClientThread = ILLEGAL_THREAD_ID;
SharedRingBuffer* mUpMessageQueue;
+ std::mutex mLockUpMessageQueue;
- int32_t mSampleRate = 0;
- int32_t mBytesPerFrame = 0;
+ AAudioThread mAAudioThread;
+ // This is used by one thread to tell another thread to exit. So it must be atomic.
+ std::atomic<bool> mThreadEnabled;
+
+
+ int mAudioDataFileDescriptor = -1;
+
+ aaudio_audio_format_t mAudioFormat = AAUDIO_FORMAT_UNSPECIFIED;
int32_t mFramesPerBurst = 0;
- int32_t mCapacityInFrames = 0;
- int32_t mCapacityInBytes = 0;
-
- android::Mutex mLockUpMessageQueue;
+ int32_t mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+ int32_t mSampleRate = AAUDIO_UNSPECIFIED;
+ int32_t mCapacityInFrames = AAUDIO_UNSPECIFIED;
};
} /* namespace aaudio */
diff --git a/services/oboeservice/AAudioServiceStreamExclusive.h b/services/oboeservice/AAudioServiceStreamExclusive.h
new file mode 100644
index 0000000..db382a3
--- /dev/null
+++ b/services/oboeservice/AAudioServiceStreamExclusive.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_AAUDIO_SERVICE_STREAM_EXCLUSIVE_H
+#define AAUDIO_AAUDIO_SERVICE_STREAM_EXCLUSIVE_H
+
+#include "AAudioServiceStreamMMAP.h"
+
+namespace aaudio {
+
+/**
+ * Exclusive mode stream in the AAudio service.
+ *
+ * This is currently a stub.
+ * We may move code from AAudioServiceStreamMMAP into this class.
+ * If not, then it will be removed.
+ */
+class AAudioServiceStreamExclusive : public AAudioServiceStreamMMAP {
+
+public:
+ AAudioServiceStreamExclusive() {};
+ virtual ~AAudioServiceStreamExclusive() = default;
+};
+
+} /* namespace aaudio */
+
+#endif //AAUDIO_AAUDIO_SERVICE_STREAM_EXCLUSIVE_H
diff --git a/services/oboeservice/AAudioServiceStreamFakeHal.cpp b/services/oboeservice/AAudioServiceStreamFakeHal.cpp
deleted file mode 100644
index 71d3542..0000000
--- a/services/oboeservice/AAudioServiceStreamFakeHal.cpp
+++ /dev/null
@@ -1,203 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "AAudioService"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-
-#include <atomic>
-
-#include "AudioClock.h"
-#include "AudioEndpointParcelable.h"
-
-#include "AAudioServiceStreamBase.h"
-#include "AAudioServiceStreamFakeHal.h"
-
-#include "FakeAudioHal.h"
-
-using namespace android;
-using namespace aaudio;
-
-// HACK values for Marlin
-#define CARD_ID 0
-#define DEVICE_ID 19
-
-/**
- * Construct the audio message queuues and message queues.
- */
-
-AAudioServiceStreamFakeHal::AAudioServiceStreamFakeHal()
- : AAudioServiceStreamBase()
- , mStreamId(nullptr)
- , mPreviousFrameCounter(0)
- , mAAudioThread()
-{
-}
-
-AAudioServiceStreamFakeHal::~AAudioServiceStreamFakeHal() {
- ALOGD("AAudioServiceStreamFakeHal::~AAudioServiceStreamFakeHal() call close()");
- close();
-}
-
-aaudio_result_t AAudioServiceStreamFakeHal::open(aaudio::AAudioStreamRequest &request,
- aaudio::AAudioStreamConfiguration &configurationOutput) {
- // Open stream on HAL and pass information about the ring buffer to the client.
- mmap_buffer_info mmapInfo;
- aaudio_result_t error;
-
- // Open HAL
- int bufferCapacity = request.getConfiguration().getBufferCapacity();
- error = fake_hal_open(CARD_ID, DEVICE_ID, bufferCapacity, &mStreamId);
- if(error < 0) {
- ALOGE("Could not open card %d, device %d", CARD_ID, DEVICE_ID);
- return error;
- }
-
- // Get information about the shared audio buffer.
- error = fake_hal_get_mmap_info(mStreamId, &mmapInfo);
- if (error < 0) {
- ALOGE("fake_hal_get_mmap_info returned %d", error);
- fake_hal_close(mStreamId);
- mStreamId = nullptr;
- return error;
- }
- mHalFileDescriptor = mmapInfo.fd;
- mFramesPerBurst = mmapInfo.burst_size_in_frames;
- mCapacityInFrames = mmapInfo.buffer_capacity_in_frames;
- mCapacityInBytes = mmapInfo.buffer_capacity_in_bytes;
- mSampleRate = mmapInfo.sample_rate;
- mBytesPerFrame = mmapInfo.channel_count * sizeof(int16_t); // FIXME based on data format
- ALOGD("AAudioServiceStreamFakeHal::open() mmapInfo.burst_size_in_frames = %d",
- mmapInfo.burst_size_in_frames);
- ALOGD("AAudioServiceStreamFakeHal::open() mmapInfo.buffer_capacity_in_frames = %d",
- mmapInfo.buffer_capacity_in_frames);
- ALOGD("AAudioServiceStreamFakeHal::open() mmapInfo.buffer_capacity_in_bytes = %d",
- mmapInfo.buffer_capacity_in_bytes);
-
- // Fill in AAudioStreamConfiguration
- configurationOutput.setSampleRate(mSampleRate);
- configurationOutput.setSamplesPerFrame(mmapInfo.channel_count);
- configurationOutput.setAudioFormat(AAUDIO_FORMAT_PCM_I16);
-
- return AAUDIO_OK;
-}
-
-/**
- * Get an immutable description of the in-memory queues
- * used to communicate with the underlying HAL or Service.
- */
-aaudio_result_t AAudioServiceStreamFakeHal::getDescription(AudioEndpointParcelable &parcelable) {
- // Gather information on the message queue.
- mUpMessageQueue->fillParcelable(parcelable,
- parcelable.mUpMessageQueueParcelable);
-
- // Gather information on the data queue.
- // TODO refactor into a SharedRingBuffer?
- int fdIndex = parcelable.addFileDescriptor(mHalFileDescriptor, mCapacityInBytes);
- parcelable.mDownDataQueueParcelable.setupMemory(fdIndex, 0, mCapacityInBytes);
- parcelable.mDownDataQueueParcelable.setBytesPerFrame(mBytesPerFrame);
- parcelable.mDownDataQueueParcelable.setFramesPerBurst(mFramesPerBurst);
- parcelable.mDownDataQueueParcelable.setCapacityInFrames(mCapacityInFrames);
- return AAUDIO_OK;
-}
-
-/**
- * Start the flow of data.
- */
-aaudio_result_t AAudioServiceStreamFakeHal::start() {
- if (mStreamId == nullptr) return AAUDIO_ERROR_NULL;
- aaudio_result_t result = fake_hal_start(mStreamId);
- sendServiceEvent(AAUDIO_SERVICE_EVENT_STARTED);
- mState = AAUDIO_STREAM_STATE_STARTED;
- if (result == AAUDIO_OK) {
- mThreadEnabled.store(true);
- result = mAAudioThread.start(this);
- }
- return result;
-}
-
-/**
- * Stop the flow of data such that start() can resume with loss of data.
- */
-aaudio_result_t AAudioServiceStreamFakeHal::pause() {
- if (mStreamId == nullptr) return AAUDIO_ERROR_NULL;
- sendCurrentTimestamp();
- aaudio_result_t result = fake_hal_pause(mStreamId);
- sendServiceEvent(AAUDIO_SERVICE_EVENT_PAUSED);
- mState = AAUDIO_STREAM_STATE_PAUSED;
- mFramesRead.reset32();
- ALOGD("AAudioServiceStreamFakeHal::pause() sent AAUDIO_SERVICE_EVENT_PAUSED");
- mThreadEnabled.store(false);
- result = mAAudioThread.stop();
- return result;
-}
-
-/**
- * Discard any data held by the underlying HAL or Service.
- */
-aaudio_result_t AAudioServiceStreamFakeHal::flush() {
- if (mStreamId == nullptr) return AAUDIO_ERROR_NULL;
- // TODO how do we flush an MMAP/NOIRQ buffer? sync pointers?
- ALOGD("AAudioServiceStreamFakeHal::pause() send AAUDIO_SERVICE_EVENT_FLUSHED");
- sendServiceEvent(AAUDIO_SERVICE_EVENT_FLUSHED);
- mState = AAUDIO_STREAM_STATE_FLUSHED;
- return AAUDIO_OK;
-}
-
-aaudio_result_t AAudioServiceStreamFakeHal::close() {
- aaudio_result_t result = AAUDIO_OK;
- if (mStreamId != nullptr) {
- result = fake_hal_close(mStreamId);
- mStreamId = nullptr;
- }
- return result;
-}
-
-void AAudioServiceStreamFakeHal::sendCurrentTimestamp() {
- int frameCounter = 0;
- int error = fake_hal_get_frame_counter(mStreamId, &frameCounter);
- if (error < 0) {
- ALOGE("AAudioServiceStreamFakeHal::sendCurrentTimestamp() error %d",
- error);
- } else if (frameCounter != mPreviousFrameCounter) {
- AAudioServiceMessage command;
- command.what = AAudioServiceMessage::code::TIMESTAMP;
- mFramesRead.update32(frameCounter);
- command.timestamp.position = mFramesRead.get();
- ALOGD("AAudioServiceStreamFakeHal::sendCurrentTimestamp() HAL frames = %d, pos = %d",
- frameCounter, (int)mFramesRead.get());
- command.timestamp.timestamp = AudioClock::getNanoseconds();
- mUpMessageQueue->getFifoBuffer()->write(&command, 1);
- mPreviousFrameCounter = frameCounter;
- }
-}
-
-// implement Runnable
-void AAudioServiceStreamFakeHal::run() {
- TimestampScheduler timestampScheduler;
- timestampScheduler.setBurstPeriod(mFramesPerBurst, mSampleRate);
- timestampScheduler.start(AudioClock::getNanoseconds());
- while(mThreadEnabled.load()) {
- int64_t nextTime = timestampScheduler.nextAbsoluteTime();
- if (AudioClock::getNanoseconds() >= nextTime) {
- sendCurrentTimestamp();
- } else {
- // Sleep until it is time to send the next timestamp.
- AudioClock::sleepUntilNanoTime(nextTime);
- }
- }
-}
-
diff --git a/services/oboeservice/AAudioServiceStreamFakeHal.h b/services/oboeservice/AAudioServiceStreamFakeHal.h
deleted file mode 100644
index e9480fb..0000000
--- a/services/oboeservice/AAudioServiceStreamFakeHal.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AAUDIO_AAUDIO_SERVICE_STREAM_FAKE_HAL_H
-#define AAUDIO_AAUDIO_SERVICE_STREAM_FAKE_HAL_H
-
-#include "AAudioServiceDefinitions.h"
-#include "AAudioServiceStreamBase.h"
-#include "FakeAudioHal.h"
-#include "MonotonicCounter.h"
-#include "AudioEndpointParcelable.h"
-#include "TimestampScheduler.h"
-
-namespace aaudio {
-
-class AAudioServiceStreamFakeHal
- : public AAudioServiceStreamBase
- , public Runnable {
-
-public:
- AAudioServiceStreamFakeHal();
- virtual ~AAudioServiceStreamFakeHal();
-
- virtual aaudio_result_t getDescription(AudioEndpointParcelable &parcelable) override;
-
- virtual aaudio_result_t open(aaudio::AAudioStreamRequest &request,
- aaudio::AAudioStreamConfiguration &configurationOutput) override;
-
- /**
- * Start the flow of data.
- */
- virtual aaudio_result_t start() override;
-
- /**
- * Stop the flow of data such that start() can resume with loss of data.
- */
- virtual aaudio_result_t pause() override;
-
- /**
- * Discard any data held by the underlying HAL or Service.
- */
- virtual aaudio_result_t flush() override;
-
- virtual aaudio_result_t close() override;
-
- void sendCurrentTimestamp();
-
- virtual void run() override; // to implement Runnable
-
-private:
- fake_hal_stream_ptr mStreamId; // Move to HAL
-
- MonotonicCounter mFramesWritten;
- MonotonicCounter mFramesRead;
- int mHalFileDescriptor = -1;
- int mPreviousFrameCounter = 0; // from HAL
-
- aaudio_stream_state_t mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
-
- AAudioThread mAAudioThread;
- std::atomic<bool> mThreadEnabled;
-};
-
-} // namespace aaudio
-
-#endif //AAUDIO_AAUDIO_SERVICE_STREAM_FAKE_HAL_H
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
new file mode 100644
index 0000000..b70c625
--- /dev/null
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -0,0 +1,269 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AAudioService"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <atomic>
+#include <stdint.h>
+
+#include <utils/String16.h>
+#include <media/nbaio/AudioStreamOutSink.h>
+#include <media/MmapStreamInterface.h>
+
+#include "AAudioServiceStreamBase.h"
+#include "AAudioServiceStreamMMAP.h"
+#include "binding/AudioEndpointParcelable.h"
+#include "SharedMemoryProxy.h"
+#include "utility/AAudioUtilities.h"
+
+using namespace android;
+using namespace aaudio;
+
+#define AAUDIO_BUFFER_CAPACITY_MIN 4 * 512
+#define AAUDIO_SAMPLE_RATE_DEFAULT 48000
+
+/**
+ * Stream that uses an MMAP buffer.
+ */
+
+AAudioServiceStreamMMAP::AAudioServiceStreamMMAP()
+ : AAudioServiceStreamBase()
+ , mMmapStreamCallback(new MyMmapStreamCallback(*this))
+ , mPreviousFrameCounter(0)
+ , mMmapStream(nullptr) {
+}
+
+AAudioServiceStreamMMAP::~AAudioServiceStreamMMAP() {
+ close();
+}
+
+aaudio_result_t AAudioServiceStreamMMAP::close() {
+ ALOGD("AAudioServiceStreamMMAP::close() called, %p", mMmapStream.get());
+ mMmapStream.clear(); // TODO review. Is that all we have to do?
+ return AAudioServiceStreamBase::close();
+}
+
+// Open stream on HAL and pass information about the shared memory buffer back to the client.
+aaudio_result_t AAudioServiceStreamMMAP::open(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) {
+ const audio_attributes_t attributes = {
+ .content_type = AUDIO_CONTENT_TYPE_MUSIC,
+ .usage = AUDIO_USAGE_MEDIA,
+ .source = AUDIO_SOURCE_DEFAULT,
+ .flags = AUDIO_FLAG_LOW_LATENCY,
+ .tags = ""
+ };
+ audio_config_base_t config;
+
+ aaudio_result_t result = AAudioServiceStreamBase::open(request, configurationOutput);
+ if (result != AAUDIO_OK) {
+ ALOGE("AAudioServiceStreamBase open returned %d", result);
+ return result;
+ }
+
+ const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
+ audio_port_handle_t deviceId = configurationInput.getDeviceId();
+
+ ALOGI("open request dump()");
+ request.dump();
+
+ mMmapClient.clientUid = request.getUserId();
+ mMmapClient.clientPid = request.getProcessId();
+ aaudio_direction_t direction = request.getDirection();
+
+ // Fill in config
+ aaudio_audio_format_t aaudioFormat = configurationInput.getAudioFormat();
+ if (aaudioFormat == AAUDIO_UNSPECIFIED || aaudioFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+ ALOGI("open forcing use of AAUDIO_FORMAT_PCM_I16");
+ aaudioFormat = AAUDIO_FORMAT_PCM_I16;
+ }
+ config.format = AAudioConvert_aaudioToAndroidDataFormat(aaudioFormat);
+
+ int32_t aaudioSampleRate = configurationInput.getSampleRate();
+ if (aaudioSampleRate == AAUDIO_UNSPECIFIED) {
+ aaudioSampleRate = AAUDIO_SAMPLE_RATE_DEFAULT;
+ }
+ config.sample_rate = aaudioSampleRate;
+
+ int32_t aaudioSamplesPerFrame = configurationInput.getSamplesPerFrame();
+
+ if (direction == AAUDIO_DIRECTION_OUTPUT) {
+ config.channel_mask = (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
+ ? AUDIO_CHANNEL_OUT_STEREO
+ : audio_channel_out_mask_from_count(aaudioSamplesPerFrame);
+ } else if (direction == AAUDIO_DIRECTION_INPUT) {
+ config.channel_mask = (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
+ ? AUDIO_CHANNEL_IN_STEREO
+ : audio_channel_in_mask_from_count(aaudioSamplesPerFrame);
+ } else {
+ ALOGE("openMmapStream - invalid direction = %d", direction);
+ return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ }
+
+ mMmapClient.packageName.setTo(String16("aaudio_service")); // FIXME what should we do here?
+
+ MmapStreamInterface::stream_direction_t streamDirection = (direction == AAUDIO_DIRECTION_OUTPUT)
+ ? MmapStreamInterface::DIRECTION_OUTPUT : MmapStreamInterface::DIRECTION_INPUT;
+
+ // Open HAL stream.
+ status_t status = MmapStreamInterface::openMmapStream(streamDirection,
+ &attributes,
+ &config,
+ mMmapClient,
+ &deviceId,
+ mMmapStreamCallback,
+ mMmapStream);
+ if (status != OK) {
+ ALOGE("openMmapStream returned status %d", status);
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+
+ // Create MMAP/NOIRQ buffer.
+ int32_t minSizeFrames = configurationInput.getBufferCapacity();
+ if (minSizeFrames == 0) { // zero will get rejected
+ minSizeFrames = AAUDIO_BUFFER_CAPACITY_MIN;
+ }
+ status = mMmapStream->createMmapBuffer(minSizeFrames, &mMmapBufferinfo);
+ if (status != OK) {
+ ALOGE("%s: createMmapBuffer() returned status %d, return AAUDIO_ERROR_UNAVAILABLE",
+ __FILE__, status);
+ return AAUDIO_ERROR_UNAVAILABLE;
+ } else {
+ ALOGD("createMmapBuffer status %d shared_address = %p buffer_size %d burst_size %d",
+ status, mMmapBufferinfo.shared_memory_address,
+ mMmapBufferinfo.buffer_size_frames,
+ mMmapBufferinfo.burst_size_frames);
+ }
+
+ // Get information about the stream and pass it back to the caller.
+ mSamplesPerFrame = (direction == AAUDIO_DIRECTION_OUTPUT)
+ ? audio_channel_count_from_out_mask(config.channel_mask)
+ : audio_channel_count_from_in_mask(config.channel_mask);
+
+ mAudioDataFileDescriptor = mMmapBufferinfo.shared_memory_fd;
+ mFramesPerBurst = mMmapBufferinfo.burst_size_frames;
+ mCapacityInFrames = mMmapBufferinfo.buffer_size_frames;
+ mAudioFormat = AAudioConvert_androidToAAudioDataFormat(config.format);
+ mSampleRate = config.sample_rate;
+
+ // Fill in AAudioStreamConfiguration
+ configurationOutput.setSampleRate(mSampleRate);
+ configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
+ configurationOutput.setAudioFormat(mAudioFormat);
+ configurationOutput.setDeviceId(deviceId);
+
+ return AAUDIO_OK;
+}
+
+
+/**
+ * Start the flow of data.
+ */
+aaudio_result_t AAudioServiceStreamMMAP::start() {
+ if (mMmapStream == nullptr) return AAUDIO_ERROR_NULL;
+ aaudio_result_t result = mMmapStream->start(mMmapClient, &mPortHandle);
+ if (result != AAUDIO_OK) {
+ ALOGE("AAudioServiceStreamMMAP::start() mMmapStream->start() returned %d", result);
+ processError();
+ } else {
+ result = AAudioServiceStreamBase::start();
+ }
+ return result;
+}
+
+/**
+ * Stop the flow of data such that start() can resume with loss of data.
+ */
+aaudio_result_t AAudioServiceStreamMMAP::pause() {
+ if (mMmapStream == nullptr) return AAUDIO_ERROR_NULL;
+
+ aaudio_result_t result1 = AAudioServiceStreamBase::pause();
+ aaudio_result_t result2 = mMmapStream->stop(mPortHandle);
+ mFramesRead.reset32();
+ return (result1 != AAUDIO_OK) ? result1 : result2;
+}
+
+/**
+ * Discard any data held by the underlying HAL or Service.
+ */
+aaudio_result_t AAudioServiceStreamMMAP::flush() {
+ if (mMmapStream == nullptr) return AAUDIO_ERROR_NULL;
+ // TODO how do we flush an MMAP/NOIRQ buffer? sync pointers?
+ ALOGD("AAudioServiceStreamMMAP::pause() send AAUDIO_SERVICE_EVENT_FLUSHED");
+ sendServiceEvent(AAUDIO_SERVICE_EVENT_FLUSHED);
+ mState = AAUDIO_STREAM_STATE_FLUSHED;
+ return AAUDIO_OK;
+}
+
+
+aaudio_result_t AAudioServiceStreamMMAP::getFreeRunningPosition(int64_t *positionFrames,
+ int64_t *timeNanos) {
+ struct audio_mmap_position position;
+ if (mMmapStream == nullptr) {
+ processError();
+ return AAUDIO_ERROR_NULL;
+ }
+ status_t status = mMmapStream->getMmapPosition(&position);
+ if (status != OK) {
+ ALOGE("sendCurrentTimestamp(): getMmapPosition() returned %d", status);
+ processError();
+ return AAudioConvert_androidToAAudioResult(status);
+ } else {
+ mFramesRead.update32(position.position_frames);
+ *positionFrames = mFramesRead.get();
+ *timeNanos = position.time_nanoseconds;
+ }
+ return AAUDIO_OK;
+}
+
+void AAudioServiceStreamMMAP::onTearDown() {
+ ALOGD("AAudioServiceStreamMMAP::onTearDown() called - TODO");
+};
+
+void AAudioServiceStreamMMAP::onVolumeChanged(audio_channel_mask_t channels,
+ android::Vector<float> values) {
+ // TODO do we really need a different volume for each channel?
+ float volume = values[0];
+ ALOGD("AAudioServiceStreamMMAP::onVolumeChanged() volume[0] = %f", volume);
+ sendServiceEvent(AAUDIO_SERVICE_EVENT_VOLUME, volume);
+};
+
+void AAudioServiceStreamMMAP::onRoutingChanged(audio_port_handle_t deviceId) {
+ ALOGD("AAudioServiceStreamMMAP::onRoutingChanged() called with %d, old = %d",
+ deviceId, mPortHandle);
+ if (mPortHandle > 0 && mPortHandle != deviceId) {
+ sendServiceEvent(AAUDIO_SERVICE_EVENT_DISCONNECTED);
+ }
+ mPortHandle = deviceId;
+};
+
+/**
+ * Get an immutable description of the data queue from the HAL.
+ */
+aaudio_result_t AAudioServiceStreamMMAP::getDownDataDescription(AudioEndpointParcelable &parcelable)
+{
+ // Gather information on the data queue based on HAL info.
+ int32_t bytesPerFrame = calculateBytesPerFrame();
+ int32_t capacityInBytes = mCapacityInFrames * bytesPerFrame;
+ int fdIndex = parcelable.addFileDescriptor(mAudioDataFileDescriptor, capacityInBytes);
+ parcelable.mDownDataQueueParcelable.setupMemory(fdIndex, 0, capacityInBytes);
+ parcelable.mDownDataQueueParcelable.setBytesPerFrame(bytesPerFrame);
+ parcelable.mDownDataQueueParcelable.setFramesPerBurst(mFramesPerBurst);
+ parcelable.mDownDataQueueParcelable.setCapacityInFrames(mCapacityInFrames);
+ return AAUDIO_OK;
+}
\ No newline at end of file
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.h b/services/oboeservice/AAudioServiceStreamMMAP.h
new file mode 100644
index 0000000..f121c5c
--- /dev/null
+++ b/services/oboeservice/AAudioServiceStreamMMAP.h
@@ -0,0 +1,138 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_AAUDIO_SERVICE_STREAM_MMAP_H
+#define AAUDIO_AAUDIO_SERVICE_STREAM_MMAP_H
+
+#include <atomic>
+
+#include <media/audiohal/StreamHalInterface.h>
+#include <media/MmapStreamCallback.h>
+#include <media/MmapStreamInterface.h>
+#include <utils/RefBase.h>
+#include <utils/String16.h>
+#include <utils/Vector.h>
+
+#include "binding/AAudioServiceMessage.h"
+#include "AAudioServiceStreamBase.h"
+#include "binding/AudioEndpointParcelable.h"
+#include "SharedMemoryProxy.h"
+#include "TimestampScheduler.h"
+#include "utility/MonotonicCounter.h"
+
+namespace aaudio {
+
+ /**
+ * Manage one memory mapped buffer that originated from a HAL.
+ */
+class AAudioServiceStreamMMAP
+ : public AAudioServiceStreamBase
+ , public android::MmapStreamCallback {
+
+public:
+ AAudioServiceStreamMMAP();
+ virtual ~AAudioServiceStreamMMAP();
+
+
+ aaudio_result_t open(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) override;
+
+ /**
+ * Start the flow of audio data.
+ *
+ * This is not guaranteed to be synchronous but it currently is.
+ * An AAUDIO_SERVICE_EVENT_STARTED will be sent to the client when complete.
+ */
+ aaudio_result_t start() override;
+
+ /**
+ * Stop the flow of data so that start() can resume without loss of data.
+ *
+ * This is not guaranteed to be synchronous but it currently is.
+ * An AAUDIO_SERVICE_EVENT_PAUSED will be sent to the client when complete.
+ */
+ aaudio_result_t pause() override;
+
+ /**
+ * Discard any data held by the underlying HAL or Service.
+ *
+ * This is not guaranteed to be synchronous but it currently is.
+ * An AAUDIO_SERVICE_EVENT_FLUSHED will be sent to the client when complete.
+ */
+ aaudio_result_t flush() override;
+
+ aaudio_result_t close() override;
+
+ /**
+ * Send a MMAP/NOIRQ buffer timestamp to the client.
+ */
+ aaudio_result_t sendCurrentTimestamp();
+
+ // -------------- Callback functions ---------------------
+ void onTearDown() override;
+
+ void onVolumeChanged(audio_channel_mask_t channels,
+ android::Vector<float> values) override;
+
+ void onRoutingChanged(audio_port_handle_t deviceId) override;
+
+protected:
+
+ aaudio_result_t getDownDataDescription(AudioEndpointParcelable &parcelable) override;
+
+ aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) override;
+
+private:
+ // This proxy class was needed to prevent a crash in AudioFlinger
+ // when the stream was closed.
+ class MyMmapStreamCallback : public android::MmapStreamCallback {
+ public:
+ explicit MyMmapStreamCallback(android::MmapStreamCallback &serviceCallback)
+ : mServiceCallback(serviceCallback){}
+ virtual ~MyMmapStreamCallback() = default;
+
+ void onTearDown() override {
+ mServiceCallback.onTearDown();
+ };
+
+ void onVolumeChanged(audio_channel_mask_t channels, android::Vector<float> values) override
+ {
+ mServiceCallback.onVolumeChanged(channels, values);
+ };
+
+ void onRoutingChanged(audio_port_handle_t deviceId) override {
+ mServiceCallback.onRoutingChanged(deviceId);
+ };
+
+ private:
+ android::MmapStreamCallback &mServiceCallback;
+ };
+
+ android::sp<MyMmapStreamCallback> mMmapStreamCallback;
+ MonotonicCounter mFramesWritten;
+ MonotonicCounter mFramesRead;
+ int32_t mPreviousFrameCounter = 0; // from HAL
+
+ // Interface to the AudioFlinger MMAP support.
+ android::sp<android::MmapStreamInterface> mMmapStream;
+ struct audio_mmap_buffer_info mMmapBufferinfo;
+ android::MmapStreamInterface::Client mMmapClient;
+ audio_port_handle_t mPortHandle = -1; // TODO review best default
+};
+
+} // namespace aaudio
+
+#endif //AAUDIO_AAUDIO_SERVICE_STREAM_MMAP_H
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
new file mode 100644
index 0000000..cd9336b
--- /dev/null
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -0,0 +1,198 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AAudioService"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <mutex>
+
+#include <aaudio/AAudio.h>
+
+#include "binding/IAAudioService.h"
+
+#include "binding/AAudioServiceMessage.h"
+#include "AAudioServiceStreamBase.h"
+#include "AAudioServiceStreamShared.h"
+#include "AAudioEndpointManager.h"
+#include "AAudioService.h"
+#include "AAudioServiceEndpoint.h"
+
+using namespace android;
+using namespace aaudio;
+
+#define MIN_BURSTS_PER_BUFFER 2
+#define MAX_BURSTS_PER_BUFFER 32
+
+AAudioServiceStreamShared::AAudioServiceStreamShared(AAudioService &audioService)
+ : mAudioService(audioService)
+ {
+}
+
+AAudioServiceStreamShared::~AAudioServiceStreamShared() {
+ close();
+}
+
+aaudio_result_t AAudioServiceStreamShared::open(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) {
+
+ aaudio_result_t result = AAudioServiceStreamBase::open(request, configurationOutput);
+ if (result != AAUDIO_OK) {
+ ALOGE("AAudioServiceStreamBase open returned %d", result);
+ return result;
+ }
+
+ const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
+ int32_t deviceId = configurationInput.getDeviceId();
+ aaudio_direction_t direction = request.getDirection();
+
+ ALOGD("AAudioServiceStreamShared::open(), direction = %d", direction);
+ AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
+ mServiceEndpoint = mEndpointManager.findEndpoint(mAudioService, deviceId, direction);
+ ALOGD("AAudioServiceStreamShared::open(), mServiceEndPoint = %p", mServiceEndpoint);
+ if (mServiceEndpoint == nullptr) {
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+
+ // Is the request compatible with the shared endpoint?
+ mAudioFormat = configurationInput.getAudioFormat();
+ if (mAudioFormat == AAUDIO_FORMAT_UNSPECIFIED) {
+ mAudioFormat = AAUDIO_FORMAT_PCM_FLOAT;
+ } else if (mAudioFormat != AAUDIO_FORMAT_PCM_FLOAT) {
+ return AAUDIO_ERROR_INVALID_FORMAT;
+ }
+
+ mSampleRate = configurationInput.getSampleRate();
+ if (mSampleRate == AAUDIO_FORMAT_UNSPECIFIED) {
+ mSampleRate = mServiceEndpoint->getSampleRate();
+ } else if (mSampleRate != mServiceEndpoint->getSampleRate()) {
+ return AAUDIO_ERROR_INVALID_RATE;
+ }
+
+ mSamplesPerFrame = configurationInput.getSamplesPerFrame();
+ if (mSamplesPerFrame == AAUDIO_FORMAT_UNSPECIFIED) {
+ mSamplesPerFrame = mServiceEndpoint->getSamplesPerFrame();
+ } else if (mSamplesPerFrame != mServiceEndpoint->getSamplesPerFrame()) {
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+ }
+
+ // Determine this stream's shared memory buffer capacity.
+ mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
+ int32_t minCapacityFrames = configurationInput.getBufferCapacity();
+ int32_t numBursts = (minCapacityFrames + mFramesPerBurst - 1) / mFramesPerBurst;
+ if (numBursts < MIN_BURSTS_PER_BUFFER) {
+ numBursts = MIN_BURSTS_PER_BUFFER;
+ } else if (numBursts > MAX_BURSTS_PER_BUFFER) {
+ numBursts = MAX_BURSTS_PER_BUFFER;
+ }
+ mCapacityInFrames = numBursts * mFramesPerBurst;
+ ALOGD("AAudioServiceStreamShared::open(), mCapacityInFrames = %d", mCapacityInFrames);
+
+ // Create audio data shared memory buffer for client.
+ mAudioDataQueue = new SharedRingBuffer();
+ mAudioDataQueue->allocate(calculateBytesPerFrame(), mCapacityInFrames);
+
+ // Fill in configuration for client.
+ configurationOutput.setSampleRate(mSampleRate);
+ configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
+ configurationOutput.setAudioFormat(mAudioFormat);
+ configurationOutput.setDeviceId(deviceId);
+
+ mServiceEndpoint->registerStream(this);
+
+ return AAUDIO_OK;
+}
+
+/**
+ * Start the flow of audio data.
+ *
+ * An AAUDIO_SERVICE_EVENT_STARTED will be sent to the client when complete.
+ */
+aaudio_result_t AAudioServiceStreamShared::start() {
+ // Add this stream to the mixer.
+ aaudio_result_t result = mServiceEndpoint->startStream(this);
+ if (result != AAUDIO_OK) {
+ ALOGE("AAudioServiceStreamShared::start() mServiceEndpoint returned %d", result);
+ processError();
+ } else {
+ result = AAudioServiceStreamBase::start();
+ }
+ return AAUDIO_OK;
+}
+
+/**
+ * Stop the flow of data so that start() can resume without loss of data.
+ *
+ * An AAUDIO_SERVICE_EVENT_PAUSED will be sent to the client when complete.
+*/
+aaudio_result_t AAudioServiceStreamShared::pause() {
+ // Add this stream to the mixer.
+ aaudio_result_t result = mServiceEndpoint->stopStream(this);
+ if (result != AAUDIO_OK) {
+ ALOGE("AAudioServiceStreamShared::stop() mServiceEndpoint returned %d", result);
+ processError();
+ } else {
+ result = AAudioServiceStreamBase::start();
+ }
+ return AAUDIO_OK;
+}
+
+/**
+ * Discard any data held by the underlying HAL or Service.
+ *
+ * An AAUDIO_SERVICE_EVENT_FLUSHED will be sent to the client when complete.
+ */
+aaudio_result_t AAudioServiceStreamShared::flush() {
+ // TODO make sure we are paused
+ return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceStreamShared::close() {
+ pause();
+ // TODO wait for pause() to synchronize
+ mServiceEndpoint->unregisterStream(this);
+ mServiceEndpoint->close();
+ mServiceEndpoint = nullptr;
+ return AAudioServiceStreamBase::close();
+}
+
+/**
+ * Get an immutable description of the data queue created by this service.
+ */
+aaudio_result_t AAudioServiceStreamShared::getDownDataDescription(AudioEndpointParcelable &parcelable)
+{
+ // Gather information on the data queue.
+ mAudioDataQueue->fillParcelable(parcelable,
+ parcelable.mDownDataQueueParcelable);
+ parcelable.mDownDataQueueParcelable.setFramesPerBurst(getFramesPerBurst());
+ return AAUDIO_OK;
+}
+
+void AAudioServiceStreamShared::onStop() {
+}
+
+void AAudioServiceStreamShared::onDisconnect() {
+ mServiceEndpoint->close();
+ mServiceEndpoint = nullptr;
+}
+
+
+aaudio_result_t AAudioServiceStreamShared::getFreeRunningPosition(int64_t *positionFrames,
+ int64_t *timeNanos) {
+ *positionFrames = mAudioDataQueue->getFifoBuffer()->getReadCounter();
+ *timeNanos = AudioClock::getNanoseconds();
+ return AAUDIO_OK;
+}
diff --git a/services/oboeservice/AAudioServiceStreamShared.h b/services/oboeservice/AAudioServiceStreamShared.h
new file mode 100644
index 0000000..f6df7ce
--- /dev/null
+++ b/services/oboeservice/AAudioServiceStreamShared.h
@@ -0,0 +1,98 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_AAUDIO_SERVICE_STREAM_SHARED_H
+#define AAUDIO_AAUDIO_SERVICE_STREAM_SHARED_H
+
+#include "fifo/FifoBuffer.h"
+#include "binding/AAudioServiceMessage.h"
+#include "binding/AAudioStreamRequest.h"
+#include "binding/AAudioStreamConfiguration.h"
+
+#include "AAudioService.h"
+#include "AAudioServiceStreamBase.h"
+
+namespace aaudio {
+
+// We expect the queue to only have a few commands.
+// This should be way more than we need.
+#define QUEUE_UP_CAPACITY_COMMANDS (128)
+
+class AAudioEndpointManager;
+class AAudioServiceEndpoint;
+class SharedRingBuffer;
+
+/**
+ * One of these is created for every MODE_SHARED stream in the AAudioService.
+ *
+ * Each Shared stream will register itself with an AAudioServiceEndpoint when it is opened.
+ */
+class AAudioServiceStreamShared : public AAudioServiceStreamBase {
+
+public:
+ AAudioServiceStreamShared(android::AAudioService &aAudioService);
+ virtual ~AAudioServiceStreamShared();
+
+ aaudio_result_t open(const aaudio::AAudioStreamRequest &request,
+ aaudio::AAudioStreamConfiguration &configurationOutput) override;
+
+ /**
+ * Start the flow of audio data.
+ *
+ * This is not guaranteed to be synchronous but it currently is.
+ * An AAUDIO_SERVICE_EVENT_STARTED will be sent to the client when complete.
+ */
+ aaudio_result_t start() override;
+
+ /**
+ * Stop the flow of data so that start() can resume without loss of data.
+ *
+ * This is not guaranteed to be synchronous but it currently is.
+ * An AAUDIO_SERVICE_EVENT_PAUSED will be sent to the client when complete.
+ */
+ aaudio_result_t pause() override;
+
+ /**
+ * Discard any data held by the underlying HAL or Service.
+ *
+ * This is not guaranteed to be synchronous but it currently is.
+ * An AAUDIO_SERVICE_EVENT_FLUSHED will be sent to the client when complete.
+ */
+ aaudio_result_t flush() override;
+
+ aaudio_result_t close() override;
+
+ android::FifoBuffer *getDataFifoBuffer() { return mAudioDataQueue->getFifoBuffer(); }
+
+ void onStop();
+
+ void onDisconnect();
+
+protected:
+
+ aaudio_result_t getDownDataDescription(AudioEndpointParcelable &parcelable) override;
+
+ aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) override;
+
+private:
+ android::AAudioService &mAudioService;
+ AAudioServiceEndpoint *mServiceEndpoint = nullptr;
+ SharedRingBuffer *mAudioDataQueue;
+};
+
+} /* namespace aaudio */
+
+#endif //AAUDIO_AAUDIO_SERVICE_STREAM_SHARED_H
diff --git a/services/oboeservice/AAudioThread.cpp b/services/oboeservice/AAudioThread.cpp
index f5e5784..b1b563d 100644
--- a/services/oboeservice/AAudioThread.cpp
+++ b/services/oboeservice/AAudioThread.cpp
@@ -21,14 +21,17 @@
#include <pthread.h>
#include <aaudio/AAudioDefinitions.h>
+#include <utility/AAudioUtilities.h>
#include "AAudioThread.h"
using namespace aaudio;
-AAudioThread::AAudioThread() {
- // mThread is a pthread_t of unknown size so we need memset.
+AAudioThread::AAudioThread()
+ : mRunnable(nullptr)
+ , mHasThread(false) {
+ // mThread is a pthread_t of unknown size so we need memset().
memset(&mThread, 0, sizeof(mThread));
}
@@ -50,14 +53,16 @@
aaudio_result_t AAudioThread::start(Runnable *runnable) {
if (mHasThread) {
+ ALOGE("AAudioThread::start() - mHasThread.load() already true");
return AAUDIO_ERROR_INVALID_STATE;
}
- mRunnable = runnable; // TODO use atomic?
+ // mRunnable will be read by the new thread when it starts.
+ // pthread_create() forces a memory synchronization so mRunnable does not need to be atomic.
+ mRunnable = runnable;
int err = pthread_create(&mThread, nullptr, AAudioThread_internalThreadProc, this);
if (err != 0) {
- ALOGE("AAudioThread::pthread_create() returned %d", err);
- // TODO convert errno to aaudio_result_t
- return AAUDIO_ERROR_INTERNAL;
+ ALOGE("AAudioThread::start() - pthread_create() returned %d %s", err, strerror(err));
+ return AAudioConvert_androidToAAudioResult(-err);
} else {
mHasThread = true;
return AAUDIO_OK;
@@ -70,7 +75,11 @@
}
int err = pthread_join(mThread, nullptr);
mHasThread = false;
- // TODO convert errno to aaudio_result_t
- return err ? AAUDIO_ERROR_INTERNAL : AAUDIO_OK;
+ if (err != 0) {
+ ALOGE("AAudioThread::stop() - pthread_join() returned %d %s", err, strerror(err));
+ return AAudioConvert_androidToAAudioResult(-err);
+ } else {
+ return AAUDIO_OK;
+ }
}
diff --git a/services/oboeservice/AAudioThread.h b/services/oboeservice/AAudioThread.h
index a5d43a4..dd9f640 100644
--- a/services/oboeservice/AAudioThread.h
+++ b/services/oboeservice/AAudioThread.h
@@ -24,16 +24,20 @@
namespace aaudio {
+/**
+ * Abstract class similar to Java Runnable.
+ */
class Runnable {
public:
Runnable() {};
virtual ~Runnable() = default;
- virtual void run() {}
+ virtual void run() = 0;
};
/**
- * Abstraction for a host thread.
+ * Abstraction for a host dependent thread.
+ * TODO Consider using Android "Thread" class or std::thread instead.
*/
class AAudioThread
{
@@ -62,9 +66,9 @@
void dispatch(); // called internally from 'C' thread wrapper
private:
- Runnable* mRunnable = nullptr; // TODO make atomic with memory barrier?
- bool mHasThread = false;
- pthread_t mThread; // initialized in constructor
+ Runnable *mRunnable;
+ bool mHasThread;
+ pthread_t mThread; // initialized in constructor
};
diff --git a/services/oboeservice/Android.mk b/services/oboeservice/Android.mk
index 5cd9121..a9c80ae 100644
--- a/services/oboeservice/Android.mk
+++ b/services/oboeservice/Android.mk
@@ -3,52 +3,54 @@
# AAudio Service
include $(CLEAR_VARS)
-LOCAL_MODULE := aaudioservice
+LOCAL_MODULE := libaaudioservice
LOCAL_MODULE_TAGS := optional
LIBAAUDIO_DIR := ../../media/libaaudio
LIBAAUDIO_SRC_DIR := $(LIBAAUDIO_DIR)/src
LOCAL_C_INCLUDES := \
+ $(TOPDIR)frameworks/av/services/audioflinger \
$(call include-path-for, audio-utils) \
frameworks/native/include \
system/core/base/include \
$(TOP)/frameworks/native/media/libaaudio/include/include \
$(TOP)/frameworks/av/media/libaaudio/include \
+ $(TOP)/frameworks/av/media/utils/include \
frameworks/native/include \
$(TOP)/external/tinyalsa/include \
- $(TOP)/frameworks/av/media/libaaudio/src \
- $(TOP)/frameworks/av/media/libaaudio/src/binding \
- $(TOP)/frameworks/av/media/libaaudio/src/client \
- $(TOP)/frameworks/av/media/libaaudio/src/core \
- $(TOP)/frameworks/av/media/libaaudio/src/fifo \
- $(TOP)/frameworks/av/media/libaaudio/src/utility
+ $(TOP)/frameworks/av/media/libaaudio/src
-# TODO These could be in a libaaudio_common library
LOCAL_SRC_FILES += \
$(LIBAAUDIO_SRC_DIR)/utility/HandleTracker.cpp \
- $(LIBAAUDIO_SRC_DIR)/utility/AAudioUtilities.cpp \
- $(LIBAAUDIO_SRC_DIR)/fifo/FifoBuffer.cpp \
- $(LIBAAUDIO_SRC_DIR)/fifo/FifoControllerBase.cpp \
- $(LIBAAUDIO_SRC_DIR)/binding/SharedMemoryParcelable.cpp \
- $(LIBAAUDIO_SRC_DIR)/binding/SharedRegionParcelable.cpp \
- $(LIBAAUDIO_SRC_DIR)/binding/RingBufferParcelable.cpp \
- $(LIBAAUDIO_SRC_DIR)/binding/AudioEndpointParcelable.cpp \
- $(LIBAAUDIO_SRC_DIR)/binding/AAudioStreamRequest.cpp \
- $(LIBAAUDIO_SRC_DIR)/binding/AAudioStreamConfiguration.cpp \
- $(LIBAAUDIO_SRC_DIR)/binding/IAAudioService.cpp \
+ SharedMemoryProxy.cpp \
SharedRingBuffer.cpp \
- FakeAudioHal.cpp \
+ AAudioEndpointManager.cpp \
+ AAudioMixer.cpp \
AAudioService.cpp \
+ AAudioServiceEndpoint.cpp \
AAudioServiceStreamBase.cpp \
- AAudioServiceStreamFakeHal.cpp \
+ AAudioServiceStreamMMAP.cpp \
+ AAudioServiceStreamShared.cpp \
TimestampScheduler.cpp \
- AAudioServiceMain.cpp \
AAudioThread.cpp
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
+# LOCAL_CFLAGS += -fvisibility=hidden
LOCAL_CFLAGS += -Wno-unused-parameter
LOCAL_CFLAGS += -Wall -Werror
-LOCAL_SHARED_LIBRARIES := libbinder libcutils libutils liblog libtinyalsa
+LOCAL_SHARED_LIBRARIES := \
+ libaaudio \
+ libaudioflinger \
+ libbinder \
+ libcutils \
+ libmediautils \
+ libutils \
+ liblog \
+ libtinyalsa
-include $(BUILD_EXECUTABLE)
+include $(BUILD_SHARED_LIBRARY)
+
+
diff --git a/services/oboeservice/FakeAudioHal.cpp b/services/oboeservice/FakeAudioHal.cpp
deleted file mode 100644
index 122671e..0000000
--- a/services/oboeservice/FakeAudioHal.cpp
+++ /dev/null
@@ -1,235 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-/**
- * Simple fake HAL that supports ALSA MMAP/NOIRQ mode.
- */
-
-#include <iostream>
-#include <math.h>
-#include <limits>
-#include <string.h>
-#include <unistd.h>
-
-#include <sound/asound.h>
-
-#include "tinyalsa/asoundlib.h"
-
-#include "FakeAudioHal.h"
-
-//using namespace aaudio;
-
-using sample_t = int16_t;
-using std::cout;
-using std::endl;
-
-#undef SNDRV_PCM_IOCTL_SYNC_PTR
-#define SNDRV_PCM_IOCTL_SYNC_PTR 0xc0884123
-#define PCM_ERROR_MAX 128
-
-const int SAMPLE_RATE = 48000; // Hz
-const int CHANNEL_COUNT = 2;
-
-struct pcm {
- int fd;
- unsigned int flags;
- int running:1;
- int prepared:1;
- int underruns;
- unsigned int buffer_size;
- unsigned int boundary;
- char error[PCM_ERROR_MAX];
- struct pcm_config config;
- struct snd_pcm_mmap_status *mmap_status;
- struct snd_pcm_mmap_control *mmap_control;
- struct snd_pcm_sync_ptr *sync_ptr;
- void *mmap_buffer;
- unsigned int noirq_frames_per_msec;
- int wait_for_avail_min;
-};
-
-static int pcm_sync_ptr(struct pcm *pcm, int flags) {
- if (pcm->sync_ptr) {
- pcm->sync_ptr->flags = flags;
- if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_SYNC_PTR, pcm->sync_ptr) < 0)
- return -1;
- }
- return 0;
-}
-
-int pcm_get_hw_ptr(struct pcm* pcm, unsigned int* hw_ptr) {
- if (!hw_ptr || !pcm) return -EINVAL;
-
- int result = pcm_sync_ptr(pcm, SNDRV_PCM_SYNC_PTR_HWSYNC);
- if (!result) {
- *hw_ptr = pcm->sync_ptr->s.status.hw_ptr;
- }
-
- return result;
-}
-
-typedef struct stream_tracker {
- struct pcm * pcm;
- int framesPerBurst;
- sample_t * hwBuffer;
- int32_t capacityInFrames;
- int32_t capacityInBytes;
-} stream_tracker_t;
-
-#define FRAMES_PER_BURST_QUALCOMM 192
-#define FRAMES_PER_BURST_NVIDIA 128
-
-int fake_hal_open(int card_id, int device_id,
- int frameCapacity,
- fake_hal_stream_ptr *streamPP) {
- int framesPerBurst = FRAMES_PER_BURST_QUALCOMM; // TODO update as needed
- int periodCountRequested = frameCapacity / framesPerBurst;
- int periodCount = 32;
- unsigned int offset1;
- unsigned int frames1;
- void *area = nullptr;
- int mmapAvail = 0;
-
- // Try to match requested size with a power of 2.
- while (periodCount < periodCountRequested && periodCount < 1024) {
- periodCount *= 2;
- }
- std::cout << "fake_hal_open() requested frameCapacity = " << frameCapacity << std::endl;
- std::cout << "fake_hal_open() periodCountRequested = " << periodCountRequested << std::endl;
- std::cout << "fake_hal_open() periodCount = " << periodCount << std::endl;
-
- // Configuration for an ALSA stream.
- pcm_config cfg;
- memset(&cfg, 0, sizeof(cfg));
- cfg.channels = CHANNEL_COUNT;
- cfg.format = PCM_FORMAT_S16_LE;
- cfg.rate = SAMPLE_RATE;
- cfg.period_count = periodCount;
- cfg.period_size = framesPerBurst;
- cfg.start_threshold = 0; // for NOIRQ, should just start, was framesPerBurst;
- cfg.stop_threshold = INT32_MAX;
- cfg.silence_size = 0;
- cfg.silence_threshold = 0;
- cfg.avail_min = framesPerBurst;
-
- stream_tracker_t *streamTracker = (stream_tracker_t *) malloc(sizeof(stream_tracker_t));
- if (streamTracker == nullptr) {
- return -1;
- }
- memset(streamTracker, 0, sizeof(stream_tracker_t));
-
- streamTracker->pcm = pcm_open(card_id, device_id, PCM_OUT | PCM_MMAP | PCM_NOIRQ, &cfg);
- if (streamTracker->pcm == nullptr) {
- cout << "Could not open device." << endl;
- free(streamTracker);
- return -1;
- }
-
- streamTracker->framesPerBurst = cfg.period_size; // Get from ALSA
- streamTracker->capacityInFrames = pcm_get_buffer_size(streamTracker->pcm);
- streamTracker->capacityInBytes = pcm_frames_to_bytes(streamTracker->pcm, streamTracker->capacityInFrames);
- std::cout << "fake_hal_open() streamTracker->framesPerBurst = " << streamTracker->framesPerBurst << std::endl;
- std::cout << "fake_hal_open() streamTracker->capacityInFrames = " << streamTracker->capacityInFrames << std::endl;
-
- if (pcm_is_ready(streamTracker->pcm) < 0) {
- cout << "Device is not ready." << endl;
- goto error;
- }
-
- if (pcm_prepare(streamTracker->pcm) < 0) {
- cout << "Device could not be prepared." << endl;
- cout << "For Marlin, please enter:" << endl;
- cout << " adb shell" << endl;
- cout << " tinymix \"QUAT_MI2S_RX Audio Mixer MultiMedia8\" 1" << endl;
- goto error;
- }
- mmapAvail = pcm_mmap_avail(streamTracker->pcm);
- if (mmapAvail <= 0) {
- cout << "fake_hal_open() mmap_avail is <=0" << endl;
- goto error;
- }
- cout << "fake_hal_open() mmap_avail = " << mmapAvail << endl;
-
- // Where is the memory mapped area?
- if (pcm_mmap_begin(streamTracker->pcm, &area, &offset1, &frames1) < 0) {
- cout << "fake_hal_open() pcm_mmap_begin failed" << endl;
- goto error;
- }
-
- // Clear the buffer.
- memset((sample_t*) area, 0, streamTracker->capacityInBytes);
- streamTracker->hwBuffer = (sample_t*) area;
- streamTracker->hwBuffer[0] = 32000; // impulse
-
- // Prime the buffer so it can start.
- if (pcm_mmap_commit(streamTracker->pcm, 0, framesPerBurst) < 0) {
- cout << "fake_hal_open() pcm_mmap_commit failed" << endl;
- goto error;
- }
-
- *streamPP = streamTracker;
- return 1;
-
-error:
- fake_hal_close(streamTracker);
- return -1;
-}
-
-int fake_hal_get_mmap_info(fake_hal_stream_ptr stream, mmap_buffer_info *info) {
- stream_tracker_t *streamTracker = (stream_tracker_t *) stream;
- info->fd = streamTracker->pcm->fd; // TODO use tinyalsa function
- info->hw_buffer = streamTracker->hwBuffer;
- info->burst_size_in_frames = streamTracker->framesPerBurst;
- info->buffer_capacity_in_frames = streamTracker->capacityInFrames;
- info->buffer_capacity_in_bytes = streamTracker->capacityInBytes;
- info->sample_rate = SAMPLE_RATE;
- info->channel_count = CHANNEL_COUNT;
- return 0;
-}
-
-int fake_hal_start(fake_hal_stream_ptr stream) {
- stream_tracker_t *streamTracker = (stream_tracker_t *) stream;
- if (pcm_start(streamTracker->pcm) < 0) {
- cout << "fake_hal_start failed" << endl;
- return -1;
- }
- return 0;
-}
-
-int fake_hal_pause(fake_hal_stream_ptr stream) {
- stream_tracker_t *streamTracker = (stream_tracker_t *) stream;
- if (pcm_stop(streamTracker->pcm) < 0) {
- cout << "fake_hal_stop failed" << endl;
- return -1;
- }
- return 0;
-}
-
-int fake_hal_get_frame_counter(fake_hal_stream_ptr stream, int *frame_counter) {
- stream_tracker_t *streamTracker = (stream_tracker_t *) stream;
- if (pcm_get_hw_ptr(streamTracker->pcm, (unsigned int *)frame_counter) < 0) {
- cout << "fake_hal_get_frame_counter failed" << endl;
- return -1;
- }
- return 0;
-}
-
-int fake_hal_close(fake_hal_stream_ptr stream) {
- stream_tracker_t *streamTracker = (stream_tracker_t *) stream;
- pcm_close(streamTracker->pcm);
- free(streamTracker);
- return 0;
-}
-
diff --git a/services/oboeservice/FakeAudioHal.h b/services/oboeservice/FakeAudioHal.h
deleted file mode 100644
index d3aa4e8..0000000
--- a/services/oboeservice/FakeAudioHal.h
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/**
- * Simple fake HAL that supports ALSA MMAP/NOIRQ mode.
- */
-
-#ifndef FAKE_AUDIO_HAL_H
-#define FAKE_AUDIO_HAL_H
-
-//namespace aaudio {
-
-using sample_t = int16_t;
-struct mmap_buffer_info {
- int fd;
- int32_t burst_size_in_frames;
- int32_t buffer_capacity_in_frames;
- int32_t buffer_capacity_in_bytes;
- int32_t sample_rate;
- int32_t channel_count;
- sample_t *hw_buffer;
-};
-
-typedef void *fake_hal_stream_ptr;
-
-//extern "C"
-//{
-
-int fake_hal_open(int card_id, int device_id,
- int frameCapacity,
- fake_hal_stream_ptr *stream_pp);
-
-int fake_hal_get_mmap_info(fake_hal_stream_ptr stream, mmap_buffer_info *info);
-
-int fake_hal_start(fake_hal_stream_ptr stream);
-
-int fake_hal_pause(fake_hal_stream_ptr stream);
-
-int fake_hal_get_frame_counter(fake_hal_stream_ptr stream, int *frame_counter);
-
-int fake_hal_close(fake_hal_stream_ptr stream);
-
-//} /* "C" */
-
-//} /* namespace aaudio */
-
-#endif // FAKE_AUDIO_HAL_H
diff --git a/services/oboeservice/SharedMemoryProxy.cpp b/services/oboeservice/SharedMemoryProxy.cpp
new file mode 100644
index 0000000..83ae1d4
--- /dev/null
+++ b/services/oboeservice/SharedMemoryProxy.cpp
@@ -0,0 +1,83 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AAudioService"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <aaudio/AAudioDefinitions.h>
+#include "SharedMemoryProxy.h"
+
+using namespace android;
+using namespace aaudio;
+
+SharedMemoryProxy::~SharedMemoryProxy()
+{
+ if (mOriginalSharedMemory != nullptr) {
+ munmap(mOriginalSharedMemory, mSharedMemorySizeInBytes);
+ mOriginalSharedMemory = nullptr;
+ }
+ if (mProxySharedMemory != nullptr) {
+ munmap(mProxySharedMemory, mSharedMemorySizeInBytes);
+ close(mProxyFileDescriptor);
+ mProxySharedMemory = nullptr;
+ }
+}
+
+aaudio_result_t SharedMemoryProxy::open(int originalFD, int32_t capacityInBytes) {
+ mOriginalFileDescriptor = originalFD;
+ mSharedMemorySizeInBytes = capacityInBytes;
+
+ mProxyFileDescriptor = ashmem_create_region("AAudioProxyDataBuffer", mSharedMemorySizeInBytes);
+ if (mProxyFileDescriptor < 0) {
+ ALOGE("SharedMemoryProxy::open() ashmem_create_region() failed %d", errno);
+ return AAUDIO_ERROR_INTERNAL;
+ }
+ int err = ashmem_set_prot_region(mProxyFileDescriptor, PROT_READ|PROT_WRITE);
+ if (err < 0) {
+ ALOGE("SharedMemoryProxy::open() ashmem_set_prot_region() failed %d", errno);
+ close(mProxyFileDescriptor);
+ mProxyFileDescriptor = -1;
+ return AAUDIO_ERROR_INTERNAL; // TODO convert errno to a better AAUDIO_ERROR;
+ }
+
+ // Get original memory address.
+ mOriginalSharedMemory = (uint8_t *) mmap(0, mSharedMemorySizeInBytes,
+ PROT_READ|PROT_WRITE,
+ MAP_SHARED,
+ mOriginalFileDescriptor, 0);
+ if (mOriginalSharedMemory == MAP_FAILED) {
+ ALOGE("SharedMemoryProxy::open() original mmap(%d) failed %d (%s)",
+ mOriginalFileDescriptor, errno, strerror(errno));
+ return AAUDIO_ERROR_INTERNAL; // TODO convert errno to a better AAUDIO_ERROR;
+ }
+
+ // Map the fd to the same memory addresses.
+ mProxySharedMemory = (uint8_t *) mmap(mOriginalSharedMemory, mSharedMemorySizeInBytes,
+ PROT_READ|PROT_WRITE,
+ MAP_SHARED,
+ mProxyFileDescriptor, 0);
+ if (mProxySharedMemory != mOriginalSharedMemory) {
+ ALOGE("SharedMemoryProxy::open() proxy mmap(%d) failed %d", mProxyFileDescriptor, errno);
+ munmap(mOriginalSharedMemory, mSharedMemorySizeInBytes);
+ mOriginalSharedMemory = nullptr;
+ close(mProxyFileDescriptor);
+ mProxyFileDescriptor = -1;
+ return AAUDIO_ERROR_INTERNAL; // TODO convert errno to a better AAUDIO_ERROR;
+ }
+
+ return AAUDIO_OK;
+}
diff --git a/services/oboeservice/SharedMemoryProxy.h b/services/oboeservice/SharedMemoryProxy.h
new file mode 100644
index 0000000..99bfdea
--- /dev/null
+++ b/services/oboeservice/SharedMemoryProxy.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_SHARED_MEMORY_PROXY_H
+#define AAUDIO_SHARED_MEMORY_PROXY_H
+
+#include <stdint.h>
+#include <cutils/ashmem.h>
+#include <sys/mman.h>
+
+#include <aaudio/AAudioDefinitions.h>
+
+namespace aaudio {
+
+/**
+ * Proxy for sharing memory between two file descriptors.
+ */
+class SharedMemoryProxy {
+public:
+ SharedMemoryProxy() {}
+
+ ~SharedMemoryProxy();
+
+ aaudio_result_t open(int fd, int32_t capacityInBytes);
+
+ int getFileDescriptor() const {
+ return mProxyFileDescriptor;
+ }
+
+private:
+ int mOriginalFileDescriptor = -1;
+ int mProxyFileDescriptor = -1;
+ uint8_t *mOriginalSharedMemory = nullptr;
+ uint8_t *mProxySharedMemory = nullptr;
+ int32_t mSharedMemorySizeInBytes = 0;
+};
+
+} /* namespace aaudio */
+
+#endif //AAUDIO_SHARED_MEMORY_PROXY_H
diff --git a/services/oboeservice/SharedRingBuffer.cpp b/services/oboeservice/SharedRingBuffer.cpp
index 9ac8fdf..efcc9d6 100644
--- a/services/oboeservice/SharedRingBuffer.cpp
+++ b/services/oboeservice/SharedRingBuffer.cpp
@@ -18,11 +18,8 @@
//#define LOG_NDEBUG 0
#include <utils/Log.h>
-#include "AudioClock.h"
-#include "AudioEndpointParcelable.h"
-
-//#include "AAudioServiceStreamBase.h"
-//#include "AAudioServiceStreamFakeHal.h"
+#include "binding/RingBufferParcelable.h"
+#include "binding/AudioEndpointParcelable.h"
#include "SharedRingBuffer.h"
diff --git a/services/oboeservice/SharedRingBuffer.h b/services/oboeservice/SharedRingBuffer.h
index 75f138b..a2c3766 100644
--- a/services/oboeservice/SharedRingBuffer.h
+++ b/services/oboeservice/SharedRingBuffer.h
@@ -22,8 +22,8 @@
#include <sys/mman.h>
#include "fifo/FifoBuffer.h"
-#include "RingBufferParcelable.h"
-#include "AudioEndpointParcelable.h"
+#include "binding/RingBufferParcelable.h"
+#include "binding/AudioEndpointParcelable.h"
namespace aaudio {
@@ -41,22 +41,22 @@
virtual ~SharedRingBuffer();
- aaudio_result_t allocate(fifo_frames_t bytesPerFrame, fifo_frames_t capacityInFrames);
+ aaudio_result_t allocate(android::fifo_frames_t bytesPerFrame, android::fifo_frames_t capacityInFrames);
void fillParcelable(AudioEndpointParcelable &endpointParcelable,
RingBufferParcelable &ringBufferParcelable);
- FifoBuffer * getFifoBuffer() {
+ android::FifoBuffer * getFifoBuffer() {
return mFifoBuffer;
}
private:
- int mFileDescriptor = -1;
- FifoBuffer * mFifoBuffer = nullptr;
- uint8_t * mSharedMemory = nullptr;
- int32_t mSharedMemorySizeInBytes = 0;
- int32_t mDataMemorySizeInBytes = 0;
- fifo_frames_t mCapacityInFrames = 0;
+ int mFileDescriptor = -1;
+ android::FifoBuffer *mFifoBuffer = nullptr;
+ uint8_t *mSharedMemory = nullptr;
+ int32_t mSharedMemorySizeInBytes = 0;
+ int32_t mDataMemorySizeInBytes = 0;
+ android::fifo_frames_t mCapacityInFrames = 0;
};
} /* namespace aaudio */
diff --git a/services/oboeservice/TimestampScheduler.h b/services/oboeservice/TimestampScheduler.h
index 91a2477..325bee4 100644
--- a/services/oboeservice/TimestampScheduler.h
+++ b/services/oboeservice/TimestampScheduler.h
@@ -17,15 +17,8 @@
#ifndef AAUDIO_TIMESTAMP_SCHEDULER_H
#define AAUDIO_TIMESTAMP_SCHEDULER_H
-
-
-#include "IAAudioService.h"
-#include "AAudioServiceDefinitions.h"
-#include "AudioStream.h"
-#include "fifo/FifoBuffer.h"
-#include "SharedRingBuffer.h"
-#include "AudioEndpointParcelable.h"
-#include "utility/AudioClock.h"
+#include <aaudio/AAudioDefinitions.h>
+#include <utility/AudioClock.h>
namespace aaudio {
@@ -47,8 +40,7 @@
void start(int64_t startTime);
/**
- * Calculate the next time that the read position should be
- * measured.
+ * Calculate the next time that the read position should be measured.
*/
int64_t nextAbsoluteTime();
@@ -68,8 +60,8 @@
private:
// Start with an arbitrary default so we do not divide by zero.
int64_t mBurstPeriod = AAUDIO_NANOS_PER_MILLISECOND;
- int64_t mStartTime;
- int64_t mLastTime;
+ int64_t mStartTime = 0;
+ int64_t mLastTime = 0;
};
} /* namespace aaudio */