AAudioService: integrated with audioserver
Call the MmapStreamInterface from AudioFlinger instead of the FakeHAL.
Fix sending timestamps from the thread.
Add shared mode in service.
Bug: 35260844
Bug: 33398120
Test: CTS test_aaudio.cpp
Change-Id: I44c7e4ecae4ce205611b6b73a72e0ae8a5b243e5
Signed-off-by: Phil Burk <philburk@google.com>
(cherry picked from commit 7f6b40d78b1976c78d1300e8a51fda36eeb50c5d)
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/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/write_sine_threaded.cpp b/media/libaaudio/examples/write_sine/src/write_sine_threaded.cpp
index 8065c48..9bc5886 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine_threaded.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine_threaded.cpp
@@ -16,7 +16,8 @@
// 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>
@@ -25,14 +26,14 @@
#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,
@@ -41,6 +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.
@@ -79,21 +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 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
@@ -104,9 +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];
@@ -117,6 +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");
@@ -149,7 +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,
@@ -159,56 +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;
}
@@ -229,7 +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) {
@@ -288,19 +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 aeccb4a..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 \
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/src/Android.mk b/media/libaaudio/src/Android.mk
index 1ee73bf..b5bb75f 100644
--- a/media/libaaudio/src/Android.mk
+++ b/media/libaaudio/src/Android.mk
@@ -26,6 +26,7 @@
$(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 \
@@ -43,13 +44,14 @@
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
@@ -96,13 +98,14 @@
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 90c619c..fe049b2 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -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 1f9ce4f..3934a6d 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -18,19 +18,24 @@
//#define LOG_NDEBUG 0
#include <utils/Log.h>
+#include <stdint.h>
#include <assert.h>
#include <binder/IServiceManager.h>
-#include <utils/Mutex.h>
#include <aaudio/AAudio.h>
#include <utils/String16.h>
-#include "utility/AudioClock.h"
-#include "AudioStreamInternal.h"
+#include "AudioClock.h"
+#include "AudioEndpointParcelable.h"
+#include "binding/AAudioStreamRequest.h"
+#include "binding/AAudioStreamConfiguration.h"
+#include "binding/IAAudioService.h"
#include "binding/AAudioServiceMessage.h"
+#include "fifo/FifoBuffer.h"
#include "core/AudioStreamBuilder.h"
+#include "AudioStreamInternal.h"
#define LOG_TIMESTAMPS 0
@@ -39,52 +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 AAUDIO_SERVICE_NAME "AAudioService"
-
#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
-// 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)
{
}
@@ -93,9 +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;
@@ -105,22 +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) {
@@ -130,17 +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);
@@ -156,12 +150,12 @@
mCallbackFrames = builder.getFramesPerDataCallback();
if (mCallbackFrames > getBufferCapacity() / 2) {
ALOGE("AudioStreamInternal.open(): framesPerCallback too large");
- service->closeStream(mServiceStreamHandle);
+ mServiceInterface.closeStream(mServiceStreamHandle);
return AAUDIO_ERROR_OUT_OF_RANGE;
} else if (mCallbackFrames < 0) {
ALOGE("AudioStreamInternal.open(): framesPerCallback negative");
- service->closeStream(mServiceStreamHandle);
+ mServiceInterface.closeStream(mServiceStreamHandle);
return AAUDIO_ERROR_OUT_OF_RANGE;
}
@@ -181,20 +175,20 @@
}
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);
+
+ mServiceInterface.closeStream(serviceStreamHandle);
delete[] mCallbackBuffer;
- return AAUDIO_OK;
+ 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;
@@ -254,19 +248,16 @@
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);
- aaudio_result_t result = aaudioService->startStream(mServiceStreamHandle);
+ aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle);;
if (result == AAUDIO_OK && getDataCallbackProc() != nullptr) {
// Launch the callback loop thread.
@@ -306,13 +297,10 @@
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()
@@ -325,20 +313,17 @@
}
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);
+ 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().
@@ -366,25 +351,22 @@
}
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());
}
aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId,
@@ -410,16 +392,16 @@
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;
@@ -438,29 +420,34 @@
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);
@@ -474,6 +461,7 @@
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
@@ -502,21 +490,26 @@
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;
}
@@ -527,18 +520,19 @@
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();
@@ -546,43 +540,52 @@
}
// 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) {
@@ -607,13 +610,68 @@
*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;
}
+
+// 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;
+ }
+ partIndex++;
+ }
+ int32_t framesWritten = numFrames - framesLeft;
+ mAudioEndpoint.advanceWriteIndex(framesWritten);
+
+ if (framesWritten > 0) {
+ incrementFramesWritten(framesWritten);
+ }
+ // ALOGD_IF(ALOG_CONDITION, "AudioStreamInternal::writeNowWithConversion() returns %d", framesWritten);
+ return framesWritten;
+}
+
void AudioStreamInternal::processTimestamp(uint64_t position, int64_t time) {
mClockModel.processTimestamp( position, time);
}
@@ -654,7 +712,7 @@
} 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 9a15a9b..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;
@@ -35,52 +37,54 @@
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 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 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 getFramesPerBurst() const override;
+ int32_t getFramesPerBurst() const override;
- virtual int64_t getFramesRead() override;
+ 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();
@@ -100,10 +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();
@@ -115,19 +119,41 @@
int64_t calculateReasonableTimeout(int32_t framesPerOperation);
private:
- IsochronousClockModel mClockModel;
- AudioEndpoint mAudioEndpoint;
- aaudio_handle_t mServiceStreamHandle;
- EndpointDescriptor mEndpointDescriptor;
+ /*
+ * 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;
- int64_t mLastFramesRead = 0;
- int32_t mFramesPerBurst;
- int32_t mXRunCount = 0;
- void processTimestamp(uint64_t position, int64_t time);
+ 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/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index bc2f281..d91d0e4 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;
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 68579fd..7c0b5ae 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -45,6 +45,7 @@
mDeviceId = builder.getDeviceId();
mFormat = builder.getFormat();
mDirection = builder.getDirection();
+ mSharingMode = builder.getSharingMode();
// callbacks
mFramesPerDataCallback = builder.getFramesPerDataCallback();
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 1485d20..da71906 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -22,8 +22,8 @@
#include <stdint.h>
#include <aaudio/AAudio.h>
-#include "AAudioUtilities.h"
-#include "MonotonicCounter.h"
+#include "utility/AAudioUtilities.h"
+#include "utility/MonotonicCounter.h"
namespace aaudio {
@@ -266,11 +266,14 @@
mState = state;
}
+ void setDeviceId(int32_t deviceId) {
+ mDeviceId = deviceId;
+ }
+
std::mutex mStreamMutex;
std::atomic<bool> mCallbackEnabled;
-
protected:
MonotonicCounter mFramesWritten;
MonotonicCounter mFramesRead;
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 858ae80..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;
/*
@@ -43,9 +49,11 @@
aaudio_result_t AudioStreamBuilder::build(AudioStream** streamPtr) {
AudioStream* audioStream = nullptr;
+ AAudioBinderClient *aaudioClient = nullptr;
const aaudio_sharing_mode_t sharingMode = getSharingMode();
- ALOGE("AudioStreamBuilder.build() sharingMode = %d", sharingMode);
+ ALOGD("AudioStreamBuilder.build() sharingMode = %d", sharingMode);
switch (getDirection()) {
+
case AAUDIO_DIRECTION_INPUT:
switch (sharingMode) {
case AAUDIO_SHARING_MODE_SHARED:
@@ -57,26 +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/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/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 3b79953..8d60678 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -22,7 +22,7 @@
#include <media/AudioTrack.h>
#include <aaudio/AAudio.h>
-#include "AudioClock.h"
+#include "utility/AudioClock.h"
#include "legacy/AudioStreamLegacy.h"
#include "legacy/AudioStreamTrack.h"
#include "utility/FixedBlockReader.h"
@@ -142,6 +142,7 @@
}
setState(AAUDIO_STREAM_STATE_OPEN);
+ setDeviceId(mAudioTrack->getRoutedDeviceId());
return AAUDIO_OK;
}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 7eb179a..2a7194b 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2133,7 +2133,7 @@
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 +2148,7 @@
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/Threads.cpp b/services/audioflinger/Threads.cpp
index 3b1edec..3a72c3c 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -7585,7 +7585,11 @@
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 +7644,7 @@
// 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 +7940,17 @@
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 +8065,9 @@
void AudioFlinger::MmapThread::threadLoop_exit()
{
- if (mCallback != 0) {
- mCallback->onTearDown();
+ sp<MmapStreamCallback> callback = mCallback.promote();
+ if (callback != 0) {
+ callback->onTearDown();
}
}
@@ -8107,8 +8115,9 @@
{
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;
}
@@ -8285,7 +8294,8 @@
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 +8306,7 @@
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..fbbfad7 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1527,7 +1527,7 @@
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/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 */