Update NBAIO to use the new audio HAL abstraction layer
Moved the HAL access abstraction layer to a separate library so it
can be used both by audioflinger and libnbaio.
Bug: 30222631
Test: manual with Loopback app, Hangouts, YouTube
Change-Id: Id622c2f1aa8f55a775d34f369a596c2c4d29d5be
diff --git a/include/media/audiohal/StreamHalInterface.h b/include/media/audiohal/StreamHalInterface.h
new file mode 100644
index 0000000..5a7b4b6
--- /dev/null
+++ b/include/media/audiohal/StreamHalInterface.h
@@ -0,0 +1,153 @@
+/*
+ * 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 ANDROID_HARDWARE_STREAM_HAL_INTERFACE_H
+#define ANDROID_HARDWARE_STREAM_HAL_INTERFACE_H
+
+#include <hardware/audio.h>
+#include <media/audiohal/EffectHalInterface.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <utils/String8.h>
+
+namespace android {
+
+class StreamHalInterface : public virtual RefBase
+{
+ public:
+ // Return the sampling rate in Hz - eg. 44100.
+ virtual status_t getSampleRate(uint32_t *rate) = 0;
+
+ // Return size of input/output buffer in bytes for this stream - eg. 4800.
+ virtual status_t getBufferSize(size_t *size) = 0;
+
+ // Return the channel mask.
+ virtual status_t getChannelMask(audio_channel_mask_t *mask) = 0;
+
+ // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
+ virtual status_t getFormat(audio_format_t *format) = 0;
+
+ // Convenience method.
+ virtual status_t getAudioProperties(
+ uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) = 0;
+
+ // Set audio stream parameters.
+ virtual status_t setParameters(const String8& kvPairs) = 0;
+
+ // Get audio stream parameters.
+ virtual status_t getParameters(const String8& keys, String8 *values) = 0;
+
+ // Return the frame size (number of bytes per sample) of a stream.
+ virtual status_t getFrameSize(size_t *size) = 0;
+
+ // Add or remove the effect on the stream.
+ virtual status_t addEffect(sp<EffectHalInterface> effect) = 0;
+ virtual status_t removeEffect(sp<EffectHalInterface> effect) = 0;
+
+ // Put the audio hardware input/output into standby mode.
+ virtual status_t standby() = 0;
+
+ virtual status_t dump(int fd) = 0;
+
+ protected:
+ // Subclasses can not be constructed directly by clients.
+ StreamHalInterface() {}
+
+ // The destructor automatically closes the stream.
+ virtual ~StreamHalInterface() {}
+};
+
+class StreamOutHalInterfaceCallback : public virtual RefBase {
+ public:
+ virtual void onWriteReady() {}
+ virtual void onDrainReady() {}
+ virtual void onError() {}
+
+ protected:
+ StreamOutHalInterfaceCallback() {}
+ virtual ~StreamOutHalInterfaceCallback() {}
+};
+
+class StreamOutHalInterface : public virtual StreamHalInterface {
+ public:
+ // Return the audio hardware driver estimated latency in milliseconds.
+ virtual status_t getLatency(uint32_t *latency) = 0;
+
+ // Use this method in situations where audio mixing is done in the hardware.
+ virtual status_t setVolume(float left, float right) = 0;
+
+ // Write audio buffer to driver.
+ virtual status_t write(const void *buffer, size_t bytes, size_t *written) = 0;
+
+ // Return the number of audio frames written by the audio dsp to DAC since
+ // the output has exited standby.
+ virtual status_t getRenderPosition(uint32_t *dspFrames) = 0;
+
+ // Get the local time at which the next write to the audio driver will be presented.
+ virtual status_t getNextWriteTimestamp(int64_t *timestamp) = 0;
+
+ // Set the callback for notifying completion of non-blocking write and drain.
+ // The callback must be owned by someone else. The output stream does not own it
+ // to avoid strong pointer loops.
+ virtual status_t setCallback(sp<StreamOutHalInterfaceCallback> callback) = 0;
+
+ // Returns whether pause and resume operations are supported.
+ virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume) = 0;
+
+ // Notifies to the audio driver to resume playback following a pause.
+ virtual status_t pause() = 0;
+
+ // Notifies to the audio driver to resume playback following a pause.
+ virtual status_t resume() = 0;
+
+ // Returns whether drain operation is supported.
+ virtual status_t supportsDrain(bool *supportsDrain) = 0;
+
+ // Requests notification when data buffered by the driver/hardware has been played.
+ virtual status_t drain(audio_drain_type_t type) = 0;
+
+ // Notifies to the audio driver to flush the queued data.
+ virtual status_t flush() = 0;
+
+ // Return a recent count of the number of audio frames presented to an external observer.
+ virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) = 0;
+
+ protected:
+ virtual ~StreamOutHalInterface() {}
+};
+
+class StreamInHalInterface : public virtual StreamHalInterface {
+ public:
+ // Set the input gain for the audio driver.
+ virtual status_t setGain(float gain) = 0;
+
+ // Read audio buffer in from driver.
+ virtual status_t read(void *buffer, size_t bytes, size_t *read) = 0;
+
+ // Return the amount of input frames lost in the audio driver.
+ virtual status_t getInputFramesLost(uint32_t *framesLost) = 0;
+
+ // Return a recent count of the number of audio frames received and
+ // the clock time associated with that frame count.
+ virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0;
+
+ protected:
+ virtual ~StreamInHalInterface() {}
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_STREAM_HAL_INTERFACE_H