| /external/webrtc/webrtc/modules/audio_device/ |
| mock_audio_device_buffer.h | 24 MOCK_METHOD1(GetPlayoutData, int32_t(void* audioBuffer));
26 int32_t(const void* audioBuffer, size_t nSamples));
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| audio_device_buffer.h | 52 virtual int32_t SetRecordedBuffer(const void* audioBuffer,
62 virtual int32_t GetPlayoutData(void* audioBuffer);
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| audio_device_buffer.cc | 383 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
405 memcpy(&_recBuffer[0], audioBuffer, _recSize);
409 int16_t* ptr16In = (int16_t*)audioBuffer;
560 int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer)
573 memcpy(audioBuffer, &_playBuffer[0], _playSize);
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| audio_device_generic.h | 174 virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
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| /system/bt/btif/include/ |
| btif_avrcp_audio_track.h | 69 int BtifAvrcpAudioTrackWriteData(void *handle, void *audioBuffer, int bufferlen);
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| /frameworks/base/media/java/android/media/ |
| MediaSync.java | 53 * public void onAudioBufferConsumed(MediaSync sync, ByteBuffer audioBuffer, int bufferId) {
126 * @param audioBuffer The returned audio buffer.
127 * @param bufferId The ID associated with audioBuffer as passed into
131 @NonNull MediaSync sync, @NonNull ByteBuffer audioBuffer, int bufferId);
172 private static class AudioBuffer {
177 public AudioBuffer(@NonNull ByteBuffer byteBuffer, int bufferId,
201 private List<AudioBuffer> mAudioBuffers = new LinkedList<AudioBuffer>();
515 mAudioBuffers.add(new AudioBuffer(audioData, bufferId, presentationTimeUs));
536 AudioBuffer audioBuffer = mAudioBuffers.get(0)
[all...] |
| /external/webrtc/webrtc/modules/media_file/ |
| media_file_utility.h | 38 // Put 10-60ms of audio data from stream into the audioBuffer depending on
39 // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
40 // The return value is the number of bytes written to audioBuffer.
44 int32_t ReadWavDataAsMono(InStream& stream, int8_t* audioBuffer,
66 // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
69 // The return value is the number of bytes written to audioBuffer.
71 const int8_t* audioBuffer,
91 // Put 10-60ms of audio data from stream into the audioBuffer depending on
92 // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
93 // The return value is the number of bytes written to audioBuffer
[all...] |
| media_file.h | 29 // Put 10-60ms of audio data from file into the audioBuffer depending on
31 // parameter. As input parameter it indicates the size of audioBuffer.
33 // audioBuffer.
38 int8_t* audioBuffer,
107 // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
112 const int8_t* audioBuffer,
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| media_file_impl.h | 33 int32_t PlayoutAudioData(int8_t* audioBuffer,
63 int32_t IncomingAudioData(const int8_t* audioBuffer,
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| /frameworks/av/media/libmedia/ |
| AudioRecord.cpp | 739 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
741 if (audioBuffer == NULL) {
748 audioBuffer->frameCount = 0;
749 audioBuffer->size = 0;
750 audioBuffer->raw = NULL;
772 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
775 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
826 buffer.mFrameCount = audioBuffer->frameCount;
832 audioBuffer->frameCount = buffer.mFrameCount;
833 audioBuffer->size = buffer.mFrameCount * mFrameSize
[all...] |
| AudioTrack.cpp | [all...] |
| /system/bt/btif/src/ |
| btif_avrcp_audio_track.cpp | 136 int BtifAvrcpAudioTrackWriteData(void *handle, void *audioBuffer, int bufferlen)
145 fwrite ((audioBuffer), 1, (size_t)bufferlen, outputPcmSampleFile);
148 retval = trackHolder->track->write(audioBuffer, (size_t)bufferlen);
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| /frameworks/av/media/libstagefright/ |
| AudioSource.cpp | 302 status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
344 CHECK_EQ(audioBuffer.size & 1, 0u);
365 if (audioBuffer.size == 0) {
370 const size_t bufferSize = audioBuffer.size;
373 audioBuffer.i16, audioBuffer.size);
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| /cts/tests/tests/media/src/android/media/cts/ |
| CodecState.java | 344 ByteBuffer audioBuffer = ByteBuffer.allocate(buffer.remaining());
345 audioBuffer.put(buffer);
346 audioBuffer.clear();
348 mAudioTrack.write(audioBuffer, info.size, info.presentationTimeUs*1000);
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| AudioHelper.java | 351 public int read(ByteBuffer audioBuffer, int sizeInBytes) {
352 int bytes = super.read(audioBuffer, sizeInBytes);
354 // read does not affect position and limit of the audioBuffer.
357 ByteBuffer copy = audioBuffer.duplicate();
368 public int read(ByteBuffer audioBuffer, int sizeInBytes, int readMode) {
369 int bytes = super.read(audioBuffer, sizeInBytes, readMode);
371 // read does not affect position and limit of the audioBuffer.
374 ByteBuffer copy = audioBuffer.duplicate();
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| opensles_player.h | 82 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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| audio_record_jni.h | 82 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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| audio_track_jni.h | 84 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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| audio_device_template.h | 431 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override {
432 output_.AttachAudioBuffer(audioBuffer);
433 input_.AttachAudioBuffer(audioBuffer);
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| audio_record_jni.cc | 181 void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
184 audio_device_buffer_ = audioBuffer;
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| audio_track_jni.cc | 198 void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
201 audio_device_buffer_ = audioBuffer;
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| /external/webrtc/webrtc/modules/utility/source/ |
| file_recorder_impl.h | 77 int32_t WriteEncodedAudioData(const int8_t* audioBuffer,
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| file_recorder_impl.cc | 256 int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
259 return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
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| /frameworks/av/include/media/ |
| AudioRecord.h | 358 * Obtains a buffer of up to "audioBuffer->frameCount" full frames.
398 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount,
454 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
459 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill.
466 void releaseBuffer(const Buffer* audioBuffer);
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| AudioTrack.h | 629 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
[all...] |
