| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| AudioBufferCallback.idl | 28 void handleEvent(AudioBuffer audioBuffer);
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| AsyncAudioDecoder.cpp | 31 #include "modules/webaudio/AudioBuffer.h"
71 // The leaked reference to audioBuffer is picked up in notifyComplete.
83 RefPtrWillBeRawPtr<AudioBuffer> audioBuffer = AudioBuffer::createFromAudioBus(audioBus);
84 if (audioBuffer.get() && successCallback)
85 successCallback->handleEvent(audioBuffer.get());
87 errorCallback->handleEvent(audioBuffer.get());
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| AudioBuffer.cpp | 33 #include "modules/webaudio/AudioBuffer.h"
44 float AudioBuffer::minAllowedSampleRate()
50 float AudioBuffer::maxAllowedSampleRate()
56 PassRefPtrWillBeRawPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
61 RefPtrWillBeRawPtr<AudioBuffer> buffer = adoptRefWillBeNoop(new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate));
68 PassRefPtrWillBeRawPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
83 if (sampleRate < AudioBuffer::minAllowedSampleRate() || sampleRate > AudioBuffer::maxAllowedSampleRate())
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_device/ |
| mock_audio_device_buffer.h | 25 MOCK_METHOD1(GetPlayoutData, int32_t(void* audioBuffer));
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| audio_device_buffer.h | 53 virtual int32_t SetRecordedBuffer(const void* audioBuffer,
63 virtual int32_t GetPlayoutData(void* audioBuffer);
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| audio_device_buffer.cc | 391 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
419 memcpy(&_recBuffer[0], audioBuffer, _recSize);
423 int16_t* ptr16In = (int16_t*)audioBuffer;
574 int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer)
586 memcpy(audioBuffer, &_playBuffer[0], _playSize);
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| audio_device_generic.h | 173 virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
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| /external/chromium_org/third_party/webrtc/examples/android/opensl_loopback/ |
| fake_audio_device_buffer.cc | 57 int32_t FakeAudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
59 assert(audioBuffer);
64 memcpy(buffer, audioBuffer, nSamples * sizeof(int16_t));
74 int32_t FakeAudioDeviceBuffer::GetPlayoutData(void* audioBuffer) {
75 assert(audioBuffer);
78 memset(audioBuffer, 0, buffer_size_bytes());
82 memcpy(audioBuffer, buffer, buffer_size_bytes());
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| fake_audio_device_buffer.h | 32 virtual int32_t SetRecordedBuffer(const void* audioBuffer,
39 virtual int32_t GetPlayoutData(void* audioBuffer);
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| /external/chromium_org/third_party/webrtc/modules/media_file/source/ |
| media_file_utility.h | 41 // The return value is the number of bytes written to audioBuffer.
64 // audioBuffer, to file. The audio frame size is determined by the
68 int32_t WriteAviAudioData(const int8_t* audioBuffer,
94 // Put 10-60ms of audio data from stream into the audioBuffer depending on
95 // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
96 // The return value is the number of bytes written to audioBuffer.
100 int32_t ReadWavDataAsMono(InStream& stream, int8_t* audioBuffer,
122 // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
125 // The return value is the number of bytes written to audioBuffer.
127 const int8_t* audioBuffer,
[all...] |
| media_file_impl.h | 34 int32_t PlayoutAudioData(int8_t* audioBuffer, uint32_t& dataLengthInBytes);
58 int32_t IncomingAudioData(const int8_t* audioBuffer,
60 int32_t IncomingAVIVideoData(const int8_t* audioBuffer,
154 // audioBuffer. As output parameter it indicates the number of bytes
155 // written to audioBuffer. If video is true the data written is a video
160 // Write one frame, i.e. the bufferLength first bytes of audioBuffer,
|
| /frameworks/av/media/libmedia/ |
| AudioRecord.cpp | 582 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
584 if (audioBuffer == NULL) {
588 audioBuffer->frameCount = 0;
589 audioBuffer->size = 0;
590 audioBuffer->raw = NULL;
609 return obtainBuffer(audioBuffer, requested);
612 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
663 buffer.mFrameCount = audioBuffer->frameCount;
669 audioBuffer->frameCount = buffer.mFrameCount;
670 audioBuffer->size = buffer.mFrameCount * mFrameSize
[all...] |
| AudioTrack.cpp | [all...] |
| /external/chromium_org/third_party/webrtc/modules/media_file/interface/ |
| media_file.h | 32 // Put 10-60ms of audio data from file into the audioBuffer depending on
34 // parameter. As input parameter it indicates the size of audioBuffer.
36 // audioBuffer.
41 int8_t* audioBuffer,
128 // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
133 const int8_t* audioBuffer,
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| /frameworks/av/media/libstagefright/ |
| AudioSource.cpp | 273 status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
311 CHECK_EQ(audioBuffer.size & 1, 0u);
332 if (audioBuffer.size == 0) {
337 const size_t bufferSize = audioBuffer.size;
340 audioBuffer.i16, audioBuffer.size);
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| /external/chromium_org/third_party/webrtc/modules/utility/source/ |
| file_recorder_impl.h | 88 const int8_t* audioBuffer,
153 const int8_t* audioBuffer,
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| file_recorder_impl.cc | 311 const int8_t* audioBuffer,
316 return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
683 const int8_t* audioBuffer,
711 _audioFramesToWrite.push_back(new AudioFrameFileInfo(audioBuffer,
716 _audioFramesToWrite.push_back(new AudioFrameFileInfo(audioBuffer,
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| /external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
| audio_device_template.h | 384 AudioDeviceBuffer* audioBuffer) {
385 output_.AttachAudioBuffer(audioBuffer);
386 input_.AttachAudioBuffer(audioBuffer);
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| audio_record_jni.h | 108 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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| audio_track_jni.h | 95 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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| opensles_input.h | 119 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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| opensles_output.h | 114 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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| /frameworks/av/include/media/ |
| AudioRecord.h | 323 /* Obtains a buffer of up to "audioBuffer->frameCount" full frames.
358 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
368 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
372 /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */
374 void releaseBuffer(Buffer* audioBuffer);
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| AudioTrack.h | 487 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
523 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
533 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
537 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
539 void releaseBuffer(Buffer* audioBuffer);
[all...] |
| /frameworks/base/media/java/android/media/ |
| AudioRecord.java | 729 * @param audioBuffer the direct buffer to which the recorded audio data is written.
736 public int read(ByteBuffer audioBuffer, int sizeInBytes) {
741 if ( (audioBuffer == null) || (sizeInBytes < 0) ) {
745 return native_read_in_direct_buffer(audioBuffer, sizeInBytes);
[all...] |
