| /external/chromium_org/third_party/webrtc/examples/android/opensl_loopback/jni/ |
| H A D | opensl_runner.cc | 34 output_.AttachAudioBuffer(&audio_buffer_);
41 input_.AttachAudioBuffer(&audio_buffer_);
63 audio_buffer_.ClearBuffer();
69 FakeAudioDeviceBuffer audio_buffer_; member in class:webrtc::OpenSlRunnerTemplate
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| /external/chromium_org/media/filters/ |
| H A D | audio_renderer_algorithm.cc | 63 // The maximum size in seconds for the |audio_buffer_|. Arbitrarily determined.
66 // The starting size in frames for |audio_buffer_|. Previous usage maintained a
156 std::min(static_cast<int>(audio_buffer_.frames() / playback_rate),
166 audio_buffer_.SeekFrames(seek_frames);
184 std::min(audio_buffer_.frames(), requested_frames);
185 const int frames_read = audio_buffer_.ReadFrames(frames_to_copy, 0, dest);
201 audio_buffer_.Clear();
216 audio_buffer_.Append(buffer_in);
220 return audio_buffer_.frames() >= capacity_;
232 const int frames = audio_buffer_
[all...] |
| H A D | audio_renderer_algorithm.h | 43 // data from our |audio_buffer_|. Data is scaled based on |playback_rate|,
46 // Data from |audio_buffer_| is consumed in proportion to the playback rate.
51 // Clears |audio_buffer_|.
58 // Returns true if |audio_buffer_| is at or exceeds capacity.
61 // Returns the capacity of |audio_buffer_| in frames.
64 // Increase the capacity of |audio_buffer_| if possible.
67 // Returns the number of frames left in |audio_buffer_|, which may be larger
69 // than |audio_buffer_| was intending to hold.
70 int frames_buffered() { return audio_buffer_.frames(); }
87 // Fill |dest| with frames from |audio_buffer_| startin
123 AudioBufferQueue audio_buffer_; member in class:media::AudioRendererAlgorithm
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
| H A D | opensles_input.cc | 62 audio_buffer_(NULL),
243 audio_buffer_ = audioBuffer;
248 audio_buffer_->SetRecordingSampleRate(rec_sampling_rate_);
249 audio_buffer_->SetRecordingChannels(kNumChannels);
517 audio_buffer_->SetRecordedBuffer(audio, buffer_size_samples());
518 audio_buffer_->SetVQEData(delay_provider_->PlayoutDelayMs(),
520 audio_buffer_->DeliverRecordedData();
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| H A D | opensles_output.cc | 61 audio_buffer_(NULL),
269 audio_buffer_ = audioBuffer;
291 if (audio_buffer_->SetPlayoutSampleRate(speaker_sampling_rate_) < 0) {
294 if (audio_buffer_->SetPlayoutChannels(kNumChannels) < 0) {
333 fine_buffer_.reset(new FineAudioBuffer(audio_buffer_, buffer_size_bytes_,
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| H A D | opensles_input.h | 206 AudioDeviceBuffer* audio_buffer_; member in class:webrtc::OpenSlesInput
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| H A D | opensles_output.h | 224 AudioDeviceBuffer* audio_buffer_; member in class:webrtc::OpenSlesOutput
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| /external/chromium_org/media/audio/alsa/ |
| H A D | alsa_input.cc | 91 audio_buffer_.reset(new uint8[bytes_per_buffer_]);
210 int frames_read = wrapper_->PcmReadi(device_handle_, audio_buffer_.get(),
213 audio_bus_->FromInterleaved(audio_buffer_.get(),
267 audio_buffer_.reset();
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| H A D | alsa_input.h | 84 scoped_ptr<uint8[]> audio_buffer_; // Buffer used for reading audio data. member in class:media::AlsaPcmInputStream
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| /external/chromium_org/media/ffmpeg/ |
| H A D | ffmpeg_unittest.cc | 94 audio_buffer_.reset(av_frame_alloc());
236 av_frame_unref(audio_buffer_.get());
237 result = avcodec_decode_audio4(av_audio_context(), audio_buffer_.get(),
249 double microseconds = 1.0L * audio_buffer_->nb_samples /
385 scoped_ptr<AVFrame, media::ScopedPtrAVFreeFrame> audio_buffer_; member in class:media::FFmpegTest
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| H A D | acm_receiver.cc | 371 ptr_audio_buffer = audio_buffer_;
412 if (ptr_audio_buffer == audio_buffer_) {
416 resampler_.Resample10Msec(audio_buffer_,
428 memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
441 audio_buffer_);
446 memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
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| H A D | acm_receiver.h | 337 int16_t audio_buffer_[AudioFrame::kMaxDataSizeSamples] GUARDED_BY(crit_sect_);
|