| /external/webrtc/modules/audio_device/ |
| D | fine_audio_buffer.cc | 65 void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer, in GetPlayoutData() argument
71 while (playout_buffer_.size() < audio_buffer.size()) { in GetPlayoutData()
91 const size_t num_bytes = audio_buffer.size() * sizeof(int16_t); in GetPlayoutData()
92 std::memset(audio_buffer.data(), 0, num_bytes); in GetPlayoutData()
98 const size_t num_bytes = audio_buffer.size() * sizeof(int16_t); in GetPlayoutData()
99 memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes); in GetPlayoutData()
101 memmove(playout_buffer_.data(), playout_buffer_.data() + audio_buffer.size(), in GetPlayoutData()
102 (playout_buffer_.size() - audio_buffer.size()) * sizeof(int16_t)); in GetPlayoutData()
103 playout_buffer_.SetSize(playout_buffer_.size() - audio_buffer.size()); in GetPlayoutData()
109 rtc::ArrayView<const int16_t> audio_buffer, in DeliverRecordedData() argument
[all …]
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| D | fine_audio_buffer.h | 51 void GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
63 void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
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| D | audio_device_buffer.cc | 230 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer, in SetRecordedBuffer() argument
234 rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer), in SetRecordedBuffer()
325 int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) { in GetPlayoutData() argument
330 int16_t* destination_r = reinterpret_cast<int16_t*>(audio_buffer); in GetPlayoutData()
344 memcpy(audio_buffer, play_buffer_.data(), in GetPlayoutData()
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| D | audio_device_buffer.h | 100 virtual int32_t SetRecordedBuffer(const void* audio_buffer,
107 virtual int32_t GetPlayoutData(void* audio_buffer);
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| /external/webrtc/modules/audio_processing/aec3/ |
| D | block_delay_buffer_unittest.cc | 77 AudioBuffer audio_buffer(rate, num_channels, rate, num_channels, rate, in TEST_P() local
80 audio_buffer.SplitIntoFrequencyBands(); in TEST_P()
85 &audio_buffer.split_bands(ch)[0]); in TEST_P()
87 delay_buffer.DelaySignal(&audio_buffer); in TEST_P()
94 EXPECT_EQ(0.f, audio_buffer.split_bands(ch)[band][i]); in TEST_P()
97 audio_buffer.split_bands(ch)[band][i]); in TEST_P()
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| /external/webrtc/modules/audio_processing/transient/ |
| D | click_annotate.cc | 69 std::unique_ptr<float[]> audio_buffer(new float[audio_buffer_length]); in main() local
74 &pcm_file, audio_buffer_length, audio_buffer.get()); in main()
78 audio_buffer[i] = 0.0; in main()
81 detector.Detect(audio_buffer.get(), audio_buffer_length, NULL, 0); in main()
92 &pcm_file, audio_buffer_length, audio_buffer.get()); in main()
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| D | transient_suppression_test.cc | 73 int16_t* audio_buffer, in ReadBuffers() argument
80 int16_t* read_ptr = audio_buffer; in ReadBuffers()
93 audio_buffer[i * audio_buffer_size + j] = in ReadBuffers()
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| /external/webrtc/modules/audio_processing/ |
| D | voice_detection_unittest.cc | 26 AudioBuffer* audio_buffer, in ProcessOneFrame() argument
29 audio_buffer->SplitIntoFrequencyBands(); in ProcessOneFrame()
32 return voice_detection->ProcessCaptureAudio(audio_buffer); in ProcessOneFrame()
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| D | high_pass_filter_unittest.cc | 29 AudioBuffer audio_buffer( in ProcessOneFrameAsAudioBuffer() local
34 test::CopyVectorToAudioBuffer(stream_config, frame_input, &audio_buffer); in ProcessOneFrameAsAudioBuffer()
35 high_pass_filter->Process(&audio_buffer, /*use_split_band_data=*/false); in ProcessOneFrameAsAudioBuffer()
37 test::ExtractVectorFromAudioBuffer(stream_config, &audio_buffer, in ProcessOneFrameAsAudioBuffer()
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| D | BUILD.gn | 73 rtc_library("audio_buffer") {
79 "audio_buffer.cc",
80 "audio_buffer.h",
109 ":audio_buffer",
166 ":audio_buffer",
232 ":audio_buffer",
308 ":audio_buffer",
351 ":audio_buffer",
601 ":audio_buffer",
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| /external/adhd/cras/src/server/ |
| D | cras_empty_iodev.c | 33 uint8_t *audio_buffer; member
85 free(empty_iodev->audio_buffer); in close_dev()
86 empty_iodev->audio_buffer = NULL; in close_dev()
99 empty_iodev->audio_buffer = calloc(1, EMPTY_BUFFER_SIZE); in configure_dev()
124 empty_iodev->audio_buffer); in get_buffer()
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| /external/webrtc/pc/test/ |
| D | fake_audio_capture_module_unittest.cc | 106 size_t GenerateZeroBuffer(void* audio_buffer, size_t audio_buffer_size) { in GenerateZeroBuffer() argument
107 memset(audio_buffer, 0, audio_buffer_size); in GenerateZeroBuffer()
110 size_t CopyFromRecBuffer(void* audio_buffer, size_t audio_buffer_size) { in CopyFromRecBuffer() argument
114 memcpy(audio_buffer, rec_buffer_, min_buffer_size); in CopyFromRecBuffer()
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| /external/adhd/cras/client/libcras/src/ |
| D | cras_stream.rs | 129 audio_buffer: CrasAudioBuffer, field
150 let (header, audio_buffer) = create_header_and_buffers(header_fd, samples_fd)?; in try_new()
162 audio_buffer, in try_new()
205 let buf = &mut self.audio_buffer.get_buffer()[offset..offset + len]; in next_playback_buffer()
220 let buf = &mut self.audio_buffer.get_buffer()[offset..offset + len]; in next_capture_buffer()
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| /external/webrtc/modules/audio_device/win/ |
| D | audio_device_module_win.h | 39 virtual void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) = 0;
63 virtual void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) = 0;
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| D | core_audio_output_win.h | 43 void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
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| D | core_audio_input_win.h | 43 void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
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| D | core_audio_output_win.cc | 87 void CoreAudioOutput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { in AttachAudioBuffer() argument
90 audio_device_buffer_ = audio_buffer; in AttachAudioBuffer()
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| D | core_audio_input_win.cc | 89 void CoreAudioInput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { in AttachAudioBuffer() argument
92 audio_device_buffer_ = audio_buffer; in AttachAudioBuffer()
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| /external/webrtc/modules/audio_processing/ns/ |
| D | BUILD.gn | 53 "..:audio_buffer",
81 "..:audio_buffer",
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| /external/webrtc/sdk/objc/unittests/ |
| D | RTCAudioDevice_xctest.mm | 88 std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
89 audio_buffer.reset(new webrtc::AudioDeviceBuffer(task_queue_factory.get()));
90 _audio_device->AttachAudioBuffer(audio_buffer.get());
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| /external/webrtc/modules/audio_device/android/ |
| D | opensles_recorder.cc | 168 void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { in AttachAudioBuffer() argument
171 RTC_CHECK(audio_buffer); in AttachAudioBuffer()
172 audio_device_buffer_ = audio_buffer; in AttachAudioBuffer()
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| D | opensles_recorder.h | 76 void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer);
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| /external/webrtc/sdk/android/src/jni/audio_device/ |
| D | opensles_recorder.cc | 178 void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { in AttachAudioBuffer() argument
181 RTC_CHECK(audio_buffer); in AttachAudioBuffer()
182 audio_device_buffer_ = audio_buffer; in AttachAudioBuffer()
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| D | opensles_recorder.h | 80 void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
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| /external/webrtc/audio/ |
| D | audio_state_unittest.cc | 265 int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels]; in TEST() local
271 kSampleRate, audio_buffer, n_samples_out, &elapsed_time_ms, in TEST()
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