/external/chromium_org/media/audio/mac/ |
H A D | audio_input_mac.cc | 179 AudioQueueBufferRef audio_buffer) { 181 return AudioQueueEnqueueBuffer(audio_queue_, audio_buffer, 0, NULL); 188 AudioQueueBufferRef audio_buffer, 193 HandleInputBuffer(audio_queue, audio_buffer, start_time, 199 AudioQueueBufferRef audio_buffer, 204 DCHECK(audio_buffer->mAudioData); 207 DCHECK_EQ(0U, audio_buffer->mAudioDataByteSize); 211 if (audio_buffer->mAudioDataByteSize) { 226 uint8* audio_data = reinterpret_cast<uint8*>(audio_buffer->mAudioData); 230 this, audio_bus_.get(), audio_buffer 178 QueueNextBuffer( AudioQueueBufferRef audio_buffer) argument 185 HandleInputBufferStatic( void* data, AudioQueueRef audio_queue, AudioQueueBufferRef audio_buffer, const AudioTimeStamp* start_time, UInt32 num_packets, const AudioStreamPacketDescription* desc) argument 197 HandleInputBuffer( AudioQueueRef audio_queue, AudioQueueBufferRef audio_buffer, const AudioTimeStamp* start_time, UInt32 num_packets, const AudioStreamPacketDescription* packet_desc) argument [all...] |
H A D | audio_input_mac.h | 51 OSStatus QueueNextBuffer(AudioQueueBufferRef audio_buffer); 57 AudioQueueBufferRef audio_buffer, 64 AudioQueueBufferRef audio_buffer,
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H A D | audio_low_latency_input_mac.cc | 79 AudioBuffer* audio_buffer = audio_buffer_list_.mBuffers; local 80 audio_buffer->mNumberChannels = input_params.channels(); 81 audio_buffer->mDataByteSize = data_byte_size; 82 audio_buffer->mData = audio_data_buffer_.get();
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/external/chromium_org/content/browser/speech/ |
H A D | speech_recognition_browsertest.cc | 137 scoped_ptr<uint8[]> audio_buffer(new uint8[buffer_size]); 140 audio_buffer[i] = static_cast<uint8>(127 * sin(i * 3.14F / 143 memset(audio_buffer.get(), 0, buffer_size); 148 audio_bus->FromInterleaved(&audio_buffer.get()[0],
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
H A D | fakeaudiocapturemodule_unittest.cc | 118 int32_t GenerateZeroBuffer(void* audio_buffer, uint32_t audio_buffer_size) { argument 119 memset(audio_buffer, 0, audio_buffer_size); 122 int32_t CopyFromRecBuffer(void* audio_buffer, uint32_t audio_buffer_size) { argument 125 memcpy(audio_buffer, rec_buffer_, min_buffer_size);
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/external/qemu/distrib/sdl-1.2.15/src/audio/paudio/ |
H A D | SDL_paudio.c | 139 audio_buffer paud_bufinfo; 240 audio_buffer paud_bufinfo;
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/external/chromium_org/content/renderer/media/ |
H A D | webrtc_audio_device_impl.cc | 82 const int16* audio_buffer = audio_data; local 102 audio_buffer, 112 audio_buffer += frames_per_10_ms * number_of_channels;
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/external/webrtc/src/modules/audio_processing/ |
H A D | Android.mk | 21 audio_buffer.cc \
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/external/sonivox/jet_tools/JetCreator/ |
H A D | eas.py | 602 self.audio_buffer = AudioBufferType()
788 result = eas_dll.EAS_RenderWaveOut(self.handle, byref(self.audio_buffer), self.buf_size, byref(samplesRendered))
790 result = eas_dll.EAS_RenderAuxMixer(self.handle, byref(self.audio_buffer), byref(samplesRendered))
801 s.writeframesraw(self.audio_buffer)
905 self.audio_buffer = AudioBufferType()
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/external/chromium_org/third_party/webrtc/modules/ |
H A D | audio_processing.target.darwin-arm.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.darwin-arm64.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.darwin-mips.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.darwin-mips64.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.darwin-x86.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.darwin-x86_64.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.linux-arm.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.linux-arm64.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.linux-mips.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.linux-mips64.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.linux-x86.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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H A D | audio_processing.target.linux-x86_64.mk | 35 third_party/webrtc/modules/audio_processing/audio_buffer.cc \
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/external/srec/srec/test/SRecTestAudio/src/ |
H A D | SRecTestAudio.c | 143 asr_int16_t audio_buffer [MAX_AUDIO_BUFFER_SIZE]; member in struct:ApplicationData_t 1831 data->num_samples_read = pfread ( data->audio_buffer, sizeof ( asr_int16_t ), data->audio_buffer_requested_size, audio_file ); 1862 esr_status = SR_RecognizerPutAudio ( data->recognizer, data->audio_buffer, &data->num_samples_read, hit_eof ); 2403 audio_status = lhs_audioinGetSamples ( audio_input_handle, &data__num_samples_read, data->audio_buffer, &input_status );
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/external/srec/srec/test/SRecTest/src/ |
H A D | SRecTest.c | 144 asr_int16_t audio_buffer [MAX_AUDIO_BUFFER_SIZE]; member in struct:ApplicationData_t 2138 data->num_samples_read = pfread ( data->audio_buffer, sizeof ( asr_int16_t ), data->audio_buffer_requested_size, audio_file ); 2169 esr_status = SR_RecognizerPutAudio ( data->recognizer, data->audio_buffer, &data->num_samples_read, hit_eof );
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/external/chromium_org/media/ |
H A D | media.target.darwin-arm.mk | 69 media/base/audio_buffer.cc \
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H A D | media.target.darwin-arm64.mk | 69 media/base/audio_buffer.cc \
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