Searched refs:output_sample_rate (Results 1 – 11 of 11) sorted by relevance
| /external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
| D | unpack.cc | 209 int output_sample_rate = msg.output_sample_rate(); in do_main() local
210 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate); in do_main()
227 if (output_sample_rate == 0) { in do_main()
228 output_sample_rate = input_sample_rate; in do_main()
233 output_samples_per_channel = output_sample_rate / 100; in do_main()
245 output_sample_rate, in do_main()
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| D | process_test.cc | 579 int output_sample_rate = 32000; in void_main() local
599 output_sample_rate = msg.sample_rate(); in void_main()
601 output_sample_rate = msg.output_sample_rate(); in void_main()
606 output_sample_rate, in void_main()
629 msg.sample_rate(), output_sample_rate); in void_main()
641 output_sample_rate, in void_main()
749 output_sample_rate, in void_main()
789 const size_t samples_per_channel = output_sample_rate / 100; in void_main()
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| D | audio_processing_unittest.cc | 1465 int output_sample_rate = msg.sample_rate(); in ProcessDebugDump() local
1467 output_sample_rate = msg.output_sample_rate(); in ProcessDebugDump()
1471 output_sample_rate, in ProcessDebugDump()
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| /external/chromium_org/content/renderer/media/ |
| D | media_stream_renderer_factory.cc | 30 int* output_sample_rate, int* output_buffer_size) { in GetDefaultOutputDeviceParams() argument
34 *output_sample_rate = hardware_config->GetOutputSampleRate(); in GetDefaultOutputDeviceParams()
48 int* output_sample_rate, in GetAuthorizedDeviceInfoForAudioRenderer() argument
56 session_id, output_sample_rate, output_frames_per_buffer); in GetAuthorizedDeviceInfoForAudioRenderer()
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| D | media_stream_audio_processor.cc | 466 const int output_sample_rate = audio_processing_ ? in InitializeCaptureFifo() local
483 int output_frames = output_sample_rate / 100; in InitializeCaptureFifo()
492 output_sample_rate, in InitializeCaptureFifo()
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| D | webrtc_audio_capturer.h | 97 int* output_sample_rate,
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| D | webrtc_audio_device_impl.cc | 517 int* output_sample_rate, in GetAuthorizedDeviceInfoForAudioRenderer() argument
526 session_id, output_sample_rate, output_frames_per_buffer); in GetAuthorizedDeviceInfoForAudioRenderer()
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| D | webrtc_audio_capturer.cc | 576 int* output_sample_rate, in GetPairedOutputParameters() argument
585 *output_sample_rate = device_info_.device.matched_output.sample_rate; in GetPairedOutputParameters()
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| D | webrtc_audio_device_impl.h | 350 int* session_id, int* output_sample_rate, int* output_buffer_size); in NON_EXPORTED_BASE()
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| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| D | debug.proto | 12 optional int32 output_sample_rate = 7; field
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| /external/chromium_org/chrome/renderer/media/ |
| D | cast_rtp_stream.cc | 343 int output_sample_rate) in CastAudioSink() argument
349 output_sample_rate_(output_sample_rate), in CastAudioSink()
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