/external/webrtc/modules/audio_coding/acm2/ |
D | acm_remixing.cc | 33 size_t num_output_channels, in ReMixFrame() argument 35 const size_t output_size = num_output_channels * input.samples_per_channel_; in ReMixFrame() 36 RTC_DCHECK(!(input.num_channels_ == 0 && num_output_channels > 0 && in ReMixFrame() 60 if (input.num_channels_ == 1 && input.num_channels_ < num_output_channels) { in ReMixFrame() 64 for (size_t j = 2; j < num_output_channels; ++j) { in ReMixFrame() 67 RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels); in ReMixFrame() 69 RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels); in ReMixFrame() 77 if (input.num_channels_ < num_output_channels) { in ReMixFrame() 82 for (size_t j = input.num_channels_; j < num_output_channels; ++j) { in ReMixFrame() 86 RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels); in ReMixFrame() [all …]
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D | acm_remixing.h | 29 size_t num_output_channels,
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/external/webrtc/modules/audio_processing/ |
D | echo_control_mobile_impl.cc | 66 size_t num_output_channels) in StreamProperties() 69 num_output_channels(num_output_channels) {} in StreamProperties() 73 size_t num_output_channels; member 115 packed_render_audio.size() / (stream_properties_->num_output_channels * in ProcessRenderAudio() 129 size_t num_output_channels, in PackRenderAudioBuffer() argument 139 for (size_t i = 0; i < num_output_channels; i++) { in PackRenderAudioBuffer() 155 size_t num_output_channels, in NumCancellersRequired() argument 157 return num_output_channels * num_reverse_channels; in NumCancellersRequired() 164 RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_output_channels); in ProcessCaptureAudio() 246 size_t num_output_channels) { in Initialize() argument [all …]
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D | echo_control_mobile_impl.h | 60 size_t num_output_channels); 63 size_t num_output_channels, 67 static size_t NumCancellersRequired(size_t num_output_channels,
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D | audio_processing_unittest.cc | 265 size_t num_output_channels, in OutputFilePath() argument 272 if (num_output_channels == 1) { in OutputFilePath() 274 } else if (num_output_channels == 2) { in OutputFilePath() 368 size_t num_output_channels, 473 size_t num_output_channels, in Init() argument 478 num_output_channels_ = num_output_channels; in Init() 504 reverse_sample_rate_hz, num_input_channels, num_output_channels, in Init() 782 {output_sample_rate_hz_, apm_->num_output_channels()}, in TestChangingReverseChannels() 820 EXPECT_EQ(j, apm_->num_output_channels()); in TEST_F() 1375 msg.num_input_channels(), msg.num_output_channels(), in ProcessDebugDump() [all …]
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D | debug.proto | 11 optional int32 num_output_channels = 4; field
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D | audio_processing_impl.cc | 683 return num_output_channels(); in num_proc_channels() 686 size_t AudioProcessingImpl::num_output_channels() const { in num_output_channels() function in webrtc::AudioProcessingImpl 900 EchoControlMobileImpl::PackRenderAudioBuffer(audio, num_output_channels(), in QueueBandedRenderAudio() 1649 use_full_band ? num_output_channels() : num_proc_channels(); in InitializeHighPassFilter() 1726 num_output_channels(), num_reverse_channels())); in InitializeEchoController() 1742 num_output_channels()); in InitializeEchoController()
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D | audio_processing_impl.h | 126 size_t num_output_channels() const override;
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/external/webrtc/modules/audio_coding/codecs/opus/test/ |
D | lapped_transform.cc | 25 size_t num_output_channels, in ProcessBlock() argument 28 RTC_CHECK_EQ(num_output_channels, parent_->num_out_channels_); in ProcessBlock() 42 num_output_channels, parent_->cplx_post_.Array()); in ProcessBlock() 44 for (size_t i = 0; i < num_output_channels; ++i) { in ProcessBlock()
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D | blocker.h | 31 size_t num_output_channels, 70 size_t num_output_channels, 79 size_t num_output_channels,
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D | blocker_unittest.cc | 26 size_t num_output_channels, in ProcessBlock() argument 28 for (size_t i = 0; i < num_output_channels; ++i) { in ProcessBlock() 42 size_t num_output_channels, in ProcessBlock() argument 44 for (size_t i = 0; i < num_output_channels; ++i) { in ProcessBlock() 69 size_t num_output_channels) { in RunTest() argument 75 num_output_channels, output_chunk); in RunTest() 76 CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk); in RunTest()
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D | blocker.cc | 102 size_t num_output_channels, in Blocker() argument 109 num_output_channels_(num_output_channels), in Blocker() 170 size_t num_output_channels, in ProcessChunk() argument 174 RTC_CHECK_EQ(num_output_channels, num_output_channels_); in ProcessChunk()
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D | lapped_transform.h | 149 size_t num_output_channels,
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/external/webrtc/rtc_tools/unpack_aecdump/ |
D | unpack.cc | 251 size_t num_output_channels = 0; in do_main() local 346 num_output_channels * output_samples_per_channel, in do_main() 354 new const float*[num_output_channels]); in do_main() 355 for (size_t i = 0; i < num_output_channels; ++i) { in do_main() 360 num_output_channels, output_wav_file.get(), in do_main() 468 num_output_channels = msg.num_output_channels(); in do_main() 470 num_output_channels); in do_main() 511 output_name.str(), output_sample_rate, num_output_channels)); in do_main()
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/external/webrtc/modules/audio_processing/test/ |
D | aec_dump_based_simulator.cc | 459 int num_output_channels; in HandleMessage() local 461 num_output_channels = *settings_.output_num_channels; in HandleMessage() 463 num_output_channels = msg.has_num_output_channels() in HandleMessage() 464 ? msg.num_output_channels() in HandleMessage() 496 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, in HandleMessage()
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D | unittest.proto | 8 optional int32 num_output_channels = 3; field
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D | debug_dump_replayer.cc | 108 StreamConfig(msg.output_sample_rate(), msg.num_output_channels()); in OnInitEvent()
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/external/webrtc/modules/audio_coding/codecs/opus/ |
D | opus_bandwidth_unittest.cc | 50 size_t num_output_channels, in ProcessAudioBlock() argument
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/external/webrtc/modules/audio_processing/include/ |
D | mock_audio_processing.h | 99 MOCK_METHOD(size_t, num_output_channels, (), (const, override));
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D | audio_processing.h | 518 virtual size_t num_output_channels() const = 0;
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