| /external/webrtc/webrtc/modules/audio_processing/test/ |
| D | unpack.cc | 85 size_t num_output_channels = 0; in do_main() local
167 num_output_channels * output_samples_per_channel, in do_main()
175 new const float* [num_output_channels]); in do_main()
176 for (size_t i = 0; i < num_output_channels; ++i) { in do_main()
182 num_output_channels, in do_main()
274 num_output_channels = msg.num_output_channels(); in do_main()
276 num_output_channels); in do_main()
308 num_output_channels)); in do_main()
|
| D | audio_processing_unittest.cc | 263 size_t num_output_channels, in OutputFilePath() argument
270 if (num_output_channels == 1) { in OutputFilePath()
272 } else if (num_output_channels == 2) { in OutputFilePath()
361 size_t num_output_channels,
486 size_t num_output_channels, in Init() argument
491 num_output_channels_ = num_output_channels; in Init()
519 reverse_sample_rate_hz, num_input_channels, num_output_channels, in Init()
843 {output_sample_rate_hz_, apm_->num_output_channels()}, in TestChangingReverseChannels()
883 EXPECT_EQ(j, apm_->num_output_channels()); in TEST_F()
1735 msg.num_output_channels(), in ProcessDebugDump()
[all …]
|
| D | process_test.cc | 606 LayoutFromChannels(static_cast<size_t>(msg.num_output_channels())); in void_main()
638 msg.num_output_channels()); in void_main()
648 static_cast<size_t>(msg.num_output_channels()))); in void_main()
754 ASSERT_TRUE(near_frame.num_channels_ == apm->num_output_channels()); in void_main()
807 apm->num_output_channels() * samples_per_channel, in void_main()
816 apm->num_output_channels(), in void_main()
1003 ASSERT_TRUE(near_frame.num_channels_ == apm->num_output_channels()); in void_main()
|
| D | unittest.proto | 8 optional int32 num_output_channels = 3; field
|
| D | debug_dump_test.cc | 308 StreamConfig(msg.output_sample_rate(), msg.num_output_channels()); in OnInitEvent()
|
| /external/webrtc/webrtc/common_audio/ |
| 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()
44 num_output_channels, in ProcessBlock()
47 for (size_t i = 0; i < num_output_channels; ++i) { in ProcessBlock()
|
| D | blocker.h | 30 size_t num_output_channels,
69 size_t num_output_channels,
77 size_t num_output_channels,
|
| D | blocker_unittest.cc | 24 size_t num_output_channels, in ProcessBlock() argument
26 for (size_t i = 0; i < num_output_channels; ++i) { in ProcessBlock()
40 size_t num_output_channels, in ProcessBlock() argument
42 for (size_t i = 0; i < num_output_channels; ++i) { in ProcessBlock()
67 size_t num_output_channels) { in RunTest() argument
75 num_output_channels, in RunTest()
77 CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk); in RunTest()
|
| D | blocker.cc | 104 size_t num_output_channels, in Blocker() argument
111 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()
|
| D | lapped_transform.h | 99 size_t num_output_channels,
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| D | echo_control_mobile_impl.cc | 107 for (size_t i = 0; i < apm_->num_output_channels(); i++) { in ProcessRenderAudio()
153 (apm_->num_output_channels() * apm_->num_reverse_channels()); in ReadQueuedRenderData()
154 for (size_t i = 0; i < apm_->num_output_channels(); i++) { in ReadQueuedRenderData()
179 assert(audio->num_channels() == apm_->num_output_channels()); in ProcessCaptureAudio()
399 return apm_->num_output_channels() * apm_->num_reverse_channels(); in num_handles_required()
|
| D | echo_cancellation_impl.cc | 102 for (size_t i = 0; i < apm_->num_output_channels(); i++) { in ProcessRenderAudio()
148 (apm_->num_output_channels() * apm_->num_reverse_channels()); in ReadQueuedRenderData()
149 for (size_t i = 0; i < apm_->num_output_channels(); i++) { in ReadQueuedRenderData()
494 return apm_->num_output_channels() * apm_->num_reverse_channels(); in num_handles_required()
|
| D | debug.proto | 11 optional int32 num_output_channels = 4; field
|
| D | audio_processing_impl.h | 106 size_t num_output_channels() const override;
|
| D | audio_processing_impl.cc | 539 return capture_nonlocked_.beamformer_enabled ? 1 : num_output_channels(); in num_proc_channels()
542 size_t AudioProcessingImpl::num_output_channels() const { in num_output_channels() function in webrtc::AudioProcessingImpl
|
| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| D | nonlinear_beamformer.h | 72 size_t num_output_channels,
|
| D | nonlinear_beamformer.cc | 413 size_t num_output_channels, in ProcessAudioBlock() argument
417 RTC_CHECK_EQ(1u, num_output_channels); in ProcessAudioBlock()
|
| /external/webrtc/webrtc/modules/audio_processing/include/ |
| D | mock_audio_processing.h | 205 MOCK_CONST_METHOD0(num_output_channels,
|
| D | audio_processing.h | 298 virtual size_t num_output_channels() const = 0;
|
| /external/webrtc/talk/media/webrtc/ |
| D | fakewebrtcvoiceengine.h | 82 size_t num_output_channels() const override { return 0; } in num_output_channels() function
|