/external/webrtc/modules/audio_coding/acm2/ |
D | acm_remixing_unittest.cc | 32 in.samples_per_channel_ = 480; in TEST() 35 for (size_t k = 0; k < in.samples_per_channel_; ++k) { in TEST() 49 in.samples_per_channel_ = 480; in TEST() 52 for (size_t k = 0; k < in.samples_per_channel_; ++k) { in TEST() 68 in.samples_per_channel_ = 480; in TEST() 71 for (size_t k = 0; k < in.samples_per_channel_; ++k) { in TEST() 80 EXPECT_THAT(out, AllOf(SizeIs(in.samples_per_channel_ * 6), Each(0))); in TEST() 87 in.samples_per_channel_ = 480; in TEST() 90 for (size_t k = 0; k < in.samples_per_channel_; ++k) { in TEST() 98 std::vector<int16_t> expected_output(in.samples_per_channel_ * 6); in TEST() [all …]
|
D | acm_remixing.cc | 19 RTC_DCHECK_EQ(output.size(), input.samples_per_channel_); in DownMixFrame() 22 std::fill(output.begin(), output.begin() + input.samples_per_channel_, 0); in DownMixFrame() 25 for (size_t n = 0; n < input.samples_per_channel_; ++n) { in DownMixFrame() 35 const size_t output_size = num_output_channels * input.samples_per_channel_; in ReMixFrame() 37 input.samples_per_channel_ > 0)); in ReMixFrame() 61 for (size_t k = 0; k < input.samples_per_channel_; ++k) { in ReMixFrame() 69 RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels); in ReMixFrame() 78 for (size_t k = 0; k < input.samples_per_channel_; ++k) { in ReMixFrame() 88 RTC_DCHECK_EQ(in_index, input.samples_per_channel_ * input.num_channels_); in ReMixFrame() 89 RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels); in ReMixFrame() [all …]
|
D | audio_coding_module.cc | 345 if (audio_frame.samples_per_channel_ == 0) { in Add10MsDataInternal() 359 audio_frame.samples_per_channel_) { in Add10MsDataInternal() 393 input_data->length_per_channel = ptr_frame->samples_per_channel_; in Add10MsDataInternal() 460 expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_); in PreprocessToAddData() 461 expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_); in PreprocessToAddData() 467 preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_; in PreprocessToAddData() 475 RTC_DCHECK_GE(audio.size(), preprocess_frame_.samples_per_channel_); in PreprocessToAddData() 476 RTC_DCHECK_GE(audio.size(), in_frame.samples_per_channel_); in PreprocessToAddData() 479 dest_ptr_audio, preprocess_frame_.samples_per_channel_)); in PreprocessToAddData() 505 preprocess_frame_.samples_per_channel_ = in PreprocessToAddData() [all …]
|
D | acm_receiver.cc | 190 audio_frame->samples_per_channel_ = in GetAudio() 195 rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100)); in GetAudio() 204 sizeof(int16_t) * audio_frame->samples_per_channel_ * in GetAudio()
|
/external/webrtc/audio/utility/ |
D | audio_frame_operations_unittest.cc | 23 frame_.samples_per_channel_ = 320; in AudioFrameOperationsTest() 38 for (size_t i = 0; i < frame->samples_per_channel_ * 4; i += 4) { in SetFrameData() 48 for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) { in SetFrameData() 56 for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_; in SetFrameData() 64 EXPECT_EQ(frame1.samples_per_channel_, frame2.samples_per_channel_); in VerifyFramesAreEqual() 67 for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_; in VerifyFramesAreEqual() 83 frame->samples_per_channel_ = samples_per_channel; in InitFrame() 94 RTC_DCHECK_LT(index, frame.samples_per_channel_); in GetChannelData() 102 for (size_t i = 0; i < frame.samples_per_channel_; ++i) { in VerifyFrameDataBounds() 112 frame_.samples_per_channel_ = AudioFrame::kMaxDataSizeSamples; in TEST_F() [all …]
|
D | audio_frame_operations.cc | 40 if (result_frame->samples_per_channel_ != frame_to_add.samples_per_channel_) { in Add() 42 RTC_DCHECK_EQ(result_frame->samples_per_channel_, 0); in Add() 43 result_frame->samples_per_channel_ = frame_to_add.samples_per_channel_; in Add() 62 frame_to_add.samples_per_channel_ * frame_to_add.num_channels_; in Add() 108 RTC_DCHECK_LE(frame->samples_per_channel_ * 4, in QuadToStereo() 112 QuadToStereo(frame->data(), frame->samples_per_channel_, in QuadToStereo() 140 RTC_DCHECK_LE(frame->samples_per_channel_ * frame->num_channels_, in DownmixChannels() 144 DownmixInterleavedToMono(frame->data(), frame->samples_per_channel_, in DownmixChannels() 160 RTC_DCHECK_LE(frame->samples_per_channel_ * target_number_of_channels, in UpmixChannels() 164 frame->samples_per_channel_ * target_number_of_channels > in UpmixChannels() [all …]
|
D | channel_mixer_unittest.cc | 34 frame_.samples_per_channel_ = kSamplesPerChannel; in ChannelMixerTest() 223 mono_frame.samples_per_channel_ = frame_.samples_per_channel(); in TEST_F() 273 mono_frame.samples_per_channel_ = frame_.samples_per_channel(); in TEST_F() 307 seven_one_frame.samples_per_channel_ = frame_.samples_per_channel(); in TEST_F() 336 stereo_frame.samples_per_channel_ = frame_.samples_per_channel(); in TEST_F() 364 stereo_frame.samples_per_channel_ = frame_.samples_per_channel(); in TEST_F() 388 five_one_frame.samples_per_channel_ = frame_.samples_per_channel(); in TEST_F()
|
/external/webrtc/api/audio/ |
D | audio_frame.cc | 32 swap(a.samples_per_channel_, b.samples_per_channel_); in swap() 40 const size_t length_a = a.samples_per_channel_ * a.num_channels_; in swap() 41 const size_t length_b = b.samples_per_channel_ * b.num_channels_; in swap() 60 samples_per_channel_ = 0; in ResetWithoutMuting() 79 samples_per_channel_ = samples_per_channel; in UpdateFrame() 108 samples_per_channel_ = src.samples_per_channel_; in CopyFrom() 116 const size_t length = samples_per_channel_ * num_channels_; in CopyFrom()
|
D | audio_frame.h | 106 size_t samples_per_channel() const { return samples_per_channel_; } in samples_per_channel() 128 size_t samples_per_channel_ = 0; variable
|
/external/webrtc/modules/audio_coding/test/ |
D | PCMFile.cc | 146 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 161 audio_frame.samples_per_channel_, pcm_file_) != in Write10MsData() 162 static_cast<size_t>(audio_frame.samples_per_channel_)) { in Write10MsData() 167 int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_]; in Write10MsData() 168 for (size_t k = 0; k < audio_frame.samples_per_channel_; k++) { in Write10MsData() 173 2 * audio_frame.samples_per_channel_, pcm_file_) != in Write10MsData() 174 static_cast<size_t>(2 * audio_frame.samples_per_channel_)) { in Write10MsData() 181 audio_frame.num_channels_ * audio_frame.samples_per_channel_, in Write10MsData() 184 audio_frame.samples_per_channel_)) { in Write10MsData()
|
/external/webrtc/audio/ |
D | remix_resample_unittest.cc | 29 src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100; in UtilityTest() 53 frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100); in SetMonoFrame() 55 for (size_t i = 0; i < frame->samples_per_channel_; i++) { in SetMonoFrame() 74 frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100); in SetStereoFrame() 76 for (size_t i = 0; i < frame->samples_per_channel_; i++) { in SetStereoFrame() 98 frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100); in SetQuadFrame() 100 for (size_t i = 0; i < frame->samples_per_channel_; i++) { in SetQuadFrame() 110 EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_); in VerifyParams() 129 i < ref_frame.samples_per_channel_ * ref_frame.num_channels_ - delay; in ComputeSNR() 153 i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) { in VerifyFramesAreEqual()
|
D | remix_resample.cc | 24 RemixAndResample(src_frame.data(), src_frame.samples_per_channel_, in RemixAndResample() 77 dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels; in RemixAndResample()
|
/external/webrtc/modules/audio_mixer/ |
D | audio_mixer_test.cc | 52 samples_per_channel_(sample_rate_hz_ / 100), in FilePlayingSource() 57 frame->samples_per_channel_ = samples_per_channel_; in GetAudioFrameWithInfo() 63 const size_t num_to_read = number_of_channels_ * samples_per_channel_; in GetAudioFrameWithInfo() 90 int samples_per_channel_; member in webrtc::test::FilePlayingSource 168 RTC_CHECK_EQ(sample_rate / 100, frame.samples_per_channel_); in main() 172 num_channels * frame.samples_per_channel_); in main()
|
D | audio_frame_manipulator.cc | 27 position < audio_frame.samples_per_channel_ * audio_frame.num_channels_; in AudioMixerCalculateEnergy() 43 size_t samples = audio_frame->samples_per_channel_; in Ramp()
|
D | audio_mixer_impl_unittest.cc | 47 frame->samples_per_channel_ = kDefaultSampleRateHz / 100; in ResetFrame() 96 audio_frame->samples_per_channel_ = in FakeAudioFrameWithInfo() 156 audio_source->fake_frame()->samples_per_channel_ = native_sample_rate / 100; in MixMonoAtGivenNativeRate() 210 const size_t n_samples = participant.fake_frame()->samples_per_channel_; in TEST() 281 participant.fake_frame()->samples_per_channel_ = expected_mix_frequency / 100; in TEST() 624 frame->samples_per_channel_ = kSamplesPerChannel; in TEST() 633 other_frame->samples_per_channel_ = kSamplesPerChannel; in TEST()
|
/external/webrtc/modules/audio_coding/neteq/test/ |
D | neteq_decoding_test.cc | 122 ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) || in Process() 123 (out_frame_.samples_per_channel_ == kBlockSize16kHz) || in Process() 124 (out_frame_.samples_per_channel_ == kBlockSize32kHz) || in Process() 125 (out_frame_.samples_per_channel_ == kBlockSize48kHz)); in Process() 158 output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_)); in DecodeAndCompare() 273 ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_); in WrapTest() 317 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); in LongCngWithClockDrift() 345 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); in LongCngWithClockDrift() 358 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); in LongCngWithClockDrift() 370 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); in LongCngWithClockDrift() [all …]
|
/external/webrtc/modules/audio_coding/neteq/ |
D | neteq_unittest.cc | 183 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); in TEST_F() 392 ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_); in CheckBgn() 410 ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_); in CheckBgn() 427 ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_); in CheckBgn() 433 k < output.num_channels_ * output.samples_per_channel_; ++k) in CheckBgn() 500 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); in TEST_F() 516 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); in TEST_F() 521 out_frame_.timestamp_ + out_frame_.samples_per_channel_); in TEST_F() 532 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); in TEST_F() 537 out_frame_.timestamp_ + out_frame_.samples_per_channel_); in TEST_F() [all …]
|
D | neteq_impl_unittest.cc | 206 ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_); in TestDtmfPacket() 496 ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_); in TEST_F() 521 output.data()[output.samples_per_channel_ - 1]), in TEST_F() 534 kPayloadLengthSamples - output.data()[output.samples_per_channel_ - 1], in TEST_F() 588 ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_); in TEST_F() 630 ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_); in TEST_F() 679 ASSERT_LE(output.samples_per_channel_, kMaxOutputSize); in TEST_F() 680 EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_); in TEST_F() 700 ASSERT_LE(output.samples_per_channel_, kMaxOutputSize); in TEST_F() 701 EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_); in TEST_F() [all …]
|
/external/webrtc/modules/audio_processing/agc2/ |
D | gain_applier.cc | 77 if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) { in ApplyGain() 98 samples_per_channel_ = static_cast<int>(samples_per_channel); in Initialize() 99 inverse_samples_per_channel_ = 1.f / samples_per_channel_; in Initialize()
|
D | gain_applier.h | 39 int samples_per_channel_ = -1; variable
|
/external/webrtc/api/audio/test/ |
D | audio_frame_unittest.cc | 118 EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_); in TEST() 161 ASSERT_EQ(kSamplesPerChannel + 1, frame1.samples_per_channel_); in TEST() 174 ASSERT_EQ(kSamplesPerChannel, frame2.samples_per_channel_); in TEST()
|
/external/webrtc/audio/voip/test/ |
D | audio_ingress_unittest.cc | 87 frame->samples_per_channel_ = kPcmuFormat.clockrate_hz / 100; // 10 ms. in GetAudioFrame() 89 frame->timestamp_ = frame->samples_per_channel_ * order; in GetAudioFrame() 130 EXPECT_EQ(audio_frame.samples_per_channel_, in TEST_F()
|
D | audio_channel_unittest.cc | 73 frame->samples_per_channel_ = kPcmuFormat.clockrate_hz / 100; // 10 ms. in GetAudioFrame() 75 frame->timestamp_ = frame->samples_per_channel_ * order; in GetAudioFrame()
|
/external/webrtc/audio/voip/ |
D | audio_egress.cc | 68 RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0); in SendAudioData() 95 rtc::dchecked_cast<uint32_t>(audio_frame->samples_per_channel_); in SendAudioData()
|
/external/webrtc/modules/audio_coding/neteq/tools/ |
D | rtp_encode.cc | 339 audio_frame.samples_per_channel_ = in RunRtpEncode() 344 while (input_file.Read(audio_frame.samples_per_channel_, in RunRtpEncode() 347 audio_frame.timestamp_ += audio_frame.samples_per_channel_; in RunRtpEncode()
|