Home
last modified time | relevance | path

Searched refs:samples_per_channel (Results 1 – 25 of 60) sorted by relevance

123

/external/webrtc/modules/audio_processing/agc2/
Dlimiter.cc49 int samples_per_channel, in ComputePerSampleSubframeFactors() argument
53 rtc::CheckedDivExact(samples_per_channel, num_subframes); in ComputePerSampleSubframeFactors()
78 const int samples_per_channel = signal.samples_per_channel(); in ScaleSamples() local
79 RTC_DCHECK_EQ(samples_per_channel, per_sample_scaling_factors.size()); in ScaleSamples()
82 for (int j = 0; j < samples_per_channel; ++j) { in ScaleSamples()
119 const int samples_per_channel = signal.samples_per_channel(); in Process() local
120 RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel); in Process()
123 &per_sample_scaling_factors_[0], samples_per_channel); in Process()
124 ComputePerSampleSubframeFactors(scaling_factors_, samples_per_channel, in Process()
Dgain_applier.cc61 for (int i = 0; i < float_frame.samples_per_channel(); ++i) { in ApplyGainWithRamping()
77 if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) { in ApplyGain()
78 Initialize(signal.samples_per_channel()); in ApplyGain()
97 void GainApplier::Initialize(int samples_per_channel) { in Initialize() argument
98 RTC_DCHECK_GT(samples_per_channel, 0); in Initialize()
99 samples_per_channel_ = static_cast<int>(samples_per_channel); in Initialize()
Dadaptive_digital_gain_applier.cc107 RTC_DCHECK_GT(src.samples_per_channel(), 0); in CopyAudio()
111 RTC_DCHECK_EQ(channel_view.size(), src.samples_per_channel()); in CopyAudio()
112 RTC_DCHECK_EQ(dst[c].size(), src.samples_per_channel()); in CopyAudio()
171 frame.samples_per_channel() == 80 || frame.samples_per_channel() == 160 || in Process()
172 frame.samples_per_channel() == 320 || frame.samples_per_channel() == 480) in Process()
235 frame.samples_per_channel()); in Process()
Dclipping_predictor.cc114 const int samples_per_channel = frame.samples_per_channel(); in Analyze() local
115 RTC_DCHECK_GT(samples_per_channel, 0); in Analyze()
124 {sum_squares / static_cast<float>(samples_per_channel), peak}); in Analyze()
253 const int samples_per_channel = frame.samples_per_channel(); in Analyze() local
254 RTC_DCHECK_GT(samples_per_channel, 0); in Analyze()
263 {sum_squares / static_cast<float>(samples_per_channel), peak}); in Analyze()
Dvector_float_frame.cc28 int samples_per_channel, in VectorFloatFrame() argument
31 std::vector<float>(samples_per_channel, start_value)), in VectorFloatFrame()
35 samples_per_channel) {} in VectorFloatFrame()
Dgain_applier_unittest.cc61 constexpr int samples_per_channel = 4; in TEST() local
62 VectorFloatFrame fake_audio(num_channels, samples_per_channel, in TEST()
80 EXPECT_NEAR(max_signal_change, total_gain_change / samples_per_channel, in TEST()
85 VectorFloatFrame next_fake_audio_frame(num_channels, samples_per_channel, in TEST()
Dnoise_level_estimator_unittest.cc35 const int samples_per_channel = in RunEstimator() local
37 VectorFloatFrame signal(1, samples_per_channel, 0.0f); in RunEstimator()
40 for (int j = 0; j < samples_per_channel; ++j) { in RunEstimator()
/external/webrtc/modules/audio_mixer/
Dframe_combiner.cc48 const size_t samples_per_channel = static_cast<size_t>( in SetAudioFrameFields() local
56 0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined, in SetAudioFrameFields()
95 size_t samples_per_channel, in MixToFloatFrame() argument
98 RTC_DCHECK_LE(samples_per_channel, FrameCombiner::kMaximumChannelSize); in MixToFloatFrame()
110 for (size_t k = 0; k < std::min(samples_per_channel, in MixToFloatFrame()
120 const size_t sample_rate = mixing_buffer_view.samples_per_channel() * 1000 / in RunLimiter()
131 const size_t samples_per_channel = mixing_buffer_view.samples_per_channel(); in InterleaveToAudioFrame() local
135 for (size_t j = 0; j < samples_per_channel; ++j) { in InterleaveToAudioFrame()
170 const size_t samples_per_channel = static_cast<size_t>( in Combine() local
174 RTC_DCHECK_EQ(samples_per_channel, frame->samples_per_channel_); in Combine()
[all …]
Daudio_frame_manipulator_unittest.cc20 void FillFrameWithConstants(size_t samples_per_channel, in FillFrameWithConstants() argument
25 frame->samples_per_channel_ = samples_per_channel; in FillFrameWithConstants()
27 std::fill(frame_data, frame_data + samples_per_channel * number_of_channels, in FillFrameWithConstants()
/external/webrtc/audio/utility/
Dchannel_mixer_unittest.cc46 for (size_t i = 0; i < frame->samples_per_channel() * frame->num_channels(); in SetFrameData()
55 for (size_t i = 0; i < frame->samples_per_channel(); ++i) { in SetMonoData()
62 ASSERT_LE(2 * frame->samples_per_channel(), frame->max_16bit_samples()); in SetStereoData()
65 for (size_t i = 0; i < frame->samples_per_channel() * 2; i += 2) { in SetStereoData()
79 ASSERT_LE(6 * frame->samples_per_channel(), frame->max_16bit_samples()); in SetFiveOneData()
82 for (size_t i = 0; i < frame->samples_per_channel() * 6; i += 6) { in SetFiveOneData()
102 ASSERT_LE(8 * frame->samples_per_channel(), frame->max_16bit_samples()); in SetSevenOneData()
105 for (size_t i = 0; i < frame->samples_per_channel() * 8; i += 8) { in SetSevenOneData()
120 for (size_t i = 0; i < frame->samples_per_channel() * frame->num_channels(); in AllSamplesEquals()
131 EXPECT_EQ(frame1.samples_per_channel(), frame2.samples_per_channel()); in VerifyFramesAreEqual()
[all …]
Dchannel_mixer.cc46 RTC_CHECK_LE(frame->samples_per_channel() * output_channels_, in Transform()
63 const size_t num_elements = frame->samples_per_channel() * output_channels_; in Transform()
73 for (size_t i = 0; i < frame->samples_per_channel(); i++) { in Transform()
96 sizeof(int16_t) * frame->samples_per_channel() * frame->num_channels()); in Transform()
/external/webrtc/modules/audio_processing/test/
Dbitexactness_tools.cc57 void ReadFloatSamplesFromStereoFile(size_t samples_per_channel, in ReadFloatSamplesFromStereoFile() argument
62 RTC_DCHECK_EQ(data.size(), samples_per_channel * num_channels); in ReadFloatSamplesFromStereoFile()
63 std::vector<int16_t> read_samples(samples_per_channel * 2); in ReadFloatSamplesFromStereoFile()
64 stereo_pcm_file->Read(samples_per_channel * 2, read_samples.data()); in ReadFloatSamplesFromStereoFile()
67 for (size_t sample = 0; sample < samples_per_channel; ++sample) { in ReadFloatSamplesFromStereoFile()
76 size_t samples_per_channel, in VerifyDeinterleavedArray() argument
90 output.begin() + channel_no * samples_per_channel, in VerifyDeinterleavedArray()
91 output.begin() + channel_no * samples_per_channel + in VerifyDeinterleavedArray()
Daudio_processing_simulator.h38 samples_per_channel = in SetFormat()
42 data.resize(num_channels * samples_per_channel); in SetFormat()
48 RTC_CHECK_EQ(samples_per_channel, dest->num_frames()); in CopyTo()
50 std::vector<float> tmp(samples_per_channel * num_channels); in CopyTo()
52 Deinterleave(tmp.data(), samples_per_channel, num_channels, in CopyTo()
58 RTC_CHECK_EQ(src.num_frames(), samples_per_channel); in CopyFrom()
59 data.resize(num_channels * samples_per_channel); in CopyFrom()
62 for (int sample = 0; sample < samples_per_channel; ++sample) { in CopyFrom()
70 int samples_per_channel; member
Dtest_utils.h42 samples_per_channel = 0; in Int16FrameData()
47 samples_per_channel = src.samples_per_channel; in CopyFrom()
51 const size_t length = samples_per_channel * num_channels; in CopyFrom()
58 size_t samples_per_channel; member
127 cb->reset(new ChannelBuffer<T>(frame->samples_per_channel, num_channels)); in SetContainerFormat()
Dsimulator_buffers.cc60 int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); in CreateConfigAndBuffer() local
67 buffer_data_samples->resize(samples_per_channel * num_channels); in CreateConfigAndBuffer()
74 (*buffer_data)[ch] = &(*buffer_data_samples)[ch * samples_per_channel]; in CreateConfigAndBuffer()
/external/webrtc/modules/audio_coding/codecs/g722/
Daudio_encoder_g722.cc38 const size_t samples_per_channel = in AudioEncoderG722Impl() local
41 encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]); in AudioEncoderG722Impl()
42 encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2); in AudioEncoderG722Impl()
109 const size_t samples_per_channel = SamplesPerChannel(); in EncodeImpl() local
113 samples_per_channel, encoders_[i].encoded_buffer.data()); in EncodeImpl()
114 RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2); in EncodeImpl()
117 const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_; in EncodeImpl()
124 for (size_t i = 0; i < samples_per_channel / 2; ++i) { in EncodeImpl()
/external/webrtc/modules/audio_device/
Daudio_device_buffer.cc233 size_t samples_per_channel) { in SetRecordedBuffer() argument
234 return SetRecordedBuffer(audio_buffer, samples_per_channel, 0); in SetRecordedBuffer()
238 size_t samples_per_channel, in SetRecordedBuffer() argument
243 rec_channels_ * samples_per_channel); in SetRecordedBuffer()
277 UpdateRecStats(max_abs, samples_per_channel); in SetRecordedBuffer()
300 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) { in RequestPlayoutData() argument
302 "samples_per_channel", samples_per_channel); in RequestPlayoutData()
307 const size_t total_samples = play_channels_ * samples_per_channel; in RequestPlayoutData()
326 samples_per_channel, bytes_per_frame, play_channels_, play_sample_rate_, in RequestPlayoutData()
503 size_t samples_per_channel) { in UpdateRecStats() argument
[all …]
Daudio_device_buffer.h103 size_t samples_per_channel);
106 size_t samples_per_channel,
112 virtual int32_t RequestPlayoutData(size_t samples_per_channel);
132 void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
133 void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
/external/webrtc/api/call/
Daudio_sink.h26 size_t samples_per_channel, in Data()
31 samples_per_channel(samples_per_channel), in Data()
37 size_t samples_per_channel; // Number of frames in the buffer. member
/external/webrtc/modules/audio_processing/aec_dump/
Dcapture_stream_info.cc37 int samples_per_channel) { in AddInput() argument
39 const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels; in AddInput()
45 int samples_per_channel) { in AddOutput() argument
47 const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels; in AddOutput()
Dmock_aec_dump.h45 int samples_per_channel),
51 int samples_per_channel),
63 int samples_per_channel),
Daec_dump_impl.h56 int samples_per_channel) override; in RTC_POP_IGNORING_WUNDEF()
59 int samples_per_channel) override; in RTC_POP_IGNORING_WUNDEF()
65 int samples_per_channel) override; in RTC_POP_IGNORING_WUNDEF()
/external/webrtc/modules/audio_processing/
Decho_control_mobile_bit_exact_unittest.cc70 const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); in RunBitexactnessTest() local
77 std::vector<float> render_input(samples_per_channel * num_channels); in RunBitexactnessTest()
85 std::vector<float> capture_input(samples_per_channel * num_channels); in RunBitexactnessTest()
88 ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels, in RunBitexactnessTest()
90 ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels, in RunBitexactnessTest()
/external/webrtc/modules/audio_processing/include/
Daec_dump.h94 int samples_per_channel) = 0;
97 int samples_per_channel) = 0;
104 int samples_per_channel) = 0;
/external/webrtc/audio/
Dremix_resample.cc34 size_t samples_per_channel, in RemixAndResample() argument
51 src_data, num_channels, samples_per_channel, dst_frame->num_channels_, in RemixAndResample()
69 const size_t src_length = samples_per_channel * audio_ptr_num_channels; in RemixAndResample()

123