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Searched refs:num_channels_ (Results 1 – 25 of 72) sorted by relevance

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/external/webrtc/webrtc/modules/audio_coding/neteq/
Daudio_multi_vector.cc27 num_channels_ = N; in AudioMultiVector()
36 num_channels_ = N; in AudioMultiVector()
48 for (size_t i = 0; i < num_channels_; ++i) { in Clear()
54 for (size_t i = 0; i < num_channels_; ++i) { in Zeros()
62 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo()
70 assert(length % num_channels_ == 0); in PushBackInterleaved()
71 if (num_channels_ == 1) { in PushBackInterleaved()
76 size_t length_per_channel = length / num_channels_; in PushBackInterleaved()
78 for (size_t channel = 0; channel < num_channels_; ++channel) { in PushBackInterleaved()
84 source_ptr += num_channels_; // Jump to next element of this channel. in PushBackInterleaved()
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Daudio_multi_vector_unittest.cc34 : num_channels_(GetParam()), // Get the test parameter. in AudioMultiVectorTest()
35 interleaved_length_(num_channels_ * array_length()) { in AudioMultiVectorTest()
36 array_interleaved_ = new int16_t[num_channels_ * array_length()]; in AudioMultiVectorTest()
53 for (size_t j = 1; j <= num_channels_; ++j) { in SetUp()
64 const size_t num_channels_; member in webrtc::AudioMultiVectorTest
73 AudioMultiVector vec1(num_channels_); in TEST_P()
75 EXPECT_EQ(num_channels_, vec1.Channels()); in TEST_P()
79 AudioMultiVector vec2(num_channels_, initial_size); in TEST_P()
81 EXPECT_EQ(num_channels_, vec2.Channels()); in TEST_P()
87 AudioMultiVector vec(num_channels_, array_length()); in TEST_P()
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Dpreemptive_expand.cc29 if (num_channels_ == 0 || in Process()
30 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ || in Process()
31 old_data_length >= input_length / num_channels_ - overlap_samples_) { in Process()
80 input, (unmodified_length + peak_index) * num_channels_); in CheckCriteriaAndStretch()
82 AudioMultiVector temp_vector(num_channels_); in CheckCriteriaAndStretch()
84 &input[(unmodified_length - peak_index) * num_channels_], in CheckCriteriaAndStretch()
85 peak_index * num_channels_); in CheckCriteriaAndStretch()
90 &input[unmodified_length * num_channels_], in CheckCriteriaAndStretch()
91 input_length - unmodified_length * num_channels_); in CheckCriteriaAndStretch()
Dbackground_noise.cc28 : num_channels_(num_channels), in BackgroundNoise()
29 channel_parameters_(new ChannelParameters[num_channels_]), in BackgroundNoise()
38 for (size_t channel = 0; channel < num_channels_; ++channel) { in Reset()
57 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) { in Update()
129 assert(channel < num_channels_); in Energy()
134 assert(channel < num_channels_); in SetMuteFactor()
139 assert(channel < num_channels_); in MuteFactor()
144 assert(channel < num_channels_); in Filter()
149 assert(channel < num_channels_); in FilterState()
155 assert(channel < num_channels_); in SetFilterState()
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Daccelerate.cc24 if (num_channels_ == 0 || in Process()
25 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) { in Process()
70 output->PushBackInterleaved(input, fs_mult_120 * num_channels_); in CheckCriteriaAndStretch()
72 AudioMultiVector temp_vector(num_channels_); in CheckCriteriaAndStretch()
73 temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_], in CheckCriteriaAndStretch()
74 peak_index * num_channels_); in CheckCriteriaAndStretch()
79 &input[(fs_mult_120 + peak_index) * num_channels_], in CheckCriteriaAndStretch()
80 input_length - (fs_mult_120 + peak_index) * num_channels_); in CheckCriteriaAndStretch()
Dneteq_stereo_unittest.cc50 : num_channels_(GetParam().num_channels), in NetEqStereoTest()
69 input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_]; in NetEqStereoTest()
71 num_channels_]; in NetEqStereoTest()
72 output_multi_channel_ = new int16_t[kMaxBlockSize * num_channels_]; in NetEqStereoTest()
94 if (num_channels_ == 2) { in SetUp()
96 } else if (num_channels_ == 5) { in SetUp()
104 if (num_channels_ == 2) { in SetUp()
112 if (num_channels_ == 2) { in SetUp()
120 if (num_channels_ == 2) { in SetUp()
151 num_channels_, in GetNewPackets()
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Dexpand_unittest.cc74 num_channels_(1), in ExpandTest()
75 background_noise_(num_channels_), in ExpandTest()
76 sync_buffer_(num_channels_, in ExpandTest()
83 num_channels_) { in ExpandTest()
98 ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels."; in SetUp()
103 size_t num_channels_; member in webrtc::ExpandTest
116 AudioMultiVector output(num_channels_); in TEST_F()
136 AudioMultiVector output(num_channels_); in TEST_F()
153 AudioMultiVector output(num_channels_); in TEST_F()
Dtime_stretch.h42 num_channels_(num_channels), in TimeStretch()
50 assert(num_channels_ > 0); in TimeStretch()
51 assert(master_channel_ < num_channels_); in TimeStretch()
94 const size_t num_channels_; variable
Dmerge.cc32 num_channels_(num_channels), in Merge()
37 expanded_(num_channels_) { in Merge()
38 assert(num_channels_ > 0); in Merge()
55 AudioMultiVector input_vector(num_channels_); in Process()
58 assert(input_length_per_channel == input_length / num_channels_); in Process()
63 for (size_t channel = 0; channel < num_channels_; ++channel) { in Process()
182 AudioMultiVector expanded_temp(num_channels_); in GetExpandedSignal()
379 return fs_hz_ / 100 * num_channels_; // 10 ms. in RequiredFutureSamples()
/external/webrtc/webrtc/modules/utility/source/
Daudio_frame_operations_unittest.cc24 frame_.num_channels_ = 2; in AudioFrameOperationsTest()
44 EXPECT_EQ(frame1.num_channels_, frame2.num_channels_); in VerifyFramesAreEqual()
48 for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_; in VerifyFramesAreEqual()
58 frame_.num_channels_ = 1; in TEST_F()
63 frame_.num_channels_ = 1; in TEST_F()
71 stereo_frame.num_channels_ = 2; in TEST_F()
79 frame_.num_channels_ = 2; // Need to set manually. in TEST_F()
84 frame_.num_channels_ = 1; in TEST_F()
96 mono_frame.num_channels_ = 1; in TEST_F()
104 frame_.num_channels_ = 1; // Need to set manually. in TEST_F()
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Daudio_frame_operations.cc26 if (frame->num_channels_ != 1) { in MonoToStereo()
38 frame->num_channels_ = 2; in MonoToStereo()
52 if (frame->num_channels_ != 2) { in StereoToMono()
57 frame->num_channels_ = 1; in StereoToMono()
63 if (frame->num_channels_ != 2) return; in SwapStereoChannels()
74 frame.samples_per_channel_ * frame.num_channels_); in Mute()
78 if (frame.num_channels_ != 2) { in Scale()
95 for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_; in ScaleWithSat()
Dfile_recorder_impl.cc144 if( incomingAudioFrame.num_channels_ == 2 && in RecordAudioToFile()
148 tempAudioFrame.num_channels_ = 1; in RecordAudioToFile()
162 else if( incomingAudioFrame.num_channels_ == 1 && in RecordAudioToFile()
166 tempAudioFrame.num_channels_ = 2; in RecordAudioToFile()
209 ptrAudioFrame->num_channels_); in RecordAudioToFile()
212 ptrAudioFrame->num_channels_, in RecordAudioToFile()
/external/webrtc/webrtc/common_audio/resampler/
Dpush_resampler.cc25 num_channels_(0) { in PushResampler()
38 num_channels == num_channels_) in InitializeIfNeeded()
48 num_channels_ = num_channels; in InitializeIfNeeded()
56 if (num_channels_ == 2) { in InitializeIfNeeded()
71 const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100; in Resample()
72 const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100; in Resample()
82 if (num_channels_ == 2) { in Resample()
83 const size_t src_length_mono = src_length / num_channels_; in Resample()
84 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample()
86 Deinterleave(src, src_length_mono, num_channels_, deinterleaved); in Resample()
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/external/webrtc/webrtc/modules/audio_coding/codecs/g722/
Daudio_encoder_g722.cc40 : num_channels_(config.num_channels), in AudioEncoderG722()
46 encoders_(new EncoderState[num_channels_]), in AudioEncoderG722()
47 interleave_buffer_(2 * num_channels_) { in AudioEncoderG722()
51 for (size_t i = 0; i < num_channels_; ++i) { in AudioEncoderG722()
64 return SamplesPerChannel() / 2 * num_channels_; in MaxEncodedBytes()
72 return num_channels_; in NumChannels()
107 for (size_t j = 0; j < num_channels_; ++j) in EncodeInternal()
108 encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j]; in EncodeInternal()
119 for (size_t i = 0; i < num_channels_; ++i) { in EncodeInternal()
130 for (size_t j = 0; j < num_channels_; ++j) { in EncodeInternal()
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/external/webrtc/webrtc/modules/include/
Dmodule_common_types.h535 size_t num_channels_; variable
563 num_channels_ = 0; in Reset()
585 num_channels_ = num_channels; in UpdateFrame()
608 num_channels_ = src.num_channels_; in CopyFrom()
612 const size_t length = samples_per_channel_ * num_channels_; in CopyFrom()
618 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t)); in Mute()
622 assert((num_channels_ > 0) && (num_channels_ < 3));
623 if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
625 for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
633 assert((num_channels_ > 0) && (num_channels_ < 3)); in Append()
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/external/webrtc/webrtc/modules/audio_processing/
Daudio_buffer.cc56 num_channels_(num_process_channels), in AudioBuffer()
153 assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1); in CopyTo()
161 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo()
169 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo()
178 for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) { in CopyTo()
188 num_channels_ = num_proc_channels_; in InitForNewData()
317 num_split_frames_, num_channels_, in mixed_low_pass_data()
345 return num_channels_; in num_channels()
349 num_channels_ = num_channels; in set_num_channels()
371 assert(frame->num_channels_ == num_input_channels_); in DeinterleaveFrom()
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Daudio_processing_impl_unittest.cc46 frame.num_channels_ = 1; in TEST()
60 frame.num_channels_ = 2; in TEST()
65 frame.num_channels_ = 2; in TEST()
/external/webrtc/webrtc/common_audio/
Dchannel_buffer.h50 num_channels_(num_channels), in data_()
52 for (size_t i = 0; i < num_channels_; ++i) { in data_()
54 channels_[j * num_channels_ + i] = in data_()
56 bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i]; in data_()
79 return &channels_[band * num_channels_]; in channels()
94 RTC_DCHECK_LT(channel, num_channels_); in bands()
107 for (size_t i = 0; i < num_channels_; ++i) in Slice()
118 size_t num_channels() const { return num_channels_; } in num_channels()
120 size_t size() const {return num_frames_ * num_channels_; } in size()
133 const size_t num_channels_; variable
Dwav_file.h54 size_t num_channels() const override { return num_channels_; } in num_channels()
60 const size_t num_channels_; variable
82 size_t num_channels() const override { return num_channels_; } in num_channels()
88 size_t num_channels_; variable
Dwav_file.cc56 RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format, in WavReader()
104 num_channels_(num_channels), in WavWriter()
108 RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat, in WavWriter()
145 WriteWavHeader(header, num_channels_, sample_rate_, kWavFormat, in Close()
/external/webrtc/webrtc/modules/audio_processing/transient/
Dtransient_suppressor.cc53 num_channels_(0), in TransientSuppressor()
111 num_channels_ = num_channels; in Initialize()
112 in_buffer_.reset(new float[analysis_length_ * num_channels_]); in Initialize()
115 analysis_length_ * num_channels_ * sizeof(in_buffer_[0])); in Initialize()
121 out_buffer_.reset(new float[analysis_length_ * num_channels_]); in Initialize()
124 analysis_length_ * num_channels_ * sizeof(out_buffer_[0])); in Initialize()
131 spectral_mean_.reset(new float[complex_analysis_length_ * num_channels_]); in Initialize()
134 complex_analysis_length_ * num_channels_ * sizeof(spectral_mean_[0])); in Initialize()
174 if (!data || data_length != data_length_ || num_channels != num_channels_ || in Suppress()
210 for (int i = 0; i < num_channels_; ++i) { in Suppress()
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/external/webrtc/webrtc/modules/audio_coding/acm2/
Dacm_send_test_oldapi.cc43 input_frame_.num_channels_ = 1; in AcmSendTestOldApi()
45 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in AcmSendTestOldApi()
61 input_frame_.num_channels_ = channels; in RegisterCodec()
62 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in RegisterCodec()
70 input_frame_.num_channels_ = external_speech_encoder->NumChannels(); in RegisterExternalCodec()
71 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in RegisterExternalCodec()
89 if (input_frame_.num_channels_ > 1) { in NextPacket()
92 input_frame_.num_channels_, in NextPacket()
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/
Daudio_decoder_pcm.h21 explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) { in AudioDecoderPcmU()
36 const size_t num_channels_;
42 explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) { in AudioDecoderPcmA()
57 const size_t num_channels_;
/external/webrtc/webrtc/voice_engine/
Dutility_unittest.cc29 src_frame_.num_channels_ = 1; in UtilityTest()
50 frame->num_channels_ = 1; in SetMonoFrame()
68 frame->num_channels_ = 2; in SetStereoFrame()
83 EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_); in VerifyParams()
100 ref_frame.num_channels_ - delay; i++) { in ComputeSNR()
121 i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) { in VerifyFramesAreEqual()
Dutility.cc28 src_frame.num_channels_, src_frame.sample_rate_hz_, in RemixAndResample()
46 if (num_channels == 2 && dst_frame->num_channels_ == 1) { in RemixAndResample()
74 if (num_channels == 1 && dst_frame->num_channels_ == 2) { in RemixAndResample()
77 dst_frame->num_channels_ = 1; in RemixAndResample()

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