/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
D | audio_multi_vector.cc | 27 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() [all …]
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D | audio_multi_vector_unittest.cc | 34 : 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() [all …]
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D | preemptive_expand.cc | 29 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() 76 input, (unmodified_length + peak_index) * num_channels_); in CheckCriteriaAndStretch() 78 AudioMultiVector temp_vector(num_channels_); in CheckCriteriaAndStretch() 80 &input[(unmodified_length - peak_index) * num_channels_], in CheckCriteriaAndStretch() 81 peak_index * num_channels_); in CheckCriteriaAndStretch() 86 &input[unmodified_length * num_channels_], in CheckCriteriaAndStretch() 87 input_length - unmodified_length * num_channels_); in CheckCriteriaAndStretch()
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D | accelerate.cc | 24 if (num_channels_ == 0 || static_cast<int>(input_length) / num_channels_ < in Process() 57 output->PushBackInterleaved(input, fs_mult_120 * num_channels_); in CheckCriteriaAndStretch() 59 AudioMultiVector temp_vector(num_channels_); in CheckCriteriaAndStretch() 60 temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_], in CheckCriteriaAndStretch() 61 peak_index * num_channels_); in CheckCriteriaAndStretch() 66 &input[(fs_mult_120 + peak_index) * num_channels_], in CheckCriteriaAndStretch() 67 input_length - (fs_mult_120 + peak_index) * num_channels_); in CheckCriteriaAndStretch()
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D | background_noise.cc | 25 : num_channels_(num_channels), in BackgroundNoise() 26 channel_parameters_(new ChannelParameters[num_channels_]), in BackgroundNoise() 35 for (size_t channel = 0; channel < num_channels_; ++channel) { in Reset() 54 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) { in Update() 126 assert(channel < num_channels_); in Energy() 131 assert(channel < num_channels_); in SetMuteFactor() 136 assert(channel < num_channels_); in MuteFactor() 141 assert(channel < num_channels_); in Filter() 146 assert(channel < num_channels_); in FilterState() 152 assert(channel < num_channels_); in SetFilterState() [all …]
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D | neteq_stereo_unittest.cc | 51 : 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() 152 num_channels_, in GetNewPackets() [all …]
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D | expand.h | 43 num_channels_(num_channels), in Expand() 50 channel_parameters_(new ChannelParameters[num_channels_]) { in Expand() 53 assert(num_channels_ > 0); in Expand() 77 assert(channel < num_channels_); in SetMuteFactor() 83 assert(channel < num_channels_); in MuteFactor() 117 const size_t num_channels_; variable
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D | merge.h | 38 num_channels_(num_channels), in Merge() 43 expanded_(num_channels_) { in Merge() 44 assert(num_channels_ > 0); in Merge() 64 const size_t num_channels_; variable
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D | time_stretch.h | 42 num_channels_(static_cast<int>(num_channels)), in TimeStretch() 50 assert(num_channels_ > 0); in TimeStretch() 51 assert(static_cast<int>(master_channel_) < num_channels_); in TimeStretch() 89 const int num_channels_; variable
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
D | audio_frame_operations_unittest.cc | 24 frame_.num_channels_ = 2; in AudioFrameOperationsTest() 44 EXPECT_EQ(frame1.num_channels_, frame2.num_channels_); in VerifyFramesAreEqual() 48 for (int 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() [all …]
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D | audio_frame_operations.cc | 26 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 (int i = 0; i < frame.samples_per_channel_ * frame.num_channels_; in ScaleWithSat()
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/external/webrtc/src/modules/audio_processing/ |
D | audio_buffer.cc | 67 num_channels_(0), in AudioBuffer() 98 assert(channel >= 0 && channel < num_channels_); in data() 107 assert(channel >= 0 && channel < num_channels_); in low_pass_split_data() 116 assert(channel >= 0 && channel < num_channels_); in high_pass_split_data() 137 assert(channel >= 0 && channel < num_channels_); in low_pass_reference() 146 assert(channel >= 0 && channel < num_channels_); in analysis_filter_state1() 151 assert(channel >= 0 && channel < num_channels_); in analysis_filter_state2() 156 assert(channel >= 0 && channel < num_channels_); in synthesis_filter_state1() 161 assert(channel >= 0 && channel < num_channels_); in synthesis_filter_state2() 178 return num_channels_; in num_channels() [all …]
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/external/chromium_org/third_party/webrtc/common_audio/resampler/ |
D | push_resampler.cc | 25 num_channels_(0) { in PushResampler() 38 num_channels == num_channels_) in InitializeIfNeeded() 48 num_channels_ = num_channels; in InitializeIfNeeded() 54 if (num_channels_ == 2) { in InitializeIfNeeded() 69 const int src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100; in Resample() 70 const int dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100; in Resample() 80 if (num_channels_ == 2) { in Resample() 81 const int src_length_mono = src_length / num_channels_; in Resample() 82 const int dst_capacity_mono = dst_capacity / num_channels_; in Resample() 84 Deinterleave(src, src_length_mono, num_channels_, deinterleaved); in Resample() [all …]
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/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
D | common.h | 45 num_channels_(num_channels) { in ChannelBuffer() 53 num_channels_(num_channels) { in ChannelBuffer() 63 num_channels_(num_channels) { in ChannelBuffer() 65 for (int i = 0; i < num_channels_; ++i) in ChannelBuffer() 72 DCHECK_LT(i, num_channels_); in CopyFrom() 81 DCHECK_LT(i, num_channels_); in channel() 93 int num_channels() const { return num_channels_; } in num_channels() 94 int length() const { return samples_per_channel_ * num_channels_; } in length() 99 for (int i = 0; i < num_channels_; ++i) in Initialize() 106 const int num_channels_; variable
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/external/chromium_org/third_party/webrtc/modules/interface/ |
D | module_common_types.h | 694 int num_channels_; variable 722 num_channels_ = 0; in Reset() 741 num_channels_ = num_channels; in UpdateFrame() 764 num_channels_ = src.num_channels_; in CopyFrom() 768 const int length = samples_per_channel_ * num_channels_; in CopyFrom() 774 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t)); in Mute() 778 assert((num_channels_ > 0) && (num_channels_ < 3)); 779 if ((num_channels_ > 2) || (num_channels_ < 1)) return *this; 781 for (int i = 0; i < samples_per_channel_ * num_channels_; i++) { 789 assert((num_channels_ > 0) && (num_channels_ < 3)); in Append() [all …]
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/external/chromium_org/media/cast/receiver/ |
D | audio_decoder.cc | 31 num_channels_(num_channels), in ImplBase() 34 if (num_channels_ <= 0 || sampling_rate <= 0 || sampling_rate % 100 != 0) in ImplBase() 81 const int num_channels_; member in media::cast::AudioDecoder::ImplBase 138 audio_bus = AudioBus::Create(num_channels_, num_samples_decoded).Pass(); in Decode() 140 for (int ch = 0; ch < num_channels_; ++ch) { in Decode() 142 const float* const src_end = src + num_samples_decoded * num_channels_; in Decode() 144 for (; src < src_end; src += num_channels_, ++dest) in Decode() 182 const int num_samples = len / sizeof(int16) / num_channels_; in Decode() 189 const int num_elements = num_samples * num_channels_; in Decode() 193 audio_bus = AudioBus::Create(num_channels_, num_samples).Pass(); in Decode()
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/external/chromium_org/third_party/webrtc/voice_engine/ |
D | utility.cc | 31 int audio_ptr_num_channels = src_frame.num_channels_; in RemixAndResample() 35 if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) { in RemixAndResample() 62 if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) { in RemixAndResample() 65 dst_frame->num_channels_ = 1; in RemixAndResample() 121 dst_af->num_channels_ = num_channels; in DownConvertToCodecFormat()
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D | utility_unittest.cc | 33 src_frame_.num_channels_ = 1; in UtilityTest() 53 frame->num_channels_ = 1; in SetMonoFrame() 71 frame->num_channels_ = 2; in SetStereoFrame() 86 EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_); in VerifyParams() 103 ref_frame.num_channels_ - delay; i++) { in ComputeSNR() 123 for (int i = 0; i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; in VerifyFramesAreEqual() 175 src_frame_.num_channels_, in RunResampleTest() 177 dst_frame_.num_channels_, in RunResampleTest()
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
D | acm_send_test_oldapi.cc | 41 input_frame_.num_channels_ = 1; in AcmSendTestOldApi() 43 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in AcmSendTestOldApi() 59 input_frame_.num_channels_ = channels; in RegisterCodec() 60 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in RegisterCodec() 77 if (input_frame_.num_channels_ > 1) { in NextPacket() 80 input_frame_.num_channels_, in NextPacket()
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D | acm_send_test.cc | 44 input_frame_.num_channels_ = 1; in AcmSendTest() 46 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in AcmSendTest() 56 input_frame_.num_channels_ = channels; in RegisterCodec() 57 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in RegisterCodec() 74 if (input_frame_.num_channels_ > 1) { in NextPacket() 77 input_frame_.num_channels_, in NextPacket()
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D | acm_g7221.cc | 182 if (num_channels_ == 2) { 195 if (num_channels_ == 2) { 205 if (num_channels_ == 2) { 215 if (num_channels_ == 2) { 233 in_audio_ix_read_ += 320 * num_channels_;
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D | acm_g7221c.cc | 186 if (num_channels_ == 2) { 199 if (num_channels_ == 2) { 209 if (num_channels_ == 2) { 219 if (num_channels_ == 2) { 238 in_audio_ix_read_ += 640 * num_channels_;
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/external/chromium_org/media/cast/sender/ |
D | audio_encoder.cc | 53 num_channels_(num_channels), in ImplBase() 63 if (num_channels_ <= 0 || samples_per_frame_ <= 0 || in ImplBase() 65 samples_per_frame_ * num_channels_ > kMaxSamplesTimesChannelsPerFrame) { in ImplBase() 117 DCHECK_EQ(audio_bus->channels(), num_channels_); in EncodeAudio() 164 const int num_channels_; member in media::cast::AudioEncoder::ImplBase 250 float* dest = buffer_.get() + buffer_fill_offset * num_channels_ + ch; in TransferSamplesIntoBuffer() 251 for (; src < src_end; ++src, dest += num_channels_) in TransferSamplesIntoBuffer() 320 buffer_.get() + buffer_fill_offset * num_channels_); in TransferSamplesIntoBuffer() 325 out->resize(num_channels_ * samples_per_frame_ * sizeof(int16)); in EncodeFromFilledBuffer() 327 const int16* const src_end = src + num_channels_ * samples_per_frame_; in EncodeFromFilledBuffer()
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/external/chromium_org/media/base/android/ |
D | audio_decoder_job.cc | 42 num_channels_(0), in AudioDecoderJob() 58 num_channels_ = configs.audio_channels; in SetDemuxerConfigs() 62 bytes_per_frame_ = kBytesPerAudioOutputSample * num_channels_; in SetDemuxerConfigs() 111 num_channels_ != configs.audio_channels || in AreDemuxerConfigsChanged() 126 audio_codec_, sampling_rate_, num_channels_, &audio_extra_data_[0], in CreateMediaCodecBridgeInternal()
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/external/chromium_org/third_party/webrtc/common_audio/ |
D | wav_writer.cc | 29 num_channels_(num_channels), in WavFile() 33 CHECK(CheckWavParameters(num_channels_, in WavFile() 59 CHECK(CheckWavParameters(num_channels_, in WriteSamples() 79 WriteWavHeader(header, num_channels_, sample_rate_, kWavFormat, in Close()
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