| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| audio_multi_vector.cc | 27 num_channels_ = N;
36 num_channels_ = N;
48 for (size_t i = 0; i < num_channels_; ++i) {
54 for (size_t i = 0; i < num_channels_; ++i) {
62 for (size_t i = 0; i < num_channels_; ++i) {
70 assert(length % num_channels_ == 0);
71 if (num_channels_ == 1) {
76 size_t length_per_channel = length / num_channels_;
78 for (size_t channel = 0; channel < num_channels_; ++channel) {
84 source_ptr += num_channels_; // Jump to next element of this channel
[all...] |
| audio_multi_vector_unittest.cc | 34 : num_channels_(GetParam()), // Get the test parameter.
35 interleaved_length_(num_channels_ * array_length()) {
36 array_interleaved_ = new int16_t[num_channels_ * array_length()];
53 for (size_t j = 1; j <= num_channels_; ++j) {
64 const size_t num_channels_; member in class:webrtc::AudioMultiVectorTest
73 AudioMultiVector vec1(num_channels_);
75 EXPECT_EQ(num_channels_, vec1.Channels());
79 AudioMultiVector vec2(num_channels_, initial_size);
81 EXPECT_EQ(num_channels_, vec2.Channels());
87 AudioMultiVector vec(num_channels_, array_length())
[all...] |
| accelerate.cc | 24 if (num_channels_ == 0 ||
25 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) {
70 output->PushBackInterleaved(input, fs_mult_120 * num_channels_);
72 AudioMultiVector temp_vector(num_channels_);
73 temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_],
74 peak_index * num_channels_);
79 &input[(fs_mult_120 + peak_index) * num_channels_],
80 input_length - (fs_mult_120 + peak_index) * num_channels_);
|
| preemptive_expand.cc | 29 if (num_channels_ == 0 ||
30 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
31 old_data_length >= input_length / num_channels_ - overlap_samples_) {
80 input, (unmodified_length + peak_index) * num_channels_);
82 AudioMultiVector temp_vector(num_channels_);
84 &input[(unmodified_length - peak_index) * num_channels_],
85 peak_index * num_channels_);
90 &input[unmodified_length * num_channels_],
91 input_length - unmodified_length * num_channels_);
|
| time_stretch.h | 42 num_channels_(num_channels),
50 assert(num_channels_ > 0);
51 assert(master_channel_ < num_channels_);
94 const size_t num_channels_;
|
| expand_unittest.cc | 74 num_channels_(1),
75 background_noise_(num_channels_),
76 sync_buffer_(num_channels_,
83 num_channels_) {
98 ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
103 size_t num_channels_; member in class:webrtc::ExpandTest
116 AudioMultiVector output(num_channels_);
136 AudioMultiVector output(num_channels_);
153 AudioMultiVector output(num_channels_);
|
| background_noise.cc | 28 : num_channels_(num_channels),
29 channel_parameters_(new ChannelParameters[num_channels_]),
38 for (size_t channel = 0; channel < num_channels_; ++channel) {
57 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
129 assert(channel < num_channels_);
134 assert(channel < num_channels_);
139 assert(channel < num_channels_);
144 assert(channel < num_channels_);
149 assert(channel < num_channels_);
155 assert(channel < num_channels_);
[all...] |
| neteq_stereo_unittest.cc | 50 : num_channels_(GetParam().num_channels),
69 input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
71 num_channels_];
72 output_multi_channel_ = new int16_t[kMaxBlockSize * num_channels_];
94 if (num_channels_ == 2) {
96 } else if (num_channels_ == 5) {
104 if (num_channels_ == 2) {
112 if (num_channels_ == 2) {
120 if (num_channels_ == 2) {
151 num_channels_,
242 const size_t num_channels_; member in class:webrtc::NetEqStereoTest
[all...] |
| /external/webrtc/webrtc/common_audio/resampler/ |
| push_resampler.cc | 25 num_channels_(0) {
38 num_channels == num_channels_)
48 num_channels_ = num_channels;
56 if (num_channels_ == 2) {
71 const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
72 const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
82 if (num_channels_ == 2) {
83 const size_t src_length_mono = src_length / num_channels_;
84 const size_t dst_capacity_mono = dst_capacity / num_channels_;
86 Deinterleave(src, src_length_mono, num_channels_, deinterleaved)
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_decoder_pcm.h | 21 explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
36 const size_t num_channels_; member in class:webrtc::final
42 explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
57 const size_t num_channels_; member in class:webrtc::final
|
| /external/webrtc/webrtc/modules/utility/source/ |
| audio_frame_operations_unittest.cc | 24 frame_.num_channels_ = 2;
44 EXPECT_EQ(frame1.num_channels_, frame2.num_channels_);
48 for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_;
58 frame_.num_channels_ = 1;
63 frame_.num_channels_ = 1;
71 stereo_frame.num_channels_ = 2;
79 frame_.num_channels_ = 2; // Need to set manually.
84 frame_.num_channels_ = 1;
96 mono_frame.num_channels_ = 1
[all...] |
| audio_frame_operations.cc | 26 if (frame->num_channels_ != 1) {
38 frame->num_channels_ = 2;
52 if (frame->num_channels_ != 2) {
57 frame->num_channels_ = 1;
63 if (frame->num_channels_ != 2) return;
74 frame.samples_per_channel_ * frame.num_channels_);
78 if (frame.num_channels_ != 2) {
95 for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
|
| file_recorder_impl.cc | 144 if( incomingAudioFrame.num_channels_ == 2 &&
148 tempAudioFrame.num_channels_ = 1;
162 else if( incomingAudioFrame.num_channels_ == 1 &&
166 tempAudioFrame.num_channels_ = 2;
209 ptrAudioFrame->num_channels_);
212 ptrAudioFrame->num_channels_,
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| audio_encoder_g722.cc | 40 : num_channels_(config.num_channels),
46 encoders_(new EncoderState[num_channels_]),
47 interleave_buffer_(2 * num_channels_) {
51 for (size_t i = 0; i < num_channels_; ++i) {
64 return SamplesPerChannel() / 2 * num_channels_;
72 return num_channels_;
107 for (size_t j = 0; j < num_channels_; ++j)
108 encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
119 for (size_t i = 0; i < num_channels_; ++i) {
130 for (size_t j = 0; j < num_channels_; ++j)
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| audio_decoder_pcm16b.cc | 19 : num_channels_(num_channels) {
26 return num_channels_;
|
| audio_decoder_pcm16b.h | 34 const size_t num_channels_; member in class:webrtc::final
|
| /external/webrtc/webrtc/common_audio/ |
| channel_buffer.h | 50 num_channels_(num_channels),
52 for (size_t i = 0; i < num_channels_; ++i) {
54 channels_[j * num_channels_ + i] =
56 bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i];
65 // 0 <= channel < |num_channels_|
75 // 0 <= channel < |num_channels_|
79 return &channels_[band * num_channels_];
90 // 0 <= channel < |num_channels_|
94 RTC_DCHECK_LT(channel, num_channels_);
107 for (size_t i = 0; i < num_channels_; ++i
133 const size_t num_channels_; member in class:webrtc::ChannelBuffer
[all...] |
| wav_file.h | 54 size_t num_channels() const override { return num_channels_; }
60 const size_t num_channels_; member in class:webrtc::final
82 size_t num_channels() const override { return num_channels_; }
88 size_t num_channels_; member in class:webrtc::final
|
| /external/webrtc/webrtc/modules/include/ |
| module_common_types.h | 475 * samples_per_channel_ * num_channels_
535 size_t num_channels_; member in class:webrtc::AudioFrame
563 num_channels_ = 0;
585 num_channels_ = num_channels;
608 num_channels_ = src.num_channels_;
612 const size_t length = samples_per_channel_ * num_channels_;
618 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
622 assert((num_channels_ > 0) && (num_channels_ < 3))
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_send_test_oldapi.cc | 43 input_frame_.num_channels_ = 1;
45 assert(input_block_size_samples_ * input_frame_.num_channels_ <=
61 input_frame_.num_channels_ = channels;
62 assert(input_block_size_samples_ * input_frame_.num_channels_ <=
70 input_frame_.num_channels_ = external_speech_encoder->NumChannels();
71 assert(input_block_size_samples_ * input_frame_.num_channels_ <=
89 if (input_frame_.num_channels_ > 1) {
92 input_frame_.num_channels_,
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| audio_processing_impl_unittest.cc | 46 frame.num_channels_ = 1;
60 frame.num_channels_ = 2;
64 // ProcessStream sets num_channels_ == num_output_channels.
65 frame.num_channels_ = 2;
|
| audio_buffer.cc | 56 num_channels_(num_process_channels),
153 assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1);
161 for (size_t i = 0; i < num_channels_; ++i) {
169 for (size_t i = 0; i < num_channels_; ++i) {
178 for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
188 num_channels_ = num_proc_channels_;
317 num_split_frames_, num_channels_, local
345 return num_channels_;
349 num_channels_ = num_channels
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| transient_suppressor.cc | 53 num_channels_(0),
111 num_channels_ = num_channels;
112 in_buffer_.reset(new float[analysis_length_ * num_channels_]);
115 analysis_length_ * num_channels_ * sizeof(in_buffer_[0]));
121 out_buffer_.reset(new float[analysis_length_ * num_channels_]);
124 analysis_length_ * num_channels_ * sizeof(out_buffer_[0]));
131 spectral_mean_.reset(new float[complex_analysis_length_ * num_channels_]);
134 complex_analysis_length_ * num_channels_ * sizeof(spectral_mean_[0]));
174 if (!data || data_length != data_length_ || num_channels != num_channels_ ||
210 for (int i = 0; i < num_channels_; ++i)
[all...] |
| /external/webrtc/webrtc/common_audio/resampler/include/ |
| push_resampler.h | 43 size_t num_channels_; member in class:webrtc::PushResampler
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| audio_sink.h | 37 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
|
