| /frameworks/ex/variablespeed/jni/ |
| ring_buffer.cc | 25 num_channels_ = num_channels;
44 samples_ = new float[size_ * num_channels_];
46 size_ * num_channels_ * sizeof(samples_[0]));
50 temp_read_buffer_ = new float[temp_read_buffer_size_ * num_channels_];
52 temp_read_buffer_size_ * num_channels_ * sizeof(samples_[0]));
87 memcpy(samples_ + head_ * num_channels_, samples,
88 num_frames * num_channels_ * sizeof(samples[0]));
92 memcpy(samples_ + head_ * num_channels_, samples,
93 num_channels_ * overhead * sizeof(samples[0]));
95 memcpy(samples_, samples + overhead * num_channels_,
[all...] |
| sola_time_scaler.h | 48 num_channels_ = num_channels;
65 int num_channels_; member in class:video_editing::SolaAnalyzer
125 int num_channels() const { return num_channels_; }
133 int num_channels_; // channel valence of audio stream member in class:video_editing::SolaTimeScaler
|
| sola_time_scaler.cc | 37 num_frames *= num_channels_;
51 return num_frames * num_channels_;
60 num_channels_ = 0;
88 num_channels_ = num_channels;
162 (sample_rate_ * duration), num_channels_, 1); local
167 (sample_rate_ * ratio_ * duration), num_channels_, 2); local
176 analyzer_->Init(sample_rate_, num_channels_);
285 output_pointer + ((merge_offset + i) * num_channels_),
290 if (score == (num_overlap_frames_ * num_channels_)) {
296 output_pointer + ((merge_offset - i) * num_channels_),
[all...] |
| /external/chromium_org/third_party/webrtc/common_audio/resampler/ |
| push_resampler.cc | 25 num_channels_(0) {
38 num_channels == num_channels_)
48 num_channels_ = num_channels;
54 if (num_channels_ == 2) {
69 const int src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
70 const int dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
80 if (num_channels_ == 2) {
81 const int src_length_mono = src_length / num_channels_;
82 const int dst_capacity_mono = dst_capacity / num_channels_;
84 Deinterleave(src, src_length_mono, num_channels_, deinterleaved)
[all...] |
| /external/chromium_org/third_party/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...] |
| accelerate.cc | 24 if (num_channels_ == 0 || static_cast<int>(input_length) / num_channels_ <
57 output->PushBackInterleaved(input, fs_mult_120 * num_channels_);
59 AudioMultiVector temp_vector(num_channels_);
60 temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_],
61 peak_index * num_channels_);
66 &input[(fs_mult_120 + peak_index) * num_channels_],
67 input_length - (fs_mult_120 + peak_index) * num_channels_);
|
| 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...] |
| 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_) {
76 input, (unmodified_length + peak_index) * num_channels_);
78 AudioMultiVector temp_vector(num_channels_);
80 &input[(unmodified_length - peak_index) * num_channels_],
81 peak_index * num_channels_);
86 &input[unmodified_length * num_channels_],
87 input_length - unmodified_length * num_channels_);
|
| merge.h | 38 num_channels_(num_channels),
43 expanded_(num_channels_) {
44 assert(num_channels_ > 0);
55 // must have |num_channels_| elements.
64 const size_t num_channels_; member in class:webrtc::Merge
|
| time_stretch.h | 42 num_channels_(static_cast<int>(num_channels)),
50 assert(num_channels_ > 0);
51 assert(static_cast<int>(master_channel_) < num_channels_);
89 const int num_channels_; member in class:webrtc::TimeStretch
|
| background_noise.cc | 25 : num_channels_(num_channels),
26 channel_parameters_(new ChannelParameters[num_channels_]),
35 for (size_t channel = 0; channel < num_channels_; ++channel) {
54 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
126 assert(channel < num_channels_);
131 assert(channel < num_channels_);
136 assert(channel < num_channels_);
141 assert(channel < num_channels_);
146 assert(channel < num_channels_);
152 assert(channel < num_channels_);
[all...] |
| expand.h | 43 num_channels_(num_channels),
50 channel_parameters_(new ChannelParameters[num_channels_]) {
53 assert(num_channels_ > 0);
77 assert(channel < num_channels_);
83 assert(channel < num_channels_);
117 const size_t num_channels_;
|
| neteq_stereo_unittest.cc | 51 : 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) {
152 num_channels_,
241 const int num_channels_; member in class:webrtc::NetEqStereoTest
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| common.h | 44 num_channels_(num_channels) {
52 assert(i < num_channels_);
58 assert(i < num_channels_);
64 int num_channels() { return num_channels_; }
65 int length() { return samples_per_channel_ * num_channels_; }
71 int num_channels_;
|
| audio_processing_impl_unittest.cc | 44 frame.num_channels_ = 1;
58 frame.num_channels_ = 2;
62 // ProcessStream sets num_channels_ == num_output_channels.
63 frame.num_channels_ = 2;
|
| /external/chromium_org/third_party/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 (int 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 (int i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
|
| /external/webrtc/src/modules/audio_processing/ |
| audio_buffer.cc | 67 num_channels_(0),
98 assert(channel >= 0 && channel < num_channels_);
107 assert(channel >= 0 && channel < num_channels_);
116 assert(channel >= 0 && channel < num_channels_);
137 assert(channel >= 0 && channel < num_channels_);
146 assert(channel >= 0 && channel < num_channels_);
151 assert(channel >= 0 && channel < num_channels_);
156 assert(channel >= 0 && channel < num_channels_);
161 assert(channel >= 0 && channel < num_channels_);
178 return num_channels_;
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_pcma.cc | 30 NULL, &in_audio_[in_audio_ix_read_], frame_len_smpl_ * num_channels_,
34 in_audio_ix_read_ += frame_len_smpl_ * num_channels_;
|
| acm_pcmu.cc | 30 NULL, &in_audio_[in_audio_ix_read_], frame_len_smpl_ * num_channels_,
35 in_audio_ix_read_ += frame_len_smpl_ * num_channels_;
|
| acm_pcm16b.cc | 59 frame_len_smpl_ * num_channels_,
63 in_audio_ix_read_ += frame_len_smpl_ * num_channels_;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| audio_sink.h | 37 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
|
| /external/chromium_org/third_party/webrtc/modules/interface/ |
| module_common_types.h | 642 * samples_per_channel_ * num_channels_
702 int num_channels_; member in class:webrtc::AudioFrame
730 num_channels_ = 0;
749 num_channels_ = num_channels;
772 num_channels_ = src.num_channels_;
776 const int length = samples_per_channel_ * num_channels_;
782 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
786 assert((num_channels_ > 0) && (num_channels_ < 3))
[all...] |
| /external/chromium_org/third_party/webrtc/common_audio/resampler/include/ |
| push_resampler.h | 43 int num_channels_; member in class:webrtc::PushResampler
|
| /external/chromium_org/media/base/android/ |
| audio_decoder_job.cc | 42 num_channels_(0),
101 num_channels_ = configs.audio_channels;
105 bytes_per_frame_ = kBytesPerAudioOutputSample * num_channels_;
111 num_channels_ != configs.audio_channels ||
126 audio_codec_, sampling_rate_, num_channels_, &audio_extra_data_[0],
|
