| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| comfort_noise.h | 37 : fs_hz_(fs_hz),
39 overlap_length_(5 * fs_hz_ / 8000),
63 int fs_hz_; member in class:webrtc::ComfortNoise
|
| normal.h | 38 : fs_hz_(fs_hz),
59 int fs_hz_; member in class:webrtc::Normal
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| merge.h | 37 : fs_hz_(fs_hz),
39 fs_mult_(fs_hz_ / 8000),
40 timestamps_per_call_(fs_hz_ / 100),
63 const int fs_hz_; member in class:webrtc::Merge
98 const int fs_mult_; // fs_hz_ / 8000.
|
| comfort_noise.cc | 55 assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
56 fs_hz_ == 48000);
93 if (fs_hz_ == 8000) {
98 } else if (fs_hz_ == 16000) {
103 } else if (fs_hz_ == 32000) {
108 } else { // fs_hz_ == 48000
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| merge.cc | 31 assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
32 fs_hz_ == 48000);
33 assert(fs_hz_ <= kMaxSampleRate); // Should not be possible.
250 int decimation_factor = fs_hz_ / 4000;
252 int length_limit = fs_hz_ / 100; // 10 ms in samples.
253 if (fs_hz_ == 8000) {
256 } else if (fs_hz_ == 16000) {
259 } else if (fs_hz_ == 32000)
[all...] |
| expand.h | 42 fs_hz_(fs),
116 const int fs_hz_;
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| expand.cc | 49 int fs_mult = fs_hz_ / 8000;
106 if (fs_hz_ == 8000) {
111 } else if (fs_hz_ == 16000) {
116 } else if (fs_hz_ == 32000) {
324 int fs_mult = fs_hz_ / 8000;
719 if (fs_hz_ == 8000) {
723 } else if (fs_hz_ == 16000) {
727 } else if (fs_hz_ == 32000) {
[all...] |
| neteq_impl.cc | 103 fs_hz_ = fs;
303 stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
634 if (decoder_info->fs_hz != fs_hz_ ||
652 delay_manager_->SetPacketAudioLength((1000 * temp_bufsize) / fs_hz_);
661 fs_hz_);
699 sid_frame_available, fs_hz_);
863 stats_.IncreaseCounter(output_size_samples_, fs_hz_);
[all...] |
| normal.cc | 48 const unsigned fs_mult = fs_hz_ / 8000;
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| neteq_impl.h | 367 int fs_hz_ GUARDED_BY(crit_sect_);
|
