| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| accelerate.h | 32 Accelerate(int sample_rate_hz, size_t num_channels,
34 : TimeStretch(sample_rate_hz, num_channels, background_noise) {
75 virtual Accelerate* Create(int sample_rate_hz,
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| neteq.cc | 33 ss << "sample_rate_hz=" << sample_rate_hz << ", enable_audio_classifier="
53 DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz);
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| preemptive_expand.h | 32 PreemptiveExpand(int sample_rate_hz,
36 : TimeStretch(sample_rate_hz, num_channels, background_noise),
81 int sample_rate_hz,
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| time_stretch.h | 38 TimeStretch(int sample_rate_hz, size_t num_channels,
40 : sample_rate_hz_(sample_rate_hz),
41 fs_mult_(sample_rate_hz / 8000),
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| dtmf_buffer_unittest.cc | 30 static int sample_rate_hz = 8000; member in namespace:webrtc
57 DtmfBuffer* buffer = new DtmfBuffer(sample_rate_hz);
91 DtmfBuffer buffer(sample_rate_hz);
126 DtmfBuffer buffer(sample_rate_hz);
152 DtmfBuffer buffer(sample_rate_hz);
196 DtmfBuffer buffer(sample_rate_hz);
239 DtmfBuffer buffer(sample_rate_hz);
273 DtmfBuffer buffer(sample_rate_hz);
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| /external/webrtc/webrtc/common_audio/vad/mock/ |
| mock_vad.h | 28 int sample_rate_hz));
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_encoder_pcm.cc | 38 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
39 : sample_rate_hz_(sample_rate_hz),
45 config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
47 RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| audio_decoder_g722.cc | 35 int sample_rate_hz,
38 RTC_DCHECK_EQ(sample_rate_hz, 16000);
74 int sample_rate_hz,
77 RTC_DCHECK_EQ(sample_rate_hz, 16000);
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| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| audio_decoder_ilbc.h | 32 int sample_rate_hz,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| audio_decoder_pcm16b.h | 29 int sample_rate_hz,
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| /external/webrtc/webrtc/modules/audio_processing/agc/ |
| mock_agc.h | 25 int sample_rate_hz));
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| agc.cc | 52 int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
53 vad_.ProcessChunk(audio, length, sample_rate_hz);
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| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| mock_nonlinear_beamformer.h | 26 MOCK_METHOD2(Initialize, void(int chunk_size_ms, int sample_rate_hz));
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| /external/webrtc/webrtc/modules/audio_processing/ |
| high_pass_filter_impl.cc | 25 explicit BiquadFilter(int sample_rate_hz) :
26 ba_(sample_rate_hz == AudioProcessing::kSampleRate8kHz ?
95 void HighPassFilterImpl::Initialize(size_t channels, int sample_rate_hz) {
98 new_filters[i].reset(new BiquadFilter(sample_rate_hz));
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| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| click_annotate.cc | 63 int sample_rate_hz = atoi(argv[4]); local
64 if (sample_rate_hz <= 0) {
69 TransientDetector detector(sample_rate_hz);
71 size_t audio_buffer_length = chunk_size_ms * sample_rate_hz / 1000;
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| /external/webrtc/webrtc/voice_engine/ |
| utility.cc | 38 int sample_rate_hz,
53 if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
55 LOG(LS_ERROR) << "InitializeIfNeeded failed: sample_rate_hz = "
56 << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
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| utility_unittest.cc | 48 void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) {
51 frame->sample_rate_hz_ = sample_rate_hz;
52 frame->samples_per_channel_ = sample_rate_hz / 100;
66 int sample_rate_hz) {
69 frame->sample_rate_hz_ = sample_rate_hz;
70 frame->samples_per_channel_ = sample_rate_hz / 100;
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| constant_pcm_packet_source.cc | 24 int sample_rate_hz,
28 samples_per_ms_(sample_rate_hz / 1000),
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| audio_encoder_isac_t_impl.h | 27 config.sample_rate_hz = codec_inst.plfreq;
29 rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
44 switch (sample_rate_hz) {
165 RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
183 RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| initial_delay_manager.h | 51 // since the last time |new_codec| should be true. |sample_rate_hz| is the
61 int sample_rate_hz,
88 int sample_rate_hz);
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| initial_delay_manager.cc | 39 int sample_rate_hz,
79 UpdatePlayoutTimestamp(*current_header, sample_rate_hz);
93 buffered_audio_ms_ += timestamp_increase * 1000 / sample_rate_hz;
96 UpdatePlayoutTimestamp(*current_header, sample_rate_hz);
235 const RTPHeader& current_header, int sample_rate_hz) {
237 initial_delay_ms_ * sample_rate_hz / 1000);
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| acm_receiver.cc | 183 const int sample_rate_hz = [&decoder] { local
187 receive_timestamp = NowInTimestamp(sample_rate_hz);
198 last_packet_sample_rate_hz_ = rtc::Optional<int>(decoder->sample_rate_hz);
305 int sample_rate_hz,
329 decoder.sample_rate_hz == sample_rate_hz) {
349 audio_decoder, neteq_decoder, name, payload_type, sample_rate_hz);
362 decoder.sample_rate_hz = sample_rate_hz;
442 codec->plfreq = last_audio_decoder_->sample_rate_hz;
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| test_utils.h | 92 int sample_rate_hz);
95 void SetContainerFormat(int sample_rate_hz,
99 SetFrameSampleRate(frame, sample_rate_hz);
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| /external/webrtc/webrtc/modules/audio_coding/neteq/mock/ |
| mock_external_decoder_pcm16b.h | 35 int sample_rate_hz,
73 int sample_rate_hz,
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| /external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
| intelligibility_enhancer.cc | 73 RealFourier::FftOrder(config.sample_rate_hz * kWindowSizeMs / 1000))),
76 static_cast<size_t>(config.sample_rate_hz * kChunkSizeMs / 1000)),
77 bank_size_(GetBankSize(config.sample_rate_hz, kErbResolution)),
78 sample_rate_hz_(config.sample_rate_hz),
131 int sample_rate_hz,
133 RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
149 int sample_rate_hz,
151 RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
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