| /external/chromium_org/third_party/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) {
71 virtual Accelerate* Create(int 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),
80 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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| accelerate.cc | 82 int sample_rate_hz,
85 return new Accelerate(sample_rate_hz, num_channels, background_noise);
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| neteq.cc | 38 DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz);
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| preemptive_expand.cc | 102 int sample_rate_hz,
107 sample_rate_hz, num_channels, background_noise, overlap_samples);
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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);
97 DtmfBuffer buffer(sample_rate_hz);
132 DtmfBuffer buffer(sample_rate_hz);
158 DtmfBuffer buffer(sample_rate_hz);
202 DtmfBuffer buffer(sample_rate_hz);
245 DtmfBuffer buffer(sample_rate_hz);
279 DtmfBuffer buffer(sample_rate_hz);
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| delay_manager.h | 49 int sample_rate_hz);
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| /external/chromium_org/third_party/webrtc/voice_engine/ |
| utility.h | 47 int sample_rate_hz,
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| transmit_mixer_unittest.cc | 24 int sample_rate_hz, bool is_stereo) {
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| utility.cc | 77 int sample_rate_hz,
90 int destination_rate = std::min(codec_rate_hz, sample_rate_hz);
102 sample_rate_hz, destination_rate, num_channels) != 0) {
105 sample_rate_hz,
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| utility_unittest.cc | 51 void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) {
54 frame->sample_rate_hz_ = sample_rate_hz;
55 frame->samples_per_channel_ = sample_rate_hz / 100;
69 int sample_rate_hz) {
72 frame->sample_rate_hz_ = sample_rate_hz;
73 frame->samples_per_channel_ = sample_rate_hz / 100;
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| 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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| 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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| audio_coding_module_impl.cc | 2056 int sample_rate_hz; local
2101 int sample_rate_hz; local
[all...] |
| nack.h | 77 void UpdateSampleRate(int sample_rate_hz);
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| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| audio_processing_impl.h | 45 explicit AudioRate(int sample_rate_hz)
46 : rate_(sample_rate_hz),
65 AudioFormat(int sample_rate_hz, int num_channels)
66 : AudioRate(sample_rate_hz),
97 virtual int sample_rate_hz() const OVERRIDE;
116 int sample_rate_hz,
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| /external/chromium_org/third_party/webrtc/voice_engine/include/mock/ |
| fake_voe_external_media.h | 53 int samples_per_channel, int sample_rate_hz,
66 it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
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| /external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
| test_utils.h | 99 int sample_rate_hz) {
100 frame->sample_rate_hz_ = sample_rate_hz;
102 sample_rate_hz / 1000;
106 void SetContainerFormat(int sample_rate_hz,
110 SetFrameSampleRate(frame, sample_rate_hz);
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/ |
| mock_delay_manager.h | 29 int(uint16_t sequence_number, uint32_t timestamp, int sample_rate_hz));
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| delay_test.cc | 32 DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
47 int sample_rate_hz; member in struct:webrtc::__anon20462::CodecSettings
141 ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
153 config.name, &my_codec_param, config.sample_rate_hz,
257 test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
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| /external/webrtc/src/modules/audio_processing/ |
| high_pass_filter_impl.cc | 36 int InitializeFilter(FilterState* hpf, int sample_rate_hz) {
39 if (sample_rate_hz == AudioProcessingImpl::kSampleRate8kHz) {
164 apm_->sample_rate_hz());
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| audio_processing_impl.h | 62 virtual int sample_rate_hz() const;
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
| neteq.h | 72 : sample_rate_hz(16000),
79 int sample_rate_hz; // Initial vale. Will change with input data. member in struct:webrtc::NetEq::Config
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| /external/webrtc/src/modules/audio_processing/test/ |
| process_test.cc | 87 printf(" -fs SAMPLE_RATE_HZ\n");
155 int32_t sample_rate_hz = 16000; local
162 int samples_per_channel = sample_rate_hz / 100;
198 ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz));
199 samples_per_channel = sample_rate_hz / 100;
202 apm->set_sample_rate_hz(sample_rate_hz));
379 printf("Sample rate: %d Hz\n", sample_rate_hz);
703 far_frame._frequencyInHz = sample_rate_hz;
706 near_frame._frequencyInHz = sample_rate_hz;
711 fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file))
[all...] |
