Searched refs:sample_rate_hz (Results 1 - 25 of 60) sorted by relevance

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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Daccelerate.h32 Accelerate(int sample_rate_hz, size_t num_channels, argument
34 : TimeStretch(sample_rate_hz, num_channels, background_noise) {
71 virtual Accelerate* Create(int sample_rate_hz,
H A Dpreemptive_expand.h32 PreemptiveExpand(int sample_rate_hz, argument
36 : TimeStretch(sample_rate_hz, num_channels, background_noise),
80 int sample_rate_hz,
H A Dtime_stretch.h38 TimeStretch(int sample_rate_hz, size_t num_channels, argument
40 : sample_rate_hz_(sample_rate_hz),
41 fs_mult_(sample_rate_hz / 8000),
H A Daccelerate.cc82 int sample_rate_hz,
85 return new Accelerate(sample_rate_hz, num_channels, background_noise);
81 Create( int sample_rate_hz, size_t num_channels, const BackgroundNoise& background_noise) const argument
H A Dneteq.cc38 DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz);
H A Dpreemptive_expand.cc102 int sample_rate_hz,
107 sample_rate_hz, num_channels, background_noise, overlap_samples);
101 Create( int sample_rate_hz, size_t num_channels, const BackgroundNoise& background_noise, int overlap_samples) const argument
H A Ddtmf_buffer_unittest.cc30 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);
H A Ddelay_manager.h49 int sample_rate_hz);
/external/chromium_org/third_party/webrtc/voice_engine/
H A Dutility.h47 int sample_rate_hz,
H A Dtransmit_mixer_unittest.cc24 int sample_rate_hz, bool is_stereo) {
22 Process(int channel, ProcessingTypes type, int16_t audio[], int samples_per_channel, int sample_rate_hz, bool is_stereo) argument
H A Dutility.cc77 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,
74 DownConvertToCodecFormat(const int16_t* src_data, int samples_per_channel, int num_channels, int sample_rate_hz, int codec_num_channels, int codec_rate_hz, int16_t* mono_buffer, PushResampler<int16_t>* resampler, AudioFrame* dst_af) argument
H A Dutility_unittest.cc51 void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) { argument
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;
68 SetStereoFrame(AudioFrame* frame, float left, float right, int sample_rate_hz) argument
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/
H A Dinitial_delay_manager.cc39 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);
34 UpdateLastReceivedPacket( const WebRtcRTPHeader& rtp_info, uint32_t receive_timestamp, PacketType type, bool new_codec, int sample_rate_hz, SyncStream* sync_stream) argument
234 UpdatePlayoutTimestamp( const RTPHeader& current_header, int sample_rate_hz) argument
H A Dinitial_delay_manager.h51 // since the last time |new_codec| should be true. |sample_rate_hz| is the
61 int sample_rate_hz,
88 int sample_rate_hz);
H A Daudio_coding_module_impl.cc2056 int sample_rate_hz; local
2059 encoder_type, &codec_name, &sample_rate_hz, &channels)) {
2064 codec_name.c_str(), &codec, sample_rate_hz, channels);
2101 int sample_rate_hz; local
2104 decoder_type, &codec_name, &sample_rate_hz, &channels)) {
2109 codec_name.c_str(), &codec, sample_rate_hz, channels);
2179 int* sample_rate_hz,
2185 *sample_rate_hz = 8000;
2190 *sample_rate_hz = 16000;
2195 *sample_rate_hz
2177 MapCodecTypeToParameters(int codec_type, std::string* codec_name, int* sample_rate_hz, int* channels) argument
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H A Dnack.h77 void UpdateSampleRate(int sample_rate_hz);
/external/chromium_org/third_party/webrtc/voice_engine/include/mock/
H A Dfake_voe_external_media.h53 int samples_per_channel, int sample_rate_hz,
66 it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
52 CallProcess(ProcessingTypes type, int16_t* audio, int samples_per_channel, int sample_rate_hz, int num_channels) argument
/external/chromium_org/third_party/webrtc/modules/audio_processing/test/
H A Dtest_utils.h99 int sample_rate_hz) {
100 frame->sample_rate_hz_ = sample_rate_hz;
102 sample_rate_hz / 1000;
106 void SetContainerFormat(int sample_rate_hz, argument
110 SetFrameSampleRate(frame, sample_rate_hz);
98 SetFrameSampleRate(AudioFrame* frame, int sample_rate_hz) argument
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/
H A Dmock_delay_manager.h29 int(uint16_t sequence_number, uint32_t timestamp, int sample_rate_hz));
/external/chromium_org/third_party/webrtc/modules/audio_processing/
H A Daudio_processing_impl.h45 explicit AudioRate(int sample_rate_hz) argument
46 : rate_(sample_rate_hz),
65 AudioFormat(int sample_rate_hz, int num_channels) argument
66 : AudioRate(sample_rate_hz),
97 virtual int sample_rate_hz() const OVERRIDE;
116 int sample_rate_hz,
H A Dhigh_pass_filter_impl.cc35 int InitializeFilter(FilterState* hpf, int sample_rate_hz) { argument
38 if (sample_rate_hz == AudioProcessing::kSampleRate8kHz) {
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A Ddelay_test.cc32 DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
47 int sample_rate_hz; member in struct:webrtc::__anon15844::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;
/external/webrtc/src/modules/audio_processing/
H A Dhigh_pass_filter_impl.cc36 int InitializeFilter(FilterState* hpf, int sample_rate_hz) { argument
39 if (sample_rate_hz == AudioProcessingImpl::kSampleRate8kHz) {
164 apm_->sample_rate_hz());
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/
H A Dneteq.h72 : sample_rate_hz(16000),
79 int sample_rate_hz; // Initial vale. Will change with input data. member in struct:webrtc::NetEq::Config
/external/webrtc/src/modules/audio_processing/test/
H A Dprocess_test.cc155 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),
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