Searched defs:sample_rate_hz (Results 1 - 25 of 34) 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 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 Ddelay_manager.cc74 int sample_rate_hz) {
75 if (sample_rate_hz <= 0) {
99 packet_len_ms = (1000 * packet_len_samp) / sample_rate_hz;
72 Update(uint16_t sequence_number, uint32_t timestamp, int sample_rate_hz) 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 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 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),
/external/chromium_org/third_party/webrtc/voice_engine/
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 Doutput_mixer.cc489 int OutputMixer::GetMixedAudio(int sample_rate_hz, argument
493 "OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%d)",
494 sample_rate_hz, num_channels);
504 frame->sample_rate_hz_ = sample_rate_hz;
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/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_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 Dnack.cc47 void Nack::UpdateSampleRate(int sample_rate_hz) { argument
48 assert(sample_rate_hz > 0);
49 sample_rate_khz_ = sample_rate_hz / 1000;
/external/chromium_org/third_party/webrtc/modules/audio_processing/aec/
H A Dsystem_delay_unittest.cc28 // Initialization of AEC handle with respect to |sample_rate_hz|. Since the
30 void Init(int sample_rate_hz);
94 void SystemDelayTest::Init(int sample_rate_hz) { argument
96 EXPECT_EQ(0, WebRtcAec_Init(handle_, sample_rate_hz, 48000));
99 samples_per_frame_ = sample_rate_hz / 100;
/external/chromium_org/third_party/webrtc/modules/audio_processing/
H A Dhigh_pass_filter_impl.cc35 int InitializeFilter(FilterState* hpf, int sample_rate_hz) { argument
38 if (sample_rate_hz == AudioProcessing::kSampleRate8kHz) {
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,
/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/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());
H A Daudio_processing_impl.cc205 int AudioProcessingImpl::sample_rate_hz() const { function in class:webrtc::AudioProcessingImpl
/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/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/
H A Dneteq_rtpplay.cc168 int sample_rate_hz = 16000; local
170 config.sample_rate_hz = sample_rate_hz;
194 input_frame_size_timestamps = 30 * sample_rate_hz / 1000;
237 packet->time_ms() * sample_rate_hz / 1000);
276 sample_rate_hz = 1000 * samples_per_channel / kOutputBlockSizeMs;
/external/chromium_org/third_party/webrtc/modules/audio_device/android/
H A Dopensles_input.h63 int32_t SetRecordingSampleRate(uint32_t sample_rate_hz) { return 0; } argument
H A Dopensles_output.h64 int32_t SetPlayoutSampleRate(uint32_t sample_rate_hz) { return 0; } argument
/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

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