/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | accelerate.h | 32 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,
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H A D | preemptive_expand.h | 32 PreemptiveExpand(int sample_rate_hz, argument 36 : TimeStretch(sample_rate_hz, num_channels, background_noise), 80 int sample_rate_hz,
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H A D | accelerate.cc | 82 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
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H A D | delay_manager.cc | 74 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
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H A D | 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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H A D | preemptive_expand.cc | 102 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
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H A D | time_stretch.h | 38 TimeStretch(int sample_rate_hz, size_t num_channels, argument 40 : sample_rate_hz_(sample_rate_hz), 41 fs_mult_(sample_rate_hz / 8000),
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | transmit_mixer_unittest.cc | 24 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
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H A D | 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, 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
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H A D | output_mixer.cc | 489 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;
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H A D | utility_unittest.cc | 51 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
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/external/chromium_org/third_party/webrtc/voice_engine/include/mock/ |
H A D | 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, 52 CallProcess(ProcessingTypes type, int16_t* audio, int samples_per_channel, int sample_rate_hz, int num_channels) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | 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); 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
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H A D | nack.cc | 47 void Nack::UpdateSampleRate(int sample_rate_hz) { argument 48 assert(sample_rate_hz > 0); 49 sample_rate_khz_ = sample_rate_hz / 1000;
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/external/chromium_org/third_party/webrtc/modules/audio_processing/aec/ |
H A D | system_delay_unittest.cc | 28 // 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;
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/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
H A D | high_pass_filter_impl.cc | 35 int InitializeFilter(FilterState* hpf, int sample_rate_hz) { argument 38 if (sample_rate_hz == AudioProcessing::kSampleRate8kHz) {
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H A D | audio_processing_impl.h | 45 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,
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/external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
H A D | 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, argument 110 SetFrameSampleRate(frame, sample_rate_hz); 98 SetFrameSampleRate(AudioFrame* frame, int sample_rate_hz) argument
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/external/webrtc/src/modules/audio_processing/ |
H A D | high_pass_filter_impl.cc | 36 int InitializeFilter(FilterState* hpf, int sample_rate_hz) { argument 39 if (sample_rate_hz == AudioProcessingImpl::kSampleRate8kHz) { 164 apm_->sample_rate_hz());
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H A D | audio_processing_impl.cc | 205 int AudioProcessingImpl::sample_rate_hz() const { function in class:webrtc::AudioProcessingImpl
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | delay_test.cc | 32 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;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | neteq_rtpplay.cc | 168 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;
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/external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
H A D | opensles_input.h | 63 int32_t SetRecordingSampleRate(uint32_t sample_rate_hz) { return 0; } argument
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H A D | opensles_output.h | 64 int32_t SetPlayoutSampleRate(uint32_t sample_rate_hz) { return 0; } argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
H A D | 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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