/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | isac.c | 2208 * - sample_rate_hz : sampling rate in Hertz, valid values are 16000, 2215 uint16_t sample_rate_hz) { 2219 if ((sample_rate_hz != 16000) && (sample_rate_hz != 32000) && 2220 (sample_rate_hz != 48000)) { 2225 if (sample_rate_hz == 16000) { 2292 instISAC->in_sample_rate_hz = sample_rate_hz; 2305 * - sample_rate_hz : sampling rate in Hertz, valid values are 16000 2312 uint16_t sample_rate_hz) { 2316 if (sample_rate_hz 2214 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst, uint16_t sample_rate_hz) argument 2311 WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst, uint16_t sample_rate_hz) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | audio_coding_module_impl.cc | 2056 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 [all...] |
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_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/ |
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 | 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 | 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 | neteq_impl.cc | 96 int fs = config.sample_rate_hz; 213 const int sample_rate_hz = AudioDecoder::CodecSampleRateHz(codec); local 215 sample_rate_hz, decoder);
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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 | 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 | 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/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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/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_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 | audio_processing_impl.cc | 311 int AudioProcessingImpl::sample_rate_hz() const { function in class:webrtc::AudioProcessingImpl 517 int sample_rate_hz, 527 sample_rate_hz, 515 AnalyzeReverseStream(const float* const* data, int samples_per_channel, int sample_rate_hz, ChannelLayout layout) argument
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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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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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/external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
H A D | audio_processing_unittest.cc | 236 std::string ResourceFilePath(std::string name, int sample_rate_hz) { argument 239 ss << name << sample_rate_hz / 1000 << "_stereo"; local 295 void Init(int sample_rate_hz, 409 void ApmTest::Init(int sample_rate_hz, argument 416 SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_); 427 std::string filename = ResourceFilePath("far", sample_rate_hz); 435 filename = ResourceFilePath("near", sample_rate_hz); 445 sample_rate_hz,
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H A D | process_test.cc | 159 int32_t sample_rate_hz = 16000; local 165 int samples_per_channel = sample_rate_hz / 100; 202 ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz)); 203 samples_per_channel = sample_rate_hz / 100; 448 printf("Sample rate: %d Hz\n", sample_rate_hz); 844 far_frame.sample_rate_hz_ = sample_rate_hz; 847 near_frame.sample_rate_hz_ = sample_rate_hz; 852 fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file)); 853 samples_per_channel = sample_rate_hz / 10 [all...] |
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/chromium_org/third_party/webrtc/modules/interface/ |
H A D | module_common_types.h | 668 int samples_per_channel, int sample_rate_hz, 731 int samples_per_channel, int sample_rate_hz, 738 sample_rate_hz_ = sample_rate_hz; 729 UpdateFrame(int id, uint32_t timestamp, const int16_t* data, int samples_per_channel, int sample_rate_hz, SpeechType speech_type, VADActivity vad_activity, int num_channels, uint32_t energy) argument
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