/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | decision_logic_unittest.cc | 24 int fs_hz = 8000; local 25 int output_size_samples = fs_hz / 100; // Samples per 10 ms. 31 DecisionLogic* logic = DecisionLogic::Create(fs_hz, output_size_samples, 36 logic = DecisionLogic::Create(fs_hz, output_size_samples, 42 logic = DecisionLogic::Create(fs_hz, output_size_samples, 48 logic = DecisionLogic::Create(fs_hz, output_size_samples,
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H A D | post_decode_vad.cc | 50 int fs_hz) { 56 fs_hz > 16000) { 57 // TODO(hlundin): Remove restriction on fs_hz. 75 int vad_frame_size_samples = vad_frame_size_ms * fs_hz / 1000; 78 vad_instance_, fs_hz, &signal[vad_sample_index], 47 Update(int16_t* signal, int length, AudioDecoder::SpeechType speech_type, bool sid_frame, int fs_hz) argument
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H A D | decision_logic.cc | 26 DecisionLogic* DecisionLogic::Create(int fs_hz, argument 36 return new DecisionLogicNormal(fs_hz, 45 return new DecisionLogicFax(fs_hz, 58 DecisionLogic::DecisionLogic(int fs_hz, argument 78 SetSampleRate(fs_hz, output_size_samples); 98 void DecisionLogic::SetSampleRate(int fs_hz, int output_size_samples) { argument 100 assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz [all...] |
H A D | comfort_noise.h | 35 ComfortNoise(int fs_hz, DecoderDatabase* decoder_database, argument 37 : fs_hz_(fs_hz),
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H A D | decision_logic_fax.h | 25 DecisionLogicFax(int fs_hz, argument 32 : DecisionLogic(fs_hz, output_size_samples, playout_mode,
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H A D | normal.h | 35 Normal(int fs_hz, DecoderDatabase* decoder_database, argument 38 : fs_hz_(fs_hz),
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H A D | dtmf_tone_generator_unittest.cc | 31 void TestAllTones(int fs_hz, int channels) { argument 36 ss << "Checking event " << event << " at sample rate " << fs_hz; local 39 ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, kAttenuation)); 48 double x = k3dbAttenuation * sin(2.0 * pi * f1 / fs_hz * (-n - 1)) + 49 sin(2.0 * pi * f2 / fs_hz * (-n - 1)); 62 void TestAmplitudes(int fs_hz, int channels) { argument 70 ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, 0)); // 0 attenuation. 75 ss << "Checking event " << event << " at sample rate " << fs_hz; local 78 ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, attenuation));
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H A D | statistics_calculator.cc | 83 void StatisticsCalculator::IncreaseCounter(int num_samples, int fs_hz) { argument 86 static_cast<uint32_t>(fs_hz * kMaxReportPeriod)) { 106 int fs_hz, 112 if (fs_hz <= 0 || !stats) { 118 stats->current_buffer_size_ms = num_samples_in_buffers * 1000 / fs_hz; 120 (fs_hz / 1000); 105 GetNetworkStatistics( int fs_hz, int num_samples_in_buffers, int samples_per_packet, const DelayManager& delay_manager, const DecisionLogic& decision_logic, NetEqNetworkStatistics *stats) argument
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H A D | statistics_calculator.h | 67 // of |fs_hz|. 68 void IncreaseCounter(int num_samples, int fs_hz); 74 // is |fs_hz|, the total number of samples in packet buffer and sync buffer 77 void GetNetworkStatistics(int fs_hz,
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H A D | decoder_database.h | 44 fs_hz(8000), 50 fs_hz(fs), 58 int fs_hz; member in struct:webrtc::DecoderDatabase::DecoderInfo 92 int fs_hz, AudioDecoder* decoder);
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H A D | dtmf_buffer.h | 57 // Set up the buffer for use at sample rate |fs_hz|. 58 explicit DtmfBuffer(int fs_hz) { argument 59 SetSampleRate(fs_hz); 90 virtual int SetSampleRate(int fs_hz);
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H A D | dtmf_buffer.cc | 176 int DtmfBuffer::SetSampleRate(int fs_hz) { argument 177 if (fs_hz != 8000 && 178 fs_hz != 16000 && 179 fs_hz != 32000 && 180 fs_hz != 48000) { 183 max_extrapolation_samples_ = 7 * fs_hz / 100; 184 frame_len_samples_ = fs_hz / 100;
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H A D | decision_logic.h | 36 static DecisionLogic* Create(int fs_hz, 45 DecisionLogic(int fs_hz, 63 void SetSampleRate(int fs_hz, int output_size_samples);
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H A D | post_decode_vad.h | 48 // samples. The data is of type |speech_type|, at the sample rate |fs_hz|. 50 AudioDecoder::SpeechType speech_type, bool sid_frame, int fs_hz);
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H A D | decision_logic_normal.h | 25 DecisionLogicNormal(int fs_hz, argument 32 : DecisionLogic(fs_hz, output_size_samples, playout_mode,
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H A D | decoder_database.cc | 47 int fs_hz = AudioDecoder::CodecSampleRateHz(codec_type); local 49 DecoderInfo info(codec_type, fs_hz, NULL, false); 60 int fs_hz, 68 if (fs_hz != 8000 && fs_hz != 16000 && fs_hz != 32000 && fs_hz != 48000) { 76 DecoderInfo info(codec_type, fs_hz, decoder, true); 58 InsertExternal(uint8_t rtp_payload_type, NetEqDecoder codec_type, int fs_hz, AudioDecoder* decoder) argument
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H A D | merge.h | 36 Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer) argument 37 : fs_hz_(fs_hz),
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H A D | neteq_impl.cc | 634 if (decoder_info->fs_hz != fs_hz_ || 636 SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels()); 1138 if (decoder_info->fs_hz != fs_hz_ || 1141 SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels()); 1828 void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) { argument 1832 fs_hz, channels)); 1833 merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get())); 1836 void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) { argument 1837 LOG_API2(fs_hz, channels); 1839 assert(fs_hz [all...] |
H A D | decoder_database_unittest.cc | 54 EXPECT_EQ(8000, info->fs_hz); 128 EXPECT_EQ(8000, info->fs_hz);
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H A D | neteq_impl.h | 318 // |fs_hz| and |channels| number audio channels. 319 void SetSampleRateAndChannels(int fs_hz, size_t channels) 327 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/ |
H A D | mock_decoder_database.h | 33 int(uint8_t rtp_payload_type, NetEqDecoder codec_type, int fs_hz,
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