/external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
D | audio_decoder_pcm16b.cc | 31 int sample_rate_hz, in DecodeInternal() argument 34 RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 || in DecodeInternal() 35 sample_rate_hz == 32000 || sample_rate_hz == 48000) in DecodeInternal() 36 << "Unsupported sample rate " << sample_rate_hz; in DecodeInternal()
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D | audio_encoder_pcm16b.cc | 33 config.sample_rate_hz = codec_inst.plfreq; in CreateConfig() 35 codec_inst.pacsize, rtc::CheckedDivExact(config.sample_rate_hz, 1000)); in CreateConfig() 42 if ((sample_rate_hz != 8000) && (sample_rate_hz != 16000) && in IsOk() 43 (sample_rate_hz != 32000) && (sample_rate_hz != 48000)) in IsOk()
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D | audio_encoder_pcm16b.h | 25 Config() : AudioEncoderPcm::Config(107), sample_rate_hz(8000) {} in Config() 28 int sample_rate_hz; member 32 : AudioEncoderPcm(config, config.sample_rate_hz) {} in AudioEncoderPcm16B()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
D | unittest.cc | 80 BoundedCapacityChannel(int sample_rate_hz, int rate_bits_per_second) in BoundedCapacityChannel() argument 83 (8.0 * sample_rate_hz)) {} in BoundedCapacityChannel() 106 int sample_rate_hz, in TestGetSetBandwidthInfo() argument 115 ASSERT_EQ(0, T::SetEncSampRate(encdec, sample_rate_hz)); in TestGetSetBandwidthInfo() 125 ASSERT_EQ(0, T::SetEncSampRate(enc, sample_rate_hz)); in TestGetSetBandwidthInfo() 134 T::SetEncSampRateInDecoder(dec, sample_rate_hz); in TestGetSetBandwidthInfo() 139 BoundedCapacityChannel channel1(sample_rate_hz, rate_bits_per_second), in TestGetSetBandwidthInfo() 140 channel2(sample_rate_hz, rate_bits_per_second); in TestGetSetBandwidthInfo() 163 const int send_time = elapsed_time_ms * (sample_rate_hz / 1000); in TestGetSetBandwidthInfo() 172 ASSERT_EQ(0, T::SetDecSampRate(encdec, sample_rate_hz)); in TestGetSetBandwidthInfo() [all …]
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D | audio_decoder_isac_t_impl.h | 44 int sample_rate_hz, in DecodeInternal() argument 47 RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000) in DecodeInternal() 48 << "Unsupported sample rate " << sample_rate_hz; in DecodeInternal() 49 if (sample_rate_hz != decoder_sample_rate_hz_) { in DecodeInternal() 50 RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz)); in DecodeInternal() 51 decoder_sample_rate_hz_ = sample_rate_hz; in DecodeInternal()
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/external/webrtc/webrtc/modules/audio_coding/codecs/ |
D | audio_decoder.cc | 21 int sample_rate_hz, size_t max_decoded_bytes, in Decode() argument 29 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, in Decode() 34 int sample_rate_hz, size_t max_decoded_bytes, in DecodeRedundant() argument 42 return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded, in DecodeRedundant() 48 int sample_rate_hz, int16_t* decoded, in DecodeRedundantInternal() argument 50 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, in DecodeRedundantInternal()
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D | audio_decoder.h | 46 int sample_rate_hz, 55 int sample_rate_hz, 108 int sample_rate_hz, 114 int sample_rate_hz,
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/external/webrtc/webrtc/modules/audio_processing/transient/ |
D | transient_detector.cc | 30 TransientDetector::TransientDetector(int sample_rate_hz) in TransientDetector() argument 31 : samples_per_chunk_(sample_rate_hz * ts::kChunkSizeMs / 1000), in TransientDetector() 37 assert(sample_rate_hz == ts::kSampleRate8kHz || in TransientDetector() 38 sample_rate_hz == ts::kSampleRate16kHz || in TransientDetector() 39 sample_rate_hz == ts::kSampleRate32kHz || in TransientDetector() 40 sample_rate_hz == ts::kSampleRate48kHz); in TransientDetector() 41 int samples_per_transient = sample_rate_hz * kTransientLengthMs / 1000; in TransientDetector()
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D | transient_detector_unittest.cc | 45 int sample_rate_hz = kSampleRatesHz[i]; in TEST() local 50 << (sample_rate_hz / 1000) << "kHz"; in TEST() 67 << (sample_rate_hz / 1000) << "kHz"; in TEST() 78 TransientDetector detector(sample_rate_hz); in TEST() 80 const size_t buffer_length = sample_rate_hz * ts::kChunkSizeMs / 1000; in TEST()
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D | click_annotate.cc | 63 int sample_rate_hz = atoi(argv[4]); in main() local 64 if (sample_rate_hz <= 0) { in main() 69 TransientDetector detector(sample_rate_hz); in main() 71 size_t audio_buffer_length = chunk_size_ms * sample_rate_hz / 1000; in main()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
D | isac_fix_type.h | 84 uint16_t sample_rate_hz) { in SetDecSampRate() argument 85 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); in SetDecSampRate() 89 uint16_t sample_rate_hz) { in SetEncSampRate() argument 90 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); in SetEncSampRate() 94 uint16_t sample_rate_hz) { in SetEncSampRateInDecoder() argument 95 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); in SetEncSampRateInDecoder()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
D | isac_float_type.h | 83 uint16_t sample_rate_hz) { in SetDecSampRate() 84 return WebRtcIsac_SetDecSampRate(inst, sample_rate_hz); in SetDecSampRate() 87 uint16_t sample_rate_hz) { in SetEncSampRate() 88 return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz); in SetEncSampRate() 91 uint16_t sample_rate_hz) { in SetEncSampRateInDecoder() 92 WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz); in SetEncSampRateInDecoder()
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/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
D | audio_decoder_opus.cc | 30 int sample_rate_hz, in DecodeInternal() argument 33 RTC_DCHECK_EQ(sample_rate_hz, 48000); in DecodeInternal() 45 int sample_rate_hz, in DecodeRedundantInternal() argument 50 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, in DecodeRedundantInternal() 54 RTC_DCHECK_EQ(sample_rate_hz, 48000); in DecodeRedundantInternal()
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/external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
D | audio_decoder_pcm.cc | 25 int sample_rate_hz, in DecodeInternal() argument 28 RTC_DCHECK_EQ(sample_rate_hz, 8000); in DecodeInternal() 49 int sample_rate_hz, in DecodeInternal() argument 52 RTC_DCHECK_EQ(sample_rate_hz, 8000); in DecodeInternal()
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D | audio_encoder_pcm.cc | 38 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz) in AudioEncoderPcm() argument 39 : sample_rate_hz_(sample_rate_hz), in AudioEncoderPcm() 45 config.num_channels * config.frame_size_ms * sample_rate_hz / 1000), in AudioEncoderPcm() 47 RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz"; in AudioEncoderPcm()
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/external/webrtc/webrtc/modules/audio_processing/vad/ |
D | voice_activity_detector.cc | 39 int sample_rate_hz) { in ProcessChunk() argument 40 RTC_DCHECK_EQ(static_cast<int>(length), sample_rate_hz / 100); in ProcessChunk() 44 if (sample_rate_hz != kSampleRateHz) { in ProcessChunk() 46 resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels), in ProcessChunk()
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
D | initial_delay_manager.cc | 39 int sample_rate_hz, in UpdateLastReceivedPacket() argument 79 UpdatePlayoutTimestamp(*current_header, sample_rate_hz); in UpdateLastReceivedPacket() 93 buffered_audio_ms_ += timestamp_increase * 1000 / sample_rate_hz; in UpdateLastReceivedPacket() 96 UpdatePlayoutTimestamp(*current_header, sample_rate_hz); in UpdateLastReceivedPacket() 235 const RTPHeader& current_header, int sample_rate_hz) { in UpdatePlayoutTimestamp() argument 237 initial_delay_ms_ * sample_rate_hz / 1000); in UpdatePlayoutTimestamp()
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D | acm_receiver.cc | 183 const int sample_rate_hz = [&decoder] { in InsertPacket() local 187 receive_timestamp = NowInTimestamp(sample_rate_hz); in InsertPacket() 198 last_packet_sample_rate_hz_ = rtc::Optional<int>(decoder->sample_rate_hz); in InsertPacket() 305 int sample_rate_hz, in AddCodec() argument 329 decoder.sample_rate_hz == sample_rate_hz) { in AddCodec() 349 audio_decoder, neteq_decoder, name, payload_type, sample_rate_hz); in AddCodec() 362 decoder.sample_rate_hz = sample_rate_hz; in AddCodec() 442 codec->plfreq = last_audio_decoder_->sample_rate_hz; in LastAudioCodec() 483 codec->plfreq = decoder.sample_rate_hz; in DecoderByPayloadType()
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/external/webrtc/webrtc/modules/audio_processing/ |
D | noise_suppression_impl.cc | 33 explicit Suppressor(int sample_rate_hz) { in Suppressor() argument 36 int error = NS_INIT(state_, sample_rate_hz); in Suppressor() 55 void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) { in Initialize() argument 58 sample_rate_hz_ = sample_rate_hz; in Initialize() 63 new_suppressors[i].reset(new Suppressor(sample_rate_hz)); in Initialize()
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D | high_pass_filter_impl.cc | 25 explicit BiquadFilter(int sample_rate_hz) : in BiquadFilter() argument 26 ba_(sample_rate_hz == AudioProcessing::kSampleRate8kHz ? in BiquadFilter() 95 void HighPassFilterImpl::Initialize(size_t channels, int sample_rate_hz) { in Initialize() argument 98 new_filters[i].reset(new BiquadFilter(sample_rate_hz)); in Initialize()
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | accelerate.h | 32 Accelerate(int sample_rate_hz, size_t num_channels, in Accelerate() argument 34 : TimeStretch(sample_rate_hz, num_channels, background_noise) { in Accelerate() 75 virtual Accelerate* Create(int sample_rate_hz,
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D | dtmf_buffer_unittest.cc | 30 static int sample_rate_hz = 8000; variable 57 DtmfBuffer* buffer = new DtmfBuffer(sample_rate_hz); in TEST() 91 DtmfBuffer buffer(sample_rate_hz); in TEST() 126 DtmfBuffer buffer(sample_rate_hz); in TEST() 152 DtmfBuffer buffer(sample_rate_hz); in TEST() 196 DtmfBuffer buffer(sample_rate_hz); in TEST() 239 DtmfBuffer buffer(sample_rate_hz); in TEST() 273 DtmfBuffer buffer(sample_rate_hz); in TEST()
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D | preemptive_expand.h | 32 PreemptiveExpand(int sample_rate_hz, in PreemptiveExpand() argument 36 : TimeStretch(sample_rate_hz, num_channels, background_noise), in PreemptiveExpand() 81 int sample_rate_hz,
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/external/webrtc/webrtc/voice_engine/ |
D | utility_unittest.cc | 48 void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) { in SetMonoFrame() argument 51 frame->sample_rate_hz_ = sample_rate_hz; in SetMonoFrame() 52 frame->samples_per_channel_ = sample_rate_hz / 100; in SetMonoFrame() 66 int sample_rate_hz) { in SetStereoFrame() argument 69 frame->sample_rate_hz_ = sample_rate_hz; in SetStereoFrame() 70 frame->samples_per_channel_ = sample_rate_hz / 100; in SetStereoFrame()
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/external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
D | audio_decoder_g722.cc | 35 int sample_rate_hz, in DecodeInternal() argument 38 RTC_DCHECK_EQ(sample_rate_hz, 16000); in DecodeInternal() 74 int sample_rate_hz, in DecodeInternal() argument 77 RTC_DCHECK_EQ(sample_rate_hz, 16000); in DecodeInternal()
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