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Searched refs:sample_rate_hz (Results 1 – 25 of 125) sorted by relevance

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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/
Dunittest.cc80 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 …]
Daudio_decoder_isac_t_impl.h44 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()
Daudio_encoder_isac_t_impl.h27 config.sample_rate_hz = codec_inst.plfreq; in CreateIsacConfig()
29 rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz); in CreateIsacConfig()
44 switch (sample_rate_hz) { in IsOk()
165 RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz)); in RecreateEncoderInstance()
183 RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz)); in RecreateEncoderInstance()
/external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/
Daudio_decoder_pcm16b.cc31 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()
Daudio_encoder_pcm16b.cc33 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()
Daudio_encoder_pcm16b.h25 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()
/external/webrtc/webrtc/modules/audio_coding/codecs/
Daudio_decoder.cc21 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()
Daudio_decoder.h46 int sample_rate_hz,
55 int sample_rate_hz,
108 int sample_rate_hz,
114 int sample_rate_hz,
/external/webrtc/webrtc/modules/audio_processing/transient/
Dtransient_detector.cc30 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()
Dtransient_detector_unittest.cc45 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()
Dclick_annotate.cc63 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()
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/
Disac_fix_type.h84 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()
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/
Disac_float_type.h83 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()
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/
Daudio_decoder_opus.cc30 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()
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/
Daudio_decoder_pcm.cc25 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()
Daudio_encoder_pcm.cc38 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()
/external/webrtc/webrtc/modules/audio_processing/vad/
Dvoice_activity_detector.cc39 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()
/external/webrtc/webrtc/modules/audio_coding/acm2/
Dacm_receiver.cc183 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()
Dinitial_delay_manager.cc39 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()
/external/webrtc/webrtc/modules/audio_processing/
Dnoise_suppression_impl.cc33 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()
Daudio_processing_impl.cc413 if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) { in InitializeLocked()
437 std::min(formats_.api_format.input_stream().sample_rate_hz(), in InitializeLocked()
438 formats_.api_format.output_stream().sample_rate_hz()); in InitializeLocked()
456 if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) { in InitializeLocked()
460 if (formats_.api_format.reverse_input_stream().sample_rate_hz() == in InitializeLocked()
472 if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate32kHz || in InitializeLocked()
473 capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate48kHz) { in InitializeLocked()
477 capture_nonlocked_.fwd_proc_format.sample_rate_hz(); in InitializeLocked()
514 return formats_.api_format.input_stream().sample_rate_hz(); in input_sample_rate_hz()
519 return capture_nonlocked_.fwd_proc_format.sample_rate_hz(); in proc_sample_rate_hz()
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Dhigh_pass_filter_impl.cc25 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()
/external/webrtc/webrtc/modules/audio_coding/neteq/
Ddtmf_buffer_unittest.cc30 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()
Daccelerate.h32 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,
/external/webrtc/webrtc/voice_engine/
Dutility_unittest.cc48 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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