| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| 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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| 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],
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| comfort_noise.h | 35 ComfortNoise(int fs_hz, DecoderDatabase* decoder_database,
37 : fs_hz_(fs_hz),
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| decision_logic_fax.h | 25 DecisionLogicFax(int fs_hz,
32 : DecisionLogic(fs_hz, output_size_samples, playout_mode,
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| normal.h | 35 Normal(int fs_hz, DecoderDatabase* decoder_database,
38 : fs_hz_(fs_hz),
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| decision_logic.cc | 26 DecisionLogic* DecisionLogic::Create(int fs_hz,
36 return new DecisionLogicNormal(fs_hz,
45 return new DecisionLogicFax(fs_hz,
58 DecisionLogic::DecisionLogic(int fs_hz,
78 SetSampleRate(fs_hz, output_size_samples);
98 void DecisionLogic::SetSampleRate(int fs_hz, int output_size_samples) {
100 assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000)
[all...] |
| 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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| dtmf_tone_generator_unittest.cc | 31 void TestAllTones(int fs_hz, int channels) {
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) {
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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| statistics_calculator.cc | 83 void StatisticsCalculator::IncreaseCounter(int num_samples, int fs_hz) {
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);
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| 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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| 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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| dtmf_buffer.h | 57 // Set up the buffer for use at sample rate |fs_hz|.
58 explicit DtmfBuffer(int fs_hz) {
59 SetSampleRate(fs_hz);
90 virtual int SetSampleRate(int fs_hz);
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| dtmf_buffer.cc | 176 int DtmfBuffer::SetSampleRate(int fs_hz) {
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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| 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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| decision_logic_normal.h | 25 DecisionLogicNormal(int fs_hz,
32 : DecisionLogic(fs_hz, output_size_samples, playout_mode,
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| 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);
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| merge.h | 36 Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer)
37 : fs_hz_(fs_hz),
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| neteq_impl.cc | 634 if (decoder_info->fs_hz != fs_hz_ ||
636 SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels());
[all...] |
| decoder_database_unittest.cc | 54 EXPECT_EQ(8000, info->fs_hz);
128 EXPECT_EQ(8000, info->fs_hz);
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| 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/ |
| mock_decoder_database.h | 33 int(uint8_t rtp_payload_type, NetEqDecoder codec_type, int fs_hz,
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