/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtcp_sender.h | 57 uint32_t frequency_hz; member in struct:webrtc::RTCPSender::FeedbackState
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H A D | rtp_sender.cc | 1403 uint32_t frequency_hz = SendPayloadFrequency(); local 1404 uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz);
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/external/chromium_org/third_party/webrtc/voice_engine/include/ |
H A D | voe_codec.h | 130 // rate the receiver will render: |frequency_hz| (in Hz). 133 virtual int SetOpusMaxPlaybackRate(int channel, int frequency_hz) { argument
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/external/chromium_org/remoting/codec/ |
H A D | audio_encoder_opus_unittest.cc | 50 // Return test signal value at the specified position |pos|. |frequency_hz| 56 double frequency_hz, 59 double angle = pos * 2 * M_PI * frequency_hz / rate + 65 // |frequency_hz|. 69 double frequency_hz, 73 data[i * kChannels] = GetSampleValue(rate, frequency_hz, i + pos, 0); 74 data[i * kChannels + 1] = GetSampleValue(rate, frequency_hz, i + pos, 1); 108 double frequency_hz, 115 GetSampleValue(rate, frequency_hz, i - shift, 0); 118 GetSampleValue(rate, frequency_hz, 54 GetSampleValue( AudioPacket::SamplingRate rate, double frequency_hz, double pos, int channel) argument 66 CreatePacket( int samples, AudioPacket::SamplingRate rate, double frequency_hz, int pos) argument 106 ValidateReceivedData(int samples, AudioPacket::SamplingRate rate, double frequency_hz, const std::vector<int16>& received_data) argument 127 TestEncodeDecode(int packet_size, double frequency_hz, AudioPacket::SamplingRate rate) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | PCMFile.cc | 50 uint16_t* frequency_hz) { 86 *frequency_hz); 90 *frequency_hz = tmp_frequency; 49 ChooseFile(std::string* file_name, int16_t max_len, uint16_t* frequency_hz) argument
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H A D | TestStereo.cc | 141 uint16_t frequency_hz; local 153 frequency_hz = 32000; 156 in_file_stereo_->Open(file_name_stereo, frequency_hz, "rb"); 158 in_file_mono_->Open(file_name_mono, frequency_hz, "rb");
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | voe_codec_impl.cc | 421 int VoECodecImpl::SetOpusMaxPlaybackRate(int channel, int frequency_hz) { argument 423 "SetOpusMaxPlaybackRate(channel=%d, frequency_hz=%d)", channel, 424 frequency_hz); 436 return channelPtr->SetOpusMaxPlaybackRate(frequency_hz);
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H A D | channel.cc | 1699 int Channel::SetOpusMaxPlaybackRate(int frequency_hz) { argument 1703 if (audio_coding_->SetOpusMaxPlaybackRate(frequency_hz) != 0) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/ |
H A D | opus_interface.c | 102 int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz) { argument 108 if (frequency_hz <= 8000) { 110 } else if (frequency_hz <= 12000) { 112 } else if (frequency_hz <= 16000) { 114 } else if (frequency_hz <= 24000) {
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | fakewebrtcvoiceengine.h | 654 WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) { argument 660 if (frequency_hz <= 8000) 662 else if (frequency_hz <= 12000) 664 else if (frequency_hz <= 16000) 666 else if (frequency_hz <= 24000)
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | audio_coding_module_impl.cc | 1907 int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) { argument 1912 return codecs_[current_send_codec_idx_]->SetOpusMaxPlaybackRate(frequency_hz);
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