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Copy audio to UDP/RTP: Opus environment setting
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f4exb
parent
ef27776a48
commit
c1fb4bb46c
4
debian/control
vendored
4
debian/control
vendored
@ -10,13 +10,13 @@ Homepage: https://github.com/f4exb/sdrangel
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Package: sdrangel
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Architecture: any
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Depends: libc6, libasound2, libfftw3-single3, libgcc1, libgl1-mesa-glx, libqt5core5a, libqt5gui5, libqt5multimedia5, libqt5network5, libqt5opengl5, libqt5widgets5, libqt5multimedia5-plugins, libstdc++6, libusb-1.0-0, libopencv-dev, pulseaudio, libxml2, ffmpeg, libavcodec-dev, libavformat-dev, ${shlibs:Depends}, ${misc:Depends}
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Depends: libc6, libasound2, libfftw3-single3, libgcc1, libgl1-mesa-glx, libqt5core5a, libqt5gui5, libqt5multimedia5, libqt5network5, libqt5opengl5, libqt5widgets5, libqt5multimedia5-plugins, libstdc++6, libusb-1.0-0, libopencv-dev, pulseaudio, libxml2, ffmpeg, libavcodec-dev, libavformat-dev, libopus-dev, ${shlibs:Depends}, ${misc:Depends}
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Description: SDR/Analyzer/Generator front-end for various hardware
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SDR/Analyzer/Generator front-end for Airspy, BladeRF, HackRF, RTL-SDR, FunCube, LimeSDR, PlutoSDR.
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Also File source and sink for I/Q samples, network I/Q sources with remote instance.
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Based on Qt5 framework and OpenGL for the spectrum and scope rendering.
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Builds on Linux, Windows and Mac O/S
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Reception modes supported:
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Reception modes supported:
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Analog: AM, ATV, NFM, WFM, SSB, broadcast FM
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Digital: D-Star, Yaesu SF, DMR, dPMR, LoRa
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Analyzer: Generic channel
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@ -136,6 +136,9 @@ void AudioNetSink::setParameters(Codec codec, bool stereo, int sampleRate)
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case CodecG722:
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m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadG722, sampleRate/2);
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break;
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case CodecOpus:
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m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadOpus, sampleRate);
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break;
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case CodecL16: // actually no codec
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default:
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m_rtpBufferAudio->setPayloadInformation(stereo ? RTPSink::PayloadL16Stereo : RTPSink::PayloadL16Mono, sampleRate);
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@ -165,6 +168,7 @@ void AudioNetSink::setDecimationFilters()
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case CodecG722:
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m_audioFilter.setDecimFilters(m_sampleRate, decimatedSampleRate, 7000.0, 50.0);
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break;
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case CodecOpus:
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case CodecL8:
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case CodecL16:
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default:
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@ -46,7 +46,8 @@ public:
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CodecL8, //!< Linear 8 bit samples
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CodecPCMA, //!< PCM A-law 8 bit samples
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CodecPCMU, //!< PCM Mu-law 8 bit samples
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CodecG722 //!< G722 compressed 8 bit samples 16kS/s in 8kS/s out
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CodecG722, //!< G722 compressed 8 bit samples 16kS/s in 8kS/s out
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CodecOpus //!< Opus compressed 8 bit samples at 64kbits/s (8kS/s out). Various input sample rates
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} Codec;
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AudioNetSink(QObject *parent); //!< without RTP
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@ -70,6 +71,7 @@ public:
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static const int m_udpBlockSize;
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static const int m_dataBlockSize = 65536; // room for G722 conversion (64000 = 12800*5 largest to date)
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static const int m_g722BlockSize = 12800; // number of resulting G722 bytes (80*20ms frames)
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static const int m_opusBlockSize = 960*4; // provision for 20ms of 2 int16 channels at 48 kS/s
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protected:
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void setDecimationFilters();
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@ -47,7 +47,8 @@ public:
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UDPCodecL8, //!< Linear 8 bit
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UDPCodecALaw, //!< PCM A-law 8 bit
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UDPCodecULaw, //!< PCM Mu-law 8 bit
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UDPCodecG722 //!< G722 compression
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UDPCodecG722, //!< G722 compression
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UDPCodecOpus //!< Opus compression
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};
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AudioOutput();
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@ -110,6 +110,12 @@ void RTPSink::setPayloadInformation(PayloadType payloadType, int sampleRate)
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m_packetSamples = m_sampleRate / 50; // 20ms packet samples
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timestampinc = m_sampleRate / 50; // 1 channel
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break;
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case PayloadOpus:
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m_sampleBytes = 1;
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m_rtpSession.SetDefaultPayloadType(101);
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m_packetSamples = 160; // Fixed 20ms @ 64 kbits/s packet samples
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timestampinc = 160; // 1 channel
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break;
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case PayloadL16Mono:
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default:
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m_sampleBytes = 2;
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@ -45,7 +45,8 @@ public:
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PayloadL8,
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PayloadPCMA8,
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PayloadPCMU8,
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PayloadG722
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PayloadG722,
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PayloadOpus
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} PayloadType;
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RTPSink(QUdpSocket *udpSocket, int sampleRate, bool stereo);
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@ -67,10 +67,11 @@ This is the codec applied before sending the stream via UDP. The following are a
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- `PCMA`: A-law 8 bit PCM (requires 8000 Hz sample rate mono)
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- `PCMU`: Mu-law 8 bit PCM (requires 8000 Hz sample rate mono)
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- `G722`: G722 64 kbit/s (requires 16000 Hz sample rate mono)
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- `OPUS` : Opus 64 kbit/s
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<h3>1.10 SDP string</h3>
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This is the SDP string representatiopn of the stream sent via UDP (RTP). In SDP files it is used on the `a=rtpmap`line (See 1.14). It can be used to check the effect of settings 1.5, 1.8 and 1.9.
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This is the SDP string representatiopn of the stream sent via UDP (RTP). In SDP files it is used on the `a=rtpmap`line (See 1.14). It can be used to check the effect of settings 1.5, 1.8 and 1.9.
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<h3>1.11 UDP address</h3>
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@ -88,18 +89,19 @@ Use this button to activate or de-activate the copy of the audio stream to UDP s
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Check this box to activate the RTP protocol over UDP. RTP parameters are as follows:
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- Payload type:
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- Payload type:
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- codec `L16`, `L8`: 96
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- codec `OPUS`: 101
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- codec `PCMA`: 8
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- codec `PCMU`: 0
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- codec `G722`: 9
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- Sample rate: the resulting stream sample rate after decimation and possible compression:
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- codec `PCMA`, `PCMU`: must be 8000 S/s
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- codec `G722`: must be 8000 S/s (16000 S/s input before compression)
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- codec `G722`: must be 8000 S/s (16000 S/s input before compression)
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- Sample format:
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- codec `L16`: 16 bit integer signed (S16LE)
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- codec `L8`, `PCMA`, `PCMU`: 8 bit integer signed (S8)
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- codec `G722`: 8 bit unsigned integer. Note that this is the stream compressed to 64 kbits/s.
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- codec `G722`, `OPUS`: 8 bit unsigned integer. Note that this is the stream compressed to 64 kbits/s.
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- Channels: 1 for mono (Left, Right and Mixed copy channels mode); 2 for stereo (Stereo copy channels mode)
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- Address and port: destination address and port (local on the client machine)
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@ -127,6 +129,14 @@ m=audio 9998 RTP/AVP 9
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a=rtpmap:9 G722/8000/1
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```
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For Opus mono:
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```
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c=IN IP4 192.168.0.34
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m=audio 9998 RTP/AVP 101
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a=rtpmap:101 opus/48000/1
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```
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☞ Note that on Android clients VLC has trouble working with the RTP stream (choppy audio, hanging unexpectedly...) therefore [MX player](https://play.google.com/store/apps/details?id=com.mxtech.videoplayer.ad&hl=en) is recommended.
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☞ With PCMA and PCMU and more recently G722 codecs it is possible to listen to the RTP stream directly in the browser using a [Janus WebRTC server](https://janus.conf.meetecho.com/). Please refer to the Wiki page "Networking audio" for detailed instructions.
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@ -315,6 +315,9 @@ void AudioDialogX::updateOutputSDPString()
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case AudioOutput::UDPCodecL8:
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format = "L8";
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break;
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case AudioOutput::UDPCodecOpus:
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format = "opus";
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break;
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case AudioOutput::UDPCodecL16:
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default:
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format = "L16";
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@ -260,6 +260,11 @@
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<string>G722</string>
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</property>
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</item>
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<item>
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<property name="text">
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<string>Opus</string>
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</property>
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</item>
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</widget>
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</item>
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<item>
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@ -46,7 +46,7 @@ def getInputOptions():
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parser.add_option("--audio-address", dest="audio_address", help="Audio: UDP destination address", metavar="IP_ADDRESS", type="string")
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parser.add_option("--audio-port", dest="audio_port", help="Audio: UDP destination port", metavar="IP_PORT", type="int")
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parser.add_option("--audio-channels", dest="audio_channels", help="Audio: UDP mode (0: L only 1: R only 2: L+R mono 3: LR stereo)", metavar="ENUM_INT", type="int")
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parser.add_option("--audio-codec", dest="audio_codec", help="Audio: codec to use for UDP (0: L16, 1: L8, 2: PCMA, 3: PCMU, 4: G722)", metavar="ENUM_INT", type="int")
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parser.add_option("--audio-codec", dest="audio_codec", help="Audio: codec to use for UDP (0: L16, 1: L8, 2: PCMA, 3: PCMU, 4: G722, 5: Opus)", metavar="ENUM_INT", type="int")
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parser.add_option("--audio-decim", dest="audio_decim", help="Audio. decimation to apply for UDP (1 to 6)", metavar="INT", type="int")
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parser.add_option("--baud-rate", dest="baud_rate", help="DSD: baud rate in Baud", metavar="BAUD", type="int", default=4800)
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parser.add_option("--fm-dev", dest="fm_deviation", help="DSD: expected FM deviation", metavar="FREQ", type="int", default=5400)
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