Copy audio to UDP/RTP: Opus environment setting

2019-02-18 02:30:43 +01:00 · 2019-02-18 02:30:43 +01:00 · c1fb4bb46c
commit c1fb4bb46c
--- a/debian/control
+++ b/debian/control
@ -10,7 +10,7 @@ Homepage: https://github.com/f4exb/sdrangel

 Package: sdrangel
 Architecture: any
-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}
+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}
 Description: SDR/Analyzer/Generator front-end for various hardware
 SDR/Analyzer/Generator front-end for Airspy, BladeRF, HackRF, RTL-SDR, FunCube, LimeSDR, PlutoSDR.
 Also File source and sink for I/Q samples, network I/Q sources with remote instance.
--- a/sdrbase/audio/audionetsink.cpp
+++ b/sdrbase/audio/audionetsink.cpp
@ -136,6 +136,9 @@ void AudioNetSink::setParameters(Codec codec, bool stereo, int sampleRate)
        case CodecG722:
            m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadG722, sampleRate/2);
            break;
+        case CodecOpus:
+            m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadOpus, sampleRate);
+            break;
        case CodecL16: // actually no codec
        default:
            m_rtpBufferAudio->setPayloadInformation(stereo ? RTPSink::PayloadL16Stereo : RTPSink::PayloadL16Mono, sampleRate);
@ -165,6 +168,7 @@ void AudioNetSink::setDecimationFilters()
    case CodecG722:
        m_audioFilter.setDecimFilters(m_sampleRate, decimatedSampleRate, 7000.0, 50.0);
        break;
+    case CodecOpus:
    case CodecL8:
    case CodecL16:
    default:
--- a/sdrbase/audio/audionetsink.h
+++ b/sdrbase/audio/audionetsink.h
@ -46,7 +46,8 @@ public:
        CodecL8,   //!< Linear 8 bit samples
        CodecPCMA, //!< PCM A-law 8 bit samples
        CodecPCMU, //!< PCM Mu-law 8 bit samples
-        CodecG722  //!< G722 compressed 8 bit samples 16kS/s in 8kS/s out
+        CodecG722, //!< G722 compressed 8 bit samples 16kS/s in 8kS/s out
+        CodecOpus  //!< Opus compressed 8 bit samples at 64kbits/s (8kS/s out). Various input sample rates
    } Codec;

    AudioNetSink(QObject *parent); //!< without RTP
@ -70,6 +71,7 @@ public:
    static const int m_udpBlockSize;
    static const int m_dataBlockSize = 65536; // room for G722 conversion (64000 = 12800*5 largest to date)
    static const int m_g722BlockSize = 12800; // number of resulting G722 bytes (80*20ms frames)
+    static const int m_opusBlockSize = 960*4; // provision for 20ms of 2 int16 channels at 48 kS/s

 protected:
    void setDecimationFilters();
--- a/sdrbase/audio/audiooutput.h
+++ b/sdrbase/audio/audiooutput.h
@ -47,7 +47,8 @@ public:
 		UDPCodecL8,    //!< Linear 8 bit
        UDPCodecALaw,  //!< PCM A-law 8 bit
        UDPCodecULaw,  //!< PCM Mu-law 8 bit
-        UDPCodecG722   //!< G722 compression
+        UDPCodecG722,  //!< G722 compression
+		UDPCodecOpus   //!< Opus compression
    };

 	AudioOutput();
--- a/sdrbase/util/rtpsink.cpp
+++ b/sdrbase/util/rtpsink.cpp
@ -110,6 +110,12 @@ void RTPSink::setPayloadInformation(PayloadType payloadType, int sampleRate)
        m_packetSamples = m_sampleRate / 50; // 20ms packet samples
        timestampinc = m_sampleRate / 50;    // 1 channel
        break;
+    case PayloadOpus:
+        m_sampleBytes = 1;
+        m_rtpSession.SetDefaultPayloadType(101);
+        m_packetSamples = 160; // Fixed 20ms @ 64 kbits/s packet samples
+        timestampinc = 160;    // 1 channel
+        break;
    case PayloadL16Mono:
    default:
        m_sampleBytes = 2;
--- a/sdrbase/util/rtpsink.h
+++ b/sdrbase/util/rtpsink.h
@ -45,7 +45,8 @@ public:
        PayloadL8,
        PayloadPCMA8,
        PayloadPCMU8,
-        PayloadG722
+        PayloadG722,
+        PayloadOpus
    } PayloadType;

    RTPSink(QUdpSocket *udpSocket, int sampleRate, bool stereo);
--- a/sdrgui/audio.md
+++ b/sdrgui/audio.md
@ -67,6 +67,7 @@ This is the codec applied before sending the stream via UDP. The following are a
  - `PCMA`: A-law 8 bit PCM (requires 8000 Hz sample rate mono)
  - `PCMU`: Mu-law 8 bit PCM (requires 8000 Hz sample rate mono)
  - `G722`: G722 64 kbit/s (requires 16000 Hz sample rate mono)
+  - `OPUS` : Opus 64 kbit/s

 <h3>1.10 SDP string</h3>

@ -90,6 +91,7 @@ Check this box to activate the RTP protocol over UDP. RTP parameters are as foll

  - Payload type:
    - codec `L16`, `L8`: 96
+    - codec `OPUS`: 101
    - codec `PCMA`: 8
    - codec `PCMU`: 0
    - codec `G722`: 9
@ -99,7 +101,7 @@ Check this box to activate the RTP protocol over UDP. RTP parameters are as foll
  - Sample format:
    - codec `L16`: 16 bit integer signed (S16LE)
    - codec `L8`, `PCMA`, `PCMU`: 8 bit integer signed (S8)
-    - codec `G722`: 8 bit unsigned integer. Note that this is the stream compressed to 64 kbits/s.
+    - codec `G722`, `OPUS`: 8 bit unsigned integer. Note that this is the stream compressed to 64 kbits/s.
  - Channels: 1 for mono (Left, Right and Mixed copy channels mode); 2 for stereo (Stereo copy channels mode)
  - Address and port: destination address and port (local on the client machine)

@ -127,6 +129,14 @@ m=audio 9998 RTP/AVP 9
 a=rtpmap:9 G722/8000/1
 ```

+For Opus mono:
+
+```
+c=IN IP4 192.168.0.34
+m=audio 9998 RTP/AVP 101
+a=rtpmap:101 opus/48000/1
+```
+
 &#9758; 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.

 &#9758; 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.
--- a/sdrgui/gui/audiodialog.cpp
+++ b/sdrgui/gui/audiodialog.cpp
@ -315,6 +315,9 @@ void AudioDialogX::updateOutputSDPString()
    case AudioOutput::UDPCodecL8:
        format = "L8";
        break;
+    case AudioOutput::UDPCodecOpus:
+        format = "opus";
+        break;
    case AudioOutput::UDPCodecL16:
    default:
        format = "L16";
--- a/sdrgui/gui/audiodialog.ui
+++ b/sdrgui/gui/audiodialog.ui
@ -260,6 +260,11 @@
             <string>G722</string>
            </property>
           </item>
+           <item>
+            <property name="text">
+             <string>Opus</string>
+            </property>
+           </item>
          </widget>
         </item>
         <item>
--- a/swagger/sdrangel/examples/rx_test.py
+++ b/swagger/sdrangel/examples/rx_test.py
@ -46,7 +46,7 @@ def getInputOptions():
    parser.add_option("--audio-address", dest="audio_address", help="Audio: UDP destination address", metavar="IP_ADDRESS", type="string")
    parser.add_option("--audio-port", dest="audio_port", help="Audio: UDP destination port", metavar="IP_PORT", type="int")
    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")
-    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")
+    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")
    parser.add_option("--audio-decim", dest="audio_decim", help="Audio. decimation to apply for UDP (1 to 6)", metavar="INT", type="int")
    parser.add_option("--baud-rate", dest="baud_rate", help="DSD: baud rate in Baud", metavar="BAUD", type="int", default=4800)
    parser.add_option("--fm-dev", dest="fm_deviation", help="DSD: expected FM deviation", metavar="FREQ", type="int", default=5400)