mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-12-23 01:55:48 -05:00
Copy audio to UDP/RTP: Opus environment setting
This commit is contained in:
f4exb
parent
ef27776a48
commit
c1fb4bb46c
2
debian/control
vendored
2
debian/control
vendored
@ -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.
|
||||
|
||||
@ -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:
|
||||
|
||||
@ -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();
|
||||
|
||||
@ -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();
|
||||
|
||||
@ -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;
|
||||
|
||||
@ -45,7 +45,8 @@ public:
|
||||
PayloadL8,
|
||||
PayloadPCMA8,
|
||||
PayloadPCMU8,
|
||||
PayloadG722
|
||||
PayloadG722,
|
||||
PayloadOpus
|
||||
} PayloadType;
|
||||
|
||||
RTPSink(QUdpSocket *udpSocket, int sampleRate, bool stereo);
|
||||
|
||||
@ -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
|
||||
```
|
||||
|
||||
☞ 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.
|
||||
|
||||
☞ 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.
|
||||
|
||||
@ -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";
|
||||
|
||||
@ -260,6 +260,11 @@
|
||||
<string>G722</string>
|
||||
</property>
|
||||
</item>
|
||||
<item>
|
||||
<property name="text">
|
||||
<string>Opus</string>
|
||||
</property>
|
||||
</item>
|
||||
</widget>
|
||||
</item>
|
||||
<item>
|
||||
|
||||
@ -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)
|
||||
|
||||
Loading…
Reference in New Issue
Block a user