I want to implement multicast streaming in embedded-linux (Yocto) system.
I thought that Gstreamer is easy to implement it, but the received data is choppy and like as if it passes low-pass filter when the filesrc is mp3. When the filesrc is wav, the recived data is like as if it passes choppy and high-pass filter.
Here is the gst-launch command (mp3).
Tx:
GST_DEBUG=3 gst-launch-1.0 filesrc location="background.mp3" ! decodebin ! \
audioconvert ! rtpL16pay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004
Rx:
GST_DEBUG=3 gst-launch-1.0 udpsrc multicast-group=239.0.0.1 auto-multicast=true port=5004 \
caps="application/x-rtp, media=audio, clock-rate=44100, payload=0" ! rtpL16depay !\
audioconvert ! alsasink
GST_DEBUG3 result is as follows:
Tx:
Setting pipeline to PAUSED ...
0:00:00.115165875 936 0x7b8c40 WARN basesrc gstbasesrc.c:3486:gst_base_src_start_complete:<filesrc0> pad not activated yet
Pipeline is PREROLLING ...
====== BEEP: 4.1.4 build on Feb 14 2017 13:39:18. ======
Core: MP3 decoder Wrapper build on Mar 21 2014 15:04:50
file: /usr/lib/imx-mm/audio-codec/wrap/lib_mp3d_wrap_arm12_elinux.so.3
CODEC: BLN_MAD-MMCODECS_MP3D_ARM_02.13.00_CORTEX-A8 build on Jul 12 2016 13:15:30.
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Rx:
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
0:00:00.269585916 1232 0x772320 WARN alsa conf.c:4974:snd_config_expand: alsalib error: Unknown parameters {AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
0:00:00.269914500 1232 0x772320 WARN alsa pcm.c:2495:snd_pcm_open_noupdate: alsalib error: Unknown PCM default:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
0:00:00.283770666 1232 0x772320 WARN alsa pcm_hw.c:1250:snd_pcm_hw_get_chmap: alsalib error: Cannot read Channel Map ctl
: No such file or directory
Redistribute latency...
0:00:06.335845459 1232 0x772320 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of -0:00:00.120430839, resyncing
0:00:07.167036751 1232 0x772320 WARN audiobasesink gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<alsasink0> correct clock skew -0:00:00.020498109 < -+0:00:00.020000000
0:00:07.178596167 1232 0x772320 WARN audiobasesink gstaudiobasesink.c:1484:gst_audio_base_sink_skew_slaving:<alsasink0> correct clock skew +0:00:00.020102330 > +0:00:00.020000000
0:00:08.215633667 1232 0x772320 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of -0:00:00.128480725, resyncing
0:00:08.962452751 1232 0x772320 WARN audiobasesink gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<alsasink0> correct clock skew -0:00:00.020283552 < -+0:00:00.020000000
0:00:09.095737543 1232 0x772320 WARN audiobasesink gstaudiobasesink.c:1484:gst_audio_base_sink_skew_slaving:<alsasink0> correct clock skew +0:00:00.020221135 > +0:00:00.020000000
0:00:10.135542001 1232 0x772320 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of -0:00:00.125238095, resyncing
Here is the gst-command (wav)
Tx:
GST_DEBUG=3 gst-launch-1.0 filesrc location="background.wav" ! decodebin ! \
audioconvert ! rtpL16pay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004
Rx:
GST_DEBUG=3 gst-launch-1.0 udpsrc multicast-group=239.0.0.1 auto-multicast=true port=5004 \
caps="application/x-rtp, media=audio, clock-rate=44100, payload=0" ! rtpL16depay !\
audioconvert ! alsasink
GST_DEBUG3 result is as follows:
Tx:
Setting pipeline to PAUSED ...
0:00:00.116759125 958 0x1c0cc40 WARN basesrc gstbasesrc.c:3486:gst_base_src_start_complete:<filesrc0> pad not activated yet
Pipeline is PREROLLING ...
0:00:00.136465125 958 0x1c1f460 FIXME default gstutils.c:3764:gst_pad_create_stream_id_internal:<wavparse0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:00:00.137230750 958 0x1c1f460 WARN riff riff-read.c:794:gst_riff_parse_info:<wavparse0> Unknown INFO (metadata) tag entry IPRT
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
0:00:00.152916625 958 0x1c0cc40 WARN bin gstbin.c:2597:gst_bin_do_latency_func:<pipeline0> did not really configure latency of 0:00:00.000000000
New clock: GstSystemClock
^Chandling interrupt.
Interrupt: Stopping pipeline ...
Execution ended after 0:00:03.435631250
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
Rx:
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
0:00:00.270927792 1238 0x120d320 WARN alsa conf.c:4974:snd_config_expand: alsalib error: Unknown parameters {AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
0:00:00.271261625 1238 0x120d320 WARN alsa pcm.c:2495:snd_pcm_open_noupdate: alsalib error: Unknown PCM default:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
0:00:00.284991583 1238 0x120d320 WARN alsa pcm_hw.c:1250:snd_pcm_hw_get_chmap: alsalib error: Cannot read Channel Map ctl
: No such file or directory
Redistribute latency...
0:00:04.227007167 1238 0x120d320 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of +0:00:00.053514739, resyncing
0:00:04.314387751 1238 0x120d320 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of +0:00:00.055510204, resyncing
0:00:04.396900334 1238 0x120d320 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of +0:00:00.052607709, resyncing
0:00:04.483605876 1238 0x120d320 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of +0:00:00.055215419, resyncing
0:00:04.570297626 1238 0x120d320 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of +0:00:00.055215419, resyncing
If I use pulsesink instead of alsasink, following is appeared.
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Redistribute latency...
0:00:00.410499500 1255 0x70813120 WARN pulse pulsesink.c:702:gst_pulsering_stream_underflow_cb:<pulsesink0> Got underflow
0:00:00.423478917 1255 0x7e7920 WARN audiobasesink gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<pulsesink0> Unexpected discontinuity in audio timestamps of +0:00:00.038095238, resyncing
0:00:00.450453459 1255 0x70813120 WARN pulse pulsesink.c:702:gst_pulsering_stream_underflow_cb:<pulsesink0> Got underflow
What is the problem ? Can anybody solve this ? I hope your kind reply.
Thank you for reading.
I reckon that the problem is that the application/x-rtp parameters don't match between the transmitter and receiver.
This can be easily solved putting verbose in the transmitter and then using the same parameters in the receiver:
Let's see in an example:
TX:
gst-launch-1.0 -v filesrc location="test.mp3" ! decodebin ! audioconvert ! rtpL16pay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004
Which last lines (thanks to -v) are next:
/GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)1806894235\,\ timestamp-offset\=(uint)468998694\,\ seqnum-offset\=(uint)20785" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink: caps = "audio/x-raw\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ format\=(string)S16BE\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = "audio/x-raw\,\ format\=(string)S32LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstDecodePad:src_0.GstProxyPad:proxypad1: caps = "audio/x-raw\,\ format\=(string)S32LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: timestamp = 468998694 /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: seqnum = 20785 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock
Using the same parameters in the player or receiver: RX:
gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)2, channels=(int)2, payload=(int)96' ! rtpL16depay ! pulsesink
And this plays perfectly.
Going to the .wav file in my case the transmitter is:
gst-launch-1.0 -v filesrc location="test.wav" ! decodebin ! audioconvert ! rtpL16pay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004
Last lines output:
/GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)1\,\ channels\=(int)1\,\ payload\=(int)96\,\ ssrc\=(uint)620824608\,\ timestamp-offset\=(uint)433377669\,\ seqnum-offset\=(uint)7103" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink: caps = "audio/x-raw\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ format\=(string)S16BE\,\ channels\=(int)1" /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ channels\=(int)1\,\ rate\=(int)44100" /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstDecodePad:src_0.GstProxyPad:proxypad1: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ channels\=(int)1\,\ rate\=(int)44100" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: timestamp = 433377669 /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: seqnum = 7103 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock
Using this information in the receiver:
gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
The audio also plays smoothly.
Hope this helps.