使用 GStreamer 的音乐文件（wav、mp3、...等）的多播流：可以接收但数据是间歇的
Multicast streaming of music file (wav, mp3, ...etc) with GStreamer: can receive but the data is Intermittent
我想在嵌入式linux (Yocto) 系统中实现多播流。
我以为Gstreamer很容易实现，但是当filesrc是mp3时，接收到的数据很不稳定，好像通过了低通滤波器。
当filesrc为wav时，接收到的数据就像是通过了断断续续的高通滤波器。
这是 gst 启动命令 (mp3)。
发送：
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
接收：
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结果如下：
发送：
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
接收：
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
这是 gst 命令 (wav)
发送：
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
接收：
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结果如下：
发送：
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 ...
接收：
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
如果我使用 pulsesink 而不是 alsasink，则会出现以下内容。
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
有什么问题吗？有人可以解决这个问题吗？
希望得到您的回复。
感谢您的阅读。
我认为问题是application/x-rtp参数在发射器和接收器之间不匹配。
这可以很容易地解决，将 verbose 放入发送器中，然后在接收器中使用相同的参数：
让我们看一个例子：
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
接下来是最后几行（感谢 -v）：
/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
在播放器或接收器中使用相同的参数：
接收：
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
这玩起来很完美。
在我的例子中转到 .wav 文件 transmitter 是：
gst-launch-1.0 -v filesrc location="test.wav" ! decodebin ! audioconvert ! rtpL16pay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004
最后一行输出：
/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
在接收器中使用此信息:
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
音频播放也很流畅。
希望对您有所帮助。
我想在嵌入式linux (Yocto) 系统中实现多播流。
我以为Gstreamer很容易实现，但是当filesrc是mp3时，接收到的数据很不稳定，好像通过了低通滤波器。 当filesrc为wav时，接收到的数据就像是通过了断断续续的高通滤波器。
这是 gst 启动命令 (mp3)。
发送：
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
接收：
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结果如下：
发送：
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
接收：
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
这是 gst 命令 (wav)
发送：
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
接收：
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结果如下：
发送：
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 ...
接收：
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
如果我使用 pulsesink 而不是 alsasink，则会出现以下内容。
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
有什么问题吗？有人可以解决这个问题吗？ 希望得到您的回复。
感谢您的阅读。
我认为问题是application/x-rtp参数在发射器和接收器之间不匹配。
这可以很容易地解决，将 verbose 放入发送器中，然后在接收器中使用相同的参数：
让我们看一个例子：
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
接下来是最后几行（感谢 -v）：
/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
在播放器或接收器中使用相同的参数： 接收：
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
这玩起来很完美。
在我的例子中转到 .wav 文件 transmitter 是：
gst-launch-1.0 -v filesrc location="test.wav" ! decodebin ! audioconvert ! rtpL16pay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004
最后一行输出：
/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
在接收器中使用此信息:
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
音频播放也很流畅。
希望对您有所帮助。