iMX6ULL & iMX7D Audio Streaming Issue

[subbaiahnv]

we are capturing microphone audio with iMX6ULL based board and sending to iMX7D based board over RTP protocol using g-streamer pipeline. Both the boards are using same codec.

We are seeing the packet loss at receving side iMX7D side. Whereas iMX7D capturing mic data and sending to iMX6ULL over RTP working fine.

If we record the mic data in iMX6ULL and copy the data to iMX7D then we are not seeing any issue. With this We are confident that there are no issues with the codec. Also the same codec working fine with ATMEL SAMA5D27 series processor.

We are facing the issue with NXP devices. What could be the issue. Kindly support.

Sender

root@imx6ull:~# gst-launch-1.0 -v alsasrc device=hw:0,0 ! audioconvert ! audioresample ! audio/x-raw, format=S16LE, channels=1, rate=44100 ! tcpclientsink host=192.168.0.134 port=5001

Setting pipeline to PAUSED ...

Pipeline is live and does not need PREROLL ...

Setting pipeline to PLAYING ...

New clock: GstAudioSrcClock

/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000

/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000

Redistribute latency...

/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = audio/x-raw, rate=(int)44100, format=(string)S16LE, channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003

/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw, rate=(int)44100, format=(string)S16LE, channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003

/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:src: caps = audio/x-raw, rate=(int)44100, format=(string)S16LE, channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003

/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x-raw, rate=(int)44100, format=(string)S16LE, channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003

/GstPipeline:pipeline0/GstTCPClientSink:tcpclientsink0.GstPad:sink: caps = audio/x-raw, rate=(int)44100, format=(string)S16LE, channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003

/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = audio/x-raw, rate=(int)44100, format=(string)S16LE, channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003

/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:sink: caps = audio/x-raw, rate=(int)44100, format=(string)S16LE, channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003

/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw, rate=(int)44100, format=(string)S16LE, channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003

WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough

Additional debug info:

../git/gst-libs/gst/audio/gstaudiobasesrc.c(841): gst_audio_base_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:

Dropped 28665 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.

WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough

Additional debug info:

../git/gst-libs/gst/audio/gstaudiobasesrc.c(841): gst_audio_base_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:

Dropped 22050 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.

WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough

Additional debug info:

../git/gst-libs/gst/audio/gstaudiobasesrc.c(841): gst_audio_base_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:

Dropped 51156 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.

WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough

Additional debug info:

../git/gst-libs/gst/audio/gstaudiobasesrc.c(841): gst_audio_base_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:

Dropped 48951 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.

WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough

[subbaiahnv]

Hi JorgeCas,

It is coming frequently for every 3 or 4 seconds capturing. There will be 50 msec data loss in the plying of IMX7D. 

[JorgeCas]

Hello,

How often does this warning appear?

All the time? At the beginning?

Best regards.