Not working audio recode in SABRA-SD

[nkkang]

Hello ...

i got a problem about audio recode that is not recoding anything in SABRA board

anyone help me this problem?

[test environments]

    board : SABRA-SDP board x 2

    root_img : oneric Ubuntu  / LTIB both

    version : L3.0.35_4.1.0_130816 / L3.0.35_4.0.0_130816 both version

    MIC input : from PC audio output

[test command]

$ arecord -d 5 test-mic.wav

$ aplay test-mic.wav

not play anything

or

$ gst-launch -v --gst-debug=*:2 alsasrc num-buffers=240 blocksize=44100 ! mfw_mp3encoder ! filesink location=output.mp3

--> log

0:00:00.047024000  7831    0x17050 WARN            GST_REGISTRY gstregistry.c:1178:gst_registry_scan_path_level:<registry0> ignoring old plugin /usr/lib/gstreamer-0.10/libgstselector.so which has been merged into the corelements plugin
0:00:00.048362000  7831    0x17050 WARN            GST_REGISTRY gstregistry.c:1178:gst_registry_scan_path_level:<registry0> ignoring old plugin /usr/lib/gstreamer-0.10/libgstvalve.so which has been merged into the corelements plugin
BLN_MAD-MMCODECS_MP3E_ARM_02.02.00_ARM12  build on Jan 18 2013 15:29:26.
MFW_GST_MP3_ENCODER_PLUGIN 3.0.7 build on Jun 26 2013 19:20:25.
Setting pipeline to PAUSED ...
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 197369
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 11609
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
/GstPipeline:pipeline0/MfwGstMp3EncInfo:mfwgstmp3encinfo0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
/GstPipeline:pipeline0/MfwGstMp3EncInfo:mfwgstmp3encinfo0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
/GstPipeline:pipeline0/MfwGstMp3EncInfo:mfwgstmp3encinfo0.GstPad:src: caps = audio/mpeg, mpegversion=(int)1, layer=(int)3, rate=(int)44100, channels=(int)2
/GstPipeline:pipeline0/MfwGstMp3EncInfo:mfwgstmp3encinfo0.GstPad:src: caps = audio/mpeg, mpegversion=(int)1, layer=(int)3, rate=(int)44100, channels=(int)2
/GstPipeline:pipeline0/GstFileSink:filesink0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1, layer=(int)3, rate=(int)44100, channels=(int)2

$ gst-launch alsasrc num-buffers=240 blocksize=44100 ! mfw_mp3encoder ! filesink location=output.mp3

Not play anything

or

$ gst-launch -v --gst-debug=*:2, alsasrc ! alsasink

   -> log output

0:00:00.048177667  6457    0x17050 WARN            GST_REGISTRY gstregistry.c:1178:gst_registry_scan_path_level:<registry0> ignoring old plugin /usr/lib/gstreamer-0.10/libgstselector.so which has been merged into the corelements plugin
0:00:00.049521667  6457    0x17050 WARN            GST_REGISTRY gstregistry.c:1178:gst_registry_scan_path_level:<registry0> ignoring old plugin /usr/lib/gstreamer-0.10/libgstvalve.so which has been merged into the corelements plugin
Setting pipeline to PAUSED ...
0:00:00.232022333  6457    0x17050 WARN                    alsa gstalsa.c:124:gst_alsa_detect_formats:<alsasink0> skipping non-int format
0:00:00.238361334  6457    0x17050 WARN                    alsa conf.c:4630:snd_config_expand: alsalib error: Unknown parameters {AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
0:00:00.238849667  6457    0x17050 WARN                    alsa pcm.c:2212:snd_pcm_open_noupdate: alsalib error: Unknown PCM iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 197369
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 11609
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
0:00:00.261248000  6457    0x17050 WARN                     bin gstbin.c:2384:gst_bin_do_latency_func:<pipeline0> failed to query latency
New clock: GstAudioSrcClock
/GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2
0:00:00.312843667  6457    0xd2f88 WARN           baseaudiosink gstbaseaudiosink.c:1365:gst_base_audio_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of -0:00:00.046439909, resyncing
0:00:00.359269667  6457    0xd2f88 WARN           baseaudiosink gstbaseaudiosink.c:1365:gst_base_audio_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of -0:00:00.046439909, resyncing
0:00:00.405803333  6457    0xd2f88 WARN           baseaudiosink gstbaseaudiosink.c:1365:gst_base_audio_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of -0:00:00.046439909, resyncing
0:00:00.452161333  6457    0xd2f88 WARN           baseaudiosink gstbaseaudiosink.c:1365:gst_base_audio_sink_get_alignment:<alsasink0> Unexpected discontinuity in audio timestamps of -0:00:00.046439909, resyncing

[Yuri · Accepted solution]

Please try to use alsamixer to configure capture / record parameters, such as volume