Resampling audio from 88200 to 96000 samples per second
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I have an audio file recorded with a sample rate
of 88200 that I want to play on a USB DAC that supports 96000 but
not 88200.
The following pipeline plays the file but re-samples to 48000: % gst-launch-0.10 filesrc location=Song.flac ! decodebin ! alsasink, device=hw:0,0 -v /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:src0: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)24, rate=(int)88200, channels=(int)2I confirmed the sampling rate with 'cat /proc/asound/card0/stream0': Playback: This mailing list post suggests something like this: % gst-launch-0.10 filesrc location=Song.flac ! decodebin ! audioresample ! 'audio/x-raw-int, rate=96000' ! alsasink, device=hw:0,0 -v However, this does not produce sound and provides the following message: /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:src0: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)24, rate=(int)88200, channels=(int)2 I'm sure the NULL caps is not a good sign. Can someone help me understand what I'm doing wrong? Thanks in advance, n _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Re: Resampling audio from 88200 to 96000 samples per second
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On Fri, 2012-11-23 at 08:17 -0800, Nick Okasinski wrote:
> This mailing list post suggests something like this: > > % gst-launch-0.10 filesrc location=Song.flac ! decodebin ! > audioresample ! 'audio/x-raw-int, rate=96000' ! alsasink, > device=hw:0,0 -v > > However, this does not produce sound and provides the following > message: > > > /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:src0: > > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > width=(int)32, depth=(int)24, rate=(int)88200, channels=(int)2 > > /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstFlacDec:flacdec0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)24, rate=(int)88200, channels=(int)2 > > ERROR: from > > element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstFlacParse:flacparse0: GStreamer encountered a general stream error. > > Additional debug info: > > gstbaseparse.c(2890): gst_base_parse_loop > > (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstFlacParse:flacparse0: > > streaming stopped, reason not-negotiated > > ERROR: pipeline doesn't want to preroll. > > (snip) > > I'm sure the NULL caps is not a good sign. Can someone help me > understand what I'm doing wrong? The NULL caps are normal when shutting down the pipeline. Try: ... ! decodebin2 ! audioconvert ! audioresample ! alsasink ... Cheers -Tim _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Tim, thanks for your astonishingly fast response. I've tried the following: % gst-launch-0.10 filesrc location=Song.flac ! decodebin2 ! audioconvert ! audioresample ! alsasink, device=hw:0,0 -v It produces sound but still at 48000 Hz rather than the desired 96000 Hz. One thing that puzzles me is how the pipeline knows that the output should be 96000 samples/sec. I tried many things: (1)% gst-launch-0.10 filesrc location=Song.flac ! decodebin2 ! audioconvert ! audioresample ! 'alsasink, device=rate96000' -v ... where: /etc/asound.conf contains "pcm.rate96000 { type plug slave { pcm "hw:0,0" rate 96000 }}" The idea being that since 'alsasink' doesn't have a "rate" option, to do it indirectly in the alsa config. (2)% gst-launch-0.10 filesrc location=Song.flac ! decodebin2 ! audioconvert ! audioresample ! 'alsasink, slave-method=0, device=rate96000' -v ... where "slave-method=0" signifies "resample". I've tried dozens of variations with arguments to decodebin and decodebin2; inserting audio/x-raw elements with rate=96000... all to no avail. Here is some more information about the capabilities of the DAC I'm trying to drive. My goal is to see it using "Altset 3" at "Momentary freq" of 96000. > GFEC ASSP DigiHug USB Audio at usb-0000:00:13.0-1, full speed : USB Audio > > Playback: > Status: Running > Interface = 3 > Altset = 1 > URBs = 3 [ 7 7 8 ] > Packet Size = 388 > Momentary freq = 48000 Hz (0x30.0000) > Interface 3 > Altset 1 > Format: S16_LE > Channels: 2 > Endpoint: 3 OUT (ADAPTIVE) > Rates: 8000, 16000, 32000, 44100, 48000, 96000 > Interface 3 > Altset 2 > Format: S24_3LE > Channels: 2 > Endpoint: 3 OUT (ADAPTIVE) > Rates: 8000, 16000, 32000, 44100, 48000, 96000 > Thanks again for your help. n. |
Re: Resampling audio from 88200 to 96000 samples per second
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On Fri, 2012-11-23 at 11:02 -0800, Nick wrote:
> I've tried the following: > > % gst-launch-0.10 filesrc location=Song.flac ! decodebin2 ! audioconvert ! > audioresample ! alsasink, device=hw:0,0 -v > > It produces sound but still at 48000 Hz rather than the desired 96000 Hz. > > One thing that puzzles me is how the pipeline knows that the output should > be 96000 samples/sec. I tried many things: Sorry, what I actually meant was: Try: ... ! decodebin2 ! audioconvert ! audioresample ! audio/x-raw-int,rate=96000 ! alsasink ... Cheers -Tim _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Yes, the following command: % gst-launch-0.10 filesrc location=Song.flac .flac ! decodebin2 ! audioconvert ! audioresample,quality=10 ! audio/x-raw-int,rate=96000 ! alsasink correctly selects "Altset 2" at 96000 Hz. This resolves my issue. Thanks!! PS: (In case someone in the future is having a similar issue), be advised that I had to add the following to my /etc/asound.conf file: """ pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } """ ... and then restart ALSA. I had no success with the ... ! alsasink,device="..." syntax. |
Re: Resampling audio from 88200 to 96000 samples per second
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On Fri, 2012-11-23 at 12:33 -0800, Nick wrote:
> I had no success with the ... ! alsasink,device="..." syntax. That's possibly because of the comma. It should be: ... ! alsasink device=hw:1,0 foo=bar et=cetera Cheers -Tim _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Hi does anybody know if gstreamer can be use for music aggregation? From: Tim-Philipp Müller <[hidden email]> To: [hidden email] Sent: Friday, November 23, 2012 1:44 PM Subject: Re: Resampling audio from 88200 to 96000 samples per second On Fri, 2012-11-23 at 12:33 -0800, Nick wrote: > I had no success with the ... ! alsasink,device="..." syntax. That's possibly because of the comma. It should be: ... ! alsasink device=hw:1,0 foo=bar et=cetera Cheers -Tim _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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