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Re: [Discuss-gnuradio] Audio sink does not throttle sample flow

From: Tom Rondeau
Subject: Re: [Discuss-gnuradio] Audio sink does not throttle sample flow
Date: Mon, 13 Aug 2012 07:33:39 -0400

On Mon, Aug 13, 2012 at 5:05 AM, Felix Wunsch
<address@hidden> wrote:
> Am 12.08.2012 19:19, schrieb Tom Rondeau:
>> On Tue, Aug 7, 2012 at 6:22 AM, Felix Wunsch
>> <address@hidden> wrote:
>>> Hi all,
>>> I recently wrote a block for decoding DRM AAC streams. For testing I put
>>> together a small flow graph consisting of a wav source, encoder block,
>>> decoder block, (rational) resampler and an audio sink (an image of the
>>> flow
>>> graph is attached).
>>> When I now run this flow graph, the audio is correctly decoded, but the
>>> output is not throttled to 44.1 kHz as it should be. It seems to be
>>> running
>>> at full speed. I also connected a resampler and an audio sink directly to
>>> the wav source and there the output is correct.
>>> I use default values for the audio sink (alsa, 44.1 kHz), GNU Radio 3.5.1
>>> and xubuntu 11.10.
>>> Any hints why this is happening and how to solve this?
>>> Best regards,
>>> Felix Wunsch
>> Felix,
>> I know that the audio sink will throttle the flow graph. What evidence
>> do you have that it's not or that it's running at full speed? Is it
>> the sound coming from the audio? My first guess is that there's a
>> misunderstanding somewhere of the actual sample rate. You're
>> resampling by almost 2x, which means you expect the signal coming from
>> the decoder to be 44.1e3/2. Is that right?
>> Tom
> Hi Tom,
> thanks for your reply.
> My first evidence was in fact the sound coming from the audio that is
> running at a very high speed. However, the pitch seems to be normal. The
> flow graph is set to "run to completion" and processes a 3 min wav-file in
> about 15 sec.
> The signal coming from the decoder has a sample rate of 24 kHz. I verified
> that by writing the decoder output into a file and using aplay for playback
> with -r 24000. At this point, the sound is still normal.
> The next steps are in detail:
> - Type conversion short->float
> - multiply const (1/32768) for range adjustment
> - rational resampler( interp: 441, decim: 240)
> - audio sink (44.1kHz)
> Did I configure the resampler correctly? I left taps blank and fractional
> bandwidth at 0.
> I also attached a file sink to the output of the resampler and tried to play
> it with aplay using -r 44100. It shows the same behaviour like the audio
> sink (normal pitch, very fast playback).
> Best regards,
> Felix


That looks like you're doing everything correctly. I wonder if it's an
issue with the sound card, seeing as you're getting the same behavior
with aplay. Can you set the output device? If you can use pulseaudio,
set the device string to 'pulse' and see what that does. Otherwise,
try 'plughw:0,0'. They might handle the sampling rate settings better.


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