[Top][All Lists]

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

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

From: Felix Wunsch
Subject: Re: [Discuss-gnuradio] Audio sink does not throttle sample flow
Date: Mon, 13 Aug 2012 11:05:59 +0200
User-agent: Mozilla/5.0 (X11; Linux x86_64; rv:14.0) Gecko/20120714 Thunderbird/14.0

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

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?


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,

reply via email to

[Prev in Thread] Current Thread [Next in Thread]