From: "Alexander E. Patrakov" <patrakov@gmail.com>
To: Petr Kulhavy <petr@barix.com>, alsa-devel@alsa-project.org
Subject: Re: How to properly handle variable samplerate?
Date: Sat, 06 Sep 2014 16:33:06 +0600 [thread overview]
Message-ID: <540AE2E2.3050000@gmail.com> (raw)
In-Reply-To: <5409E2DA.2090403@barix.com>
05.09.2014 22:20, Petr Kulhavy wrote:
> Hi,
>
> I'm developing an audio stream player application with ALSA output, the
> player has a large software buffer. The trouble is that the stream can
> potentially change the sampling rate (e.g. streaming RTP from a VLC
> playlist created from different MP3 songs). How to properly handle this?
>
> I'm thinking of the following concept:
> The codec feeds data into the software buffer together with markers for
> sample rate changes. The ALSA playback is asynchronous, using a callback
> function feeding the data from the software buffer into ALSA. If a
> samplerate marker is hit the new rate is set.
>
> Can the sample rate be set from a callback? What other constraints are
> there? And how does the HW FIFO handle samplerate changes?
> Or is there a better way of implementing?
ALSA does not support variable sample rate at all. So you either have to
do the variable-rate resampling yourself in your player, or depend on
PulseAudio which does support this feature.
--
Alexander E. Patrakov
prev parent reply other threads:[~2014-09-06 10:33 UTC|newest]
Thread overview: 2+ messages / expand[flat|nested] mbox.gz Atom feed top
2014-09-05 16:20 How to properly handle variable samplerate? Petr Kulhavy
2014-09-06 10:33 ` Alexander E. Patrakov [this message]
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