From: "Volker Rümelin" <vr_qemu@t-online.de>
To: Christian Schoenebeck <qemu_oss@crudebyte.com>,
Gerd Hoffmann <kraxel@redhat.com>,
qemu-devel@nongnu.org
Cc: "Thomas Huth" <thuth@redhat.com>,
"Marc-André Lureau" <marcandre.lureau@gmail.com>
Subject: Re: [PATCH 10/11] alsaaudio: change default playback settings
Date: Mon, 26 Dec 2022 16:08:37 +0100 [thread overview]
Message-ID: <a257ab88-a779-bb84-e96e-664a3434b417@t-online.de> (raw)
In-Reply-To: <2230283.NDGgU1aIbp@silver>
Am 21.12.22 um 12:03 schrieb Christian Schoenebeck:
> On Sunday, December 18, 2022 6:15:38 PM CET Volker Rümelin wrote:
>> The currently used default playback settings in the ALSA audio
>> backend are a bit unfortunate. With a few emulated audio devices,
>> audio playback does not work properly. Here is a short part of
>> the debug log while audio is playing (elapsed time in seconds).
> Which emulated devices are these?
The hda device and sb16. When I wrote this patch two months ago ac97
also had occasional dropouts, but at the moment ac97 works without issues.
>> audio: Elapsed since last alsa run (running): 0.046244
>> audio: Elapsed since last alsa run (running): 0.023137
>> audio: Elapsed since last alsa run (running): 0.023170
>> audio: Elapsed since last alsa run (running): 0.023650
>> audio: Elapsed since last alsa run (running): 0.060802
>> audio: Elapsed since last alsa run (running): 0.031931
>>
>> For some audio devices the time of more than 23ms between updates
>> is too long.
>>
>> Set the period time to 5.8ms so that the maximum time between
>> two updates typically does not exceed 11ms. This roughly matches
>> the 10ms period time when doing playback with the audio timer.
>> After this patch the debug log looks like this.
> And what about dynamically adapting that value instead of reducing period time
> for everyone by default?
It seems this would be only needed for the ALSA backend. All other
backends with the exception of OSS are fine with a 10ms period, and the
ALSA audio backend also uses 10ms with -audiodev
alsa,out.try-poll=off,in.try-poll=off.
> 23ms is usually a good trade off between low latency, CPU load and potential
> for audio dropouts.
Quite often it's longer than 23ms. For the rest of the audio backends a
timer period of 10ms was selected as a good trade off between CPU load
and audio dropouts. But you are right, this patch increases the CPU load.
On my system the CPU load is increased by 0.9%. This was measured with a
Linux guest using rhythmbox for audio playback. The guest was configured
to use pulseaudio as sound server. The measurement was done with top -b
-d 10 -n 14 over a period of two minutes. The first and last measurement
was dropped. The average QEMU CPU load was 10.7% with and 9.8% without
this patch.
I would prefer a system with a 0.9% increased CPU load where audio just
works over a system where you have to fine tune audio parameters.
>> audio: Elapsed since last alsa run (running): 0.011919
>> audio: Elapsed since last alsa run (running): 0.005788
>> audio: Elapsed since last alsa run (running): 0.005995
>> audio: Elapsed since last alsa run (running): 0.011069
>> audio: Elapsed since last alsa run (running): 0.005901
>> audio: Elapsed since last alsa run (running): 0.006084
>>
>> Signed-off-by: Volker Rümelin<vr_qemu@t-online.de>
>> ---
>> audio/alsaaudio.c | 11 ++++-------
>> 1 file changed, 4 insertions(+), 7 deletions(-)
>>
>> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
>> index 5f50dfa0bf..0cc982e61f 100644
>> --- a/audio/alsaaudio.c
>> +++ b/audio/alsaaudio.c
>> @@ -913,17 +913,14 @@ static void *alsa_audio_init(Audiodev *dev)
>> alsa_init_per_direction(aopts->in);
>> alsa_init_per_direction(aopts->out);
>>
>> - /*
>> - * need to define them, as otherwise alsa produces no sound
>> - * doesn't set has_* so alsa_open can identify it wasn't set by the user
>> - */
>> + /* don't set has_* so alsa_open can identify it wasn't set by the user */
>> if (!dev->u.alsa.out->has_period_length) {
>> - /* 1024 frames assuming 44100Hz */
>> - dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
>> + /* 256 frames assuming 44100Hz */
>> + dev->u.alsa.out->period_length = 5805;
>> }
>> if (!dev->u.alsa.out->has_buffer_length) {
>> /* 4096 frames assuming 44100Hz */
>> - dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
>> + dev->u.alsa.out->buffer_length = 92880;
> Not a big fan of magic numbers, as it makes code less readable.
I can't see how this can be improved. The buffer length is unchanged. I
just evaluated the constant expression to have a time in microseconds
like the rest of the audio backends. And libasound tells me to use
5804us for the period length which I rounded up to 5805us. I would
prefer a period length of 5000us.
./qemu-system-x86_64 -device ich9-intel-hda -device
hda-duplex,audiodev=audio0 -audiodev
alsa,id=audio0,out.period-length=5000,out.dev=PCH,,0
alsa: Requested period time 5000 was rejected, using 5804
>> }
>>
>> /*
>>
next prev parent reply other threads:[~2022-12-26 15:09 UTC|newest]
Thread overview: 39+ messages / expand[flat|nested] mbox.gz Atom feed top
2022-12-18 17:13 [PATCH 00/11] audio: more improvements Volker Rümelin
2022-12-18 17:15 ` [PATCH 01/11] audio: log unimplemented audio device sample rates Volker Rümelin
2022-12-18 20:26 ` Christian Schoenebeck
2022-12-19 7:22 ` Volker Rümelin
2022-12-19 13:38 ` Christian Schoenebeck
2022-12-18 17:15 ` [PATCH 02/11] audio: don't show unnecessary error messages Volker Rümelin
2022-12-18 17:20 ` Philippe Mathieu-Daudé
2022-12-18 17:15 ` [PATCH 03/11] audio: rename hardware store to backend Volker Rümelin
2022-12-29 9:39 ` Thomas Huth
2022-12-18 17:15 ` [PATCH 04/11] audio: remove unused #define AUDIO_STRINGIFY Volker Rümelin
2022-12-18 17:31 ` Philippe Mathieu-Daudé
2022-12-29 9:39 ` Thomas Huth
2022-12-18 17:15 ` [PATCH 05/11] audio/mixeng: use g_new0() instead of audio_calloc() Volker Rümelin
2022-12-18 20:56 ` Richard Henderson
2022-12-18 17:15 ` [PATCH 06/11] audio/alsaaudio: " Volker Rümelin
2022-12-18 17:24 ` Philippe Mathieu-Daudé
2022-12-18 20:57 ` Richard Henderson
2022-12-18 17:15 ` [PATCH 07/11] audio/audio_template: use g_malloc0() to replace audio_calloc() Volker Rümelin
2022-12-18 17:26 ` Philippe Mathieu-Daudé
2022-12-18 17:39 ` Volker Rümelin
2022-12-18 20:05 ` Christian Schoenebeck
2022-12-18 20:34 ` Philippe Mathieu-Daudé
2023-01-16 8:58 ` Daniel P. Berrangé
2023-01-17 7:05 ` Volker Rümelin
2022-12-18 17:15 ` [PATCH 08/11] audio/audio_template: use g_new0() " Volker Rümelin
2022-12-18 21:02 ` Richard Henderson
2023-01-16 9:03 ` Daniel P. Berrangé
2023-01-17 7:02 ` Volker Rümelin
2022-12-18 17:15 ` [PATCH 09/11] audio: remove audio_calloc() function Volker Rümelin
2022-12-18 17:29 ` Philippe Mathieu-Daudé
2022-12-18 17:15 ` [PATCH 10/11] alsaaudio: change default playback settings Volker Rümelin
2022-12-21 11:03 ` Christian Schoenebeck
2022-12-26 15:08 ` Volker Rümelin [this message]
2022-12-26 15:37 ` Volker Rümelin
2022-12-28 13:52 ` Christian Schoenebeck
2022-12-29 9:08 ` Volker Rümelin
2022-12-30 9:01 ` Volker Rümelin
2022-12-30 14:05 ` Christian Schoenebeck
2022-12-18 17:15 ` [PATCH 11/11] alsaaudio: reintroduce default recording settings Volker Rümelin
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