[alsa-devel] Noise Problem with rate convert plugins
Hello,
I'm new to this list.I hope someone could help me with a problem concerning the samplerate converter plugins of the alsa driver.
Firstly some facts about my application.
- Envy24 (VT1270) pci audio chip on a custom mainboard - playback of sounds/music with different sample rates from 8kHz to 48kHz (sample rate of the files) - the hardware (Envy24 chip) must always run with 44.1kHz or 48kHz (selectable by user), because the audio is outputed by a DAC and SPDIF and the SPDIF clock must not be changed according to the sample rate of the files. - Kernel is 2.6.22 and alsa is 1.0.14
So I'm using the samplerate (libsample rate based) plugin from the alsa-plugin package. Generaly it works very good.
My problem is: Every time a playback is started a short plop noise is heard. The noise is on analog and SPDIF output. If I don't use the samplerate converter the noise is not heard at all. So it comes definitely from the rate converter plugin, I think. Changing the quality level of the plugin don't solve the problem. I've also switched to the new Speex rate converter plugin. With this plugin there is no noise at the start of playback. But here I have trouble with awful noise at some rate conversions (6kHz(source) -> 48kHz(output); 11.025kHz -> 44.1kHz; 22.05kHz -> 44.1kHz). So it's even worse than using the libsamplerate plugin. Finaly I've tried the libavcodec plugin, which is the worst according to noise.
Could any comment this problem, please?
This is my asound.conf: ############################################## # Playback over Line-Out and SPDIF simultaniously as default
pcm.!default { type plug slave.pcm { type multi slaves.a.pcm "plughw:0,0" slaves.a.channels 2 slaves.b.pcm "hw:0,1" slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1 } ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 }
pcm_slave.s1 { pcm default rate 44100 }
pcm_slave.s2 { pcm default rate 48000 }
pcm.rate_convert_44100 { type rate slave s1 converter samplerate }
pcm.rate_convert_48000 { type rate slave s2 converter samplerate } #####################################################
Regards Andreas Rumpler
At Wed, 11 Jul 2007 17:41:21 +0200, Andreas Rumpler wrote:
Hello,
I'm new to this list.I hope someone could help me with a problem concerning the samplerate converter plugins of the alsa driver.
Firstly some facts about my application.
- Envy24 (VT1270) pci audio chip on a custom mainboard
- playback of sounds/music with different sample rates from 8kHz to
48kHz (sample rate of the files)
- the hardware (Envy24 chip) must always run with 44.1kHz or 48kHz
(selectable by user), because the audio is outputed by a DAC and SPDIF and the SPDIF clock must not be changed according to the sample rate of the files.
- Kernel is 2.6.22 and alsa is 1.0.14
So I'm using the samplerate (libsample rate based) plugin from the alsa-plugin package. Generaly it works very good.
My problem is: Every time a playback is started a short plop noise is heard. The noise is on analog and SPDIF output. If I don't use the samplerate converter the noise is not heard at all. So it comes definitely from the rate converter plugin, I think. Changing the quality level of the plugin don't solve the problem.
Could you try the patch below for alsa-plugins?
I've also switched to the new Speex rate converter plugin. With this plugin there is no noise at the start of playback. But here I have trouble with awful noise at some rate conversions (6kHz(source) -> 48kHz(output); 11.025kHz -> 44.1kHz; 22.05kHz -> 44.1kHz). So it's even worse than using the libsamplerate plugin. Finaly I've tried the libavcodec plugin, which is the worst according to noise.
I can confirm the noise (like flanger effect) at 11025 -> 44100 conversion, too, but no at others. Since 11024 -> 44100 works fine, it appears specific to quater or so.
Takashi
diff -r 83b528a8ca2e rate/rate_samplerate.c --- a/rate/rate_samplerate.c Mon Jun 04 15:23:44 2007 +0200 +++ b/rate/rate_samplerate.c Thu Jul 12 15:36:38 2007 +0200 @@ -116,6 +116,7 @@ static void pcm_src_convert_s16(void *ob const int16_t *src, unsigned int src_frames) { struct rate_src *rate = obj; + unsigned int ofs;
rate->data.input_frames = src_frames; rate->data.output_frames = dst_frames; @@ -123,7 +124,12 @@ static void pcm_src_convert_s16(void *ob src_short_to_float_array(src, rate->src_buf, src_frames * rate->channels); src_process(rate->state, &rate->data); - src_float_to_short_array(rate->dst_buf, dst, dst_frames * rate->channels); + if (rate->data.output_frames_gen < dst_frames) + ofs = dst_frames - rate->data.output_frames_gen; + else + ofs = 0; + src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels, + rate->data.output_frames_gen * rate->channels); }
static void pcm_src_close(void *obj)
Dear Takashi,
Thanks for your response. I've tried the patch. It's seems be better now. But something seems still to be wrong.
At the start of the playback a short sequence seems to be outputed twice. Please have a look at the attached screenshot from my oszi. The screenshot shows playback of a sine with 440Hz.
The length of the sequence outputed twice is different for every samplerate conversion factor. If I go from 8kHz to 48kHz, it's 2.6 millisec long. If I go from 22.025kHz to 48kHz, it's 900 microsec long. So it seems to be a fixed count of samples.
Regards Andreas
Takashi Iwai schrieb:
At Wed, 11 Jul 2007 17:41:21 +0200, Andreas Rumpler wrote:
Hello,
I'm new to this list.I hope someone could help me with a problem concerning the samplerate converter plugins of the alsa driver.
Firstly some facts about my application.
- Envy24 (VT1270) pci audio chip on a custom mainboard
- playback of sounds/music with different sample rates from 8kHz to
48kHz (sample rate of the files)
- the hardware (Envy24 chip) must always run with 44.1kHz or 48kHz
(selectable by user), because the audio is outputed by a DAC and SPDIF and the SPDIF clock must not be changed according to the sample rate of the files.
- Kernel is 2.6.22 and alsa is 1.0.14
So I'm using the samplerate (libsample rate based) plugin from the alsa-plugin package. Generaly it works very good.
My problem is: Every time a playback is started a short plop noise is heard. The noise is on analog and SPDIF output. If I don't use the samplerate converter the noise is not heard at all. So it comes definitely from the rate converter plugin, I think. Changing the quality level of the plugin don't solve the problem.
Could you try the patch below for alsa-plugins?
I've also switched to the new Speex rate converter plugin. With this plugin there is no noise at the start of playback. But here I have trouble with awful noise at some rate conversions (6kHz(source) -> 48kHz(output); 11.025kHz -> 44.1kHz; 22.05kHz -> 44.1kHz). So it's even worse than using the libsamplerate plugin. Finaly I've tried the libavcodec plugin, which is the worst according to noise.
I can confirm the noise (like flanger effect) at 11025 -> 44100 conversion, too, but no at others. Since 11024 -> 44100 works fine, it appears specific to quater or so.
Takashi
diff -r 83b528a8ca2e rate/rate_samplerate.c --- a/rate/rate_samplerate.c Mon Jun 04 15:23:44 2007 +0200 +++ b/rate/rate_samplerate.c Thu Jul 12 15:36:38 2007 +0200 @@ -116,6 +116,7 @@ static void pcm_src_convert_s16(void *ob const int16_t *src, unsigned int src_frames) { struct rate_src *rate = obj;
unsigned int ofs;
rate->data.input_frames = src_frames; rate->data.output_frames = dst_frames;
@@ -123,7 +124,12 @@ static void pcm_src_convert_s16(void *ob
src_short_to_float_array(src, rate->src_buf, src_frames * rate->channels); src_process(rate->state, &rate->data);
- src_float_to_short_array(rate->dst_buf, dst, dst_frames * rate->channels);
- if (rate->data.output_frames_gen < dst_frames)
ofs = dst_frames - rate->data.output_frames_gen;- else
ofs = 0;- src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels,
rate->data.output_frames_gen * rate->channels);}
static void pcm_src_close(void *obj) _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
At Mon, 16 Jul 2007 16:25:58 +0200, Andreas Rumpler wrote:
[1 <text/plain; ISO-8859-1 (7bit)>] Dear Takashi,
Thanks for your response. I've tried the patch. It's seems be better now. But something seems still to be wrong.
At the start of the playback a short sequence seems to be outputed twice. Please have a look at the attached screenshot from my oszi. The screenshot shows playback of a sine with 440Hz.
Hm, it's weird. I'll check this later.
The length of the sequence outputed twice is different for every samplerate conversion factor. If I go from 8kHz to 48kHz, it's 2.6 millisec long. If I go from 22.025kHz to 48kHz, it's 900 microsec long. So it seems to be a fixed count of samples.
BTW, a temporary fix against the problem with speex resampler was committed to HG tree today. It'll always use the interpolation, thus more CPU time, but it's still not that bad at all. Give it a try.
Takashi
participants (2)
-
Andreas Rumpler -
Takashi Iwai