[alsa-devel] Noise Problem with rate convert plugins
andreas.rumpler at audivo.com
Mon Jul 16 16:25:58 CEST 2007
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.
Takashi Iwai schrieb:
>At Wed, 11 Jul 2007 17:41:21 +0200,
>Andreas Rumpler wrote:
>>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
>>- 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.
>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;
>+ 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 at alsa-project.org
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