Re: Noise Problem with rate convert plugins

[Andreas Rumpler <andreas.rumpler@audivo.com>]

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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)
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>
>  
>

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