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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