Here is what I have in my asound.conf...
pcm.record_left {
type dsnoop
ipc_key 4096
ipc_perm 0666
slave {
pcm "hw:0,0"
channels 2
}
bindings.0 0
}
pcm.record_right {
type dsnoop
ipc_key 8192
ipc_perm 0666
slave {
pcm "hw:0,0"
channels 2
}
bindings.0 1
}
pcm.my_pcm {
type speex
slave.pcm "record_right"
frames 128
denoise 0
agc 1
echo 1
filter_length 8192
agc_level 8000
dereverb 0
dereverb_decay 0
dereverb_level 0
}
Best Regards,
--
Rob Krakora
Senior Software Engineer
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext. 206
(317)663-0808 Fax
2009/9/17 Frédéric COIFFIER <fcoiffie.coiffier@xxxxxxx>
Hello Mr Krakora, Mr Iwai,
Have you succeeded to get a generic asound.conf which is able to convert in
mono before using the speex plugin ?
I don't succeed to get a default device using speex plugin for capture (and
ALSA configuration seems really complex and not well documented).
Frederic Coiffier
Le jeudi 4 juin 2009 15:46:19, Takashi Iwai a écrit :
> ---------------------------------------------------------------------------> At Tue, 2 Jun 2009 16:39:46 -0400,
>
> Robert Krakora wrote:
> > Hello,
> >
> > Has anyone successfully employed the ALSA 1.0.20 Speex PCM Plugin? I
> > followed the "speexdsp.txt" document under the 'doc' directory but the
> > result was the following error:
> >
> > [root@vizioroom105 ~]# arecord -Dplug:mic poopy.wav
> > Recording WAVE 'poopy.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
> > ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not
> > usable arecord: set_params:957: Broken configuration for this PCM: no
> > configurations available
>
> It's because the slave plugin of pcm.mic is "hw" and that doesn't
> support mono streams but only stereo. speex plugin requires a mono
> stream explicitly.
>
> Wrap speex plugin over the default or add plug layer inside it, too.
>
> BTW, I found that I didn't put any echo-cancelling code in the speex
> plugin. It was just written for denoising.
>
> The below is a quick hack to add the echo-cancelling part. Give it a try
> (although it's totally untested :)
>
>
> Takashi
>
> ---
> diff --git a/doc/speexdsp.txt b/doc/speexdsp.txt
> index 875fc19..1937de6 100644
> --- a/doc/speexdsp.txt
> +++ b/doc/speexdsp.txt
> @@ -12,7 +12,7 @@ using libspeex DSP API. You can use the plugin with the
> plugin type
>
> Then record like
>
> - % arecord -fdat -c1 -Dplug:speex foo.wav
> + % arecord -fdat -c1 -Dplug:my_pcm foo.wav
>
> so that you'll get 48kHz mono stream with the denoising effect.
>
> @@ -44,6 +44,16 @@ The following parameters can be set optionally:
>
> A boolean value to enable/disable dereverb function. Default is no.
>
> +* echo
> +
> + A boolean value to enable/disable echo-cancellation function.
> + Default is no.
> +
> +* filter_length
> +
> + Number of samples of echo to cancel. As default it's 256.
> +
> +
> For example, you can enable agc like
>
> pcm.my_pcm {
> diff --git a/speex/pcm_speex.c b/speex/pcm_speex.c
> index 7bb9213..38b3582 100644
> --- a/speex/pcm_speex.c
> +++ b/speex/pcm_speex.c
> @@ -1,5 +1,5 @@
> /*
> - * Speex preprocess plugin
> + * Speex DSP plugin
> *
> * Copyright (c) 2009 by Takashi Iwai <tiwai@xxxxxxx>
> *
> @@ -21,12 +21,15 @@
> #include <alsa/asoundlib.h>
> #include <alsa/pcm_external.h>
> #include <speex/speex_preprocess.h>
> +#include <speex/speex_echo.h>
>
> -/* preprocessing parameters */
> +/* DSP parameters */
> struct spx_parms {
> int frames;
> int denoise;
> int agc;
> + int echo;
> + int filter_length;
> float agc_level;
> int dereverb;
> float dereverb_decay;
> @@ -38,7 +41,9 @@ typedef struct {
> struct spx_parms parms;
> /* instance and intermedate buffer */
> SpeexPreprocessState *state;
> + SpeexEchoState *echo_state;
> short *buf;
> + short *outbuf;
> /* running states */
> unsigned int filled;
> unsigned int processed;
> @@ -64,6 +69,12 @@ spx_transfer(snd_pcm_extplug_t *ext,
> short *src = "" src_offset);
> short *dst = area_addr(dst_areas, dst_offset);
> unsigned int count = size;
> + short *databuf;
> +
> + if (spx->parms.echo)
> + databuf = spx->outbuf;
> + else
> + databuf = spx->buf;
>
> while (count > 0) {
> unsigned int chunk;
> @@ -72,14 +83,19 @@ spx_transfer(snd_pcm_extplug_t *ext,
> else
> chunk = count;
> if (spx->processed)
> - memcpy(dst, spx->buf + spx->filled, chunk * 2);
> + memcpy(dst, databuf + spx->filled, chunk * 2);
> else
> memset(dst, 0, chunk * 2);
> dst += chunk;
> memcpy(spx->buf + spx->filled, src, chunk * 2);
> spx->filled += chunk;
> if (spx->filled == spx->parms.frames) {
> - speex_preprocess_run(spx->state, spx->buf);
> + if (spx->parms.echo)
> + speex_echo_capture(spx->echo_state, spx->buf,
> + spx->outbuf);
> + speex_preprocess_run(spx->state, databuf);
> + if (spx->parms.echo)
> + speex_echo_playback(spx->echo_state, databuf);
> spx->processed = 1;
> spx->filled = 0;
> }
> @@ -101,13 +117,34 @@ static int spx_init(snd_pcm_extplug_t *ext)
> }
> memset(spx->buf, 0, spx->parms.frames * 2);
>
> - if (spx->state)
> + if (spx->state) {
> speex_preprocess_state_destroy(spx->state);
> + spx->state = NULL;
> + }
> + if (spx->echo_state) {
> + speex_echo_state_destroy(spx->echo_state);
> + spx->echo_state = NULL;
> + }
> +
> + if (spx->parms.echo) {
> + spx->echo_state = speex_echo_state_init(spx->parms.frames,
> + spx->parms.filter_length);
> + if (!spx->echo_state)
> + return -EIO;
> + speex_echo_ctl(spx->echo_state, SPEEX_ECHO_SET_SAMPLING_RATE,
> + &spx->ext.rate);
> + }
> +
> spx->state = speex_preprocess_state_init(spx->parms.frames,
> spx->ext.rate);
> if (!spx->state)
> return -EIO;
>
> + if (spx->parms.echo)
> + speex_preprocess_ctl(spx->state,
> + SPEEX_PREPROCESS_SET_ECHO_STATE,
> + spx->echo_state);
> +
> speex_preprocess_ctl(spx->state, SPEEX_PREPROCESS_SET_DENOISE,
> &spx->parms.denoise);
> speex_preprocess_ctl(spx->state, SPEEX_PREPROCESS_SET_AGC,
> @@ -132,6 +169,8 @@ static int spx_close(snd_pcm_extplug_t *ext)
> free(spx->buf);
> if (spx->state)
> speex_preprocess_state_destroy(spx->state);
> + if (spx->echo_state)
> + speex_echo_state_destroy(spx->echo_state);
> return 0;
> }
>
> @@ -205,6 +244,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex)
> .dereverb = 0,
> .dereverb_decay = 0,
> .dereverb_level = 0,
> + .echo = 0,
> + .filter_length = 256,
> };
>
> snd_config_for_each(i, next, conf) {
> @@ -242,6 +283,12 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex)
> &parms.dereverb_level);
> if (err)
> goto ok;
> + err = get_bool_parm(n, id, "echo", &parms.echo);
> + if (err)
> + goto ok;
> + err = get_int_parm(n, id, "filter_length", &parms.filter_length);
> + if (err)
> + goto ok;
> SNDERR("Unknown field %s", id);
> err = -EINVAL;
> ok:
> @@ -259,7 +306,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex)
> return -ENOMEM;
>
> spx->ext.version = SND_PCM_EXTPLUG_VERSION;
> - spx->ext.name = "Speex Denoise Plugin";
> + spx->ext.name = "Speex DSP Plugin";
> spx->ext.callback = &speex_callback;
> spx->ext.private_data = spx;
> spx->parms = parms;
>
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