[asterisk-commits] bbryant: branch bbryant/noise_reduction_and_agc r114910 - /team/bbryant/noise...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Apr 30 15:29:56 CDT 2008
Author: bbryant
Date: Wed Apr 30 15:29:55 2008
New Revision: 114910
URL: http://svn.digium.com/view/asterisk?view=rev&rev=114910
Log:
Update code in the func_speex.c file
Modified:
team/bbryant/noise_reduction_and_agc/funcs/func_speex.c
Modified: team/bbryant/noise_reduction_and_agc/funcs/func_speex.c
URL: http://svn.digium.com/view/asterisk/team/bbryant/noise_reduction_and_agc/funcs/func_speex.c?view=diff&rev=114910&r1=114909&r2=114910
==============================================================================
--- team/bbryant/noise_reduction_and_agc/funcs/func_speex.c (original)
+++ team/bbryant/noise_reduction_and_agc/funcs/func_speex.c Wed Apr 30 15:29:55 2008
@@ -4,7 +4,7 @@
* Copyright (C) 2008, Digium, Inc.
*
* Brian Degenhardt <bmd at digium.com>
- * Russell Bryant <russell at digium.com>
+ * Brett Bryant <bbryant at digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
@@ -22,7 +22,7 @@
* \brief Noise reduction and automatic gain control (AGC)
*
* \author Brian Degenhardt <bmd at digium.com>
- * \author Russell Bryant <russell at digium.com>
+ * \author Brett Bryant <bbryant at digium.com>
*
* \ingroup functions
*
@@ -47,14 +47,16 @@
#define DEFAULT_AGC_LEVEL 8000.0
struct speex_direction_info {
- int denoise;
- float agclevel;
- SpeexPreprocessState *preprocess_state;
+ SpeexPreprocessState *state; /*! speex preprocess state object */
+ int agc; /*! audio gain control is enabled or not */
+ int denoise; /*! denoise is enabled or not */
+ int samples; /*! n of 8Khz samples in last frame */
+ float agclevel; /*! audio gain control level [1.0 - 32768.0] */
};
struct speex_info {
struct ast_audiohook audiohook;
- struct speex_direction_info tx, rx;
+ struct speex_direction_info *tx, *rx;
};
static void destroy_callback(void *data)
@@ -62,6 +64,22 @@
struct speex_info *si = data;
ast_audiohook_destroy(&si->audiohook);
+
+ if (si->rx && si->rx->state) {
+ speex_preprocess_state_destroy(si->rx->state);
+ }
+
+ if (si->tx && si->tx->state) {
+ speex_preprocess_state_destroy(si->tx->state);
+ }
+
+ if (si->rx) {
+ ast_free(si->rx);
+ }
+
+ if (si->tx) {
+ ast_free(si->tx);
+ }
ast_free(data);
};
@@ -74,33 +92,48 @@
static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
{
struct ast_datastore *datastore = NULL;
+ struct speex_direction_info *sdi = NULL;
struct speex_info *si = NULL;
- struct speex_direction_info *sdi = NULL;
- int agcenabled;
/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
- if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
+ if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
return 0;
-
- if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL)))
+ }
+
+ ast_channel_lock(chan);
+ if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
+ ast_channel_unlock(chan);
return 0;
+ }
+ ast_channel_unlock(chan);
si = datastore->data;
- /* If we don't have a voice frame, or there's no agc level for the direction of this frame... exit */
- if (frame->frametype != AST_FRAME_VOICE)
+ sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
+
+ if (!sdi) {
return 0;
-
- sdi = (direction = AST_AUDIOHOOK_DIRECTION_READ) ? &si->rx : &si->tx;
-
- agcenabled = !!sdi->agclevel;
-
- speex_preprocess_ctl(sdi->preprocess_state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
- speex_preprocess_ctl(sdi->preprocess_state, SPEEX_PREPROCESS_SET_AGC, &agcenabled);
- if (agcenabled)
- speex_preprocess_ctl(sdi->preprocess_state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &sdi->agclevel);
-
- speex_preprocess(sdi->preprocess_state, frame->data, NULL);
+ }
+
+ if (sdi->samples != frame->samples) {
+ if (sdi->state) {
+ speex_preprocess_state_destroy(sdi->state);
+ }
+
+ if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), 8000))) {
+ return -1;
+ }
+
+ speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
+
+ if (sdi->agc) {
+ speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &sdi->agclevel);
+ }
+
+ speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
+ }
+
+ speex_preprocess(sdi->state, frame->data, NULL);
return 0;
}
@@ -109,12 +142,17 @@
{
struct ast_datastore *datastore = NULL;
struct speex_info *si = NULL;
+ struct speex_direction_info **sdi = NULL;
int is_new = 0;
- float *agcval = NULL;
-
+
+ ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
- if (!(datastore = ast_channel_datastore_alloc(&speex_datastore, NULL)))
+ ast_channel_unlock(chan);
+
+ if (!(datastore = ast_channel_datastore_alloc(&speex_datastore, NULL))) {
return 0;
+ }
+
if (!(si = ast_calloc(1, sizeof(*si)))) {
ast_channel_datastore_free(datastore);
return 0;
@@ -125,29 +163,86 @@
is_new = 1;
} else {
+ ast_channel_unlock(chan);
si = datastore->data;
}
+
+ if (!strcasecmp(data, "rx")) {
+ sdi = &si->rx;
+ } else if (!strcasecmp(data, "tx")) {
+ sdi = &si->tx;
+ } else {
+ ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
+
+ if (is_new) {
+ ast_channel_datastore_free(datastore);
+ return -1;
+ }
+ }
+
+ if (!*sdi) {
+ if (!(*sdi = ast_calloc(1, sizeof(*sdi)))) {
+ return 0;
+ }
+ /* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
+ * audio. When it supports 16 kHz (or any other sample rates, we will
+ * have to take that into account here. */
+ (*sdi)->samples = -1;
+ }
+
+ if (!strcasecmp(cmd, "agc")) {
+ if (!sscanf(value, "%f", &(*sdi)->agclevel))
+ (*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
- if (!strcasecmp(cmd, "agc")) {
- if (!strcasecmp(data, "tx") && !sscanf(value, "%f", agcval))
- agcval = &si->tx.agclevel;
- else if (!strcasecmp(data, "rx"))
- agcval = &si->rx.agclevel;
-
- if (agcval && !sscanf(value, "%f", agcval))
- *agcval = (ast_true(value)) ? DEFAULT_AGC_LEVEL : 0;
+ if ((*sdi)->agclevel > 32768.0) {
+ ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n",
+ ((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
+ (*sdi)->agclevel = 32768.0;
+ }
+
+ (*sdi)->agc = !!((*sdi)->agclevel);
+
+ if ((*sdi)->state) {
+ speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
+ if ((*sdi)->agc) {
+ speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
+ }
+ }
} else if (!strcasecmp(cmd, "denoise")) {
- if (!strcasecmp(data, "tx"))
- si->tx.denoise = ast_true(value);
- else if (!strcasecmp(data, "rx"))
- si->rx.denoise = ast_true(value);
+ (*sdi)->denoise = ast_true(value);
+
+ if ((*sdi)->state) {
+ speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
+ }
+ }
+
+ if (!(*sdi)->agc && !(*sdi)->denoise) {
+ if ((*sdi)->state)
+ speex_preprocess_state_destroy((*sdi)->state);
+
+ ast_free(*sdi);
+ *sdi = NULL;
+ }
+
+ if (!si->rx && !si->tx) {
+ if (is_new) {
+ is_new = 0;
+ } else {
+ ast_channel_lock(chan);
+ ast_channel_datastore_remove(chan, datastore);
+ ast_channel_unlock(chan);
+ }
+
+ ast_audiohook_detach(&si->audiohook);
+
+ ast_channel_datastore_free(datastore);
}
if (is_new) {
- si->rx.preprocess_state = speex_preprocess_state_init(160, 8000);
- si->tx.preprocess_state = speex_preprocess_state_init(160, 8000);
datastore->data = si;
+ ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
+ ast_channel_unlock(chan);
ast_audiohook_attach(chan, &si->audiohook);
}
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