[asterisk-commits] tilghman: branch group/codec_bits r112560 - in /team/group/codec_bits: format...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Apr 2 19:23:21 CDT 2008
Author: tilghman
Date: Wed Apr 2 19:23:21 2008
New Revision: 112560
URL: http://svn.digium.com/view/asterisk?view=rev&rev=112560
Log:
Bunches more done
Modified:
team/group/codec_bits/formats/format_pcm.c
team/group/codec_bits/formats/format_sln.c
team/group/codec_bits/formats/format_sln16.c
team/group/codec_bits/formats/format_vox.c
team/group/codec_bits/formats/format_wav.c
team/group/codec_bits/formats/format_wav_gsm.c
team/group/codec_bits/funcs/func_channel.c
team/group/codec_bits/include/asterisk/abstract_jb.h
team/group/codec_bits/include/asterisk/mod_format.h
team/group/codec_bits/main/abstract_jb.c
team/group/codec_bits/main/app.c
team/group/codec_bits/main/audiohook.c
team/group/codec_bits/main/channel.c
team/group/codec_bits/main/dial.c
team/group/codec_bits/main/dsp.c
team/group/codec_bits/main/features.c
team/group/codec_bits/main/file.c
team/group/codec_bits/main/frame.c
team/group/codec_bits/res/res_adsi.c
team/group/codec_bits/res/res_agi.c
team/group/codec_bits/res/res_clioriginate.c
team/group/codec_bits/res/res_musiconhold.c
team/group/codec_bits/res/res_speech.c
Modified: team/group/codec_bits/formats/format_pcm.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/formats/format_pcm.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/formats/format_pcm.c (original)
+++ team/group/codec_bits/formats/format_pcm.c Wed Apr 2 19:23:21 2008
@@ -80,7 +80,8 @@
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
- s->fr.subclass = s->fmt->format;
+ s->fr.subclass = 0;
+ s->fr.codec = s->fmt->format;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) < 1) {
@@ -89,7 +90,7 @@
return NULL;
}
s->fr.datalen = res;
- if (s->fmt->format == AST_FORMAT_G722)
+ if (s->fmt->format.audio[0] == AST_FORMAT_AUDIO_G722)
*whennext = s->fr.samples = res * 2;
else
*whennext = s->fr.samples = res;
@@ -126,7 +127,7 @@
}
if (whence == SEEK_FORCECUR && offset > max) { /* extend the file */
size_t left = offset - max;
- const char *src = (fs->fmt->format == AST_FORMAT_ALAW) ? alaw_silence : ulaw_silence;
+ const char *src = (fs->fmt->format.audio[0] == AST_FORMAT_AUDIO_ALAW) ? alaw_silence : ulaw_silence;
while (left) {
size_t written = fwrite(src, 1, (left > BUF_SIZE) ? BUF_SIZE : left, fs->f);
@@ -163,8 +164,8 @@
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
}
- if (f->subclass != fs->fmt->format) {
- ast_log(LOG_WARNING, "Asked to write incompatible format frame (%d)!\n", f->subclass);
+ if (!FMT_EQ(f->codec, fs->fmt->format)) {
+ ast_log(LOG_WARNING, "Asked to write incompatible format frame (%s)!\n", ast_getformatname(f->codec));
return -1;
}
@@ -371,8 +372,9 @@
{
off_t min, max, cur;
long offset = 0, bytes;
-
- if (fs->fmt->format == AST_FORMAT_G722)
+ const struct ast_extended_codec g722 = { .audio = { AST_FORMAT_AUDIO_G722 } };
+
+ if (FMT_EQ(fs->fmt->format, g722))
bytes = sample_offset / 2;
else
bytes = sample_offset;
@@ -415,7 +417,7 @@
static const struct ast_format alaw_f = {
.name = "alaw",
.exts = "alaw|al",
- .format = AST_FORMAT_ALAW,
+ .format = { .audio = { AST_FORMAT_AUDIO_ALAW } },
.write = pcm_write,
.seek = pcm_seek,
.trunc = pcm_trunc,
@@ -432,7 +434,7 @@
static const struct ast_format pcm_f = {
.name = "pcm",
.exts = "pcm|ulaw|ul|mu",
- .format = AST_FORMAT_ULAW,
+ .format = { .audio = { AST_FORMAT_AUDIO_ULAW } },
.write = pcm_write,
.seek = pcm_seek,
.trunc = pcm_trunc,
@@ -444,7 +446,7 @@
static const struct ast_format g722_f = {
.name = "g722",
.exts = "g722",
- .format = AST_FORMAT_G722,
+ .format = { .audio = { AST_FORMAT_AUDIO_G722 } },
.write = pcm_write,
.seek = pcm_seek,
.trunc = pcm_trunc,
@@ -456,7 +458,7 @@
static const struct ast_format au_f = {
.name = "au",
.exts = "au",
- .format = AST_FORMAT_ULAW,
+ .format = { .audio = { AST_FORMAT_AUDIO_ULAW } },
.open = au_open,
.rewrite = au_rewrite,
.write = pcm_write,
Modified: team/group/codec_bits/formats/format_sln.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/formats/format_sln.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/formats/format_sln.c (original)
+++ team/group/codec_bits/formats/format_sln.c Wed Apr 2 19:23:21 2008
@@ -36,10 +36,12 @@
static struct ast_frame *slinear_read(struct ast_filestream *s, int *whennext)
{
int res;
+ const struct ast_extended_codec slin = { .audio = { AST_FORMAT_AUDIO_SLINEAR } };
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
- s->fr.subclass = AST_FORMAT_SLINEAR;
+ s->fr.subclass = 0;
+ s->fr.codec = slin;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) < 1) {
@@ -59,13 +61,13 @@
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
}
- if (f->subclass != AST_FORMAT_SLINEAR) {
+ if (f->codec.audio[0] != AST_FORMAT_AUDIO_SLINEAR) {
ast_log(LOG_WARNING, "Asked to write non-slinear frame (%d)!\n", f->subclass);
return -1;
}
if ((res = fwrite(f->data, 1, f->datalen, fs->f)) != f->datalen) {
- ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
- return -1;
+ ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
+ return -1;
}
return 0;
}
@@ -106,7 +108,7 @@
static const struct ast_format slin_f = {
.name = "sln",
.exts = "sln|raw",
- .format = AST_FORMAT_SLINEAR,
+ .format = { .audio = { AST_FORMAT_AUDIO_SLINEAR } },
.write = slinear_write,
.seek = slinear_seek,
.trunc = slinear_trunc,
Modified: team/group/codec_bits/formats/format_sln16.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/formats/format_sln16.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/formats/format_sln16.c (original)
+++ team/group/codec_bits/formats/format_sln16.c Wed Apr 2 19:23:21 2008
@@ -37,10 +37,12 @@
static struct ast_frame *slinear_read(struct ast_filestream *s, int *whennext)
{
int res;
+ const struct ast_extended_codec sln16 = { .audio = { AST_FORMAT_AUDIO_SLINEAR16 } };
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
- s->fr.subclass = AST_FORMAT_SLINEAR16;
+ s->fr.subclass = 0;
+ s->fr.codec = sln16;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) < 1) {
@@ -61,8 +63,8 @@
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
}
- if (f->subclass != AST_FORMAT_SLINEAR16) {
- ast_log(LOG_WARNING, "Asked to write non-slinear16 frame (%d)!\n", f->subclass);
+ if (f->codec.audio[0] != AST_FORMAT_AUDIO_SLINEAR16) {
+ ast_log(LOG_WARNING, "Asked to write non-slinear16 frame (%s)!\n", ast_getformatname(f->codec));
return -1;
}
if ((res = fwrite(f->data, 1, f->datalen, fs->f)) != f->datalen) {
@@ -113,7 +115,7 @@
static const struct ast_format slin_f = {
.name = "sln16",
.exts = "sln16",
- .format = AST_FORMAT_SLINEAR16,
+ .format = { .audio = { AST_FORMAT_AUDIO_SLINEAR16 } },
.write = slinear_write,
.seek = slinear_seek,
.trunc = slinear_trunc,
Modified: team/group/codec_bits/formats/format_vox.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/formats/format_vox.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/formats/format_vox.c (original)
+++ team/group/codec_bits/formats/format_vox.c Wed Apr 2 19:23:21 2008
@@ -38,10 +38,12 @@
static struct ast_frame *vox_read(struct ast_filestream *s, int *whennext)
{
int res;
+ const struct ast_extended_codec vox = { .audio = { AST_FORMAT_AUDIO_ADPCM } };
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
- s->fr.subclass = AST_FORMAT_ADPCM;
+ s->fr.subclass = 0;
+ s->fr.codec = vox;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) < 1) {
@@ -61,57 +63,56 @@
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
}
- if (f->subclass != AST_FORMAT_ADPCM) {
+ if (f->codec.audio[0] != AST_FORMAT_AUDIO_ADPCM) {
ast_log(LOG_WARNING, "Asked to write non-ADPCM frame (%d)!\n", f->subclass);
return -1;
}
if ((res = fwrite(f->data, 1, f->datalen, s->f)) != f->datalen) {
- ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
- return -1;
+ ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
+ return -1;
}
return 0;
}
static int vox_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
{
- off_t offset=0,min,cur,max,distance;
-
- min = 0;
- cur = ftello(fs->f);
- fseeko(fs->f, 0, SEEK_END);
- max = ftello(fs->f);
-
- /* have to fudge to frame here, so not fully to sample */
- distance = sample_offset/2;
- if(whence == SEEK_SET)
- offset = distance;
- else if(whence == SEEK_CUR || whence == SEEK_FORCECUR)
- offset = distance + cur;
- else if(whence == SEEK_END)
- offset = max - distance;
- if (whence != SEEK_FORCECUR) {
- offset = (offset > max)?max:offset;
- offset = (offset < min)?min:offset;
- }
- return fseeko(fs->f, offset, SEEK_SET);
+ off_t offset=0,min,cur,max,distance;
+
+ min = 0;
+ cur = ftello(fs->f);
+ fseeko(fs->f, 0, SEEK_END);
+ max = ftello(fs->f);
+
+ /* have to fudge to frame here, so not fully to sample */
+ distance = sample_offset / 2;
+ if (whence == SEEK_SET) {
+ offset = distance;
+ } else if (whence == SEEK_CUR || whence == SEEK_FORCECUR) {
+ offset = distance + cur;
+ } else if (whence == SEEK_END) {
+ offset = max - distance;
+ }
+ if (whence != SEEK_FORCECUR) {
+ offset = (offset > max) ? max : offset;
+ offset = (offset < min) ? min : offset;
+ }
+ return fseeko(fs->f, offset, SEEK_SET);
}
static int vox_trunc(struct ast_filestream *fs)
{
- return ftruncate(fileno(fs->f), ftello(fs->f));
+ return ftruncate(fileno(fs->f), ftello(fs->f));
}
static off_t vox_tell(struct ast_filestream *fs)
{
- off_t offset;
- offset = ftello(fs->f) << 1;
- return offset;
+ return ftello(fs->f) << 1;
}
static const struct ast_format vox_f = {
.name = "vox",
.exts = "vox",
- .format = AST_FORMAT_ADPCM,
+ .format = { .audio = { AST_FORMAT_AUDIO_ADPCM } },
.write = vox_write,
.seek = vox_seek,
.trunc = vox_trunc,
Modified: team/group/codec_bits/formats/format_wav.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/formats/format_wav.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/formats/format_wav.c (original)
+++ team/group/codec_bits/formats/format_wav.c Wed Apr 2 19:23:21 2008
@@ -345,6 +345,7 @@
off_t here;
/* Send a frame from the file to the appropriate channel */
struct wav_desc *fs = (struct wav_desc *)s->_private;
+ const struct ast_extended_codec wav = { .audio = { AST_FORMAT_AUDIO_SLINEAR } };
here = ftello(s->f);
if (fs->maxlen - here < bytes) /* truncate if necessary */
@@ -353,7 +354,8 @@
bytes = 0;
/* ast_debug(1, "here: %d, maxlen: %d, bytes: %d\n", here, s->maxlen, bytes); */
s->fr.frametype = AST_FRAME_VOICE;
- s->fr.subclass = AST_FORMAT_SLINEAR;
+ s->fr.subclass = 0;
+ s->fr.codec = wav;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, bytes);
@@ -389,7 +391,7 @@
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
}
- if (f->subclass != AST_FORMAT_SLINEAR) {
+ if (f->codec.audio[0] != AST_FORMAT_AUDIO_SLINEAR) {
ast_log(LOG_WARNING, "Asked to write non-SLINEAR frame (%d)!\n", f->subclass);
return -1;
}
@@ -403,7 +405,7 @@
return -1;
}
tmpi = f->data;
- for (x=0; x < f->datalen/2; x++)
+ for (x = 0; x < f->datalen / 2; x++)
tmp[x] = (tmpi[x] << 8) | ((tmpi[x] & 0xff00) >> 8);
if ((res = fwrite(tmp, 1, f->datalen, fs->f)) != f->datalen ) {
@@ -431,14 +433,15 @@
cur = ftello(fs->f);
fseeko(fs->f, 0, SEEK_END);
max = ftello(fs->f);
- if (whence == SEEK_SET)
+ if (whence == SEEK_SET) {
offset = samples + min;
- else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
+ } else if (whence == SEEK_CUR || whence == SEEK_FORCECUR) {
offset = samples + cur;
- else if (whence == SEEK_END)
+ } else if (whence == SEEK_END) {
offset = max - samples;
- if (whence != SEEK_FORCECUR) {
- offset = (offset > max)?max:offset;
+ }
+ if (whence != SEEK_FORCECUR) {
+ offset = (offset > max) ? max : offset;
}
/* always protect the header space. */
offset = (offset < min)?min:offset;
@@ -463,7 +466,7 @@
static const struct ast_format wav_f = {
.name = "wav",
.exts = "wav",
- .format = AST_FORMAT_SLINEAR,
+ .format = { .audio = { AST_FORMAT_AUDIO_SLINEAR } },
.open = wav_open,
.rewrite = wav_rewrite,
.write = wav_write,
Modified: team/group/codec_bits/formats/format_wav_gsm.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/formats/format_wav_gsm.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/formats/format_wav_gsm.c (original)
+++ team/group/codec_bits/formats/format_wav_gsm.c Wed Apr 2 19:23:21 2008
@@ -393,9 +393,11 @@
{
/* Send a frame from the file to the appropriate channel */
struct wavg_desc *fs = (struct wavg_desc *)s->_private;
+ const struct ast_extended_codec gsm = { .audio = { AST_FORMAT_AUDIO_GSM } };
s->fr.frametype = AST_FRAME_VOICE;
- s->fr.subclass = AST_FORMAT_GSM;
+ s->fr.subclass = 0;
+ s->fr.codec = gsm;
s->fr.offset = AST_FRIENDLY_OFFSET;
s->fr.samples = GSM_SAMPLES;
s->fr.mallocd = 0;
@@ -432,7 +434,7 @@
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
}
- if (f->subclass != AST_FORMAT_GSM) {
+ if (f->codec.audio[0] != AST_FORMAT_AUDIO_GSM) {
ast_log(LOG_WARNING, "Asked to write non-GSM frame (%d)!\n", f->subclass);
return -1;
}
@@ -522,7 +524,7 @@
static const struct ast_format wav49_f = {
.name = "wav49",
.exts = "WAV|wav49",
- .format = AST_FORMAT_GSM,
+ .format = { .audio = { AST_FORMAT_AUDIO_GSM } },
.open = wav_open,
.rewrite = wav_rewrite,
.write = wav_write,
Modified: team/group/codec_bits/funcs/func_channel.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/funcs/func_channel.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/funcs/func_channel.c (original)
+++ team/group/codec_bits/funcs/func_channel.c Wed Apr 2 19:23:21 2008
@@ -62,11 +62,11 @@
if (!strcasecmp(data, "audionativeformat"))
/* use the _multiple version when chan->nativeformats holds multiple formats */
/* ast_getformatname_multiple(buf, len, chan->nativeformats & AST_FORMAT_AUDIO_MASK); */
- ast_copy_string(buf, ast_getformatname(chan->nativeformats & AST_FORMAT_AUDIO_MASK), len);
+ ast_copy_string(buf, ast_getformatname(FMT_AND(chan->nativeformats, AST_FMT_AUDIO_MASK)), len);
else if (!strcasecmp(data, "videonativeformat"))
/* use the _multiple version when chan->nativeformats holds multiple formats */
/* ast_getformatname_multiple(buf, len, chan->nativeformats & AST_FORMAT_VIDEO_MASK); */
- ast_copy_string(buf, ast_getformatname(chan->nativeformats & AST_FORMAT_VIDEO_MASK), len);
+ ast_copy_string(buf, ast_getformatname(FMT_AND(chan->nativeformats, AST_FMT_VIDEO_MASK)), len);
else if (!strcasecmp(data, "audioreadformat"))
ast_copy_string(buf, ast_getformatname(chan->readformat), len);
else if (!strcasecmp(data, "audiowriteformat"))
Modified: team/group/codec_bits/include/asterisk/abstract_jb.h
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/include/asterisk/abstract_jb.h?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/include/asterisk/abstract_jb.h (original)
+++ team/group/codec_bits/include/asterisk/abstract_jb.h Wed Apr 2 19:23:21 2008
@@ -31,6 +31,7 @@
#define _ABSTRACT_JB_H_
#include <sys/time.h>
+#include "asterisk/frame.h"
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
@@ -92,7 +93,7 @@
/*! \brief The time the next frame should be played. */
long next;
/*! \brief Voice format of the last frame in. */
- int last_format;
+ struct ast_extended_codec last_format;
/*! \brief File for frame timestamp tracing. */
FILE *logfile;
/*! \brief Jitterbuffer internal state flags. */
Modified: team/group/codec_bits/include/asterisk/mod_format.h
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/include/asterisk/mod_format.h?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/include/asterisk/mod_format.h (original)
+++ team/group/codec_bits/include/asterisk/mod_format.h Wed Apr 2 19:23:21 2008
@@ -109,7 +109,7 @@
/*! Transparently translate from another format -- just once */
struct ast_trans_pvt *trans;
struct ast_tranlator_pvt *tr;
- int lastwriteformat;
+ struct ast_extended_codec lastwriteformat;
int lasttimeout;
struct ast_channel *owner;
FILE *f;
Modified: team/group/codec_bits/main/abstract_jb.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/main/abstract_jb.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/main/abstract_jb.c (original)
+++ team/group/codec_bits/main/abstract_jb.c Wed Apr 2 19:23:21 2008
@@ -403,14 +403,15 @@
case JB_IMPL_DROP:
jb_framelog("\tJB_GET {now=%ld}: %s frame with ts=%ld and len=%ld\n",
now, jb_get_actions[res], f->ts, f->len);
- jb->last_format = f->subclass;
+ jb->last_format = f->codec;
ast_frfree(f);
break;
case JB_IMPL_INTERP:
/* interpolate a frame */
f = &finterp;
f->frametype = AST_FRAME_VOICE;
- f->subclass = jb->last_format;
+ f->subclass = 0;
+ f->codec = jb->last_format;
f->datalen = 0;
f->samples = interpolation_len * 8;
f->mallocd = 0;
@@ -474,7 +475,7 @@
jb->next = jbimpl->next(jbobj);
/* Init last format for a first time. */
- jb->last_format = frr->subclass;
+ jb->last_format = frr->codec;
/* Create a frame log file */
if (ast_test_flag(jbconf, AST_JB_LOG)) {
Modified: team/group/codec_bits/main/app.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/main/app.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/main/app.c (original)
+++ team/group/codec_bits/main/app.c Wed Apr 2 19:23:21 2008
@@ -300,15 +300,15 @@
int fd;
int autoclose;
int allowoverride;
- int origwfmt;
+ struct ast_extended_codec origwfmt;
};
static void linear_release(struct ast_channel *chan, void *params)
{
struct linear_state *ls = params;
- if (ls->origwfmt && ast_set_write_format(chan, ls->origwfmt))
- ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%d'\n", chan->name, ls->origwfmt);
+ if (FMT_NZ(ls->origwfmt) && ast_set_write_format(chan, ls->origwfmt))
+ ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%s'\n", chan->name, ast_getformatname(ls->origwfmt));
if (ls->autoclose)
close(ls->fd);
@@ -322,7 +322,8 @@
struct linear_state *ls = data;
struct ast_frame f = {
.frametype = AST_FRAME_VOICE,
- .subclass = AST_FORMAT_SLINEAR,
+ .subclass = 0,
+ .codec = AST_FMT_SLINEAR,
.data = buf + AST_FRIENDLY_OFFSET / 2,
.offset = AST_FRIENDLY_OFFSET,
};
@@ -359,7 +360,7 @@
ls->origwfmt = chan->writeformat;
- if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
+ if (ast_set_write_format(chan, AST_FMT_SLINEAR)) {
ast_log(LOG_WARNING, "Unable to set '%s' to linear format (write)\n", chan->name);
ast_free(ls);
ls = params = NULL;
@@ -572,7 +573,7 @@
time_t start, end;
struct ast_dsp *sildet = NULL; /* silence detector dsp */
int totalsilence = 0;
- int rfmt = 0;
+ struct ast_extended_codec rfmt = { { 0 } };
struct ast_silence_generator *silgen = NULL;
char prependfile[80];
@@ -640,7 +641,7 @@
}
ast_dsp_set_threshold(sildet, silencethreshold);
rfmt = chan->readformat;
- res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
+ res = ast_set_read_format(chan, AST_FMT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
ast_dsp_free(sildet);
@@ -812,7 +813,7 @@
ast_filedelete(prependfile, sfmt[x]);
}
}
- if (rfmt && ast_set_read_format(chan, rfmt)) {
+ if (FMT_NZ(rfmt) && ast_set_read_format(chan, rfmt)) {
ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(rfmt), chan->name);
}
if (outmsg == 2) {
Modified: team/group/codec_bits/main/audiohook.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/main/audiohook.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/main/audiohook.c (original)
+++ team/group/codec_bits/main/audiohook.c Wed Apr 2 19:23:21 2008
@@ -40,7 +40,7 @@
struct ast_audiohook_translate {
struct ast_trans_pvt *trans_pvt;
- int format;
+ struct ast_extended_codec format;
};
struct ast_audiohook_list {
@@ -148,7 +148,8 @@
short buf[samples];
struct ast_frame frame = {
.frametype = AST_FRAME_VOICE,
- .subclass = AST_FORMAT_SLINEAR,
+ .subclass = 0,
+ .codec = AST_FMT_SLINEAR,
.data = buf,
.datalen = sizeof(buf),
.samples = samples,
@@ -175,7 +176,8 @@
short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
struct ast_frame frame = {
.frametype = AST_FRAME_VOICE,
- .subclass = AST_FORMAT_SLINEAR,
+ .subclass = 0,
+ .codec = AST_FMT_SLINEAR,
.data = NULL,
.datalen = sizeof(buf1),
.samples = samples,
@@ -267,7 +269,7 @@
* \param format Format of frame remote side wants back
* \return Returns frame on success, NULL on failure
*/
-struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format)
+struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_extended_codec format)
{
struct ast_frame *read_frame = NULL, *final_frame = NULL;
@@ -275,15 +277,15 @@
return NULL;
/* If they don't want signed linear back out, we'll have to send it through the translation path */
- if (format != AST_FORMAT_SLINEAR) {
+ if (format.audio[0] != AST_FORMAT_AUDIO_SLINEAR) {
/* Rebuild translation path if different format then previously */
- if (audiohook->format != format) {
+ if (!FMT_EQ(audiohook->format, format)) {
if (audiohook->trans_pvt) {
ast_translator_free_path(audiohook->trans_pvt);
audiohook->trans_pvt = NULL;
}
/* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */
- if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
+ if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FMT_SLINEAR))) {
ast_frfree(read_frame);
return NULL;
}
@@ -491,13 +493,13 @@
int samples = frame->samples;
/* If the frame coming in is not signed linear we have to send it through the in_translate path */
- if (frame->subclass != AST_FORMAT_SLINEAR) {
- if (in_translate->format != frame->subclass) {
+ if (frame->codec.audio[0] != AST_FORMAT_AUDIO_SLINEAR) {
+ if (!FMT_EQ(in_translate->format, frame->codec)) {
if (in_translate->trans_pvt)
ast_translator_free_path(in_translate->trans_pvt);
- if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass)))
+ if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FMT_SLINEAR, frame->codec)))
return frame;
- in_translate->format = frame->subclass;
+ in_translate->format = frame->codec;
}
if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
return frame;
@@ -570,15 +572,15 @@
if (middle_frame == end_frame) {
/* Middle frame was modified and became the end frame... let's see if we need to transcode */
if (end_frame->subclass != start_frame->subclass) {
- if (out_translate->format != start_frame->subclass) {
+ if (!FMT_EQ(out_translate->format, start_frame->codec)) {
if (out_translate->trans_pvt)
ast_translator_free_path(out_translate->trans_pvt);
- if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass, AST_FORMAT_SLINEAR))) {
+ if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->codec, AST_FMT_SLINEAR))) {
/* We can't transcode this... drop our middle frame and return the original */
ast_frfree(middle_frame);
return start_frame;
}
- out_translate->format = start_frame->subclass;
+ out_translate->format = start_frame->codec;
}
/* Transcode from our middle (signed linear) frame to new format of the frame that came in */
if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
Modified: team/group/codec_bits/main/channel.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/main/channel.c?view=diff&rev=112560&r1=112559&r2=112560
==============================================================================
--- team/group/codec_bits/main/channel.c (original)
+++ team/group/codec_bits/main/channel.c Wed Apr 2 19:23:21 2008
@@ -268,6 +268,7 @@
static char *handle_cli_core_show_channeltype(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chanlist *cl = NULL;
+ struct ast_str *str = ast_str_alloca(BITSTRING_SIZE);
switch (cmd) {
case CLI_INIT:
@@ -302,7 +303,7 @@
" Device State: %s\n"
" Indication: %s\n"
" Transfer : %s\n"
- " Capabilities: %d\n"
+ " Capabilities: %s\n"
" Digit Begin: %s\n"
" Digit End: %s\n"
" Send HTML : %s\n"
@@ -312,7 +313,7 @@
(cl->tech->devicestate) ? "yes" : "no",
(cl->tech->indicate) ? "yes" : "no",
(cl->tech->transfer) ? "yes" : "no",
- (cl->tech->capabilities) ? cl->tech->capabilities : -1,
+ FMT_NZ(cl->tech->capabilities) ? ast_codec2bitstring(cl->tech->capabilities, &str) : "ALL",
(cl->tech->send_digit_begin) ? "yes" : "no",
(cl->tech->send_digit_end) ? "yes" : "no",
(cl->tech->send_html) ? "yes" : "no",
@@ -702,56 +703,62 @@
}
/*! \brief Pick the best audio codec */
-int ast_best_codec(int fmts)
+struct ast_extended_codec ast_best_codec(struct ast_extended_codec fmts)
{
/* This just our opinion, expressed in code. We are asked to choose
the best codec to use, given no information */
int x;
- static const int prefs[] =
- {
+ struct ast_extended_codec result = { { 0 } };
+ struct ast_str *str = ast_str_alloca(BITSTRING_SIZE);
+ static const struct {
+ int page;
+ int codec;
+ } prefs[] = {
/*! Okay, ulaw is used by all telephony equipment, so start with it */
- AST_FORMAT_ULAW,
+ { 0, AST_FORMAT_AUDIO_ULAW },
/*! Unless of course, you're a silly European, so then prefer ALAW */
- AST_FORMAT_ALAW,
+ { 0, AST_FORMAT_AUDIO_ALAW },
/*! G.722 is better then all below, but not as common as the above... so give ulaw and alaw priority */
- AST_FORMAT_G722,
+ { 0, AST_FORMAT_AUDIO_G722 },
/*! Okay, well, signed linear is easy to translate into other stuff */
- AST_FORMAT_SLINEAR16,
- AST_FORMAT_SLINEAR,
+ { 0, AST_FORMAT_AUDIO_SLINEAR16 },
+ { 0, AST_FORMAT_AUDIO_SLINEAR },
/*! G.726 is standard ADPCM, in RFC3551 packing order */
- AST_FORMAT_G726,
+ { 0, AST_FORMAT_AUDIO_G726 },
/*! G.726 is standard ADPCM, in AAL2 packing order */
- AST_FORMAT_G726_AAL2,
+ { 0, AST_FORMAT_AUDIO_G726_AAL2 },
/*! ADPCM has great sound quality and is still pretty easy to translate */
- AST_FORMAT_ADPCM,
+ { 0, AST_FORMAT_AUDIO_ADPCM },
/*! Okay, we're down to vocoders now, so pick GSM because it's small and easier to
translate and sounds pretty good */
- AST_FORMAT_GSM,
+ { 0, AST_FORMAT_AUDIO_GSM },
/*! iLBC is not too bad */
- AST_FORMAT_ILBC,
+ { 0, AST_FORMAT_AUDIO_ILBC },
/*! Speex is free, but computationally more expensive than GSM */
- AST_FORMAT_SPEEX,
+ { 0, AST_FORMAT_AUDIO_SPEEX },
/*! Ick, LPC10 sounds terrible, but at least we have code for it, if you're tacky enough
to use it */
- AST_FORMAT_LPC10,
+ { 0, AST_FORMAT_AUDIO_LPC10 },
/*! G.729a is faster than 723 and slightly less expensive */
- AST_FORMAT_G729A,
+ { 0, AST_FORMAT_AUDIO_G729A },
/*! Down to G.723.1 which is proprietary but at least designed for voice */
- AST_FORMAT_G723_1,
+ { 0, AST_FORMAT_AUDIO_G723_1 },
};
/* Strip out video */
- fmts &= AST_FORMAT_AUDIO_MASK;
+ fmts = FMT_AND(fmts, AST_FMT_AUDIO_MASK);
/* Find the first preferred codec in the format given */
for (x = 0; x < (sizeof(prefs) / sizeof(prefs[0]) ); x++) {
- if (fmts & prefs[x])
- return prefs[x];
- }
-
- ast_log(LOG_WARNING, "Don't know any of 0x%x formats\n", fmts);
-
- return 0;
+ if (fmts.audio[prefs[x].page] & prefs[x].codec) {
+ result.audio[prefs[x].page] = prefs[x].codec;
+ return result;
+ }
+ }
+
+ ast_log(LOG_WARNING, "Don't know any of %s formats\n", ast_codec2bitstring(fmts, &str));
+
+ return AST_FMT_NULL_MASK;
}
static const struct ast_channel_tech null_tech = {
@@ -1724,7 +1731,7 @@
if (!tmp || !generate)
return 0;
- res = generate(chan, tmp, 0, ast_format_rate(chan->writeformat & AST_FORMAT_AUDIO_MASK) / 50);
+ res = generate(chan, tmp, 0, ast_format_rate(FMT_AND(chan->writeformat, AST_FMT_AUDIO_MASK)) / 50);
chan->generatordata = tmp;
@@ -2266,9 +2273,9 @@
chan->generatordata = NULL; /* reset, to let writes go through */
- if (f->subclass != chan->writeformat) {
+ if (!FMT_EQ(f->codec, chan->writeformat)) {
float factor;
- factor = ((float) ast_format_rate(chan->writeformat)) / ((float) ast_format_rate(f->subclass));
+ factor = ((float) ast_format_rate(chan->writeformat)) / ((float) ast_format_rate(f->codec));
samples = (int) ( ((float) f->samples) * factor );
} else {
samples = f->samples;
@@ -2620,11 +2627,11 @@
ast_frfree(f);
f = &ast_null_frame;
}
- } else if ((f->frametype == AST_FRAME_VOICE) && !(f->subclass & chan->nativeformats)) {
+ } else if ((f->frametype == AST_FRAME_VOICE) && FMT_NOT(FMT_AND(f->codec, chan->nativeformats))) {
/* This frame can't be from the current native formats -- drop it on the
floor */
ast_log(LOG_NOTICE, "Dropping incompatible voice frame on %s of format %s since our native format has changed to %s\n",
- chan->name, ast_getformatname(f->subclass), ast_getformatname(chan->nativeformats));
+ chan->name, ast_getformatname(f->codec), ast_getformatname(chan->nativeformats));
ast_frfree(f);
f = &ast_null_frame;
} else if ((f->frametype == AST_FRAME_VOICE)) {
@@ -2906,7 +2913,8 @@
/* Send an empty audio frame to get things moving */
if (chan->_state != AST_STATE_UP) {
ast_debug(1, "Prodding channel '%s'\n", chan->name);
- a.subclass = chan->rawwriteformat;
+ a.subclass = 0;
+ a.codec = chan->rawwriteformat;
a.data = nothing + AST_FRIENDLY_OFFSET;
a.src = "ast_prod";
if (ast_write(chan, &a))
@@ -3014,7 +3022,7 @@
CHECK_BLOCKING(chan);
break;
case AST_FRAME_TEXT:
- if (fr->subclass == AST_FORMAT_T140) {
+ if (fr->codec.text[0] == AST_FORMAT_TEXT_T140) {
res = (chan->tech->write_text == NULL) ? 0 :
chan->tech->write_text(chan, fr);
} else {
@@ -3048,10 +3056,11 @@
}
/* If the frame is in the raw write format, then it's easy... just use the frame - otherwise we will have to translate */
- if (fr->subclass == chan->rawwriteformat)
+ if (FMT_EQ(fr->codec, chan->rawwriteformat)) {
f = fr;
- else
+ } else {
f = (chan->writetrans) ? ast_translate(chan->writetrans, fr, 0) : fr;
+ }
if (!f) {
res = 0;
@@ -3119,27 +3128,29 @@
return res;
}
-static int set_format(struct ast_channel *chan, int fmt, int *rawformat, int *format,
+static int set_format(struct ast_channel *chan, struct ast_extended_codec fmt, struct ast_extended_codec *rawformat, struct ast_extended_codec *format,
struct ast_trans_pvt **trans, const int direction)
{
- int native;
+ struct ast_extended_codec native;
int res;
/* Make sure we only consider audio */
- fmt &= AST_FORMAT_AUDIO_MASK;
+ fmt = FMT_AND(fmt, AST_FMT_AUDIO_MASK);
native = chan->nativeformats;
- if (!fmt || !native) /* No audio requested */
+ if (FMT_NOT(fmt) || FMT_NOT(native)) { /* No audio requested */
return 0; /* Let's try a call without any sounds (video, text) */
-
+ }
+
/* Find a translation path from the native format to one of the desired formats */
- if (!direction)
+ if (!direction) {
/* reading */
res = ast_translator_best_choice(&fmt, &native);
- else
+ } else {
/* writing */
res = ast_translator_best_choice(&native, &fmt);
+ }
if (res < 0) {
ast_log(LOG_WARNING, "Unable to find a codec translation path from %s to %s\n",
@@ -3150,7 +3161,7 @@
/* Now we have a good choice for both. */
ast_channel_lock(chan);
- if ((*rawformat == native) && (*format == fmt) && ((*rawformat == *format) || (*trans))) {
+ if (FMT_EQ(*rawformat, native) && FMT_EQ(*format, fmt) && (FMT_EQ(*rawformat, *format) || (*trans))) {
/* the channel is already in these formats, so nothing to do */
ast_channel_unlock(chan);
return 0;
@@ -3163,25 +3174,26 @@
if (*trans)
ast_translator_free_path(*trans);
/* Build a translation path from the raw format to the desired format */
- if (!direction)
+ if (!direction) {
/* reading */
*trans = ast_translator_build_path(*format, *rawformat);
- else
+ } else {
/* writing */
*trans = ast_translator_build_path(*rawformat, *format);
+ }
ast_channel_unlock(chan);
ast_debug(1, "Set channel %s to %s format %s\n", chan->name,
direction ? "write" : "read", ast_getformatname(fmt));
return 0;
}
-int ast_set_read_format(struct ast_channel *chan, int fmt)
+int ast_set_read_format(struct ast_channel *chan, struct ast_extended_codec fmt)
{
return set_format(chan, fmt, &chan->rawreadformat, &chan->readformat,
&chan->readtrans, 0);
}
-int ast_set_write_format(struct ast_channel *chan, int fmt)
+int ast_set_write_format(struct ast_channel *chan, struct ast_extended_codec fmt)
{
return set_format(chan, fmt, &chan->rawwriteformat, &chan->writeformat,
&chan->writetrans, 1);
@@ -3210,7 +3222,7 @@
}
}
-struct ast_channel *__ast_request_and_dial(const char *type, int format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name, struct outgoing_helper *oh)
+struct ast_channel *__ast_request_and_dial(const char *type, struct ast_extended_codec format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name, struct outgoing_helper *oh)
{
int dummy_outstate;
int cause = 0;
@@ -3341,21 +3353,21 @@
return chan;
}
-struct ast_channel *ast_request_and_dial(const char *type, int format, void *data, int timeout, int *outstate, const char *cidnum, const char *cidname)
+struct ast_channel *ast_request_and_dial(const char *type, struct ast_extended_codec format, void *data, int timeout, int *outstate, const char *cidnum, const char *cidname)
{
return __ast_request_and_dial(type, format, data, timeout, outstate, cidnum, cidname, NULL);
}
-struct ast_channel *ast_request(const char *type, int format, void *data, int *cause)
+struct ast_channel *ast_request(const char *type, struct ast_extended_codec format, void *data, int *cause)
{
struct chanlist *chan;
struct ast_channel *c;
- int capabilities;
- int fmt;
+ struct ast_extended_codec capabilities, fmt;
int res;
int foo;
- int videoformat = format & AST_FORMAT_VIDEO_MASK;
- int textformat = format & AST_FORMAT_TEXT_MASK;
+ struct ast_extended_codec videoformat = FMT_AND(format, AST_FMT_VIDEO_MASK);
+ struct ast_extended_codec textformat = FMT_AND(format, AST_FMT_TEXT_MASK);
+ struct ast_str *str[2] = { ast_str_alloca(BITSTRING_SIZE), ast_str_alloca(BITSTRING_SIZE) };
if (!cause)
cause = &foo;
@@ -3371,14 +3383,13 @@
continue;
capabilities = chan->tech->capabilities;
- fmt = format & AST_FORMAT_AUDIO_MASK;
- if (fmt) {
+ if (FMT_NZ(fmt = FMT_AND(format, AST_FMT_AUDIO_MASK))) {
/* We have audio - is it possible to connect the various calls to each other?
(Avoid this check for calls without audio, like text+video calls)
*/
res = ast_translator_best_choice(&fmt, &capabilities);
if (res < 0) {
- ast_log(LOG_WARNING, "No translator path exists for channel type %s (native 0x%x) to 0x%x\n", type, chan->tech->capabilities, format);
+ ast_log(LOG_WARNING, "No translator path exists for channel type %s (native %s) to %s\n", type, ast_codec2bitstring(chan->tech->capabilities, &str[0]), ast_codec2bitstring(format, &str[1]));
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
AST_RWLIST_UNLOCK(&channels);
return NULL;
@@ -3388,7 +3399,7 @@
if (!chan->tech->requester)
return NULL;
- if (!(c = chan->tech->requester(type, capabilities | videoformat | textformat, data, cause)))
+ if (!(c = chan->tech->requester(type, FMT_OR(capabilities, FMT_OR(videoformat, textformat)), data, cause)))
return NULL;
/* no need to generate a Newchannel event here; it is done in the channel_alloc call */
@@ -3512,10 +3523,11 @@
/*! \brief Set up translation from one channel to another */
static int ast_channel_make_compatible_helper(struct ast_channel *from, struct ast_channel *to)
{
- int src;
- int dst;
-
- if (from->readformat == to->writeformat && from->writeformat == to->readformat) {
+ struct ast_extended_codec src;
+ struct ast_extended_codec dst;
+ struct ast_str *str[2] = { ast_str_alloca(BITSTRING_SIZE), ast_str_alloca(BITSTRING_SIZE) };
+
+ if (FMT_EQ(from->readformat, to->writeformat) && FMT_EQ(from->writeformat, to->readformat)) {
/* Already compatible! Moving on ... */
return 0;
}
@@ -3525,11 +3537,12 @@
dst = to->nativeformats;
/* If there's no audio in this call, don't bother with trying to find a translation path */
- if ((src & AST_FORMAT_AUDIO_MASK) == 0 || (dst & AST_FORMAT_AUDIO_MASK) == 0)
+ if (FMT_NOT(FMT_AND(src, AST_FMT_AUDIO_MASK)) || FMT_NOT(FMT_AND(dst, AST_FMT_AUDIO_MASK))) {
return 0;
+ }
if (ast_translator_best_choice(&dst, &src) < 0) {
- ast_log(LOG_WARNING, "No path to translate from %s(%d) to %s(%d)\n", from->name, src, to->name, dst);
+ ast_log(LOG_WARNING, "No path to translate from %s(%s) to %s(%s)\n", from->name, ast_codec2bitstring(src, &str[0]), to->name, ast_codec2bitstring(dst, &str[1]));
return -1;
}
@@ -3537,15 +3550,15 @@
transcoding is needed; if desired, force transcode path
to use SLINEAR between channels, but only if there is
no direct conversion available */
- if ((src != dst) && ast_opt_transcode_via_slin &&
[... 1271 lines stripped ...]
More information about the asterisk-commits
mailing list