[svn-commits] tilghman: trunk r89071 - in /trunk: channels/ codecs/ include/asterisk/ main/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Nov 6 16:51:48 CST 2007
Author: tilghman
Date: Tue Nov 6 16:51:48 2007
New Revision: 89071
URL: http://svn.digium.com/view/asterisk?view=rev&rev=89071
Log:
Commit some cleanups to the format type code.
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
Modified:
trunk/channels/chan_gtalk.c
trunk/channels/chan_h323.c
trunk/channels/chan_jingle.c
trunk/channels/chan_mgcp.c
trunk/channels/chan_phone.c
trunk/channels/chan_sip.c
trunk/channels/chan_skinny.c
trunk/codecs/codec_g722.c
trunk/include/asterisk/frame.h
trunk/include/asterisk/translate.h
trunk/main/file.c
trunk/main/frame.c
trunk/main/rtp.c
trunk/main/translate.c
Modified: trunk/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_gtalk.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/channels/chan_gtalk.c (original)
+++ trunk/channels/chan_gtalk.c Tue Nov 6 16:51:48 2007
@@ -198,7 +198,7 @@
static const struct ast_channel_tech gtalk_tech = {
.type = "Gtalk",
.description = "Gtalk Channel Driver",
- .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+ .capabilities = AST_FORMAT_AUDIO_MASK,
.requester = gtalk_request,
.send_digit_begin = gtalk_digit_begin,
.send_digit_end = gtalk_digit_end,
Modified: trunk/channels/chan_h323.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_h323.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/channels/chan_h323.c (original)
+++ trunk/channels/chan_h323.c Tue Nov 6 16:51:48 2007
@@ -249,7 +249,7 @@
static const struct ast_channel_tech oh323_tech = {
.type = "H323",
.description = tdesc,
- .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+ .capabilities = AST_FORMAT_AUDIO_MASK,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = oh323_request,
.send_digit_begin = oh323_digit_begin,
@@ -1739,7 +1739,7 @@
return NULL;
}
oldformat = format;
- format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1);
+ format &= AST_FORMAT_AUDIO_MASK;
if (!format) {
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", format);
oh323_destroy(pvt);
Modified: trunk/channels/chan_jingle.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_jingle.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/channels/chan_jingle.c (original)
+++ trunk/channels/chan_jingle.c Tue Nov 6 16:51:48 2007
@@ -198,7 +198,7 @@
static const struct ast_channel_tech jingle_tech = {
.type = "Jingle",
.description = "Jingle Channel Driver",
- .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+ .capabilities = AST_FORMAT_AUDIO_MASK,
.requester = jingle_request,
.send_digit_begin = jingle_digit_begin,
.send_digit_end = jingle_digit_end,
@@ -340,10 +340,7 @@
continue;
if (alreadysent & pref_codec)
continue;
- if (pref_codec <= AST_FORMAT_MAX_AUDIO)
- add_codec_to_answer(p, pref_codec, dcodecs);
- else
- add_codec_to_answer(p, pref_codec, dcodecs);
+ add_codec_to_answer(p, pref_codec, dcodecs);
alreadysent |= pref_codec;
}
payload_red = iks_new("payload-type");
Modified: trunk/channels/chan_mgcp.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_mgcp.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/channels/chan_mgcp.c (original)
+++ trunk/channels/chan_mgcp.c Tue Nov 6 16:51:48 2007
@@ -2076,7 +2076,7 @@
snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));
ast_copy_string(t, "t=0 0\r\n", sizeof(t));
snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port));
- for (x = 1; x <= AST_FORMAT_MAX_AUDIO; x <<= 1) {
+ for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
if (p->capability & x) {
if (mgcpdebug) {
ast_verbose("Answering with capability %d\n", x);
@@ -2142,7 +2142,7 @@
return 0;
}
ast_copy_string(local, "p:20", sizeof(local));
- for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
+ for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
if (p->capability & x) {
snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
strncat(local, tmp, sizeof(local) - strlen(local) - 1);
@@ -2172,7 +2172,7 @@
struct mgcp_endpoint *p = sub->parent;
ast_copy_string(local, "p:20", sizeof(local));
- for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
+ for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
if (p->capability & x) {
snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
strncat(local, tmp, sizeof(local) - strlen(local) - 1);
Modified: trunk/channels/chan_phone.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_phone.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/channels/chan_phone.c (original)
+++ trunk/channels/chan_phone.c Tue Nov 6 16:51:48 2007
@@ -594,7 +594,7 @@
}
p->fr.samples = 240;
p->fr.datalen = res;
- p->fr.frametype = p->lastinput <= AST_FORMAT_MAX_AUDIO ?
+ p->fr.frametype = p->lastinput <= AST_FORMAT_AUDIO_MASK ?
AST_FRAME_VOICE :
p->lastinput <= AST_FORMAT_PNG ? AST_FRAME_IMAGE
: AST_FRAME_VIDEO;
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Nov 6 16:51:48 2007
@@ -154,8 +154,6 @@
#endif
#define XMIT_ERROR -2
-
-#define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
/* #define VOCAL_DATA_HACK */
@@ -7228,20 +7226,20 @@
}
/* Now send any other common audio and video codecs, and non-codec formats: */
- for (x = 1; x <= (needtext ? AST_FORMAT_MAX_TEXT : (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO)); x <<= 1) {
+ for (x = 1; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
if (!(capability & x)) /* Codec not requested */
continue;
if (alreadysent & x) /* Already added to SDP */
continue;
- if (x <= AST_FORMAT_MAX_AUDIO)
+ if (x & AST_FORMAT_AUDIO_MASK)
add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
&m_audio, &a_audio, debug, &min_audio_packet_size);
- else if (x <= AST_FORMAT_MAX_VIDEO)
+ else if (x & AST_FORMAT_VIDEO_MASK)
add_vcodec_to_sdp(p, x, 90000,
&m_video, &a_video, debug, &min_video_packet_size);
- else if (x <= AST_FORMAT_MAX_TEXT)
+ else if (x & AST_FORMAT_TEXT_MASK)
add_tcodec_to_sdp(p, x, 1000,
&m_text, &a_text, debug, &min_text_packet_size);
}
Modified: trunk/channels/chan_skinny.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_skinny.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/channels/chan_skinny.c (original)
+++ trunk/channels/chan_skinny.c Tue Nov 6 16:51:48 2007
@@ -1239,7 +1239,7 @@
static const struct ast_channel_tech skinny_tech = {
.type = "Skinny",
.description = tdesc,
- .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+ .capabilities = AST_FORMAT_AUDIO_MASK,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = skinny_request,
.devicestate = skinny_devicestate,
@@ -5395,7 +5395,7 @@
oldformat = format;
- if (!(format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1))) {
+ if (!(format &= AST_FORMAT_AUDIO_MASK)) {
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", format);
return NULL;
}
Modified: trunk/codecs/codec_g722.c
URL: http://svn.digium.com/view/asterisk/trunk/codecs/codec_g722.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/codecs/codec_g722.c (original)
+++ trunk/codecs/codec_g722.c Tue Nov 6 16:51:48 2007
@@ -74,12 +74,30 @@
return 0;
}
+static int lin16tog722_new(struct ast_trans_pvt *pvt)
+{
+ struct g722_encoder_pvt *tmp = pvt->pvt;
+
+ g722_encode_init(&tmp->g722, 64000, 0);
+
+ return 0;
+}
+
/*! \brief init a new instance of g722_encoder_pvt. */
static int g722tolin_new(struct ast_trans_pvt *pvt)
{
struct g722_decoder_pvt *tmp = pvt->pvt;
g722_decode_init(&tmp->g722, 64000, G722_SAMPLE_RATE_8000);
+
+ return 0;
+}
+
+static int g722tolin16_new(struct ast_trans_pvt *pvt)
+{
+ struct g722_decoder_pvt *tmp = pvt->pvt;
+
+ g722_decode_init(&tmp->g722, 64000, 0);
return 0;
}
@@ -125,11 +143,39 @@
return &f;
}
+static struct ast_frame *g722tolin16_sample(void)
+{
+ static struct ast_frame f = {
+ .frametype = AST_FRAME_VOICE,
+ .subclass = AST_FORMAT_G722,
+ .datalen = sizeof(slin_g722_ex),
+ .samples = sizeof(slin_g722_ex) / sizeof(slin_g722_ex[0]),
+ .src = __PRETTY_FUNCTION__,
+ .data = slin_g722_ex,
+ };
+
+ return &f;
+}
+
static struct ast_frame *lintog722_sample (void)
{
static struct ast_frame f = {
.frametype = AST_FRAME_VOICE,
.subclass = AST_FORMAT_SLINEAR,
+ .datalen = sizeof(slin_g722_ex),
+ .samples = sizeof(slin_g722_ex) / sizeof(slin_g722_ex[0]),
+ .src = __PRETTY_FUNCTION__,
+ .data = slin_g722_ex,
+ };
+
+ return &f;
+}
+
+static struct ast_frame *lin16tog722_sample (void)
+{
+ static struct ast_frame f = {
+ .frametype = AST_FRAME_VOICE,
+ .subclass = AST_FORMAT_SLINEAR16,
.datalen = sizeof(slin_g722_ex),
.samples = sizeof(slin_g722_ex) / sizeof(slin_g722_ex[0]),
.src = __PRETTY_FUNCTION__,
@@ -159,6 +205,31 @@
.newpvt = lintog722_new, /* same for both directions */
.framein = lintog722_framein,
.sample = lintog722_sample,
+ .desc_size = sizeof(struct g722_encoder_pvt),
+ .buffer_samples = BUFFER_SAMPLES,
+ .buf_size = BUFFER_SAMPLES,
+};
+
+static struct ast_translator g722tolin16 = {
+ .name = "g722tolin16",
+ .srcfmt = AST_FORMAT_G722,
+ .dstfmt = AST_FORMAT_SLINEAR16,
+ .newpvt = g722tolin16_new, /* same for both directions */
+ .framein = g722tolin_framein,
+ .sample = g722tolin16_sample,
+ .desc_size = sizeof(struct g722_decoder_pvt),
+ .buffer_samples = BUFFER_SAMPLES,
+ .buf_size = BUFFER_SAMPLES,
+ .plc_samples = 160,
+};
+
+static struct ast_translator lin16tog722 = {
+ .name = "lin16tog722",
+ .srcfmt = AST_FORMAT_SLINEAR16,
+ .dstfmt = AST_FORMAT_G722,
+ .newpvt = lin16tog722_new, /* same for both directions */
+ .framein = lintog722_framein,
+ .sample = lin16tog722_sample,
.desc_size = sizeof(struct g722_encoder_pvt),
.buffer_samples = BUFFER_SAMPLES,
.buf_size = BUFFER_SAMPLES,
@@ -198,6 +269,8 @@
res |= ast_unregister_translator(&g722tolin);
res |= ast_unregister_translator(&lintog722);
+ res |= ast_unregister_translator(&g722tolin16);
+ res |= ast_unregister_translator(&lin16tog722);
return res;
}
@@ -205,13 +278,14 @@
static int load_module(void)
{
int res = 0;
-
if (parse_config(0))
return AST_MODULE_LOAD_DECLINE;
res |= ast_register_translator(&g722tolin);
res |= ast_register_translator(&lintog722);
+ res |= ast_register_translator(&g722tolin16);
+ res |= ast_register_translator(&lin16tog722);
if (res) {
unload_module();
Modified: trunk/include/asterisk/frame.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/frame.h?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/include/asterisk/frame.h (original)
+++ trunk/include/asterisk/frame.h Tue Nov 6 16:51:48 2007
@@ -243,8 +243,8 @@
#define AST_FORMAT_G726 (1 << 11)
/*! G.722 */
#define AST_FORMAT_G722 (1 << 12)
-/*! Maximum audio format */
-#define AST_FORMAT_MAX_AUDIO (1 << 15)
+/*! Raw 16-bit Signed Linear (16000 Hz) PCM */
+#define AST_FORMAT_SLINEAR16 (1 << 15)
/*! Maximum audio mask */
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1)
/*! JPEG Images */
@@ -261,14 +261,10 @@
#define AST_FORMAT_H264 (1 << 21)
/*! MPEG4 Video */
#define AST_FORMAT_MP4_VIDEO (1 << 22)
-/*! Maximum video format */
-#define AST_FORMAT_MAX_VIDEO (1 << 24)
#define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK))
/*! T.140 Text format - ITU T.140, RFC 4351*/
#define AST_FORMAT_T140 (1 << 25)
-/*! Maximum text mask */
-#define AST_FORMAT_MAX_TEXT (1 << 26)
-#define AST_FORMAT_TEXT_MASK (((1 << 27)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))
+#define AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))
enum ast_control_frame_type {
AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */
@@ -361,9 +357,9 @@
/*! \brief Definition of supported media formats (codecs) */
struct ast_format_list {
- int visible; /*!< Can we see this entry */
int bits; /*!< bitmask value */
char *name; /*!< short name */
+ int samplespersecond; /*!< Number of samples per second (8000/16000) */
char *desc; /*!< Description */
int fr_len; /*!< Single frame length in bytes */
int min_ms; /*!< Min value */
Modified: trunk/include/asterisk/translate.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/translate.h?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/include/asterisk/translate.h (original)
+++ trunk/include/asterisk/translate.h Tue Nov 6 16:51:48 2007
@@ -44,7 +44,7 @@
* related to run-time operation (size of buffers, auxiliary
* descriptors, etc).
*
- * A coded registers itself by filling the relevant fields
+ * A codec registers itself by filling the relevant fields
* of a structure and passing it as an argument to
* ast_register_translator(). The structure should not be
* modified after a successful registration, and its address
Modified: trunk/main/file.c
URL: http://svn.digium.com/view/asterisk/trunk/main/file.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/main/file.c (original)
+++ trunk/main/file.c Tue Nov 6 16:51:48 2007
@@ -148,7 +148,7 @@
int res = -1;
int alt = 0;
if (f->frametype == AST_FRAME_VIDEO) {
- if (fs->fmt->format < AST_FORMAT_MAX_AUDIO) {
+ if (fs->fmt->format & AST_FORMAT_AUDIO_MASK) {
/* This is the audio portion. Call the video one... */
if (!fs->vfs && fs->filename) {
const char *type = ast_getformatname(f->subclass & ~0x1);
@@ -381,7 +381,7 @@
struct ast_filestream *s;
if ( !(chan->writeformat & f->format) &&
- !(f->format >= AST_FORMAT_MAX_AUDIO && fmt)) {
+ !(f->format & AST_FORMAT_AUDIO_MASK && fmt)) {
ast_free(fn);
continue; /* not a supported format */
}
@@ -407,7 +407,7 @@
s->fmt = f;
s->trans = NULL;
s->filename = NULL;
- if (s->fmt->format < AST_FORMAT_MAX_AUDIO) {
+ if (s->fmt->format & AST_FORMAT_AUDIO_MASK) {
if (chan->stream)
ast_closestream(chan->stream);
chan->stream = s;
@@ -577,7 +577,7 @@
if (buf == NULL)
return NULL;
- for (format = AST_FORMAT_MAX_AUDIO << 1; format <= AST_FORMAT_MAX_VIDEO; format = format << 1) {
+ for (format = AST_FORMAT_AUDIO_MASK + 1; format <= AST_FORMAT_VIDEO_MASK; format = format << 1) {
int fd;
const char *fmt;
@@ -710,7 +710,7 @@
{
enum fsread_res res;
- if (s->fmt->format < AST_FORMAT_MAX_AUDIO)
+ if (s->fmt->format & AST_FORMAT_AUDIO_MASK)
res = ast_readaudio_callback(s);
else
res = ast_readvideo_callback(s);
@@ -749,7 +749,7 @@
size_t size = 0;
/* Stop a running stream if there is one */
if (f->owner) {
- if (f->fmt->format < AST_FORMAT_MAX_AUDIO) {
+ if (f->fmt->format & AST_FORMAT_AUDIO_MASK) {
f->owner->stream = NULL;
if (f->owner->streamid > -1)
ast_sched_del(f->owner->sched, f->owner->streamid);
Modified: trunk/main/frame.c
URL: http://svn.digium.com/view/asterisk/trunk/main/frame.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/main/frame.c (original)
+++ trunk/main/frame.c Tue Nov 6 16:51:48 2007
@@ -104,30 +104,28 @@
/*! \brief Definition of supported media formats (codecs) */
static struct ast_format_list AST_FORMAT_LIST[] = {
- { 1, AST_FORMAT_G723_1 , "g723" , "G.723.1", 20, 30, 300, 30, 30 }, /*!< G723.1 */
- { 1, AST_FORMAT_GSM, "gsm" , "GSM", 33, 20, 300, 20, 20 }, /*!< codec_gsm.c */
- { 1, AST_FORMAT_ULAW, "ulaw", "G.711 u-law", 80, 10, 150, 10, 20 }, /*!< codec_ulaw.c */
- { 1, AST_FORMAT_ALAW, "alaw", "G.711 A-law", 80, 10, 150, 10, 20 }, /*!< codec_alaw.c */
- { 1, AST_FORMAT_G726, "g726", "G.726 RFC3551", 40, 10, 300, 10, 20 }, /*!< codec_g726.c */
- { 1, AST_FORMAT_ADPCM, "adpcm" , "ADPCM", 40, 10, 300, 10, 20 }, /*!< codec_adpcm.c */
- { 1, AST_FORMAT_SLINEAR, "slin", "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE }, /*!< Signed linear */
- { 1, AST_FORMAT_LPC10, "lpc10", "LPC10", 7, 20, 20, 20, 20 }, /*!< codec_lpc10.c */
- { 1, AST_FORMAT_G729A, "g729", "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 }, /*!< Binary commercial distribution */
- { 1, AST_FORMAT_SPEEX, "speex", "SpeeX", 10, 10, 60, 10, 20 }, /*!< codec_speex.c */
- { 1, AST_FORMAT_ILBC, "ilbc", "iLBC", 50, 30, 30, 30, 30 }, /*!< codec_ilbc.c */ /* inc=30ms - workaround */
- { 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2", 40, 10, 300, 10, 20 }, /*!< codec_g726.c */
- { 1, AST_FORMAT_G722, "g722", "G722", 80, 10, 150, 10, 20 }, /*!< G722 Passthrough */
- { 0, AST_FORMAT_MAX_AUDIO, "maxaudio", "Maximum audio format" },
- { 1, AST_FORMAT_JPEG, "jpeg", "JPEG image"}, /*!< See format_jpeg.c */
- { 1, AST_FORMAT_PNG, "png", "PNG image"}, /*!< PNG Image format */
- { 1, AST_FORMAT_H261, "h261", "H.261 Video" }, /*!< H.261 Video Passthrough */
- { 1, AST_FORMAT_H263, "h263", "H.263 Video" }, /*!< H.263 Passthrough support, see format_h263.c */
- { 1, AST_FORMAT_H263_PLUS, "h263p", "H.263+ Video" }, /*!< H.263plus passthrough support See format_h263.c */
- { 1, AST_FORMAT_H264, "h264", "H.264 Video" }, /*!< Passthrough support, see format_h263.c */
- { 1, AST_FORMAT_MP4_VIDEO, "mpeg4", "MPEG4 Video" }, /*!< Passthrough support for MPEG4 */
- { 0, AST_FORMAT_MAX_VIDEO, "maxvideo", "Maximum video format" },
- { 1, AST_FORMAT_T140, "t140", "Passthrough T.140 Realtime Text" }, /*!< Passthrough support for T.140 Realtime Text */
- { 0, AST_FORMAT_MAX_TEXT, "maxtext", "Maximum text format" },
+ { AST_FORMAT_G723_1 , "g723", 8000, "G.723.1", 20, 30, 300, 30, 30 }, /*!< G723.1 */
+ { AST_FORMAT_GSM, "gsm", 8000, "GSM", 33, 20, 300, 20, 20 }, /*!< codec_gsm.c */
+ { AST_FORMAT_ULAW, "ulaw", 8000, "G.711 u-law", 80, 10, 150, 10, 20 }, /*!< codec_ulaw.c */
+ { AST_FORMAT_ALAW, "alaw", 8000, "G.711 A-law", 80, 10, 150, 10, 20 }, /*!< codec_alaw.c */
+ { AST_FORMAT_G726, "g726", 8000, "G.726 RFC3551", 40, 10, 300, 10, 20 }, /*!< codec_g726.c */
+ { AST_FORMAT_ADPCM, "adpcm" , 8000, "ADPCM", 40, 10, 300, 10, 20 }, /*!< codec_adpcm.c */
+ { AST_FORMAT_SLINEAR, "slin", 8000, "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE }, /*!< Signed linear */
+ { AST_FORMAT_LPC10, "lpc10", 8000, "LPC10", 7, 20, 20, 20, 20 }, /*!< codec_lpc10.c */
+ { AST_FORMAT_G729A, "g729", 8000, "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 }, /*!< Binary commercial distribution */
+ { AST_FORMAT_SPEEX, "speex", 8000, "SpeeX", 10, 10, 60, 10, 20 }, /*!< codec_speex.c */
+ { AST_FORMAT_ILBC, "ilbc", 8000, "iLBC", 50, 30, 30, 30, 30 }, /*!< codec_ilbc.c */ /* inc=30ms - workaround */
+ { AST_FORMAT_G726_AAL2, "g726aal2", 8000, "G.726 AAL2", 40, 10, 300, 10, 20 }, /*!< codec_g726.c */
+ { AST_FORMAT_G722, "g722", 16000, "G722", 80, 10, 150, 10, 20 }, /*!< codec_g722.c */
+ { AST_FORMAT_SLINEAR16, "slin16", 16000, "16 bit Signed Linear PCM (16kHz)", 320, 10, 70, 10, 20 }, /*!< Signed linear (16kHz) */
+ { AST_FORMAT_JPEG, "jpeg", 0, "JPEG image"}, /*!< See format_jpeg.c */
+ { AST_FORMAT_PNG, "png", 0, "PNG image"}, /*!< PNG Image format */
+ { AST_FORMAT_H261, "h261", 0, "H.261 Video" }, /*!< H.261 Video Passthrough */
+ { AST_FORMAT_H263, "h263", 0, "H.263 Video" }, /*!< H.263 Passthrough support, see format_h263.c */
+ { AST_FORMAT_H263_PLUS, "h263p", 0, "H.263+ Video" }, /*!< H.263plus passthrough support See format_h263.c */
+ { AST_FORMAT_H264, "h264", 0, "H.264 Video" }, /*!< Passthrough support, see format_h263.c */
+ { AST_FORMAT_MP4_VIDEO, "mpeg4", 0, "MPEG4 Video" }, /*!< Passthrough support for MPEG4 */
+ { AST_FORMAT_T140, "t140", 0, "Passthrough T.140 Realtime Text" }, /*!< Passthrough support for T.140 Realtime Text */
};
struct ast_frame ast_null_frame = { AST_FRAME_NULL, };
@@ -525,7 +523,7 @@
int x;
char *ret = "unknown";
for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
- if (AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) {
+ if (AST_FORMAT_LIST[x].bits == format) {
ret = AST_FORMAT_LIST[x].name;
break;
}
@@ -547,7 +545,7 @@
size -= len;
start = end;
for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
- if (AST_FORMAT_LIST[x].visible && (AST_FORMAT_LIST[x].bits & format)) {
+ if (AST_FORMAT_LIST[x].bits & format) {
snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name);
len = strlen(end);
end += len;
@@ -566,6 +564,7 @@
char *realname;
} ast_codec_alias_table[] = {
{ "slinear", "slin"},
+ { "slinear16", "slin16"},
{ "g723.1", "g723"},
};
@@ -586,9 +585,9 @@
all = strcasecmp(name, "all") ? 0 : 1;
for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
- if (AST_FORMAT_LIST[x].visible && (all ||
+ if (all ||
!strcasecmp(AST_FORMAT_LIST[x].name,name) ||
- !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) {
+ !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name))) {
format |= AST_FORMAT_LIST[x].bits;
if (!all)
break;
@@ -603,7 +602,7 @@
int x;
char *ret = "unknown";
for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
- if (AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) {
+ if (AST_FORMAT_LIST[x].bits == codec) {
ret = AST_FORMAT_LIST[x].desc;
break;
}
@@ -1357,7 +1356,7 @@
samples = speex_samples(f->data, f->datalen);
break;
case AST_FORMAT_G723_1:
- samples = g723_samples(f->data, f->datalen);
+ samples = g723_samples(f->data, f->datalen);
break;
case AST_FORMAT_ILBC:
samples = 240 * (f->datalen / 50);
@@ -1369,10 +1368,11 @@
samples = f->datalen * 8;
break;
case AST_FORMAT_SLINEAR:
+ case AST_FORMAT_SLINEAR16:
samples = f->datalen / 2;
break;
case AST_FORMAT_LPC10:
- /* assumes that the RTP packet contains one LPC10 frame */
+ /* assumes that the RTP packet contains one LPC10 frame */
samples = 22 * 8;
samples += (((char *)(f->data))[7] & 0x1) * 8;
break;
@@ -1411,10 +1411,12 @@
len = samples / 8;
break;
case AST_FORMAT_SLINEAR:
+ case AST_FORMAT_SLINEAR16:
len = samples * 2;
break;
case AST_FORMAT_ULAW:
case AST_FORMAT_ALAW:
+ case AST_FORMAT_G722:
len = samples;
break;
case AST_FORMAT_ADPCM:
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Tue Nov 6 16:51:48 2007
@@ -1571,7 +1571,7 @@
return f ? f : &ast_null_frame;
}
rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
- rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : (rtp->f.subclass < AST_FORMAT_MAX_VIDEO) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
+ rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
if (!rtp->lastrxts)
rtp->lastrxts = timestamp;
@@ -1586,7 +1586,7 @@
rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.seqno = seqno;
- if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
+ if (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) {
rtp->f.samples = ast_codec_get_samples(&rtp->f);
if (rtp->f.subclass == AST_FORMAT_SLINEAR)
ast_frame_byteswap_be(&rtp->f);
@@ -1595,7 +1595,7 @@
rtp->f.has_timing_info = 1;
rtp->f.ts = timestamp / 8;
rtp->f.len = rtp->f.samples / 8;
- } else if(rtp->f.subclass < AST_FORMAT_MAX_VIDEO) {
+ } else if(rtp->f.subclass & AST_FORMAT_VIDEO_MASK) {
/* Video -- samples is # of samples vs. 90000 */
if (!rtp->lastividtimestamp)
rtp->lastividtimestamp = timestamp;
@@ -2958,7 +2958,7 @@
ms = calc_txstamp(rtp, &f->delivery);
/* Default prediction */
- if (f->subclass < AST_FORMAT_MAX_AUDIO) {
+ if (f->subclass & AST_FORMAT_AUDIO_MASK) {
pred = rtp->lastts + f->samples;
/* Re-calculate last TS */
@@ -2973,7 +2973,7 @@
mark = 1;
}
}
- } else if(f->subclass < AST_FORMAT_MAX_VIDEO) {
+ } else if(f->subclass & AST_FORMAT_VIDEO_MASK) {
mark = f->subclass & 0x1;
pred = rtp->lastovidtimestamp + f->samples;
/* Re-calculate last TS */
Modified: trunk/main/translate.c
URL: http://svn.digium.com/view/asterisk/trunk/main/translate.c?view=diff&rev=89071&r1=89070&r2=89071
==============================================================================
--- trunk/main/translate.c (original)
+++ trunk/main/translate.c Tue Nov 6 16:51:48 2007
@@ -806,7 +806,7 @@
known audio formats to determine whether there exists
a translation path from the source format to the
destination format. */
- for (x = 1; src_audio && x < AST_FORMAT_MAX_AUDIO; x <<= 1) {
+ for (x = 1; src_audio && (x & AST_FORMAT_AUDIO_MASK); x <<= 1) {
/* if this is not a desired format, nothing to do */
if (!dest & x)
continue;
@@ -832,7 +832,7 @@
known video formats to determine whether there exists
a translation path from the source format to the
destination format. */
- for (; src_video && x < AST_FORMAT_MAX_VIDEO; x <<= 1) {
+ for (; src_video && (x & AST_FORMAT_VIDEO_MASK); x <<= 1) {
/* if this is not a desired format, nothing to do */
if (!dest & x)
continue;
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