[asterisk-commits] mjordan: trunk r397526 - in /trunk: channels/ include/asterisk/ main/ res/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Aug 23 10:42:33 CDT 2013
Author: mjordan
Date: Fri Aug 23 10:42:27 2013
New Revision: 397526
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=397526
Log:
Add pass through support for Opus and VP8; Opus format attribute negotiation
This patch adds pass through support for Opus and VP8. That includes:
* Format attribute negotiation for Opus. Note that unlike some other codecs,
the draft RFC specifies having spaces delimiting the attributes in addition
to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
chan_sip, so a small tweak was also included in this patch for that.
* A format attribute negotiation module for Opus, res_format_attr_opus
* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time.
Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.
Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)
Added:
trunk/include/asterisk/opus.h (with props)
trunk/res/res_format_attr_opus.c (with props)
Modified:
trunk/channels/chan_pjsip.c
trunk/channels/chan_sip.c
trunk/include/asterisk/format.h
trunk/main/channel.c
trunk/main/format.c
trunk/main/frame.c
trunk/main/rtp_engine.c
trunk/res/res_pjsip_sdp_rtp.c
trunk/res/res_rtp_asterisk.c
Modified: trunk/channels/chan_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_pjsip.c?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/channels/chan_pjsip.c (original)
+++ trunk/channels/chan_pjsip.c Fri Aug 23 10:42:27 2013
@@ -1168,10 +1168,25 @@
case AST_CONTROL_VIDUPDATE:
media = pvt->media[SIP_MEDIA_VIDEO];
if (media && media->rtp) {
- ao2_ref(channel->session, +1);
-
- if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
- ao2_cleanup(channel->session);
+ /* FIXME: Only use this for VP8. Additional work would have to be done to
+ * fully support other video codecs */
+ struct ast_format_cap *fcap = ast_channel_nativeformats(ast);
+ struct ast_format vp8;
+ ast_format_set(&vp8, AST_FORMAT_VP8, 0);
+ if (ast_format_cap_iscompatible(fcap, &vp8)) {
+ /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
+ * RTP engine would provide a way to externally write/schedule RTCP
+ * packets */
+ struct ast_frame fr;
+ fr.frametype = AST_FRAME_CONTROL;
+ fr.subclass.integer = AST_CONTROL_VIDUPDATE;
+ res = ast_rtp_instance_write(media->rtp, &fr);
+ } else {
+ ao2_ref(channel->session, +1);
+
+ if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
+ ao2_cleanup(channel->session);
+ }
}
} else {
res = -1;
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Aug 23 10:42:27 2013
@@ -1269,7 +1269,7 @@
static void start_ice(struct ast_rtp_instance *instance);
static void add_codec_to_sdp(const struct sip_pvt *p, struct ast_format *codec,
struct ast_str **m_buf, struct ast_str **a_buf,
- int debug, int *min_packet_size);
+ int debug, int *min_packet_size, int *max_packet_size);
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug);
@@ -7945,10 +7945,25 @@
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
if (p->vrtp && !p->novideo) {
- transmit_info_with_vidupdate(p);
- /* ast_rtcp_send_h261fur(p->vrtp); */
- } else
+ /* FIXME: Only use this for VP8. Additional work would have to be done to
+ * fully support other video codecs */
+ struct ast_format_cap *fcap = ast_channel_nativeformats(ast);
+ struct ast_format vp8;
+ ast_format_set(&vp8, AST_FORMAT_VP8, 0);
+ if (ast_format_cap_iscompatible(fcap, &vp8)) {
+ /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
+ * RTP engine would provide a way to externally write/schedule RTCP
+ * packets */
+ struct ast_frame fr;
+ fr.frametype = AST_FRAME_CONTROL;
+ fr.subclass.integer = AST_CONTROL_VIDUPDATE;
+ res = ast_rtp_instance_write(p->vrtp, &fr);
+ } else {
+ transmit_info_with_vidupdate(p);
+ }
+ } else {
res = -1;
+ }
break;
case AST_CONTROL_T38_PARAMETERS:
res = -1;
@@ -11167,7 +11182,7 @@
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
}
- } else if (sscanf(a, "fmtp: %30u %255s", &codec, fmtp_string) == 2) {
+ } else if (sscanf(a, "fmtp: %30u %255[^\t\n]", &codec, fmtp_string) == 2) {
struct ast_format *format;
if ((format = ast_rtp_codecs_get_payload_format(newaudiortp, codec))) {
@@ -11230,7 +11245,8 @@
/* We have a rtpmap to handle */
if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
/* Note: should really look at the '#chans' params too */
- if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {
+ if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)
+ || !strncasecmp(mimeSubtype, "VP8", 3)) {
if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate))) {
if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
@@ -12799,7 +12815,8 @@
struct ast_str **m_buf,
struct ast_str **a_buf,
int debug,
- int *min_packet_size)
+ int *min_packet_size,
+ int *max_packet_size)
{
int rtp_code;
struct ast_format_list fmt;
@@ -12821,7 +12838,12 @@
} else /* I don't see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
- ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, mime, rate);
+ /* Opus mandates 2 channels in rtpmap */
+ if ((int)format->id == AST_FORMAT_OPUS) {
+ ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d/2\r\n", rtp_code, mime, rate);
+ } else {
+ ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, mime, rate);
+ }
ast_format_sdp_generate(format, rtp_code, a_buf);
@@ -12852,12 +12874,22 @@
break;
}
- if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
+ if (max_packet_size && fmt.max_ms && (fmt.max_ms < *max_packet_size)) {
+ *max_packet_size = fmt.max_ms;
+ }
+
+ if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size)) {
*min_packet_size = fmt.cur_ms;
+ }
/* Our first codec packetization processed cannot be zero */
- if ((*min_packet_size)==0 && fmt.cur_ms)
+ if ((*min_packet_size) == 0 && fmt.cur_ms) {
*min_packet_size = fmt.cur_ms;
+ }
+
+ if ((*max_packet_size) == 0 && fmt.max_ms) {
+ *max_packet_size = fmt.max_ms;
+ }
}
/*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
@@ -12884,6 +12916,10 @@
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, subtype, rate);
+ /* VP8: add RTCP FIR support */
+ if ((int)format->id == AST_FORMAT_VP8) {
+ ast_str_append(a_buf, 0, "a=rtcp-fb:* ccm fir\r\n");
+ }
ast_format_sdp_generate(format, rtp_code, a_buf);
}
@@ -13128,6 +13164,7 @@
int needtext = FALSE;
int debug = sip_debug_test_pvt(p);
int min_audio_packet_size = 0;
+ int max_audio_packet_size = 0;
int min_video_packet_size = 0;
int min_text_packet_size = 0;
@@ -13309,7 +13346,7 @@
if (AST_FORMAT_GET_TYPE(tmp_fmt.id) != AST_FORMAT_TYPE_AUDIO) {
continue;
}
- add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size);
+ add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size);
ast_format_cap_add(alreadysent, &tmp_fmt);
}
ast_format_cap_iter_end(p->prefcaps);
@@ -13329,7 +13366,7 @@
continue;
if (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_AUDIO) {
- add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size);
+ add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size);
} else if (needvideo && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_VIDEO)) {
add_vcodec_to_sdp(p, &tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size);
} else if (needtext && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_TEXT)) {
@@ -13346,7 +13383,7 @@
continue;
if (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_AUDIO) {
- add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size);
+ add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size);
} else if (needvideo && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_VIDEO)) {
add_vcodec_to_sdp(p, &tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size);
} else if (needtext && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_TEXT)) {
@@ -13365,19 +13402,27 @@
ast_debug(3, "-- Done with adding codecs to SDP\n");
- if (!p->owner || !ast_internal_timing_enabled(p->owner))
+ if (!p->owner || !ast_internal_timing_enabled(p->owner)) {
ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");
-
- if (min_audio_packet_size)
+ }
+
+ if (min_audio_packet_size) {
ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);
+ }
/* XXX don't think you can have ptime for video */
- if (min_video_packet_size)
+ if (min_video_packet_size) {
ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);
+ }
/* XXX don't think you can have ptime for text */
- if (min_text_packet_size)
+ if (min_text_packet_size) {
ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size);
+ }
+
+ if (max_audio_packet_size) {
+ ast_str_append(&a_text, 0, "a=maxptime:%d\r\n", max_audio_packet_size);
+ }
if (!doing_directmedia) {
if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
Modified: trunk/include/asterisk/format.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/format.h?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/include/asterisk/format.h (original)
+++ trunk/include/asterisk/format.h Fri Aug 23 10:42:27 2013
@@ -29,6 +29,7 @@
#include "asterisk/astobj2.h"
#include "asterisk/silk.h"
#include "asterisk/celt.h"
+#include "asterisk/opus.h"
#define AST_FORMAT_ATTR_SIZE 64
#define AST_FORMAT_INC 100000
@@ -101,6 +102,8 @@
AST_FORMAT_SLINEAR192 = 27 + AST_FORMAT_TYPE_AUDIO,
AST_FORMAT_SPEEX32 = 28 + AST_FORMAT_TYPE_AUDIO,
AST_FORMAT_CELT = 29 + AST_FORMAT_TYPE_AUDIO,
+ /*! Opus */
+ AST_FORMAT_OPUS = 30 + AST_FORMAT_TYPE_AUDIO,
/*! H.261 Video */
AST_FORMAT_H261 = 1 + AST_FORMAT_TYPE_VIDEO,
@@ -112,6 +115,8 @@
AST_FORMAT_H264 = 4 + AST_FORMAT_TYPE_VIDEO,
/*! MPEG4 Video */
AST_FORMAT_MP4_VIDEO = 5 + AST_FORMAT_TYPE_VIDEO,
+ /*! VP8 */
+ AST_FORMAT_VP8 = 6 + AST_FORMAT_TYPE_VIDEO,
/*! JPEG Images */
AST_FORMAT_JPEG = 1 + AST_FORMAT_TYPE_IMAGE,
Added: trunk/include/asterisk/opus.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/opus.h?view=auto&rev=397526
==============================================================================
--- trunk/include/asterisk/opus.h (added)
+++ trunk/include/asterisk/opus.h Fri Aug 23 10:42:27 2013
@@ -1,0 +1,41 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * Lorenzo Miniero <lorenzo at meetecho.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief Opus Format Attributes (http://tools.ietf.org/html/draft-ietf-payload-rtp-opus)
+ *
+ * \author Lorenzo Miniero <lorenzo at meetecho.com>
+ */
+#ifndef _AST_FORMAT_OPUS_H_
+#define _AST_FORMAT_OPUS_H_
+
+/*! Opus format attribute key value pairs, all are accessible through ast_format_get_value()*/
+enum opus_attr_keys {
+ OPUS_ATTR_KEY_MAX_BITRATE, /*! value is an int (6000-510000 in spec). */
+ OPUS_ATTR_KEY_MAX_PLAYRATE, /*! value is an int (8000-48000), maximum output rate the receiver can render. */
+ OPUS_ATTR_KEY_MINPTIME, /*! value is an int (3-120 in spec, 10-60 in format.c), decoder's minimum length of time in milliseconds. */
+ OPUS_ATTR_KEY_STEREO, /*! value is an int, 1 prefer receiving stereo, 0 prefer mono. */
+ OPUS_ATTR_KEY_CBR, /*! value is an int, 1 use constant bitrate, 0 use variable bitrate. */
+ OPUS_ATTR_KEY_FEC, /*! value is an int, 1 encode with FEC, 0 do not use FEC. */
+ OPUS_ATTR_KEY_DTX, /*! value is an int, 1 dtx is enabled, 0 dtx not enabled. */
+ OPUS_ATTR_KEY_SPROP_CAPTURE_RATE, /*! value is an int (8000-48000), likely input rate we're going to produce. */
+ OPUS_ATTR_KEY_SPROP_STEREO, /*! value is an int, 1 likely to send stereo, 0 likely to send mono. */
+};
+
+#endif /* _AST_FORMAT_OPUS_H */
Propchange: trunk/include/asterisk/opus.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/include/asterisk/opus.h
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: trunk/include/asterisk/opus.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: trunk/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/channel.c?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/main/channel.c (original)
+++ trunk/main/channel.c Fri Aug 23 10:42:27 2013
@@ -802,6 +802,8 @@
AST_FORMAT_SPEEX32,
AST_FORMAT_SPEEX16,
AST_FORMAT_SPEEX,
+ /*! Opus */
+ AST_FORMAT_OPUS,
/*! SILK is pretty awesome. */
AST_FORMAT_SILK,
/*! CELT supports crazy high sample rates */
Modified: trunk/main/format.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/format.c?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/main/format.c (original)
+++ trunk/main/format.c Fri Aug 23 10:42:27 2013
@@ -430,6 +430,9 @@
/*! SpeeX Wideband (16kHz) Free Compression */
case AST_FORMAT_SPEEX16:
return (1ULL << 33);
+ /*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
+ case AST_FORMAT_OPUS:
+ return (1ULL << 34);
/*! Raw mu-law data (G.711) */
case AST_FORMAT_TESTLAW:
return (1ULL << 47);
@@ -449,6 +452,9 @@
/*! MPEG4 Video */
case AST_FORMAT_MP4_VIDEO:
return (1ULL << 22);
+ /*! VP8 Video */
+ case AST_FORMAT_VP8:
+ return (1ULL << 23);
/*! JPEG Images */
case AST_FORMAT_JPEG:
@@ -532,6 +538,9 @@
/*! SpeeX Wideband (16kHz) Free Compression */
case (1ULL << 33):
return ast_format_set(dst, AST_FORMAT_SPEEX16, 0);
+ /*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
+ case (1ULL << 34):
+ return ast_format_set(dst, AST_FORMAT_OPUS, 0);
/*! Raw mu-law data (G.711) */
case (1ULL << 47):
return ast_format_set(dst, AST_FORMAT_TESTLAW, 0);
@@ -551,6 +560,9 @@
/*! MPEG4 Video */
case (1ULL << 22):
return ast_format_set(dst, AST_FORMAT_MP4_VIDEO, 0);
+ /*! VP8 Video */
+ case (1ULL << 23):
+ return ast_format_set(dst, AST_FORMAT_VP8, 0);
/*! JPEG Images */
case (1ULL << 16):
@@ -782,6 +794,9 @@
return samplerate;
}
}
+ /* Opus */
+ case AST_FORMAT_OPUS:
+ return 48000;
default:
return 8000;
}
@@ -1067,6 +1082,10 @@
format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0), "slin48", 48000, "16 bit Signed Linear PCM (48kHz)", 960, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (48kHz) */
format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR96, 0), "slin96", 96000, "16 bit Signed Linear PCM (96kHz)", 1920, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (96kHz) */
format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR192, 0), "slin192", 192000, "16 bit Signed Linear PCM (192kHz)", 3840, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (192kHz) */
+ /* Opus (FIXME: real min is 3/5/10, real max is 120...) */
+ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), "opus", 48000, "Opus Codec", 10, 20, 60, 20, 20, 0, 0); /*!< codec_opus.c */
+ /* VP8 (passthrough) */
+ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), "vp8", 0, "VP8 Video", 0, 0, 0, 0 ,0 ,0, 0); /*!< Passthrough support, see format_h263.c */
return 0;
}
Modified: trunk/main/frame.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/frame.c?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/main/frame.c (original)
+++ trunk/main/frame.c Fri Aug 23 10:42:27 2013
@@ -1097,8 +1097,12 @@
return 160;
}
case AST_FORMAT_CELT:
- /* TODO The assumes 20ms delivery right now, which is incorrect */
+ /* TODO This assumes 20ms delivery right now, which is incorrect */
samples = ast_format_rate(&f->subclass.format) / 50;
+ break;
+ case AST_FORMAT_OPUS:
+ /* TODO This assumes 20ms delivery right now, which is incorrect */
+ samples = 960;
break;
default:
ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
Modified: trunk/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/rtp_engine.c?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/main/rtp_engine.c (original)
+++ trunk/main/rtp_engine.c Fri Aug 23 10:42:27 2013
@@ -1977,6 +1977,9 @@
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
+ /* Opus and VP8 */
+ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0, "audio", "opus", 48000);
+ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0, "video", "VP8", 90000);
/* Define the static rtp payload mappings */
add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
@@ -2018,6 +2021,9 @@
add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
add_static_payload(121, NULL, AST_RTP_CISCO_DTMF); /* Must be type 121 */
+ /* Opus and VP8 */
+ add_static_payload(100, ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0);
+ add_static_payload(107, ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0);
return 0;
}
Added: trunk/res/res_format_attr_opus.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_format_attr_opus.c?view=auto&rev=397526
==============================================================================
--- trunk/res/res_format_attr_opus.c (added)
+++ trunk/res/res_format_attr_opus.c Fri Aug 23 10:42:27 2013
@@ -1,0 +1,321 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * Lorenzo Miniero <lorenzo at meetecho.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief Opus format attribute interface
+ *
+ * \author Lorenzo Miniero <lorenzo at meetecho.com>
+ */
+
+/*** MODULEINFO
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/module.h"
+#include "asterisk/format.h"
+
+/*!
+ * \brief Opus attribute structure.
+ *
+ * \note http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-00.
+ */
+struct opus_attr {
+ unsigned int maxbitrate; /* Default 64-128 kb/s for FB stereo music */
+ unsigned int maxplayrate /* Default 48000 */;
+ unsigned int minptime; /* Default 3, but it's 10 in format.c */
+ unsigned int stereo; /* Default 0 */
+ unsigned int cbr; /* Default 0 */
+ unsigned int fec; /* Default 0 */
+ unsigned int dtx; /* Default 0 */
+ unsigned int spropmaxcapturerate; /* Default 48000 */
+ unsigned int spropstereo; /* Default 0 */
+};
+
+static int opus_sdp_parse(struct ast_format_attr *format_attr, const char *attributes)
+{
+ struct opus_attr *attr = (struct opus_attr *) format_attr;
+ const char *kvp;
+ unsigned int val;
+
+ if ((kvp = strstr(attributes, "maxplaybackrate")) && sscanf(kvp, "maxplaybackrate=%30u", &val) == 1) {
+ attr->maxplayrate = val;
+ }
+ if ((kvp = strstr(attributes, "sprop-maxcapturerate")) && sscanf(kvp, "sprop-maxcapturerate=%30u", &val) == 1) {
+ attr->spropmaxcapturerate = val;
+ }
+ if ((kvp = strstr(attributes, "minptime")) && sscanf(kvp, "minptime=%30u", &val) == 1) {
+ attr->minptime = val;
+ }
+ if ((kvp = strstr(attributes, "maxaveragebitrate")) && sscanf(kvp, "maxaveragebitrate=%30u", &val) == 1) {
+ attr->maxbitrate = val;
+ }
+ if ((kvp = strstr(attributes, " stereo")) && sscanf(kvp, " stereo=%30u", &val) == 1) {
+ attr->stereo = val;
+ }
+ if ((kvp = strstr(attributes, ";stereo")) && sscanf(kvp, ";stereo=%30u", &val) == 1) {
+ attr->stereo = val;
+ }
+ if ((kvp = strstr(attributes, "sprop-stereo")) && sscanf(kvp, "sprop-stereo=%30u", &val) == 1) {
+ attr->spropstereo = val;
+ }
+ if ((kvp = strstr(attributes, "cbr")) && sscanf(kvp, "cbr=%30u", &val) == 1) {
+ attr->cbr = val;
+ }
+ if ((kvp = strstr(attributes, "useinbandfec")) && sscanf(kvp, "useinbandfec=%30u", &val) == 1) {
+ attr->fec = val;
+ }
+ if ((kvp = strstr(attributes, "usedtx")) && sscanf(kvp, "usedtx=%30u", &val) == 1) {
+ attr->dtx = val;
+ }
+
+ return 0;
+}
+
+static void opus_sdp_generate(const struct ast_format_attr *format_attr, unsigned int payload, struct ast_str **str)
+{
+ struct opus_attr *attr = (struct opus_attr *) format_attr;
+
+ /* FIXME should we only generate attributes that were explicitly set? */
+ ast_str_append(str, 0,
+ "a=fmtp:%d "
+ "maxplaybackrate=%d;"
+ "sprop-maxcapturerate=%d;"
+ "minptime=%d;"
+ "maxaveragebitrate=%d;"
+ "stereo=%d;"
+ "sprop-stereo=%d;"
+ "cbr=%d;"
+ "useinbandfec=%d;"
+ "usedtx=%d\r\n",
+ payload,
+ attr->maxplayrate ? attr->maxplayrate : 48000, /* maxplaybackrate */
+ attr->spropmaxcapturerate ? attr->spropmaxcapturerate : 48000, /* sprop-maxcapturerate */
+ attr->minptime > 10 ? attr->minptime : 10, /* minptime */
+ attr->maxbitrate ? attr->maxbitrate : 20000, /* maxaveragebitrate */
+ attr->stereo ? 1 : 0, /* stereo */
+ attr->spropstereo ? 1 : 0, /* sprop-stereo */
+ attr->cbr ? 1 : 0, /* cbr */
+ attr->fec ? 1 : 0, /* useinbandfec */
+ attr->dtx ? 1 : 0 /* usedtx */
+ );
+}
+
+static int opus_get_val(const struct ast_format_attr *fattr, int key, void *result)
+{
+ const struct opus_attr *attr = (struct opus_attr *) fattr;
+ int *val = result;
+
+ switch (key) {
+ case OPUS_ATTR_KEY_MAX_BITRATE:
+ *val = attr->maxbitrate;
+ break;
+ case OPUS_ATTR_KEY_MAX_PLAYRATE:
+ *val = attr->maxplayrate;
+ break;
+ case OPUS_ATTR_KEY_MINPTIME:
+ *val = attr->minptime;
+ break;
+ case OPUS_ATTR_KEY_STEREO:
+ *val = attr->stereo;
+ break;
+ case OPUS_ATTR_KEY_CBR:
+ *val = attr->cbr;
+ break;
+ case OPUS_ATTR_KEY_FEC:
+ *val = attr->fec;
+ break;
+ case OPUS_ATTR_KEY_DTX:
+ *val = attr->dtx;
+ break;
+ case OPUS_ATTR_KEY_SPROP_CAPTURE_RATE:
+ *val = attr->spropmaxcapturerate;
+ break;
+ case OPUS_ATTR_KEY_SPROP_STEREO:
+ *val = attr->spropstereo;
+ break;
+ default:
+ ast_log(LOG_WARNING, "unknown attribute type %d\n", key);
+ return -1;
+ }
+ return 0;
+}
+
+static int opus_isset(const struct ast_format_attr *fattr, va_list ap)
+{
+ enum opus_attr_keys key;
+ const struct opus_attr *attr = (struct opus_attr *) fattr;
+
+ for (key = va_arg(ap, int);
+ key != AST_FORMAT_ATTR_END;
+ key = va_arg(ap, int))
+ {
+ switch (key) {
+ case OPUS_ATTR_KEY_MAX_BITRATE:
+ if (attr->maxbitrate != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ case OPUS_ATTR_KEY_MAX_PLAYRATE:
+ if (attr->maxplayrate != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ case OPUS_ATTR_KEY_MINPTIME:
+ if (attr->minptime != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ case OPUS_ATTR_KEY_STEREO:
+ if (attr->stereo != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ case OPUS_ATTR_KEY_CBR:
+ if (attr->cbr != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ case OPUS_ATTR_KEY_FEC:
+ if (attr->fec != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ case OPUS_ATTR_KEY_DTX:
+ if (attr->dtx != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ case OPUS_ATTR_KEY_SPROP_CAPTURE_RATE:
+ if (attr->spropmaxcapturerate != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ case OPUS_ATTR_KEY_SPROP_STEREO:
+ if (attr->spropstereo != (va_arg(ap, int))) {
+ return -1;
+ }
+ break;
+ default:
+ ast_log(LOG_WARNING, "unknown attribute type %d\n", key);
+ return -1;
+ }
+ }
+ return 0;
+}
+static int opus_getjoint(const struct ast_format_attr *fattr1, const struct ast_format_attr *fattr2, struct ast_format_attr *result)
+{
+ struct opus_attr *attr1 = (struct opus_attr *) fattr1;
+ struct opus_attr *attr2 = (struct opus_attr *) fattr2;
+ struct opus_attr *attr_res = (struct opus_attr *) result;
+ int joint = 0;
+
+ /* Only do dtx if both sides want it. DTX is a trade off between
+ * computational complexity and bandwidth. */
+ attr_res->dtx = attr1->dtx && attr2->dtx ? 1 : 0;
+
+ /* Only do FEC if both sides want it. If a peer specifically requests not
+ * to receive with FEC, it may be a waste of bandwidth. */
+ attr_res->fec = attr1->fec && attr2->fec ? 1 : 0;
+
+ /* Only do stereo if both sides want it. If a peer specifically requests not
+ * to receive stereo signals, it may be a waste of bandwidth. */
+ attr_res->stereo = attr1->stereo && attr2->stereo ? 1 : 0;
+
+ /* FIXME: do we need to join other attributes as well, e.g., minptime, cbr, etc.? */
+
+ return joint;
+}
+
+static void opus_set(struct ast_format_attr *fattr, va_list ap)
+{
+ enum opus_attr_keys key;
+ struct opus_attr *attr = (struct opus_attr *) fattr;
+
+ for (key = va_arg(ap, int);
+ key != AST_FORMAT_ATTR_END;
+ key = va_arg(ap, int))
+ {
+ switch (key) {
+ case OPUS_ATTR_KEY_MAX_BITRATE:
+ attr->maxbitrate = (va_arg(ap, int));
+ break;
+ case OPUS_ATTR_KEY_MAX_PLAYRATE:
+ attr->maxplayrate = (va_arg(ap, int));
+ break;
+ case OPUS_ATTR_KEY_MINPTIME:
+ attr->minptime = (va_arg(ap, int));
+ break;
+ case OPUS_ATTR_KEY_STEREO:
+ attr->stereo = (va_arg(ap, int));
+ break;
+ case OPUS_ATTR_KEY_CBR:
+ attr->cbr = (va_arg(ap, int));
+ break;
+ case OPUS_ATTR_KEY_FEC:
+ attr->fec = (va_arg(ap, int));
+ break;
+ case OPUS_ATTR_KEY_DTX:
+ attr->dtx = (va_arg(ap, int));
+ break;
+ case OPUS_ATTR_KEY_SPROP_CAPTURE_RATE:
+ attr->spropmaxcapturerate = (va_arg(ap, int));
+ break;
+ case OPUS_ATTR_KEY_SPROP_STEREO:
+ attr->spropstereo = (va_arg(ap, int));
+ break;
+ default:
+ ast_log(LOG_WARNING, "unknown attribute type %d\n", key);
+ }
+ }
+}
+
+static struct ast_format_attr_interface opus_interface = {
+ .id = AST_FORMAT_OPUS,
+ .format_attr_get_joint = opus_getjoint,
+ .format_attr_set = opus_set,
+ .format_attr_isset = opus_isset,
+ .format_attr_get_val = opus_get_val,
+ .format_attr_sdp_parse = opus_sdp_parse,
+ .format_attr_sdp_generate = opus_sdp_generate,
+};
+
+static int load_module(void)
+{
+ if (ast_format_attr_reg_interface(&opus_interface)) {
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+ ast_format_attr_unreg_interface(&opus_interface);
+ return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Opus Format Attribute Module",
+ .load = load_module,
+ .unload = unload_module,
+ .load_pri = AST_MODPRI_CHANNEL_DEPEND,
+);
Propchange: trunk/res/res_format_attr_opus.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/res/res_format_attr_opus.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: trunk/res/res_format_attr_opus.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: trunk/res/res_pjsip_sdp_rtp.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip_sdp_rtp.c?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/res/res_pjsip_sdp_rtp.c (original)
+++ trunk/res/res_pjsip_sdp_rtp.c Fri Aug 23 10:42:27 2013
@@ -849,7 +849,9 @@
char tmp[512];
pj_str_t stmp;
pjmedia_sdp_attr *attr;
- int index = 0, min_packet_size = 0, noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) ? AST_RTP_DTMF : 0;
+ int index = 0;
+ int noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) ? AST_RTP_DTMF : 0;
+ int min_packet_size = 0, max_packet_size = 0;
int rtp_code;
struct ast_format format;
RAII_VAR(struct ast_format_cap *, caps, NULL, ast_format_cap_destroy);
@@ -950,6 +952,10 @@
struct ast_format_list fmt = ast_codec_pref_getsize(pref, &format);
if (fmt.cur_ms && ((fmt.cur_ms < min_packet_size) || !min_packet_size)) {
min_packet_size = fmt.cur_ms;
+ }
+
+ if (fmt.max_ms && ((fmt.max_ms < max_packet_size) || !max_packet_size)) {
+ max_packet_size = fmt.max_ms;
}
}
}
@@ -980,6 +986,12 @@
if (min_packet_size) {
snprintf(tmp, sizeof(tmp), "%d", min_packet_size);
attr = pjmedia_sdp_attr_create(pool, "ptime", pj_cstr(&stmp, tmp));
+ media->attr[media->attr_count++] = attr;
+ }
+
+ if (max_packet_size) {
+ snprintf(tmp, sizeof(tmp), "%d", max_packet_size);
+ attr = pjmedia_sdp_attr_create(pool, "maxptime", pj_cstr(&stmp, tmp));
media->attr[media->attr_count++] = attr;
}
Modified: trunk/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_rtp_asterisk.c?view=diff&rev=397526&r1=397525&r2=397526
==============================================================================
--- trunk/res/res_rtp_asterisk.c (original)
+++ trunk/res/res_rtp_asterisk.c Fri Aug 23 10:42:27 2013
@@ -90,6 +90,8 @@
#define RTCP_PT_SDES 202
#define RTCP_PT_BYE 203
#define RTCP_PT_APP 204
+/* VP8: RTCP Feedback */
+#define RTCP_PT_PSFB 206
#define RTP_MTU 1200
@@ -350,6 +352,9 @@
double normdevrtt;
double stdevrtt;
unsigned int rtt_count;
+
+ /* VP8: sequence number for the RTCP FIR FCI */
+ int firseq;
};
struct rtp_red {
@@ -2414,7 +2419,7 @@
}
if (!res) {
- /*
+ /*
* Not being rescheduled.
*/
ao2_ref(instance, -1);
@@ -2609,6 +2614,45 @@
return 0;
}
+ /* VP8: is this a request to send a RTCP FIR? */
+ if (frame->frametype == AST_FRAME_CONTROL && frame->subclass.integer == AST_CONTROL_VIDUPDATE) {
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ unsigned int *rtcpheader;
+ char bdata[1024];
+ int len = 20;
+ int ice;
+ int res;
+
+ if (!rtp || !rtp->rtcp) {
+ return 0;
+ }
+
+ if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
+ /*
+ * RTCP was stopped.
+ */
+ return 0;
+ }
+
+ /* Prepare RTCP FIR (PT=206, FMT=4) */
+ rtp->rtcp->firseq++;
+ if(rtp->rtcp->firseq == 256) {
+ rtp->rtcp->firseq = 0;
+ }
+
+ rtcpheader = (unsigned int *)bdata;
+ rtcpheader[0] = htonl((2 << 30) | (4 << 24) | (RTCP_PT_PSFB << 16) | ((len/4)-1));
+ rtcpheader[1] = htonl(rtp->ssrc);
+ rtcpheader[2] = htonl(rtp->themssrc);
+ rtcpheader[3] = htonl(rtp->themssrc); /* FCI: SSRC */
+ rtcpheader[4] = htonl(rtp->rtcp->firseq << 24); /* FCI: Sequence number */
+ res = rtcp_sendto(instance, (unsigned int *)rtcpheader, len, 0, &rtp->rtcp->them, &ice);
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTCP FIR transmission error: %s\n", strerror(errno));
+ }
+ return 0;
+ }
+
/* If there is no data length we can't very well send the packet */
if (!frame->datalen) {
ast_debug(1, "Received frame with no data for RTP instance '%p' so dropping frame\n", instance);
@@ -2660,6 +2704,8 @@
case AST_FORMAT_SIREN7:
case AST_FORMAT_SIREN14:
case AST_FORMAT_G719:
+ /* Opus */
+ case AST_FORMAT_OPUS:
/* these are all frame-based codecs and cannot be safely run through
a smoother */
break;
@@ -3353,6 +3399,8 @@
message_blob);
break;
case RTCP_PT_FUR:
+ /* Handle RTCP FIR as FUR */
+ case RTCP_PT_PSFB:
if (rtcp_debug_test_addr(&addr)) {
ast_verbose("Received an RTCP Fast Update Request\n");
}
@@ -4174,14 +4222,14 @@
payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_CN);
level = 127 - (level & 0x7f);
-
+
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
/* Get a pointer to the header */
rtpheader = (unsigned int *)data;
rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
rtpheader[1] = htonl(rtp->lastts);
- rtpheader[2] = htonl(rtp->ssrc);
+ rtpheader[2] = htonl(rtp->ssrc);
data[12] = level;
res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 1, 0, &remote_address, &ice);
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