diff options
author | Matthew Jordan <mjordan@digium.com> | 2013-08-23 15:42:27 +0000 |
---|---|---|
committer | Matthew Jordan <mjordan@digium.com> | 2013-08-23 15:42:27 +0000 |
commit | 4d348e853cbd9ba7bc976487bfcb352a84e5ece0 (patch) | |
tree | fdf289e34cd706884aed7a262409fc3cdcba9bd1 /res | |
parent | e31bd332b83f0245ce8bd6626279e1b9c683ec18 (diff) |
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res')
-rw-r--r-- | res/res_format_attr_opus.c | 321 | ||||
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 14 | ||||
-rw-r--r-- | res/res_rtp_asterisk.c | 54 |
3 files changed, 385 insertions, 4 deletions
diff --git a/res/res_format_attr_opus.c b/res/res_format_attr_opus.c new file mode 100644 index 000000000..ed8adb77f --- /dev/null +++ b/res/res_format_attr_opus.c @@ -0,0 +1,321 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2013, Digium, Inc. + * + * Lorenzo Miniero <lorenzo@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@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, +); diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index c97c0cb40..be5d59f06 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -849,7 +849,9 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as 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); @@ -951,6 +953,10 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as 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; + } } } @@ -983,6 +989,12 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as 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; + } + /* Add the sendrecv attribute - we purposely don't keep track because pjmedia-sdp will automatically change our offer for us */ attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr); attr->name = STR_SENDRECV; diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index db07e4ec5..6383b09e3 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -90,6 +90,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #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 @@ struct ast_rtcp { 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 @@ static int ast_rtcp_write(const void *data) } if (!res) { - /* + /* * Not being rescheduled. */ ao2_ref(instance, -1); @@ -2609,6 +2614,45 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr 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 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr 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 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance) 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 @@ static int ast_rtp_sendcng(struct ast_rtp_instance *instance, int level) 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); |