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 /channels | |
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 'channels')
-rw-r--r-- | channels/chan_pjsip.c | 21 | ||||
-rw-r--r-- | channels/chan_sip.c | 79 |
2 files changed, 80 insertions, 20 deletions
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c index df6e9a385..c2fc5feba 100644 --- a/channels/chan_pjsip.c +++ b/channels/chan_pjsip.c @@ -1168,10 +1168,25 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi case AST_CONTROL_VIDUPDATE: media = pvt->media[SIP_MEDIA_VIDEO]; if (media && media->rtp) { - ao2_ref(channel->session, +1); + /* 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); + if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) { + ao2_cleanup(channel->session); + } } } else { res = -1; diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 8fb01c929..e4596a892 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1269,7 +1269,7 @@ static void add_dtls_to_sdp(struct ast_rtp_instance *instance, struct ast_str ** 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 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data 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 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ 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 @@ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_ /* 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 @@ static void add_codec_to_sdp(const struct sip_pvt *p, 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 @@ static void add_codec_to_sdp(const struct sip_pvt *p, } 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 @@ static void add_codec_to_sdp(const struct sip_pvt *p, 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 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format 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 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int 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 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int 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 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int 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 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int 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 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int 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)) { |