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Diffstat (limited to 'pjmedia/src/pjmedia/stream.c')
-rw-r--r--pjmedia/src/pjmedia/stream.c525
1 files changed, 0 insertions, 525 deletions
diff --git a/pjmedia/src/pjmedia/stream.c b/pjmedia/src/pjmedia/stream.c
index fba0c39b..b5354cd1 100644
--- a/pjmedia/src/pjmedia/stream.c
+++ b/pjmedia/src/pjmedia/stream.c
@@ -2793,531 +2793,6 @@ PJ_DEF(pj_status_t) pjmedia_stream_set_dtmf_callback(pjmedia_stream *stream,
return PJ_SUCCESS;
}
-
-static const pj_str_t ID_AUDIO = { "audio", 5};
-static const pj_str_t ID_IN = { "IN", 2 };
-static const pj_str_t ID_IP4 = { "IP4", 3};
-static const pj_str_t ID_IP6 = { "IP6", 3};
-static const pj_str_t ID_RTP_AVP = { "RTP/AVP", 7 };
-static const pj_str_t ID_RTP_SAVP = { "RTP/SAVP", 8 };
-//static const pj_str_t ID_SDP_NAME = { "pjmedia", 7 };
-static const pj_str_t ID_RTPMAP = { "rtpmap", 6 };
-static const pj_str_t ID_TELEPHONE_EVENT = { "telephone-event", 15 };
-
-static const pj_str_t STR_INACTIVE = { "inactive", 8 };
-static const pj_str_t STR_SENDRECV = { "sendrecv", 8 };
-static const pj_str_t STR_SENDONLY = { "sendonly", 8 };
-static const pj_str_t STR_RECVONLY = { "recvonly", 8 };
-
-
-/*
- * Internal function for collecting codec info and param from the SDP media.
- */
-static pj_status_t get_audio_codec_info_param(pjmedia_stream_info *si,
- pj_pool_t *pool,
- pjmedia_codec_mgr *mgr,
- const pjmedia_sdp_media *local_m,
- const pjmedia_sdp_media *rem_m)
-{
- const pjmedia_sdp_attr *attr;
- pjmedia_sdp_rtpmap *rtpmap;
- unsigned i, fmti, pt = 0;
- pj_status_t status;
-
- /* Find the first codec which is not telephone-event */
- for ( fmti = 0; fmti < local_m->desc.fmt_count; ++fmti ) {
- pjmedia_sdp_rtpmap r;
-
- if ( !pj_isdigit(*local_m->desc.fmt[fmti].ptr) )
- return PJMEDIA_EINVALIDPT;
- pt = pj_strtoul(&local_m->desc.fmt[fmti]);
-
- if (pt < 96) {
- /* This is known static PT. Skip rtpmap checking because it is
- * optional. */
- break;
- }
-
- attr = pjmedia_sdp_media_find_attr(local_m, &ID_RTPMAP,
- &local_m->desc.fmt[fmti]);
- if (attr == NULL)
- continue;
-
- status = pjmedia_sdp_attr_get_rtpmap(attr, &r);
- if (status != PJ_SUCCESS)
- continue;
-
- if (pj_strcmp(&r.enc_name, &ID_TELEPHONE_EVENT) != 0)
- break;
- }
- if ( fmti >= local_m->desc.fmt_count )
- return PJMEDIA_EINVALIDPT;
-
- /* Get payload type for receiving direction */
- si->rx_pt = pt;
-
- /* Get codec info.
- * For static payload types, get the info from codec manager.
- * For dynamic payload types, MUST get the rtpmap.
- */
- if (pt < 96) {
- pj_bool_t has_rtpmap;
-
- rtpmap = NULL;
- has_rtpmap = PJ_TRUE;
-
- attr = pjmedia_sdp_media_find_attr(local_m, &ID_RTPMAP,
- &local_m->desc.fmt[fmti]);
- if (attr == NULL) {
- has_rtpmap = PJ_FALSE;
- }
- if (attr != NULL) {
- status = pjmedia_sdp_attr_to_rtpmap(pool, attr, &rtpmap);
- if (status != PJ_SUCCESS)
- has_rtpmap = PJ_FALSE;
- }
-
- /* Build codec format info: */
- if (has_rtpmap) {
- si->fmt.type = si->type;
- si->fmt.pt = pj_strtoul(&local_m->desc.fmt[fmti]);
- pj_strdup(pool, &si->fmt.encoding_name, &rtpmap->enc_name);
- si->fmt.clock_rate = rtpmap->clock_rate;
-
-#if defined(PJMEDIA_HANDLE_G722_MPEG_BUG) && (PJMEDIA_HANDLE_G722_MPEG_BUG != 0)
- /* The session info should have the actual clock rate, because
- * this info is used for calculationg buffer size, etc in stream
- */
- if (si->fmt.pt == PJMEDIA_RTP_PT_G722)
- si->fmt.clock_rate = 16000;
-#endif
-
- /* For audio codecs, rtpmap parameters denotes the number of
- * channels.
- */
- if (si->type == PJMEDIA_TYPE_AUDIO && rtpmap->param.slen) {
- si->fmt.channel_cnt = (unsigned) pj_strtoul(&rtpmap->param);
- } else {
- si->fmt.channel_cnt = 1;
- }
-
- } else {
- const pjmedia_codec_info *p_info;
-
- status = pjmedia_codec_mgr_get_codec_info( mgr, pt, &p_info);
- if (status != PJ_SUCCESS)
- return status;
-
- pj_memcpy(&si->fmt, p_info, sizeof(pjmedia_codec_info));
- }
-
- /* For static payload type, pt's are symetric */
- si->tx_pt = pt;
-
- } else {
- pjmedia_codec_id codec_id;
- pj_str_t codec_id_st;
- const pjmedia_codec_info *p_info;
-
- attr = pjmedia_sdp_media_find_attr(local_m, &ID_RTPMAP,
- &local_m->desc.fmt[fmti]);
- if (attr == NULL)
- return PJMEDIA_EMISSINGRTPMAP;
-
- status = pjmedia_sdp_attr_to_rtpmap(pool, attr, &rtpmap);
- if (status != PJ_SUCCESS)
- return status;
-
- /* Build codec format info: */
-
- si->fmt.type = si->type;
- si->fmt.pt = pj_strtoul(&local_m->desc.fmt[fmti]);
- si->fmt.encoding_name = rtpmap->enc_name;
- si->fmt.clock_rate = rtpmap->clock_rate;
-
- /* For audio codecs, rtpmap parameters denotes the number of
- * channels.
- */
- if (si->type == PJMEDIA_TYPE_AUDIO && rtpmap->param.slen) {
- si->fmt.channel_cnt = (unsigned) pj_strtoul(&rtpmap->param);
- } else {
- si->fmt.channel_cnt = 1;
- }
-
- /* Normalize the codec info from codec manager. Note that the
- * payload type will be resetted to its default (it might have
- * been rewritten by the SDP negotiator to match to the remote
- * offer), this is intentional as currently some components may
- * prefer (or even require) the default PT in codec info.
- */
- pjmedia_codec_info_to_id(&si->fmt, codec_id, sizeof(codec_id));
-
- i = 1;
- codec_id_st = pj_str(codec_id);
- status = pjmedia_codec_mgr_find_codecs_by_id(mgr, &codec_id_st,
- &i, &p_info, NULL);
- if (status != PJ_SUCCESS)
- return status;
-
- pj_memcpy(&si->fmt, p_info, sizeof(pjmedia_codec_info));
-
- /* Determine payload type for outgoing channel, by finding
- * dynamic payload type in remote SDP that matches the answer.
- */
- si->tx_pt = 0xFFFF;
- for (i=0; i<rem_m->desc.fmt_count; ++i) {
- unsigned rpt;
- pjmedia_sdp_attr *r_attr;
- pjmedia_sdp_rtpmap r_rtpmap;
-
- rpt = pj_strtoul(&rem_m->desc.fmt[i]);
- if (rpt < 96)
- continue;
-
- r_attr = pjmedia_sdp_media_find_attr(rem_m, &ID_RTPMAP,
- &rem_m->desc.fmt[i]);
- if (!r_attr)
- continue;
-
- if (pjmedia_sdp_attr_get_rtpmap(r_attr, &r_rtpmap) != PJ_SUCCESS)
- continue;
-
- if (!pj_stricmp(&rtpmap->enc_name, &r_rtpmap.enc_name) &&
- rtpmap->clock_rate == r_rtpmap.clock_rate)
- {
- /* Found matched codec. */
- si->tx_pt = rpt;
-
- break;
- }
- }
-
- if (si->tx_pt == 0xFFFF)
- return PJMEDIA_EMISSINGRTPMAP;
- }
-
-
- /* Now that we have codec info, get the codec param. */
- si->param = PJ_POOL_ALLOC_T(pool, pjmedia_codec_param);
- status = pjmedia_codec_mgr_get_default_param(mgr, &si->fmt,
- si->param);
-
- /* Get remote fmtp for our encoder. */
- pjmedia_stream_info_parse_fmtp(pool, rem_m, si->tx_pt,
- &si->param->setting.enc_fmtp);
-
- /* Get local fmtp for our decoder. */
- pjmedia_stream_info_parse_fmtp(pool, local_m, si->rx_pt,
- &si->param->setting.dec_fmtp);
-
- /* Get the remote ptime for our encoder. */
- attr = pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr,
- "ptime", NULL);
- if (attr) {
- pj_str_t tmp_val = attr->value;
- unsigned frm_per_pkt;
-
- pj_strltrim(&tmp_val);
-
- /* Round up ptime when the specified is not multiple of frm_ptime */
- frm_per_pkt = (pj_strtoul(&tmp_val) +
- si->param->info.frm_ptime/2) /
- si->param->info.frm_ptime;
- if (frm_per_pkt != 0) {
- si->param->setting.frm_per_pkt = (pj_uint8_t)frm_per_pkt;
- }
- }
-
- /* Get remote maxptime for our encoder. */
- attr = pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr,
- "maxptime", NULL);
- if (attr) {
- pj_str_t tmp_val = attr->value;
-
- pj_strltrim(&tmp_val);
- si->tx_maxptime = pj_strtoul(&tmp_val);
- }
-
- /* When direction is NONE (it means SDP negotiation has failed) we don't
- * need to return a failure here, as returning failure will cause
- * the whole SDP to be rejected. See ticket #:
- * http://
- *
- * Thanks Alain Totouom
- */
- if (status != PJ_SUCCESS && si->dir != PJMEDIA_DIR_NONE)
- return status;
-
-
- /* Get incomming payload type for telephone-events */
- si->rx_event_pt = -1;
- for (i=0; i<local_m->attr_count; ++i) {
- pjmedia_sdp_rtpmap r;
-
- attr = local_m->attr[i];
- if (pj_strcmp(&attr->name, &ID_RTPMAP) != 0)
- continue;
- if (pjmedia_sdp_attr_get_rtpmap(attr, &r) != PJ_SUCCESS)
- continue;
- if (pj_strcmp(&r.enc_name, &ID_TELEPHONE_EVENT) == 0) {
- si->rx_event_pt = pj_strtoul(&r.pt);
- break;
- }
- }
-
- /* Get outgoing payload type for telephone-events */
- si->tx_event_pt = -1;
- for (i=0; i<rem_m->attr_count; ++i) {
- pjmedia_sdp_rtpmap r;
-
- attr = rem_m->attr[i];
- if (pj_strcmp(&attr->name, &ID_RTPMAP) != 0)
- continue;
- if (pjmedia_sdp_attr_get_rtpmap(attr, &r) != PJ_SUCCESS)
- continue;
- if (pj_strcmp(&r.enc_name, &ID_TELEPHONE_EVENT) == 0) {
- si->tx_event_pt = pj_strtoul(&r.pt);
- break;
- }
- }
-
- return PJ_SUCCESS;
-}
-
-
-
-/*
- * Create stream info from SDP media line.
- */
-PJ_DEF(pj_status_t) pjmedia_stream_info_from_sdp(
- pjmedia_stream_info *si,
- pj_pool_t *pool,
- pjmedia_endpt *endpt,
- const pjmedia_sdp_session *local,
- const pjmedia_sdp_session *remote,
- unsigned stream_idx)
-{
- pjmedia_codec_mgr *mgr;
- const pjmedia_sdp_attr *attr;
- const pjmedia_sdp_media *local_m;
- const pjmedia_sdp_media *rem_m;
- const pjmedia_sdp_conn *local_conn;
- const pjmedia_sdp_conn *rem_conn;
- int rem_af, local_af;
- pj_sockaddr local_addr;
- pj_status_t status;
-
-
- /* Validate arguments: */
- PJ_ASSERT_RETURN(pool && si && local && remote, PJ_EINVAL);
- PJ_ASSERT_RETURN(stream_idx < local->media_count, PJ_EINVAL);
- PJ_ASSERT_RETURN(stream_idx < remote->media_count, PJ_EINVAL);
-
- /* Keep SDP shortcuts */
- local_m = local->media[stream_idx];
- rem_m = remote->media[stream_idx];
-
- local_conn = local_m->conn ? local_m->conn : local->conn;
- if (local_conn == NULL)
- return PJMEDIA_SDP_EMISSINGCONN;
-
- rem_conn = rem_m->conn ? rem_m->conn : remote->conn;
- if (rem_conn == NULL)
- return PJMEDIA_SDP_EMISSINGCONN;
-
- /* Media type must be audio */
- if (pj_stricmp(&local_m->desc.media, &ID_AUDIO) != 0)
- return PJMEDIA_EINVALIMEDIATYPE;
-
- /* Get codec manager. */
- mgr = pjmedia_endpt_get_codec_mgr(endpt);
-
- /* Reset: */
-
- pj_bzero(si, sizeof(*si));
-
-#if PJMEDIA_HAS_RTCP_XR && PJMEDIA_STREAM_ENABLE_XR
- /* Set default RTCP XR enabled/disabled */
- si->rtcp_xr_enabled = PJ_TRUE;
-#endif
-
- /* Media type: */
- si->type = PJMEDIA_TYPE_AUDIO;
-
- /* Transport protocol */
-
- /* At this point, transport type must be compatible,
- * the transport instance will do more validation later.
- */
- status = pjmedia_sdp_transport_cmp(&rem_m->desc.transport,
- &local_m->desc.transport);
- if (status != PJ_SUCCESS)
- return PJMEDIA_SDPNEG_EINVANSTP;
-
- if (pj_stricmp(&local_m->desc.transport, &ID_RTP_AVP) == 0) {
-
- si->proto = PJMEDIA_TP_PROTO_RTP_AVP;
-
- } else if (pj_stricmp(&local_m->desc.transport, &ID_RTP_SAVP) == 0) {
-
- si->proto = PJMEDIA_TP_PROTO_RTP_SAVP;
-
- } else {
-
- si->proto = PJMEDIA_TP_PROTO_UNKNOWN;
- return PJ_SUCCESS;
- }
-
-
- /* Check address family in remote SDP */
- rem_af = pj_AF_UNSPEC();
- if (pj_stricmp(&rem_conn->net_type, &ID_IN)==0) {
- if (pj_stricmp(&rem_conn->addr_type, &ID_IP4)==0) {
- rem_af = pj_AF_INET();
- } else if (pj_stricmp(&rem_conn->addr_type, &ID_IP6)==0) {
- rem_af = pj_AF_INET6();
- }
- }
-
- if (rem_af==pj_AF_UNSPEC()) {
- /* Unsupported address family */
- return PJ_EAFNOTSUP;
- }
-
- /* Set remote address: */
- status = pj_sockaddr_init(rem_af, &si->rem_addr, &rem_conn->addr,
- rem_m->desc.port);
- if (status != PJ_SUCCESS) {
- /* Invalid IP address. */
- return PJMEDIA_EINVALIDIP;
- }
-
- /* Check address family of local info */
- local_af = pj_AF_UNSPEC();
- if (pj_stricmp(&local_conn->net_type, &ID_IN)==0) {
- if (pj_stricmp(&local_conn->addr_type, &ID_IP4)==0) {
- local_af = pj_AF_INET();
- } else if (pj_stricmp(&local_conn->addr_type, &ID_IP6)==0) {
- local_af = pj_AF_INET6();
- }
- }
-
- if (local_af==pj_AF_UNSPEC()) {
- /* Unsupported address family */
- return PJ_SUCCESS;
- }
-
- /* Set remote address: */
- status = pj_sockaddr_init(local_af, &local_addr, &local_conn->addr,
- local_m->desc.port);
- if (status != PJ_SUCCESS) {
- /* Invalid IP address. */
- return PJMEDIA_EINVALIDIP;
- }
-
- /* Local and remote address family must match */
- if (local_af != rem_af)
- return PJ_EAFNOTSUP;
-
- /* Media direction: */
-
- if (local_m->desc.port == 0 ||
- pj_sockaddr_has_addr(&local_addr)==PJ_FALSE ||
- pj_sockaddr_has_addr(&si->rem_addr)==PJ_FALSE ||
- pjmedia_sdp_media_find_attr(local_m, &STR_INACTIVE, NULL)!=NULL)
- {
- /* Inactive stream. */
-
- si->dir = PJMEDIA_DIR_NONE;
-
- } else if (pjmedia_sdp_media_find_attr(local_m, &STR_SENDONLY, NULL)!=NULL) {
-
- /* Send only stream. */
-
- si->dir = PJMEDIA_DIR_ENCODING;
-
- } else if (pjmedia_sdp_media_find_attr(local_m, &STR_RECVONLY, NULL)!=NULL) {
-
- /* Recv only stream. */
-
- si->dir = PJMEDIA_DIR_DECODING;
-
- } else {
-
- /* Send and receive stream. */
-
- si->dir = PJMEDIA_DIR_ENCODING_DECODING;
-
- }
-
- /* No need to do anything else if stream is rejected */
- if (local_m->desc.port == 0) {
- return PJ_SUCCESS;
- }
-
- /* If "rtcp" attribute is present in the SDP, set the RTCP address
- * from that attribute. Otherwise, calculate from RTP address.
- */
- attr = pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr,
- "rtcp", NULL);
- if (attr) {
- pjmedia_sdp_rtcp_attr rtcp;
- status = pjmedia_sdp_attr_get_rtcp(attr, &rtcp);
- if (status == PJ_SUCCESS) {
- if (rtcp.addr.slen) {
- status = pj_sockaddr_init(rem_af, &si->rem_rtcp, &rtcp.addr,
- (pj_uint16_t)rtcp.port);
- } else {
- pj_sockaddr_init(rem_af, &si->rem_rtcp, NULL,
- (pj_uint16_t)rtcp.port);
- pj_memcpy(pj_sockaddr_get_addr(&si->rem_rtcp),
- pj_sockaddr_get_addr(&si->rem_addr),
- pj_sockaddr_get_addr_len(&si->rem_addr));
- }
- }
- }
-
- if (!pj_sockaddr_has_addr(&si->rem_rtcp)) {
- int rtcp_port;
-
- pj_memcpy(&si->rem_rtcp, &si->rem_addr, sizeof(pj_sockaddr));
- rtcp_port = pj_sockaddr_get_port(&si->rem_addr) + 1;
- pj_sockaddr_set_port(&si->rem_rtcp, (pj_uint16_t)rtcp_port);
- }
-
-
- /* Get the payload number for receive channel. */
- /*
- Previously we used to rely on fmt[0] being the selected codec,
- but some UA sends telephone-event as fmt[0] and this would
- cause assert failure below.
-
- Thanks Chris Hamilton <chamilton .at. cs.dal.ca> for this patch.
-
- // And codec must be numeric!
- if (!pj_isdigit(*local_m->desc.fmt[0].ptr) ||
- !pj_isdigit(*rem_m->desc.fmt[0].ptr))
- {
- return PJMEDIA_EINVALIDPT;
- }
-
- pt = pj_strtoul(&local_m->desc.fmt[0]);
- pj_assert(PJMEDIA_RTP_PT_TELEPHONE_EVENTS==0 ||
- pt != PJMEDIA_RTP_PT_TELEPHONE_EVENTS);
- */
-
- /* Get codec info and param */
- status = get_audio_codec_info_param(si, pool, mgr, local_m, rem_m);
-
- /* Leave SSRC to random. */
- si->ssrc = pj_rand();
-
- /* Set default jitter buffer parameter. */
- si->jb_init = si->jb_max = si->jb_min_pre = si->jb_max_pre = -1;
-
- return status;
-}
-
/*
* Send RTCP SDES.
*/
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-25 22:53:25 +0000