]> git.sesse.net Git - ffmpeg/blobdiff - libavformat/rtpdec.c
riff: Move functions around to be covered by appropriate #ifdefs
[ffmpeg] / libavformat / rtpdec.c
index 0f6ed27ff6d8024e0f24f855b381bc1059a4e33e..dac367dd86097d5f39020883b82929fcf7e72e3b 100644 (file)
  */
 
 #include "libavutil/mathematics.h"
+#include "libavutil/avstring.h"
+#include "libavutil/time.h"
 #include "libavcodec/get_bits.h"
 #include "avformat.h"
 #include "mpegts.h"
 #include "url.h"
 
-#include <unistd.h>
-#include <strings.h>
 #include "network.h"
 
 #include "rtpdec.h"
          'ffio_open_dyn_packet_buf')
 */
 
-static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
+static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
     .enc_name           = "X-MP3-draft-00",
     .codec_type         = AVMEDIA_TYPE_AUDIO,
-    .codec_id           = CODEC_ID_MP3ADU,
+    .codec_id           = AV_CODEC_ID_MP3ADU,
+};
+
+static RTPDynamicProtocolHandler speex_dynamic_handler = {
+    .enc_name         = "speex",
+    .codec_type       = AVMEDIA_TYPE_AUDIO,
+    .codec_id         = AV_CODEC_ID_SPEEX,
+};
+
+static RTPDynamicProtocolHandler opus_dynamic_handler = {
+    .enc_name         = "opus",
+    .codec_type       = AVMEDIA_TYPE_AUDIO,
+    .codec_id         = AV_CODEC_ID_OPUS,
 };
 
 /* statistics functions */
@@ -66,7 +78,10 @@ void av_register_rtp_dynamic_payload_handlers(void)
     ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
@@ -74,7 +89,9 @@ void av_register_rtp_dynamic_payload_handlers(void)
     ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
-    ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
+    ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
+    ff_register_dynamic_payload_handler(&speex_dynamic_handler);
+    ff_register_dynamic_payload_handler(&opus_dynamic_handler);
 
     ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
     ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
@@ -83,6 +100,11 @@ void av_register_rtp_dynamic_payload_handlers(void)
     ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
     ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
     ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
+
+    ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
 }
 
 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
@@ -91,7 +113,7 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
     RTPDynamicProtocolHandler *handler;
     for (handler = RTPFirstDynamicPayloadHandler;
          handler; handler = handler->next)
-        if (!strcasecmp(name, handler->enc_name) &&
+        if (!av_strcasecmp(name, handler->enc_name) &&
             codec_type == handler->codec_type)
             return handler;
     return NULL;
@@ -218,7 +240,7 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
     return 1;
 }
 
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
 {
     AVIOContext *pb;
     uint8_t *buf;
@@ -294,7 +316,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
     avio_w8(pb, RTCP_SDES);
     len = strlen(s->hostname);
     avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
-    avio_wb32(pb, s->ssrc);
+    avio_wb32(pb, s->ssrc + 1);
     avio_w8(pb, 0x01);
     avio_w8(pb, len);
     avio_write(pb, s->hostname, len);
@@ -315,7 +337,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
     return 0;
 }
 
-void rtp_send_punch_packets(URLContext* rtp_handle)
+void ff_rtp_send_punch_packets(URLContext* rtp_handle)
 {
     AVIOContext *pb;
     uint8_t *buf;
@@ -357,9 +379,9 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
 /**
  * open a new RTP parse context for stream 'st'. 'st' can be NULL for
  * MPEG2TS streams to indicate that they should be demuxed inside the
- * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
+ * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
  */
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
 {
     RTPDemuxContext *s;
 
@@ -379,18 +401,21 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
             av_free(s);
             return NULL;
         }
-    } else {
+    } else if (st) {
         switch(st->codec->codec_id) {
-        case CODEC_ID_MPEG1VIDEO:
-        case CODEC_ID_MPEG2VIDEO:
-        case CODEC_ID_MP2:
-        case CODEC_ID_MP3:
-        case CODEC_ID_MPEG4:
-        case CODEC_ID_H263:
-        case CODEC_ID_H264:
+        case AV_CODEC_ID_MPEG1VIDEO:
+        case AV_CODEC_ID_MPEG2VIDEO:
+        case AV_CODEC_ID_MP2:
+        case AV_CODEC_ID_MP3:
+        case AV_CODEC_ID_MPEG4:
+        case AV_CODEC_ID_H263:
+        case AV_CODEC_ID_H264:
             st->need_parsing = AVSTREAM_PARSE_FULL;
             break;
-        case CODEC_ID_ADPCM_G722:
+        case AV_CODEC_ID_VORBIS:
+            st->need_parsing = AVSTREAM_PARSE_HEADERS;
+            break;
+        case AV_CODEC_ID_ADPCM_G722:
             /* According to RFC 3551, the stream clock rate is 8000
              * even if the sample rate is 16000. */
             if (st->codec->sample_rate == 8000)
@@ -407,8 +432,8 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
 }
 
 void
-rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
-                               RTPDynamicProtocolHandler *handler)
+ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+                                  RTPDynamicProtocolHandler *handler)
 {
     s->dynamic_protocol_context = ctx;
     s->parse_packet = handler->parse_packet;
@@ -421,7 +446,10 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
 {
     if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
         return; /* Timestamp already set by depacketizer */
-    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
+    if (timestamp == RTP_NOTS_VALUE)
+        return;
+
+    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
         int64_t addend;
         int delta_timestamp;
 
@@ -433,11 +461,16 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
                    delta_timestamp;
         return;
     }
-    if (timestamp == RTP_NOTS_VALUE)
-        return;
+
     if (!s->base_timestamp)
         s->base_timestamp = timestamp;
-    pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
+    /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
+    if (!s->timestamp)
+        s->unwrapped_timestamp += timestamp;
+    else
+        s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
+    s->timestamp = timestamp;
+    pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
 }
 
 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
@@ -519,8 +552,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
     } else {
         // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
         switch(st->codec->codec_id) {
-        case CODEC_ID_MP2:
-        case CODEC_ID_MP3:
+        case AV_CODEC_ID_MP2:
+        case AV_CODEC_ID_MP3:
             /* better than nothing: skip mpeg audio RTP header */
             if (len <= 4)
                 return -1;
@@ -530,8 +563,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
             av_new_packet(pkt, len);
             memcpy(pkt->data, buf, len);
             break;
-        case CODEC_ID_MPEG1VIDEO:
-        case CODEC_ID_MPEG2VIDEO:
+        case AV_CODEC_ID_MPEG1VIDEO:
+        case AV_CODEC_ID_MPEG2VIDEO:
             /* better than nothing: skip mpeg video RTP header */
             if (len <= 4)
                 return -1;
@@ -681,7 +714,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
 
     if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
         return -1;
-    if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
+    if (RTP_PT_IS_RTCP(buf[1])) {
         return rtcp_parse_packet(s, buf, len);
     }
 
@@ -722,8 +755,8 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
  * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
  * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
  */
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
-                     uint8_t **bufptr, int len)
+int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+                        uint8_t **bufptr, int len)
 {
     int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
     s->prev_ret = rv;
@@ -732,7 +765,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
     return rv ? rv : has_next_packet(s);
 }
 
-void rtp_parse_close(RTPDemuxContext *s)
+void ff_rtp_parse_close(RTPDemuxContext *s)
 {
     ff_rtp_reset_packet_queue(s);
     if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {