Merge remote-tracking branch 'qatar/master'

* qatar/master:
  mingw/cygwin: Stop adding -fno-common to gcc CFLAGS
  Restructure av_log_missing_feature message
  rtp: Support packetization/depacketization of opus
  file: Set the return value type for lseek to int64_t.
  ppc: fix Altivec build with old compilers
  build: add LTO support for PGI compiler
  build: add -Mdse to PGI optimisation flags
  rtpenc_vp8: Update the packetizer to the latest spec version
  rtpdec_vp8: Make the depacketizer implement the latest spec draft
  doc: allow building with old texi2html versions
  avutil: skip old_pix_fmts.h since it is just a list

Conflicts:
	libavcodec/aacdec.c
	libavcodec/h264.c
	libavcodec/ppc/fmtconvert_altivec.c
	libavcodec/utils.c
	libavformat/file.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 19 insertions, 4 deletions

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index ea678a6792..3a5bbeb398 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -77,6 +77,7 @@ static int is_supported(enum AVCodecID id)
     case AV_CODEC_ID_ILBC:
     case AV_CODEC_ID_MJPEG:
     case AV_CODEC_ID_SPEEX:
+    case AV_CODEC_ID_OPUS:
         return 1;
     default:
         return 0;
@@ -181,15 +182,21 @@ static int rtp_write_header(AVFormatContext *s1)
         s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
         s->num_frames = 0;
         goto defaultcase;
-    case AV_CODEC_ID_VP8:
-        av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
-                                 "incompatible with the latest spec drafts.\n");
-        break;
     case AV_CODEC_ID_ADPCM_G722:
         /* Due to a historical error, the clock rate for G722 in RTP is
          * 8000, even if the sample rate is 16000. See RFC 3551. */
         avpriv_set_pts_info(st, 32, 1, 8000);
         break;
+    case AV_CODEC_ID_OPUS:
+        if (st->codec->channels > 2) {
+            av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
+            goto fail;
+        }
+        /* The opus RTP RFC says that all opus streams should use 48000 Hz
+         * as clock rate, since all opus sample rates can be expressed in
+         * this clock rate, and sample rate changes on the fly are supported. */
+        avpriv_set_pts_info(st, 32, 1, 48000);
+        break;
     case AV_CODEC_ID_ILBC:
         if (st->codec->block_align != 38 && st->codec->block_align != 50) {
             av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
@@ -529,6 +536,14 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
     case AV_CODEC_ID_MJPEG:
         ff_rtp_send_jpeg(s1, pkt->data, size);
         break;
+    case AV_CODEC_ID_OPUS:
+        if (size > s->max_payload_size) {
+            av_log(s1, AV_LOG_ERROR,
+                   "Packet size %d too large for max RTP payload size %d\n",
+                   size, s->max_payload_size);
+            return AVERROR(EINVAL);
+        }
+        /* Intentional fallthrough */
     default:
         /* better than nothing : send the codec raw data */
         rtp_send_raw(s1, pkt->data, size);