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>

7 files changed, 101 insertions, 62 deletions

diff --git a/libavformat/file.c b/libavformat/file.c
index 8c41a3c372..209957b9d3 100644
--- a/libavformat/file.c
+++ b/libavformat/file.c
@@ -135,7 +135,7 @@ static int file_open(URLContext *h, const char *filename, int flags)
 static int64_t file_seek(URLContext *h, int64_t pos, int whence)
 {
     FileContext *c = h->priv_data;
-    off_t ret;
+    int64_t ret;
 
     if (whence == AVSEEK_SIZE) {
         struct stat st;
diff --git a/libavformat/oggparseskeleton.c b/libavformat/oggparseskeleton.c
index 4b02c838f2..d819e0efb1 100644
--- a/libavformat/oggparseskeleton.c
+++ b/libavformat/oggparseskeleton.c
@@ -75,8 +75,8 @@ static int skeleton_header(AVFormatContext *s, int idx)
         target_idx = ogg_find_stream(ogg, AV_RL32(buf+12));
         start_granule = AV_RL64(buf+36);
         if (os->start_granule != OGG_NOGRANULE_VALUE) {
-            av_log_missing_feature(s, "multiple fisbone for the "
-                                      "same stream\n", 0);
+            av_log_missing_feature(s,
+                                   "multiple fisbone for the same stream", 0);
             return 1;
         }
         if (target_idx >= 0 && start_granule != OGG_NOGRANULE_VALUE) {
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index e044435332..a90f0051c0 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -55,6 +55,12 @@ static RTPDynamicProtocolHandler speex_dynamic_handler = {
     .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 */
 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
 
@@ -85,6 +91,7 @@ void av_register_rtp_dynamic_payload_handlers(void)
     ff_register_dynamic_payload_handler(&ff_qcelp_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);
diff --git a/libavformat/rtpdec_vp8.c b/libavformat/rtpdec_vp8.c
index e24bbb6a6f..aef5f78132 100644
--- a/libavformat/rtpdec_vp8.c
+++ b/libavformat/rtpdec_vp8.c
@@ -1,6 +1,7 @@
 /*
  * RTP VP8 Depacketizer
  * Copyright (c) 2010 Josh Allmann
+ * Copyright (c) 2012 Martin Storsjo
  *
  * This file is part of FFmpeg.
  *
@@ -23,7 +24,7 @@
  * @file
  * @brief RTP support for the VP8 payload
  * @author Josh Allmann <joshua.allmann@gmail.com>
- * @see http://www.webmproject.org/code/specs/rtp/
+ * @see http://tools.ietf.org/html/draft-ietf-payload-vp8-05
  */
 
 #include "libavcodec/bytestream.h"
@@ -33,14 +34,12 @@
 struct PayloadContext {
     AVIOContext *data;
     uint32_t       timestamp;
-    int is_keyframe;
 };
 
 static void prepare_packet(AVPacket *pkt, PayloadContext *vp8, int stream)
 {
     av_init_packet(pkt);
     pkt->stream_index = stream;
-    pkt->flags        = vp8->is_keyframe ? AV_PKT_FLAG_KEY : 0;
     pkt->size         = avio_close_dyn_buf(vp8->data, &pkt->data);
     pkt->destruct     = av_destruct_packet;
     vp8->data         = NULL;
@@ -54,70 +53,84 @@ static int vp8_handle_packet(AVFormatContext *ctx,
                              const uint8_t *buf,
                              int len, int flags)
 {
-    int start_packet, end_packet, has_au, ret = AVERROR(EAGAIN);
-
-    if (!buf) {
-        // only called when vp8_handle_packet returns 1
-        if (!vp8->data) {
-            av_log(ctx, AV_LOG_ERROR, "Invalid VP8 data passed\n");
+    int start_partition, end_packet;
+    int extended_bits, non_ref, part_id;
+    int pictureid_present = 0, tl0picidx_present = 0, tid_present = 0,
+        keyidx_present = 0;
+    int pictureid = -1, keyidx = -1;
+
+    if (len < 1)
+        return AVERROR_INVALIDDATA;
+
+    extended_bits   = buf[0] & 0x80;
+    non_ref         = buf[0] & 0x20;
+    start_partition = buf[0] & 0x10;
+    part_id         = buf[0] & 0x0f;
+    end_packet      = flags & RTP_FLAG_MARKER;
+    buf++;
+    len--;
+    if (extended_bits) {
+        if (len < 1)
             return AVERROR_INVALIDDATA;
+        pictureid_present = buf[0] & 0x80;
+        tl0picidx_present = buf[0] & 0x40;
+        tid_present       = buf[0] & 0x20;
+        keyidx_present    = buf[0] & 0x10;
+        buf++;
+        len--;
+    }
+    if (pictureid_present) {
+        if (len < 1)
+            return AVERROR_INVALIDDATA;
+        if (buf[0] & 0x80) {
+            if (len < 2)
+                return AVERROR_INVALIDDATA;
+            pictureid = AV_RB16(buf) & 0x7fff;
+            buf += 2;
+            len -= 2;
+        } else {
+            pictureid = buf[0] & 0x7f;
+            buf++;
+            len--;
         }
-        prepare_packet(pkt, vp8, st->index);
-        *timestamp = vp8->timestamp;
-        return 0;
     }
+    if (tl0picidx_present) {
+        // Ignoring temporal level zero index
+        buf++;
+        len--;
+    }
+    if (tid_present || keyidx_present) {
+        // Ignoring temporal layer index and layer sync bit
+        if (len < 1)
+            return AVERROR_INVALIDDATA;
+        if (keyidx_present)
+            keyidx = buf[0] & 0x1f;
+        buf++;
+        len--;
+    }
+    if (len < 1)
+        return AVERROR_INVALIDDATA;
 
-    start_packet = *buf & 1;
-    end_packet   = flags & RTP_FLAG_MARKER;
-    has_au       = *buf & 2;
-    buf++;
-    len--;
-
-    if (start_packet) {
+    if (start_partition && part_id == 0) {
         int res;
-        uint32_t ts = *timestamp;
         if (vp8->data) {
-            // missing end marker; return old frame anyway. untested
-            prepare_packet(pkt, vp8, st->index);
-            *timestamp = vp8->timestamp; // reset timestamp from old frame
-
-            // if current frame fits into one rtp packet, need to hold
-            // that for the next av_get_packet call
-            ret = end_packet ? 1 : 0;
+            uint8_t *tmp;
+            avio_close_dyn_buf(vp8->data, &tmp);
+            av_free(tmp);
+            vp8->data = NULL;
         }
         if ((res = avio_open_dyn_buf(&vp8->data)) < 0)
             return res;
-        vp8->is_keyframe = *buf & 1;
-        vp8->timestamp   = ts;
+        vp8->timestamp = *timestamp;
      }
 
-    if (!vp8->data || vp8->timestamp != *timestamp && ret == AVERROR(EAGAIN)) {
+    if (!vp8->data || vp8->timestamp != *timestamp) {
         av_log(ctx, AV_LOG_WARNING,
                "Received no start marker; dropping frame\n");
         return AVERROR(EAGAIN);
     }
 
-    // cycle through VP8AU headers if needed
-    // not tested with actual VP8AUs
-    while (len) {
-        int au_len = len;
-        if (has_au && len > 2) {
-            au_len = AV_RB16(buf);
-            buf += 2;
-            len -= 2;
-            if (buf + au_len > buf + len) {
-                av_log(ctx, AV_LOG_ERROR, "Invalid VP8AU length\n");
-                return AVERROR_INVALIDDATA;
-            }
-        }
-
-        avio_write(vp8->data, buf, au_len);
-        buf += au_len;
-        len -= au_len;
-    }
-
-    if (ret != AVERROR(EAGAIN)) // did we miss a end marker?
-        return ret;
+    avio_write(vp8->data, buf, len);
 
     if (end_packet) {
         prepare_packet(pkt, vp8, st->index);
@@ -129,8 +142,6 @@ static int vp8_handle_packet(AVFormatContext *ctx,
 
 static PayloadContext *vp8_new_context(void)
 {
-    av_log(NULL, AV_LOG_ERROR, "RTP VP8 payload implementation is incompatible "
-                               "with the latest spec drafts.\n");
     return av_mallocz(sizeof(PayloadContext));
 }
 
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);
diff --git a/libavformat/rtpenc_vp8.c b/libavformat/rtpenc_vp8.c
index e865514cef..d16e6764c9 100644
--- a/libavformat/rtpenc_vp8.c
+++ b/libavformat/rtpenc_vp8.c
@@ -22,7 +22,7 @@
 #include "rtpenc.h"
 
 /* Based on a draft spec for VP8 RTP.
- * ( http://www.webmproject.org/code/specs/rtp/ ) */
+ * ( http://tools.ietf.org/html/draft-ietf-payload-vp8-05 ) */
 void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buf, int size)
 {
     RTPMuxContext *s = s1->priv_data;
@@ -32,7 +32,9 @@ void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buf, int size)
     s->timestamp    = s->cur_timestamp;
     max_packet_size = s->max_payload_size - 1; // minus one for header byte
 
-    *s->buf_ptr++ = 1; // 0b1 indicates start of frame
+    // no extended control bits, reference frame, start of partition,
+    // partition id 0
+    *s->buf_ptr++ = 0x10;
     while (size > 0) {
         len = FFMIN(size, max_packet_size);
 
diff --git a/libavformat/sdp.c b/libavformat/sdp.c
index 9a4dec4bb2..ca8c3dfdb7 100644
--- a/libavformat/sdp.c
+++ b/libavformat/sdp.c
@@ -576,6 +576,10 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
             av_strlcatf(buff, size, "a=rtpmap:%d speex/%d\r\n",
                                      payload_type, c->sample_rate);
             break;
+        case AV_CODEC_ID_OPUS:
+            av_strlcatf(buff, size, "a=rtpmap:%d opus/48000\r\n",
+                                     payload_type);
+            break;
         default:
             /* Nothing special to do here... */
             break;