Merge remote-tracking branch 'qatar/master'

* qatar/master: (40 commits)
  swf: check return values for av_get/new_packet().
  wavpack: Don't shift minclip/maxclip
  rtpenc: Expose the max packet size via an avoption
  rtpenc: Move max_packet_size to a context variable
  rtpenc: Add an option for not sending RTCP packets
  lavc: drop encode() support for video.
  snowenc: switch to encode2().
  snowenc: don't abuse input picture for storing information.
  a64multienc: switch to encode2().
  a64multienc: don't write into output buffer when there's no output.
  libxvid: switch to encode2().
  tiffenc: switch to encode2().
  tiffenc: properly forward error codes in encode_frame().
  lavc: drop libdirac encoder.
  gifenc: switch to encode2().
  libvpxenc: switch to encode2().
  flashsvenc: switch to encode2().
  Remove libpostproc.
  lcl: don't overwrite input memory.
  swscale: take first/lastline over/underflows into account for MMX.
  ...

Conflicts:
	.gitignore
	Makefile
	cmdutils.c
	configure
	doc/APIchanges
	libavcodec/Makefile
	libavcodec/allcodecs.c
	libavcodec/libdiracenc.c
	libavcodec/libxvidff.c
	libavcodec/qtrleenc.c
	libavcodec/tiffenc.c
	libavcodec/utils.c
	libavformat/mov.c
	libavformat/movenc.c
	libpostproc/Makefile
	libpostproc/postprocess.c
	libpostproc/postprocess.h
	libpostproc/postprocess_altivec_template.c
	libpostproc/postprocess_internal.h
	libpostproc/postprocess_template.c
	libswscale/swscale.c
	libswscale/utils.c

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

11 files changed, 166 insertions, 23 deletions

diff --git a/libavformat/Makefile b/libavformat/Makefile
index f58dc42..61bb112 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -258,6 +258,7 @@ OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o         \
                                             rtpenc_latm.o    \
                                             rtpenc_amr.o     \
                                             rtpenc_h263.o    \
+                                            rtpenc_h263_rfc2190.o \
                                             rtpenc_mpv.o     \
                                             rtpenc.o      \
                                             rtpenc_h264.o \
diff --git a/libavformat/mov.c b/libavformat/mov.c
index ff97a9b..15df2bf 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -25,6 +25,7 @@
 //#define DEBUG
 //#define MOV_EXPORT_ALL_METADATA
 
+#include "libavutil/audioconvert.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/intfloat.h"
 #include "libavutil/mathematics.h"
@@ -32,6 +33,7 @@
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
 #include "libavutil/timecode.h"
+#include "libavcodec/ac3tab.h"
 #include "avformat.h"
 #include "internal.h"
 #include "avio_internal.h"
@@ -570,6 +572,9 @@ static int mov_read_dac3(MOVContext *c, AVIOContext *pb, MOVAtom atom)
     acmod = (ac3info >> 11) & 0x7;
     lfeon = (ac3info >> 10) & 0x1;
     st->codec->channels = ((int[]){2,1,2,3,3,4,4,5})[acmod] + lfeon;
+    st->codec->channel_layout = avpriv_ac3_channel_layout_tab[acmod];
+    if (lfeon)
+        st->codec->channel_layout |= AV_CH_LOW_FREQUENCY;
     st->codec->audio_service_type = bsmod;
     if (st->codec->channels > 1 && bsmod == 0x7)
         st->codec->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
diff --git a/libavformat/movenc.c b/libavformat/movenc.c
index 138c00f..4722819 100644
--- a/libavformat/movenc.c
+++ b/libavformat/movenc.c
@@ -52,7 +52,7 @@ static const AVOption options[] = {
     { "separate_moof", "Write separate moof/mdat atoms for each track", 0, AV_OPT_TYPE_CONST, {.dbl = FF_MOV_FLAG_SEPARATE_MOOF}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "movflags" },
     { "frag_custom", "Flush fragments on caller requests", 0, AV_OPT_TYPE_CONST, {.dbl = FF_MOV_FLAG_FRAG_CUSTOM}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "movflags" },
     { "isml", "Create a live smooth streaming feed (for pushing to a publishing point)", 0, AV_OPT_TYPE_CONST, {.dbl = FF_MOV_FLAG_ISML}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "movflags" },
-    FF_RTP_FLAG_OPTS(MOVMuxContext, rtp_flags),
+    FF_RTP_FLAG_OPTS(MOVMuxContext, rtp_flags)
     { "skip_iods", "Skip writing iods atom.", offsetof(MOVMuxContext, iods_skip), AV_OPT_TYPE_INT, {.dbl = 1}, 0, 1, AV_OPT_FLAG_ENCODING_PARAM},
     { "iods_audio_profile", "iods audio profile atom.", offsetof(MOVMuxContext, iods_audio_profile), AV_OPT_TYPE_INT, {.dbl = -1}, -1, 255, AV_OPT_FLAG_ENCODING_PARAM},
     { "iods_video_profile", "iods video profile atom.", offsetof(MOVMuxContext, iods_video_profile), AV_OPT_TYPE_INT, {.dbl = -1}, -1, 255, AV_OPT_FLAG_ENCODING_PARAM},
diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index 98d9616..a5484ae 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -106,7 +106,9 @@ int ff_rtp_get_payload_type(AVFormatContext *fmt, AVCodecContext *codec)
     /* static payload type */
     for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
         if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
-            if (codec->codec_id == CODEC_ID_H263)
+            if (codec->codec_id == CODEC_ID_H263 && (!fmt ||
+                !fmt->oformat->priv_class ||
+                !av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190")))
                 continue;
             if (codec->codec_id == CODEC_ID_PCM_S16BE)
                 if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 4b1ff1d..3228868 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -31,8 +31,9 @@
 //#define DEBUG
 
 static const AVOption options[] = {
-    FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
+    FF_RTP_FLAG_OPTS(RTPMuxContext, flags)
     { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
+    { "max_packet_size", "Max packet size", offsetof(RTPMuxContext, max_packet_size), AV_OPT_TYPE_INT, {.dbl = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
     { NULL },
 };
 
@@ -82,11 +83,13 @@ static int is_supported(enum CodecID id)
 static int rtp_write_header(AVFormatContext *s1)
 {
     RTPMuxContext *s = s1->priv_data;
-    int max_packet_size, n;
+    int n;
     AVStream *st;
 
-    if (s1->nb_streams != 1)
-        return -1;
+    if (s1->nb_streams != 1) {
+        av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
+        return AVERROR(EINVAL);
+    }
     st = s1->streams[0];
     if (!is_supported(st->codec->codec_id)) {
         av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
@@ -107,16 +110,21 @@ static int rtp_write_header(AVFormatContext *s1)
         s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
                                  NTP_OFFSET_US;
 
-    max_packet_size = s1->pb->max_packet_size;
-    if (max_packet_size <= 12) {
-        av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", max_packet_size);
+    if (s->max_packet_size) {
+        if (s1->pb->max_packet_size)
+            s->max_packet_size = FFMIN(s->max_payload_size,
+                                       s1->pb->max_packet_size);
+    } else
+        s->max_packet_size = s1->pb->max_packet_size;
+    if (s->max_packet_size <= 12) {
+        av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s->max_packet_size);
         return AVERROR(EIO);
     }
-    s->buf = av_malloc(max_packet_size);
+    s->buf = av_malloc(s->max_packet_size);
     if (s->buf == NULL) {
         return AVERROR(ENOMEM);
     }
-    s->max_payload_size = max_packet_size - 12;
+    s->max_payload_size = s->max_packet_size - 12;
 
     s->max_frames_per_packet = 0;
     if (s1->max_delay) {
@@ -386,8 +394,9 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 
     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
         RTCP_TX_RATIO_DEN;
-    if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
-                           (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
+    if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
+                            (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
+        !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
         rtcp_send_sr(s1, ff_ntp_time());
         s->last_octet_count = s->octet_count;
         s->first_packet = 0;
@@ -443,6 +452,11 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
         ff_rtp_send_h264(s1, pkt->data, size);
         break;
     case CODEC_ID_H263:
+        if (s->flags & FF_RTP_FLAG_RFC2190) {
+            ff_rtp_send_h263_rfc2190(s1, pkt->data, size);
+            break;
+        }
+        /* Fallthrough */
     case CODEC_ID_H263P:
         ff_rtp_send_h263(s1, pkt->data, size);
         break;
diff --git a/libavformat/rtpenc.h b/libavformat/rtpenc.h
index 9013363..c28377f 100644
--- a/libavformat/rtpenc.h
+++ b/libavformat/rtpenc.h
@@ -34,6 +34,7 @@ struct RTPMuxContext {
     uint32_t timestamp;
     uint32_t base_timestamp;
     uint32_t cur_timestamp;
+    int max_packet_size;
     int max_payload_size;
     int num_frames;
 
@@ -64,15 +65,20 @@ struct RTPMuxContext {
 typedef struct RTPMuxContext RTPMuxContext;
 
 #define FF_RTP_FLAG_MP4A_LATM 1
+#define FF_RTP_FLAG_RFC2190   2
+#define FF_RTP_FLAG_SKIP_RTCP 4
 
 #define FF_RTP_FLAG_OPTS(ctx, fieldname) \
     { "rtpflags", "RTP muxer flags", offsetof(ctx, fieldname), AV_OPT_TYPE_FLAGS, {.dbl = 0}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
-    { "latm", "Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_MP4A_LATM}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" } \
+    { "latm", "Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_MP4A_LATM}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
+    { "rfc2190", "Use RFC 2190 packetization instead of RFC 4629 for H.263", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_RFC2190}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
+    { "skip_rtcp", "Don't send RTCP sender reports", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_SKIP_RTCP}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
 
 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);
 
 void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size);
 void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size);
+void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf1, int size);
 void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size);
 void ff_rtp_send_latm(AVFormatContext *s1, const uint8_t *buff, int size);
 void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size);
@@ -80,4 +86,7 @@ void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);
 void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size);
 void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buff, int size);
 
+const uint8_t *ff_h263_find_resync_marker_reverse(const uint8_t *restrict start,
+                                                  const uint8_t *restrict end);
+
 #endif /* AVFORMAT_RTPENC_H */
diff --git a/libavformat/rtpenc_h263.c b/libavformat/rtpenc_h263.c
index 84403a1..e14aaf1 100644
--- a/libavformat/rtpenc_h263.c
+++ b/libavformat/rtpenc_h263.c
@@ -23,8 +23,8 @@
 #include "avformat.h"
 #include "rtpenc.h"
 
-static const uint8_t *find_resync_marker_reverse(const uint8_t *restrict start,
-                                                 const uint8_t *restrict end)
+const uint8_t *ff_h263_find_resync_marker_reverse(const uint8_t *restrict start,
+                                                  const uint8_t *restrict end)
 {
     const uint8_t *p = end - 1;
     start += 1; /* Make sure we never return the original start. */
@@ -63,7 +63,8 @@ void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size)
 
         /* Look for a better place to split the frame into packets. */
         if (len < size) {
-            const uint8_t *end = find_resync_marker_reverse(buf1, buf1 + len);
+            const uint8_t *end = ff_h263_find_resync_marker_reverse(buf1,
+                                                                    buf1 + len);
             len = end - buf1;
         }
 
diff --git a/libavformat/rtpenc_h263_rfc2190.c b/libavformat/rtpenc_h263_rfc2190.c
new file mode 100644
index 0000000..305c1a2
--- /dev/null
+++ b/libavformat/rtpenc_h263_rfc2190.c
@@ -0,0 +1,104 @@
+/*
+ * RTP packetization for H.263 video
+ * Copyright (c) 2012 Martin Storsjo
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avformat.h"
+#include "rtpenc.h"
+#include "libavcodec/put_bits.h"
+#include "libavcodec/get_bits.h"
+
+struct H263Info {
+    int src;
+    int i;
+    int u;
+    int s;
+    int a;
+    int pb;
+    int tr;
+};
+
+static void send_mode_a(AVFormatContext *s1, const struct H263Info *info,
+                        const uint8_t *buf, int len, int m)
+{
+    RTPMuxContext *s = s1->priv_data;
+    PutBitContext pb;
+
+    init_put_bits(&pb, s->buf, 32);
+    put_bits(&pb, 1, 0); /* F - 0, mode A */
+    put_bits(&pb, 1, 0); /* P - 0, normal I/P */
+    put_bits(&pb, 3, 0); /* SBIT - 0 bits */
+    put_bits(&pb, 3, 0); /* EBIT - 0 bits */
+    put_bits(&pb, 3, info->src); /* SRC - source format */
+    put_bits(&pb, 1, info->i); /* I - inter/intra */
+    put_bits(&pb, 1, info->u); /* U - unrestricted motion vector */
+    put_bits(&pb, 1, info->s); /* S - syntax-baesd arithmetic coding */
+    put_bits(&pb, 1, info->a); /* A - advanced prediction */
+    put_bits(&pb, 4, 0); /* R - reserved */
+    put_bits(&pb, 2, 0); /* DBQ - 0 */
+    put_bits(&pb, 3, 0); /* TRB - 0 */
+    put_bits(&pb, 8, info->tr); /* TR */
+    flush_put_bits(&pb);
+    memcpy(s->buf + 4, buf, len);
+
+    ff_rtp_send_data(s1, s->buf, len + 4, m);
+}
+
+void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf, int size)
+{
+    RTPMuxContext *s = s1->priv_data;
+    int len;
+    GetBitContext gb;
+    struct H263Info info = { 0 };
+
+    s->timestamp = s->cur_timestamp;
+
+    init_get_bits(&gb, buf, size*8);
+    if (get_bits(&gb, 22) == 0x20) { /* Picture Start Code */
+        info.tr  = get_bits(&gb, 8);
+        skip_bits(&gb, 2); /* PTYPE start, H261 disambiguation */
+        skip_bits(&gb, 3); /* Split screen, document camera, freeze picture release */
+        info.src = get_bits(&gb, 3);
+        info.i   = get_bits(&gb, 1);
+        info.u   = get_bits(&gb, 1);
+        info.s   = get_bits(&gb, 1);
+        info.a   = get_bits(&gb, 1);
+        info.pb  = get_bits(&gb, 1);
+    }
+
+    while (size > 0) {
+        len = FFMIN(s->max_payload_size - 4, size);
+
+        /* Look for a better place to split the frame into packets. */
+        if (len < size) {
+            const uint8_t *end = ff_h263_find_resync_marker_reverse(buf,
+                                                                    buf + len);
+            len = end - buf;
+            if (len == s->max_payload_size - 4)
+                av_log(s1, AV_LOG_WARNING,
+                       "No GOB boundary found within MTU size, splitting at "
+                       "a random boundary\n");
+        }
+
+        send_mode_a(s1, &info, buf, len, len == size);
+
+        buf  += len;
+        size -= len;
+    }
+}
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index bc03db4..0d919ae 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -73,7 +73,7 @@
 
 const AVOption ff_rtsp_options[] = {
     { "initial_pause",  "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
-    FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
+    FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags)
     { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
     { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
     { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
diff --git a/libavformat/sdp.c b/libavformat/sdp.c
index 20cf588..77b8945 100644
--- a/libavformat/sdp.c
+++ b/libavformat/sdp.c
@@ -404,6 +404,9 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
              * actually specifies the maximum video size, but we only know
              * the current size. This is required for playback on Android
              * stagefright and on Samsung bada. */
+            if (!fmt || !fmt->oformat->priv_class ||
+                !av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190") ||
+                c->codec_id == CODEC_ID_H263P)
             av_strlcatf(buff, size, "a=rtpmap:%d H263-2000/90000\r\n"
                                     "a=framesize:%d %d-%d\r\n",
                                     payload_type,
diff --git a/libavformat/swfdec.c b/libavformat/swfdec.c
index 2a01286..a84d380 100644
--- a/libavformat/swfdec.c
+++ b/libavformat/swfdec.c
@@ -84,7 +84,7 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
     SWFContext *swf = s->priv_data;
     AVIOContext *pb = s->pb;
     AVStream *vst = NULL, *ast = NULL, *st = 0;
-    int tag, len, i, frame, v;
+    int tag, len, i, frame, v, res;
 
     for(;;) {
         uint64_t pos = avio_tell(pb);
@@ -147,7 +147,8 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
                 st = s->streams[i];
                 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO && st->id == ch_id) {
                     frame = avio_rl16(pb);
-                    av_get_packet(pb, pkt, len-2);
+                    if ((res = av_get_packet(pb, pkt, len-2)) < 0)
+                        return res;
                     pkt->pos = pos;
                     pkt->pts = frame;
                     pkt->stream_index = st->index;
@@ -160,9 +161,11 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
                 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && st->id == -1) {
             if (st->codec->codec_id == CODEC_ID_MP3) {
                 avio_skip(pb, 4);
-                av_get_packet(pb, pkt, len-4);
+                if ((res = av_get_packet(pb, pkt, len-4)) < 0)
+                    return res;
             } else { // ADPCM, PCM
-                av_get_packet(pb, pkt, len);
+                if ((res = av_get_packet(pb, pkt, len)) < 0)
+                    return res;
             }
             pkt->pos = pos;
             pkt->stream_index = st->index;
@@ -187,7 +190,8 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
                 st = vst;
             }
             avio_rl16(pb); /* BITMAP_ID */
-            av_new_packet(pkt, len-2);
+            if ((res = av_new_packet(pkt, len-2)) < 0)
+                return res;
             avio_read(pb, pkt->data, 4);
             if (AV_RB32(pkt->data) == 0xffd8ffd9 ||
                 AV_RB32(pkt->data) == 0xffd9ffd8) {