Merge remote-tracking branch 'qatar/master'

* qatar/master:
  avcodec_default_reget_buffer(): fix compilation in DEBUG mode
  fate: Overhaul WavPack coverage
  h264: fix mmxext chroma deblock to use correct TC values.
  flvdec: Remove the now redundant check for known broken metadata creator
  flvdec: Validate index entries added from metadata while reading
  rtsp: Handle requests from server to client
  movenc: use timestamps instead of frame_size for samples-per-packet
  movenc: use the first cluster duration as the tfhd default duration
  movenc: factorize calculation of cluster duration into a separate function
  doc/APIchanges: fill in missing dates and hashes.
  lavc: reorder AVCodecContext fields.
  lavc: reorder AVFrame fields.

Conflicts:
	doc/APIchanges
	libavcodec/avcodec.h
	libavformat/flvdec.c
	libavformat/movenc.c
	tests/fate/lossless-audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>

doc/APIchanges

@@ -35,19 +35,19 @@ API changes, most recent first:
 2012-01-24 - xxxxxxx - lavfi 2.60.100
   Add avfilter_graph_dump.
 
-2012-xx-xx - xxxxxxx - lavu 51.24.0 - error.h
+2012-02-25 - c9bca80 - lavu 51.24.0 - error.h
   Add AVERROR_UNKNOWN
 
-2012-xx-xx - xxxxxxx - lavc 54.x.x
+2012-02-20 - e9cda85 - lavc 54.2.0
   Add duration field to AVCodecParserContext
 
-2012-02-xx - xxxxxxx - lavu 51.23.1 - mathematics.h
+2012-02-20 - 0b42a93 - lavu 51.23.1 - mathematics.h
   Add av_rescale_q_rnd()
 
-2012-02-xx - xxxxxxx - lavu 51.22.1 - pixdesc.h
+2012-02-08 - 38d5533 - lavu 51.22.1 - pixdesc.h
   Add PIX_FMT_PSEUDOPAL flag.
 
-2012-02-01 - xxxxxxx - lavc 54.01.0
+2012-02-08 - 52f82a1 - lavc 54.01.0
   Add avcodec_encode_video2() and deprecate avcodec_encode_video().
 
 2012-02-01 - 316fc74 - lavc 54.01.0

libavcodec/pthread.c

@@ -480,7 +480,7 @@ static int update_context_from_user(AVCodecContext *dst, AVCodecContext *src)
     dst->slice_flags = src->slice_flags;
     dst->flags2      = src->flags2;
 
-    copy_fields(skip_loop_filter, bidir_refine);
+    copy_fields(skip_loop_filter, subtitle_header);
 
     dst->frame_number     = src->frame_number;
     dst->reordered_opaque = src->reordered_opaque;

libavcodec/utils.c

@@ -603,7 +603,7 @@ int avcodec_default_reget_buffer(AVCodecContext *s, AVFrame *pic){
         return s->get_buffer(s, pic);
     }
 
-    assert(s->pix_fmt == pic->pix_fmt);
+    assert(s->pix_fmt == pic->format);
 
     /* If internal buffer type return the same buffer */
     if(pic->type == FF_BUFFER_TYPE_INTERNAL) {

libavcodec/x86/h264_deblock_10bit.asm

@@ -876,7 +876,7 @@ cglobal deblock_v_chroma_10_%1, 5,7-(mmsize/16),8*(mmsize/16)
 %if mmsize < 16
     add         r0, mmsize
     add         r5, mmsize
-    add         r4, mmsize/8
+    add         r4, mmsize/4
     dec         r6
     jg .loop
     REP_RET

libavformat/flvdec.c

@@ -35,12 +35,20 @@
 #include "avio_internal.h"
 #include "flv.h"
 
+#define VALIDATE_INDEX_TS_THRESH 2500
+
 typedef struct {
     int wrong_dts; ///< wrong dts due to negative cts
     uint8_t *new_extradata[FLV_STREAM_TYPE_NB];
     int      new_extradata_size[FLV_STREAM_TYPE_NB];
     int      last_sample_rate;
     int      last_channels;
+    struct {
+        int64_t dts;
+        int64_t pos;
+    } validate_index[2];
+    int validate_next;
+    int validate_count;
 } FLVContext;
 
 static int flv_probe(AVProbeData *p)
@@ -137,6 +145,7 @@ static int amf_get_string(AVIOContext *ioc, char *buffer, int buffsize) {
 }
 
 static int parse_keyframes_index(AVFormatContext *s, AVIOContext *ioc, AVStream *vstream, int64_t max_pos) {
+    FLVContext *flv = s->priv_data;
     unsigned int timeslen = 0, fileposlen = 0, i;
     char str_val[256];
     int64_t *times = NULL;
@@ -192,18 +201,15 @@ static int parse_keyframes_index(AVFormatContext *s, AVIOContext *ioc, AVStream
     }
 
     if (timeslen == fileposlen && fileposlen>1 && max_pos <= filepositions[0]) {
-        int64_t av_unused dts, size0, size1;
-        avio_seek(ioc, filepositions[1]-4, SEEK_SET);
-        size0 = avio_rb32(ioc);
-                avio_r8(ioc);
-        size1 = avio_rb24(ioc);
-        dts   = avio_rb24(ioc);
-        dts  |= avio_r8(ioc) << 24;
-        if (size0 > filepositions[1] || FFABS(dts - times[1]*1000)>5000/*arbitraray threshold to detect invalid index*/)
-            goto invalid;
-        for(i = 0; i < timeslen; i++)
+        for (i = 0; i < fileposlen; i++) {
             av_add_index_entry(vstream, filepositions[i], times[i]*1000,
                                0, 0, AVINDEX_KEYFRAME);
+            if (i < 2) {
+                flv->validate_index[i].pos = filepositions[i];
+                flv->validate_index[i].dts = times[i] * 1000;
+                flv->validate_count = i + 1;
+            }
+        }
     } else {
 invalid:
         av_log(s, AV_LOG_WARNING, "Invalid keyframes object, skipping.\n");
@@ -444,6 +450,22 @@ static int flv_queue_extradata(FLVContext *flv, AVIOContext *pb, int stream,
     return 0;
 }
 
+static void clear_index_entries(AVFormatContext *s, int64_t pos)
+{
+    int i, j, out;
+    av_log(s, AV_LOG_WARNING, "Found invalid index entries, clearing the index.\n");
+    for (i = 0; i < s->nb_streams; i++) {
+        AVStream *st = s->streams[i];
+        /* Remove all index entries that point to >= pos */
+        out = 0;
+        for (j = 0; j < st->nb_index_entries; j++) {
+            if (st->index_entries[j].pos < pos)
+                st->index_entries[out++] = st->index_entries[j];
+        }
+        st->nb_index_entries = out;
+    }
+}
+
 static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
 {
     FLVContext *flv = s->priv_data;
@@ -467,6 +489,22 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
     avio_skip(s->pb, 3); /* stream id, always 0 */
     flags = 0;
 
+    if (flv->validate_next < flv->validate_count) {
+        int64_t validate_pos = flv->validate_index[flv->validate_next].pos;
+        if (pos == validate_pos) {
+            if (FFABS(dts - flv->validate_index[flv->validate_next].dts) <=
+                VALIDATE_INDEX_TS_THRESH) {
+                flv->validate_next++;
+            } else {
+                clear_index_entries(s, validate_pos);
+                flv->validate_count = 0;
+            }
+        } else if (pos > validate_pos) {
+            clear_index_entries(s, validate_pos);
+            flv->validate_count = 0;
+        }
+    }
+
     if(size == 0)
         continue;
 
@@ -654,6 +692,8 @@ leave:
 static int flv_read_seek(AVFormatContext *s, int stream_index,
     int64_t ts, int flags)
 {
+    FLVContext *flv = s->priv_data;
+    flv->validate_count = 0;
     return avio_seek_time(s->pb, stream_index, ts, flags);
 }
 

libavformat/movenc.c

@@ -565,6 +565,42 @@ static int mov_get_lpcm_flags(enum CodecID codec_id)
     }
 }
 
+static int get_cluster_duration(MOVTrack *track, int cluster_idx)
+{
+    int64_t next_dts;
+
+    if (cluster_idx >= track->entry)
+        return 0;
+
+    if (cluster_idx + 1 == track->entry)
+        next_dts = track->track_duration + track->start_dts;
+    else
+        next_dts = track->cluster[cluster_idx + 1].dts;
+
+    return next_dts - track->cluster[cluster_idx].dts;
+}
+
+static int get_samples_per_packet(MOVTrack *track)
+{
+    int i, first_duration;
+
+// return track->enc->frame_size;
+
+    /* use 1 for raw PCM */
+    if (!track->audio_vbr)
+        return 1;
+
+    /* check to see if duration is constant for all clusters */
+    if (!track->entry)
+        return 0;
+    first_duration = get_cluster_duration(track, 0);
+    for (i = 1; i < track->entry; i++) {
+        if (get_cluster_duration(track, i) != first_duration)
+            return 0;
+    }
+    return first_duration;
+}
+
 static int mov_write_audio_tag(AVIOContext *pb, MOVTrack *track)
 {
     int64_t pos = avio_tell(pb);
@@ -607,7 +643,7 @@ static int mov_write_audio_tag(AVIOContext *pb, MOVTrack *track)
         avio_wb32(pb, av_get_bits_per_sample(track->enc->codec_id));
         avio_wb32(pb, mov_get_lpcm_flags(track->enc->codec_id));
         avio_wb32(pb, track->sample_size);
-        avio_wb32(pb, track->enc->frame_size);
+        avio_wb32(pb, get_samples_per_packet(track));
     } else {
         if (track->mode == MODE_MOV) {
             avio_wb16(pb, track->enc->channels);
@@ -1143,9 +1179,7 @@ static int mov_write_stts_tag(AVIOContext *pb, MOVTrack *track)
                        av_malloc(track->entry * sizeof(*stts_entries)) : /* worst case */
                        NULL;
         for (i=0; i<track->entry; i++) {
-            int64_t duration = i + 1 == track->entry ?
-                track->track_duration - track->cluster[i].dts + track->start_dts : /* readjusting */
-                track->cluster[i+1].dts - track->cluster[i].dts;
+            int duration = get_cluster_duration(track, i);
             if (i && duration == stts_entries[entries].duration) {
                 stts_entries[entries].count++; /* compress */
             } else {
@@ -2262,7 +2296,7 @@ static int mov_write_tfhd_tag(AVIOContext *pb, MOVTrack *track,
     if (flags & MOV_TFHD_BASE_DATA_OFFSET)
         avio_wb64(pb, moof_offset);
     if (flags & MOV_TFHD_DEFAULT_DURATION) {
-        track->default_duration = track->audio_vbr ? track->enc->frame_size : 1;
+        track->default_duration = get_cluster_duration(track, 0);
         avio_wb32(pb, track->default_duration);
     }
     if (flags & MOV_TFHD_DEFAULT_SIZE) {
@@ -2295,10 +2329,7 @@ static int mov_write_trun_tag(AVIOContext *pb, MOVTrack *track)
     int i;
 
     for (i = 0; i < track->entry; i++) {
-        int64_t duration = i + 1 == track->entry ?
-            track->track_duration - track->cluster[i].dts + track->start_dts :
-            track->cluster[i + 1].dts - track->cluster[i].dts;
-        if (duration != track->default_duration)
+        if (get_cluster_duration(track, i) != track->default_duration)
             flags |= MOV_TRUN_SAMPLE_DURATION;
         if (track->cluster[i].size != track->default_size)
             flags |= MOV_TRUN_SAMPLE_SIZE;
@@ -2322,11 +2353,8 @@ static int mov_write_trun_tag(AVIOContext *pb, MOVTrack *track)
         avio_wb32(pb, get_sample_flags(track, &track->cluster[0]));
 
     for (i = 0; i < track->entry; i++) {
-        int64_t duration = i + 1 == track->entry ?
-            track->track_duration - track->cluster[i].dts + track->start_dts :
-            track->cluster[i + 1].dts - track->cluster[i].dts;
         if (flags & MOV_TRUN_SAMPLE_DURATION)
-            avio_wb32(pb, duration);
+            avio_wb32(pb, get_cluster_duration(track, i));
         if (flags & MOV_TRUN_SAMPLE_SIZE)
             avio_wb32(pb, track->cluster[i].size);
         if (flags & MOV_TRUN_SAMPLE_FLAGS)

libavformat/rtsp.c

@@ -900,9 +900,13 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
     char buf[4096], buf1[1024], *q;
     unsigned char ch;
     const char *p;
-    int ret, content_length, line_count = 0;
+    int ret, content_length, line_count = 0, request = 0;
     unsigned char *content = NULL;
 
+start:
+    line_count = 0;
+    request = 0;
+    content = NULL;
     memset(reply, 0, sizeof(*reply));
 
     /* parse reply (XXX: use buffers) */
@@ -938,9 +942,15 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
         if (line_count == 0) {
             /* get reply code */
             get_word(buf1, sizeof(buf1), &p);
+            if (!strncmp(buf1, "RTSP/", 5)) {
                 get_word(buf1, sizeof(buf1), &p);
                 reply->status_code = atoi(buf1);
                 av_strlcpy(reply->reason, p, sizeof(reply->reason));
+            } else {
+                av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
+                get_word(buf1, sizeof(buf1), &p); // object
+                request = 1;
+            }
         } else {
             ff_rtsp_parse_line(reply, p, rt, method);
             av_strlcat(rt->last_reply, p,    sizeof(rt->last_reply));
@@ -949,7 +959,7 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
         line_count++;
     }
 
-    if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
+    if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
         av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
 
     content_length = reply->content_length;
@@ -964,6 +974,44 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
     else
         av_free(content);
 
+    if (request) {
+        char buf[1024];
+        char base64buf[AV_BASE64_SIZE(sizeof(buf))];
+        const char* ptr = buf;
+
+        if (!strcmp(reply->reason, "OPTIONS")) {
+            snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
+            if (reply->seq)
+                av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
+            if (reply->session_id[0])
+                av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
+                                              reply->session_id);
+        } else {
+            snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
+        }
+        av_strlcat(buf, "\r\n", sizeof(buf));
+
+        if (rt->control_transport == RTSP_MODE_TUNNEL) {
+            av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
+            ptr = base64buf;
+        }
+        ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
+
+        rt->last_cmd_time = av_gettime();
+        /* Even if the request from the server had data, it is not the data
+         * that the caller wants or expects. The memory could also be leaked
+         * if the actual following reply has content data. */
+        if (content_ptr)
+            av_freep(content_ptr);
+        /* If method is set, this is called from ff_rtsp_send_cmd,
+         * where a reply to exactly this request is awaited. For
+         * callers from within packet reciving, we just want to
+         * return to the caller and go back to receiving packets. */
+        if (method)
+            goto start;
+        return 0;
+    }
+
     if (rt->seq != reply->seq) {
         av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
             rt->seq, reply->seq);

tests/fate/lossless-audio.mak

@@ -15,3 +15,4 @@ fate-lossless-tta: CMD = crc -i $(SAMPLES)/lossless-audio/inside.tta
 
 FATE_TESTS += $(FATE_LOSSLESS_AUDIO)
 fate-lossless-audio: $(FATE_LOSSLESS_AUDIO)
+