Merge commit '1cd432e167b1a80853760c89a33606e2b5f229c2'

* commit '1cd432e167b1a80853760c89a33606e2b5f229c2':
  configure: fix libcdio check
  rtsp: Allow setting the reordering buffer size via an AVOption
  rtsp: Vertically align a constant definition
  rtp: Update the check for distinguishing between RTP and RTCP
  aac: fix build with hardcoded tables
  fate: dependencies for screen codec tests
  riff: Move functions around to be covered by appropriate #ifdefs

Conflicts:
	configure
	tests/fate/screen.mak

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

4 files changed, 112 insertions, 84 deletions

diff --git a/libavformat/riff.c b/libavformat/riff.c
index 769a4d6ba5..4cdefb91cc 100644
--- a/libavformat/riff.c
+++ b/libavformat/riff.c
@@ -589,6 +589,84 @@ void ff_put_bmp_header(AVIOContext *pb, AVCodecContext *enc, const AVCodecTag *t
     if (!for_asf && enc->extradata_size & 1)
         avio_w8(pb, 0);
 }
+
+void ff_parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale)
+{
+    int gcd;
+    int audio_frame_size;
+
+    /* We use the known constant frame size for the codec if known, otherwise
+       fallback to using AVCodecContext.frame_size, which is not as reliable
+       for indicating packet duration */
+    audio_frame_size = av_get_audio_frame_duration(stream, 0);
+    if (!audio_frame_size)
+        audio_frame_size = stream->frame_size;
+
+    *au_ssize= stream->block_align;
+    if (audio_frame_size && stream->sample_rate) {
+        *au_scale = audio_frame_size;
+        *au_rate= stream->sample_rate;
+    }else if(stream->codec_type == AVMEDIA_TYPE_VIDEO ||
+             stream->codec_type == AVMEDIA_TYPE_DATA ||
+             stream->codec_type == AVMEDIA_TYPE_SUBTITLE){
+        *au_scale= stream->time_base.num;
+        *au_rate = stream->time_base.den;
+    }else{
+        *au_scale= stream->block_align ? stream->block_align*8 : 8;
+        *au_rate = stream->bit_rate ? stream->bit_rate : 8*stream->sample_rate;
+    }
+    gcd= av_gcd(*au_scale, *au_rate);
+    *au_scale /= gcd;
+    *au_rate /= gcd;
+}
+
+void ff_riff_write_info_tag(AVIOContext *pb, const char *tag, const char *str)
+{
+    int len = strlen(str);
+    if (len > 0) {
+        len++;
+        ffio_wfourcc(pb, tag);
+        avio_wl32(pb, len);
+        avio_put_str(pb, str);
+        if (len & 1)
+            avio_w8(pb, 0);
+    }
+}
+
+static int riff_has_valid_tags(AVFormatContext *s)
+{
+    int i;
+    AVDictionaryEntry *t = NULL;
+
+    for (i = 0; *ff_riff_tags[i]; i++) {
+        if ((t = av_dict_get(s->metadata, ff_riff_tags[i], NULL, AV_DICT_MATCH_CASE)))
+            return 1;
+    }
+
+    return 0;
+}
+
+void ff_riff_write_info(AVFormatContext *s)
+{
+    AVIOContext *pb = s->pb;
+    int i;
+    int64_t list_pos;
+    AVDictionaryEntry *t = NULL;
+
+    ff_metadata_conv(&s->metadata, ff_riff_info_conv, NULL);
+
+    /* writing empty LIST is not nice and may cause problems */
+    if (!riff_has_valid_tags(s))
+        return;
+
+    list_pos = ff_start_tag(pb, "LIST");
+    ffio_wfourcc(pb, "INFO");
+    for (i = 0; *ff_riff_tags[i]; i++) {
+        if ((t = av_dict_get(s->metadata, ff_riff_tags[i], NULL, AV_DICT_MATCH_CASE)))
+            ff_riff_write_info_tag(s->pb, t->key, t->value);
+    }
+    ff_end_tag(pb, list_pos);
+}
 #endif //CONFIG_MUXERS
 
 #if CONFIG_DEMUXERS
@@ -705,37 +783,6 @@ int ff_get_bmp_header(AVIOContext *pb, AVStream *st, unsigned *esize)
     avio_rl32(pb); /* ClrImportant */
     return tag1;
 }
-#endif // CONFIG_DEMUXERS
-
-void ff_parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale)
-{
-    int gcd;
-    int audio_frame_size;
-
-    /* We use the known constant frame size for the codec if known, otherwise
-       fallback to using AVCodecContext.frame_size, which is not as reliable
-       for indicating packet duration */
-    audio_frame_size = av_get_audio_frame_duration(stream, 0);
-    if (!audio_frame_size)
-        audio_frame_size = stream->frame_size;
-
-    *au_ssize= stream->block_align;
-    if (audio_frame_size && stream->sample_rate) {
-        *au_scale = audio_frame_size;
-        *au_rate= stream->sample_rate;
-    }else if(stream->codec_type == AVMEDIA_TYPE_VIDEO ||
-             stream->codec_type == AVMEDIA_TYPE_DATA ||
-             stream->codec_type == AVMEDIA_TYPE_SUBTITLE){
-        *au_scale= stream->time_base.num;
-        *au_rate = stream->time_base.den;
-    }else{
-        *au_scale= stream->block_align ? stream->block_align*8 : 8;
-        *au_rate = stream->bit_rate ? stream->bit_rate : 8*stream->sample_rate;
-    }
-    gcd= av_gcd(*au_scale, *au_rate);
-    *au_scale /= gcd;
-    *au_rate /= gcd;
-}
 
 void ff_get_guid(AVIOContext *s, ff_asf_guid *g)
 {
@@ -799,51 +846,4 @@ int ff_read_riff_info(AVFormatContext *s, int64_t size)
 
     return 0;
 }
-
-static int riff_has_valid_tags(AVFormatContext *s)
-{
-    int i;
-    AVDictionaryEntry *t = NULL;
-
-    for (i = 0; *ff_riff_tags[i]; i++) {
-        if ((t = av_dict_get(s->metadata, ff_riff_tags[i], NULL, AV_DICT_MATCH_CASE)))
-            return 1;
-    }
-
-    return 0;
-}
-
-void ff_riff_write_info_tag(AVIOContext *pb, const char *tag, const char *str)
-{
-    int len = strlen(str);
-    if (len > 0) {
-        len++;
-        ffio_wfourcc(pb, tag);
-        avio_wl32(pb, len);
-        avio_put_str(pb, str);
-        if (len & 1)
-            avio_w8(pb, 0);
-    }
-}
-
-void ff_riff_write_info(AVFormatContext *s)
-{
-    AVIOContext *pb = s->pb;
-    int i;
-    int64_t list_pos;
-    AVDictionaryEntry *t = NULL;
-
-    ff_metadata_conv(&s->metadata, ff_riff_info_conv, NULL);
-
-    /* writing empty LIST is not nice and may cause problems */
-    if (!riff_has_valid_tags(s))
-        return;
-
-    list_pos = ff_start_tag(pb, "LIST");
-    ffio_wfourcc(pb, "INFO");
-    for (i = 0; *ff_riff_tags[i]; i++) {
-        if ((t = av_dict_get(s->metadata, ff_riff_tags[i], NULL, AV_DICT_MATCH_CASE)))
-            ff_riff_write_info_tag(s->pb, t->key, t->value);
-    }
-    ff_end_tag(pb, list_pos);
-}
+#endif // CONFIG_DEMUXERS
diff --git a/libavformat/rtp.h b/libavformat/rtp.h
index fa3e1fa1bd..ab2de528a0 100644
--- a/libavformat/rtp.h
+++ b/libavformat/rtp.h
@@ -84,13 +84,24 @@ enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type);
 
 /* RTCP packet types */
 enum RTCPType {
+    RTCP_FIR    = 192,
+    RTCP_NACK, // 193
+    RTCP_SMPTETC,// 194
+    RTCP_IJ,   // 195
     RTCP_SR     = 200,
     RTCP_RR,   // 201
     RTCP_SDES, // 202
     RTCP_BYE,  // 203
-    RTCP_APP   // 204
+    RTCP_APP,  // 204
+    RTCP_RTPFB,// 205
+    RTCP_PSFB, // 206
+    RTCP_XR,   // 207
+    RTCP_AVB,  // 208
+    RTCP_RSI,  // 209
+    RTCP_TOKEN,// 210
 };
 
-#define RTP_PT_IS_RTCP(x) ((x) >= RTCP_SR && (x) <= RTCP_APP)
+#define RTP_PT_IS_RTCP(x) (((x) >= RTCP_FIR && (x) <= RTCP_IJ) || \
+                           ((x) >= RTCP_SR  && (x) <= RTCP_TOKEN))
 
 #endif /* AVFORMAT_RTP_H */
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 80f1047c5d..0328d28989 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -75,6 +75,9 @@
     { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
     { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
 
+#define RTSP_REORDERING_OPTS() \
+    { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
+
 const AVOption ff_rtsp_options[] = {
     { "initial_pause",  "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
     FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
@@ -88,17 +91,20 @@ const AVOption ff_rtsp_options[] = {
     { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
     { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
     { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
+    RTSP_REORDERING_OPTS(),
     { NULL },
 };
 
 static const AVOption sdp_options[] = {
     RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
     RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
+    RTSP_REORDERING_OPTS(),
     { NULL },
 };
 
 static const AVOption rtp_options[] = {
     RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
+    RTSP_REORDERING_OPTS(),
     { NULL },
 };
 
@@ -608,6 +614,13 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
 {
     RTSPState *rt = s->priv_data;
     AVStream *st = NULL;
+    int reordering_queue_size = rt->reordering_queue_size;
+    if (reordering_queue_size < 0) {
+        if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
+            reordering_queue_size = 0;
+        else
+            reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
+    }
 
     /* open the RTP context */
     if (rtsp_st->stream_index >= 0)
@@ -632,8 +645,7 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
     else if (CONFIG_RTPDEC)
         rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
                                          rtsp_st->sdp_payload_type,
-            (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
-            ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
+                                         reordering_queue_size);
 
     if (!rtsp_st->transport_priv) {
          return AVERROR(ENOMEM);
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index e55073c430..176fef7ee2 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -385,12 +385,17 @@ typedef struct RTSPState {
      * Timeout to wait for incoming connections.
      */
     int initial_timeout;
+
+    /**
+     * Size of RTP packet reordering queue.
+     */
+    int reordering_queue_size;
 } RTSPState;
 
 #define RTSP_FLAG_FILTER_SRC  0x1    /**< Filter incoming UDP packets -
                                           receive packets only from the right
                                           source address and port. */
-#define RTSP_FLAG_LISTEN 0x2         /**< Wait for incoming connections. */
+#define RTSP_FLAG_LISTEN      0x2    /**< Wait for incoming connections. */
 
 /**
  * Describe a single stream, as identified by a single m= line block in the