Merge remote-tracking branch 'qatar/master'

* qatar/master:
  tiertexseq: set correct block_align for audio
  tiertexseq: set audio stream start time to 0
  voc/avs: Do not change the sample rate mid-stream.
  segafilm: use the sample rate as the time base for audio streams
  ea: fix audio pts
  psx-str: fix audio pts
  vqf: set packet duration
  tta demuxer: set packet duration
  mpegaudio_parser: do not ignore information from the first parsed frame
  mpegaudio_parser: be less picky about the start position
  thp: set audio packet durations
  avcodec: add a Vorbis parser to get packet duration
  vorbisdec: read the previous window flag for long windows
  lavc: free the output packet when encoding failed or produced no output.
  lavc: preserve avpkt->destruct in ff_alloc_packet().
  lavc: clarify the meaning of AVCodecContext.frame_number.
  mpegts: Pad the packet buffer in handle_packet().
  mpegts: Do not call read_sl_header() when no bytes remain in the buffer.

Conflicts:
	libavcodec/mpegaudio_parser.c
	libavcodec/version.h
	libavformat/mpegts.c
	tests/ref/fate/pva-demux

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

12 files changed, 62 insertions, 38 deletions

diff --git a/libavformat/electronicarts.c b/libavformat/electronicarts.c
index 9719c6711a..968682af7e 100644
--- a/libavformat/electronicarts.c
+++ b/libavformat/electronicarts.c
@@ -70,7 +70,6 @@ typedef struct EaDemuxContext {
 
     enum CodecID audio_codec;
     int audio_stream_index;
-    int audio_frame_counter;
 
     int bytes;
     int sample_rate;
@@ -472,7 +471,7 @@ static int ea_read_header(AVFormatContext *s)
             st->codec->bits_per_coded_sample / 4;
         st->codec->block_align = st->codec->channels*st->codec->bits_per_coded_sample;
         ea->audio_stream_index = st->index;
-        ea->audio_frame_counter = 0;
+        st->start_time = 0;
     }
 
     return 1;
@@ -522,24 +521,26 @@ static int ea_read_packet(AVFormatContext *s,
             if (ret < 0)
                 return ret;
             pkt->stream_index = ea->audio_stream_index;
-            pkt->pts = 90000;
-            pkt->pts *= ea->audio_frame_counter;
-            pkt->pts /= ea->sample_rate;
 
             switch (ea->audio_codec) {
             case CODEC_ID_ADPCM_EA:
-                /* 2 samples/byte, 1 or 2 samples per frame depending
-                 * on stereo; chunk also has 12-byte header */
-                ea->audio_frame_counter += ((chunk_size - 12) * 2) /
-                    ea->num_channels;
+            case CODEC_ID_ADPCM_EA_R1:
+            case CODEC_ID_ADPCM_EA_R2:
+            case CODEC_ID_ADPCM_IMA_EA_EACS:
+                pkt->duration = AV_RL32(pkt->data);
+                break;
+            case CODEC_ID_ADPCM_EA_R3:
+                pkt->duration = AV_RB32(pkt->data);
+                break;
+            case CODEC_ID_ADPCM_IMA_EA_SEAD:
+                pkt->duration = ret * 2 / ea->num_channels;
                 break;
             case CODEC_ID_PCM_S16LE_PLANAR:
             case CODEC_ID_MP3:
-                ea->audio_frame_counter += num_samples;
+                pkt->duration = num_samples;
                 break;
             default:
-                ea->audio_frame_counter += chunk_size /
-                    (ea->bytes * ea->num_channels);
+                pkt->duration = chunk_size / (ea->bytes * ea->num_channels);
             }
 
             packet_read = 1;
diff --git a/libavformat/mpegts.c b/libavformat/mpegts.c
index 460372c116..2d86a263ab 100644
--- a/libavformat/mpegts.c
+++ b/libavformat/mpegts.c
@@ -714,11 +714,6 @@ static int read_sl_header(PESContext *pes, SLConfigDescr *sl, const uint8_t *buf
     int dts_flag = -1, cts_flag = -1;
     int64_t dts = AV_NOPTS_VALUE, cts = AV_NOPTS_VALUE;
 
-    if (buf_size<=0) {
-        av_log(0,AV_LOG_WARNING, "empty SL header\n");
-        return 0;
-    }
-
     init_get_bits(&gb, buf, buf_size*8);
 
     if (sl->use_au_start)
@@ -928,7 +923,7 @@ static int mpegts_push_data(MpegTSFilter *filter,
                 /* we got the full header. We parse it and get the payload */
                 pes->state = MPEGTS_PAYLOAD;
                 pes->data_index = 0;
-                if (pes->stream_type == 0x12) {
+                if (pes->stream_type == 0x12 && buf_size > 0) {
                     int sl_header_bytes = read_sl_header(pes, &pes->sl, p, buf_size);
                     pes->pes_header_size += sl_header_bytes;
                     p += sl_header_bytes;
@@ -1844,6 +1839,7 @@ static int handle_packets(MpegTSContext *ts, int nb_packets)
 
     ts->stop_parse = 0;
     packet_num = 0;
+    memset(packet + TS_PACKET_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
     for(;;) {
         packet_num++;
         if (nb_packets != 0 && packet_num >= nb_packets ||
diff --git a/libavformat/oggparsevorbis.c b/libavformat/oggparsevorbis.c
index ba9b348456..b9d9f575e0 100644
--- a/libavformat/oggparsevorbis.c
+++ b/libavformat/oggparsevorbis.c
@@ -251,6 +251,7 @@ vorbis_header (AVFormatContext * s, int idx)
 
         st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
         st->codec->codec_id = CODEC_ID_VORBIS;
+        st->need_parsing = AVSTREAM_PARSE_HEADERS;
 
         if (srate > 0) {
             st->codec->sample_rate = srate;
diff --git a/libavformat/psxstr.c b/libavformat/psxstr.c
index 24f1c08617..ff9b7edceb 100644
--- a/libavformat/psxstr.c
+++ b/libavformat/psxstr.c
@@ -258,7 +258,9 @@ static int str_read_packet(AVFormatContext *s,
             //    st->codec->bit_rate = 0; //FIXME;
                 st->codec->block_align = 128;
 
-                avpriv_set_pts_info(st, 64, 128, st->codec->sample_rate);
+                avpriv_set_pts_info(st, 64, 18 * 224 / st->codec->channels,
+                                    st->codec->sample_rate);
+                st->start_time = 0;
             }
             pkt = ret_pkt;
             if (av_new_packet(pkt, 2304))
@@ -267,6 +269,7 @@ static int str_read_packet(AVFormatContext *s,
 
             pkt->stream_index =
                 str->channels[channel].audio_stream_index;
+            pkt->duration = 1;
             return 0;
         default:
             av_log(s, AV_LOG_WARNING, "Unknown sector type %02X\n", sector[0x12]);
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index dae2e581d2..440609928f 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -396,6 +396,9 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext
         case CODEC_ID_H264:
             st->need_parsing = AVSTREAM_PARSE_FULL;
             break;
+        case CODEC_ID_VORBIS:
+            st->need_parsing = AVSTREAM_PARSE_HEADERS;
+            break;
         case CODEC_ID_ADPCM_G722:
             /* According to RFC 3551, the stream clock rate is 8000
              * even if the sample rate is 16000. */
diff --git a/libavformat/segafilm.c b/libavformat/segafilm.c
index 210dedad48..c0a5e0f81e 100644
--- a/libavformat/segafilm.c
+++ b/libavformat/segafilm.c
@@ -197,8 +197,13 @@ static int film_read_header(AVFormatContext *s)
     if (!film->sample_table)
         return AVERROR(ENOMEM);
 
-    for(i=0; i<s->nb_streams; i++)
-        avpriv_set_pts_info(s->streams[i], 33, 1, film->base_clock);
+    for (i = 0; i < s->nb_streams; i++) {
+        st = s->streams[i];
+        if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO)
+            avpriv_set_pts_info(st, 33, 1, film->base_clock);
+        else
+            avpriv_set_pts_info(st, 64, 1, film->audio_samplerate);
+    }
 
     audio_frame_counter = 0;
     for (i = 0; i < film->sample_count; i++) {
@@ -213,8 +218,6 @@ static int film_read_header(AVFormatContext *s)
         if (AV_RB32(&scratch[8]) == 0xFFFFFFFF) {
             film->sample_table[i].stream = film->audio_stream_index;
             film->sample_table[i].pts = audio_frame_counter;
-            film->sample_table[i].pts *= film->base_clock;
-            film->sample_table[i].pts /= film->audio_samplerate;
 
             if (film->audio_type == CODEC_ID_ADPCM_ADX)
                 audio_frame_counter += (film->sample_table[i].sample_size * 32 /
diff --git a/libavformat/thp.c b/libavformat/thp.c
index dc30fbaf70..e71cbc4bd4 100644
--- a/libavformat/thp.c
+++ b/libavformat/thp.c
@@ -184,6 +184,9 @@ static int thp_read_packet(AVFormatContext *s,
         }
 
         pkt->stream_index = thp->audio_stream_index;
+        if (thp->audiosize >= 8)
+            pkt->duration = AV_RB32(&pkt->data[4]);
+
         thp->audiosize = 0;
         thp->frame++;
     }
diff --git a/libavformat/tiertexseq.c b/libavformat/tiertexseq.c
index d2d4e880c2..a0c06deab6 100644
--- a/libavformat/tiertexseq.c
+++ b/libavformat/tiertexseq.c
@@ -224,6 +224,7 @@ static int seq_read_header(AVFormatContext *s)
     if (!st)
         return AVERROR(ENOMEM);
 
+    st->start_time = 0;
     avpriv_set_pts_info(st, 32, 1, SEQ_SAMPLE_RATE);
     seq->audio_stream_index = st->index;
     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
@@ -233,7 +234,7 @@ static int seq_read_header(AVFormatContext *s)
     st->codec->sample_rate = SEQ_SAMPLE_RATE;
     st->codec->bits_per_coded_sample = 16;
     st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample * st->codec->channels;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
+    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample / 8;
 
     return 0;
 }
diff --git a/libavformat/tta.c b/libavformat/tta.c
index c2f0902535..9c31432b7b 100644
--- a/libavformat/tta.c
+++ b/libavformat/tta.c
@@ -27,6 +27,8 @@
 
 typedef struct {
     int totalframes, currentframe;
+    int frame_size;
+    int last_frame_size;
 } TTAContext;
 
 static int tta_probe(AVProbeData *p)
@@ -42,7 +44,7 @@ static int tta_read_header(AVFormatContext *s)
 {
     TTAContext *c = s->priv_data;
     AVStream *st;
-    int i, channels, bps, samplerate, datalen, framelen;
+    int i, channels, bps, samplerate, datalen;
     uint64_t framepos, start_offset;
 
     if (!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX))
@@ -69,8 +71,11 @@ static int tta_read_header(AVFormatContext *s)
 
     avio_skip(s->pb, 4); // header crc
 
-    framelen = samplerate*256/245;
-    c->totalframes = datalen / framelen + ((datalen % framelen) ? 1 : 0);
+    c->frame_size      = samplerate * 256 / 245;
+    c->last_frame_size = datalen % c->frame_size;
+    if (!c->last_frame_size)
+        c->last_frame_size = c->frame_size;
+    c->totalframes = datalen / c->frame_size + (c->last_frame_size < c->frame_size);
     c->currentframe = 0;
 
     if(c->totalframes >= UINT_MAX/sizeof(uint32_t) || c->totalframes <= 0){
@@ -90,7 +95,8 @@ static int tta_read_header(AVFormatContext *s)
 
     for (i = 0; i < c->totalframes; i++) {
         uint32_t size = avio_rl32(s->pb);
-        av_add_index_entry(st, framepos, i*framelen, size, 0, AVINDEX_KEYFRAME);
+        av_add_index_entry(st, framepos, i * c->frame_size, size, 0,
+                           AVINDEX_KEYFRAME);
         framepos += size;
     }
     avio_skip(s->pb, 4); // seektable crc
@@ -132,6 +138,8 @@ static int tta_read_packet(AVFormatContext *s, AVPacket *pkt)
 
     ret = av_get_packet(s->pb, pkt, size);
     pkt->dts = st->index_entries[c->currentframe++].timestamp;
+    pkt->duration = c->currentframe == c->totalframes ? c->last_frame_size :
+                                                        c->frame_size;
     return ret;
 }
 
diff --git a/libavformat/utils.c b/libavformat/utils.c
index 580b9fea3e..7f6bfaa88a 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -759,9 +759,6 @@ static int get_audio_frame_size(AVCodecContext *enc, int size)
 {
     int frame_size;
 
-    if(enc->codec_id == CODEC_ID_VORBIS)
-        return -1;
-
     if (enc->frame_size <= 1) {
         int bits_per_sample = av_get_bits_per_sample(enc->codec_id);
 
@@ -2105,8 +2102,7 @@ static int has_codec_parameters(AVCodecContext *avctx)
     case AVMEDIA_TYPE_AUDIO:
         val = avctx->sample_rate && avctx->channels && avctx->sample_fmt != AV_SAMPLE_FMT_NONE;
         if (!avctx->frame_size &&
-            (avctx->codec_id == CODEC_ID_VORBIS ||
-             avctx->codec_id == CODEC_ID_AAC ||
+            (avctx->codec_id == CODEC_ID_AAC ||
              avctx->codec_id == CODEC_ID_MP1 ||
              avctx->codec_id == CODEC_ID_MP2 ||
              avctx->codec_id == CODEC_ID_MP3 ||
diff --git a/libavformat/vocdec.c b/libavformat/vocdec.c
index ea06b53ecf..6df4d8d01f 100644
--- a/libavformat/vocdec.c
+++ b/libavformat/vocdec.c
@@ -86,9 +86,13 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
 
         switch (type) {
         case VOC_TYPE_VOICE_DATA:
-            dec->sample_rate = 1000000 / (256 - avio_r8(pb));
-            if (sample_rate)
-                dec->sample_rate = sample_rate;
+            if (!dec->sample_rate) {
+                dec->sample_rate = 1000000 / (256 - avio_r8(pb));
+                if (sample_rate)
+                    dec->sample_rate = sample_rate;
+                avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
+            } else
+                avio_skip(pb, 1);
             dec->channels = channels;
             tmp_codec = avio_r8(pb);
             dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
@@ -110,7 +114,11 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
             break;
 
         case VOC_TYPE_NEW_VOICE_DATA:
-            dec->sample_rate = avio_rl32(pb);
+            if (!dec->sample_rate) {
+                dec->sample_rate = avio_rl32(pb);
+                avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
+            } else
+                avio_skip(pb, 4);
             dec->bits_per_coded_sample = avio_r8(pb);
             dec->channels = avio_r8(pb);
             tmp_codec = avio_rl16(pb);
diff --git a/libavformat/vqf.c b/libavformat/vqf.c
index 4664c0068f..196cd39994 100644
--- a/libavformat/vqf.c
+++ b/libavformat/vqf.c
@@ -201,7 +201,7 @@ static int vqf_read_header(AVFormatContext *s)
         return -1;
     }
     c->frame_bit_len = st->codec->bit_rate*size/st->codec->sample_rate;
-    avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+    avpriv_set_pts_info(st, 64, size, st->codec->sample_rate);
 
     /* put first 12 bytes of COMM chunk in extradata */
     if (!(st->codec->extradata = av_malloc(12 + FF_INPUT_BUFFER_PADDING_SIZE)))
@@ -225,6 +225,7 @@ static int vqf_read_packet(AVFormatContext *s, AVPacket *pkt)
 
     pkt->pos          = avio_tell(s->pb);
     pkt->stream_index = 0;
+    pkt->duration     = 1;
 
     pkt->data[0] = 8 - c->remaining_bits; // Number of bits to skip
     pkt->data[1] = c->last_frame_bits;