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>

24 files changed, 473 insertions, 83 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 44dafb9626..433ae4867e 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -698,6 +698,7 @@ OBJS-$(CONFIG_RV40_PARSER)             += rv34_parser.o
 OBJS-$(CONFIG_VC1_PARSER)              += vc1_parser.o vc1.o vc1data.o \
                                           msmpeg4.o msmpeg4data.o mpeg4video.o \
                                           h263.o mpegvideo.o error_resilience.o
+OBJS-$(CONFIG_VORBIS_PARSER)           += vorbis_parser.o xiph.o
 OBJS-$(CONFIG_VP3_PARSER)              += vp3_parser.o
 OBJS-$(CONFIG_VP8_PARSER)              += vp8_parser.o
 
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 18121ba808..c8cfbb53a8 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -452,6 +452,7 @@ void avcodec_register_all(void)
     REGISTER_PARSER  (RV30, rv30);
     REGISTER_PARSER  (RV40, rv40);
     REGISTER_PARSER  (VC1, vc1);
+    REGISTER_PARSER  (VORBIS, vorbis);
     REGISTER_PARSER  (VP3, vp3);
     REGISTER_PARSER  (VP8, vp8);
 
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 007f08248d..592c5341ea 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -2027,7 +2027,17 @@ typedef struct AVCodecContext {
      * Samples per packet, initialized when calling 'init'.
      */
     int frame_size;
-    int frame_number;   ///< audio or video frame number
+
+    /**
+     * Frame counter, set by libavcodec.
+     *
+     * - decoding: total number of frames returned from the decoder so far.
+     * - encoding: total number of frames passed to the encoder so far.
+     *
+     *   @note the counter is not incremented if encoding/decoding resulted in
+     *   an error.
+     */
+    int frame_number;
 
     /**
      * number of bytes per packet if constant and known or 0
@@ -3966,6 +3976,10 @@ int attribute_deprecated avcodec_encode_audio(AVCodecContext *avctx,
  *                  avpkt->data is NULL, the encoder will allocate it.
  *                  The encoder will set avpkt->size to the size of the
  *                  output packet.
+ *
+ *                  If this function fails or produces no output, avpkt will be
+ *                  freed using av_free_packet() (i.e. avpkt->destruct will be
+ *                  called to free the user supplied buffer).
  * @param[in] frame AVFrame containing the raw audio data to be encoded.
  *                  May be NULL when flushing an encoder that has the
  *                  CODEC_CAP_DELAY capability set.
@@ -4048,6 +4062,10 @@ int avcodec_encode_video(AVCodecContext *avctx, uint8_t *buf, int buf_size,
  *                  The encoder will set avpkt->size to the size of the
  *                  output packet. The returned data (if any) belongs to the
  *                  caller, he is responsible for freeing it.
+ *
+ *                  If this function fails or produces no output, avpkt will be
+ *                  freed using av_free_packet() (i.e. avpkt->destruct will be
+ *                  called to free the user supplied buffer).
  * @param[in] frame AVFrame containing the raw video data to be encoded.
  *                  May be NULL when flushing an encoder that has the
  *                  CODEC_CAP_DELAY capability set.
diff --git a/libavcodec/mpegaudio_parser.c b/libavcodec/mpegaudio_parser.c
index b8089b0b59..18c224f76c 100644
--- a/libavcodec/mpegaudio_parser.c
+++ b/libavcodec/mpegaudio_parser.c
@@ -66,8 +66,8 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
 
                 ret = avpriv_mpa_decode_header(avctx, state, &sr, &channels, &frame_size, &bit_rate);
                 if (ret < 4) {
-                    if(i > 4)
-                        s->header_count= -2;
+                    if (i > 4)
+                        s->header_count = -2;
                 } else {
                     if((state&SAME_HEADER_MASK) != (s->header&SAME_HEADER_MASK) && s->header)
                         s->header_count= -3;
@@ -75,7 +75,7 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
                     s->header_count++;
                     s->frame_size = ret-4;
 
-                    if(s->header_count > 1){
+                    if (s->header_count > 1) {
                         avctx->sample_rate= sr;
                         avctx->channels   = channels;
                         s1->duration      = frame_size;
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index f462bd1a73..47e29c9151 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -945,14 +945,13 @@ int ff_alloc_packet(AVPacket *avpkt, int size)
         return AVERROR(EINVAL);
 
     if (avpkt->data) {
-        uint8_t *pkt_data;
+        void *destruct = avpkt->destruct;
 
         if (avpkt->size < size)
             return AVERROR(EINVAL);
 
-        pkt_data = avpkt->data;
         av_init_packet(avpkt);
-        avpkt->data = pkt_data;
+        avpkt->destruct = destruct;
         avpkt->size = size;
         return 0;
     } else {
@@ -972,6 +971,7 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
     *got_packet_ptr = 0;
 
     if (!(avctx->codec->capabilities & CODEC_CAP_DELAY) && !frame) {
+        av_free_packet(avpkt);
         av_init_packet(avpkt);
         avpkt->size = 0;
         return 0;
@@ -1072,6 +1072,9 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
     if (!ret)
         avctx->frame_number++;
 
+    if (ret < 0 || !*got_packet_ptr)
+        av_free_packet(avpkt);
+
     /* NOTE: if we add any audio encoders which output non-keyframe packets,
              this needs to be moved to the encoders, but for now we can do it
              here to simplify things */
@@ -1203,6 +1206,7 @@ int attribute_align_arg avcodec_encode_video2(AVCodecContext *avctx,
     *got_packet_ptr = 0;
 
     if (!(avctx->codec->capabilities & CODEC_CAP_DELAY) && !frame) {
+        av_free_packet(avpkt);
         av_init_packet(avpkt);
         avpkt->size     = 0;
         return 0;
@@ -1230,6 +1234,9 @@ int attribute_align_arg avcodec_encode_video2(AVCodecContext *avctx,
         avctx->frame_number++;
     }
 
+    if (ret < 0 || !*got_packet_ptr)
+        av_free_packet(avpkt);
+
     emms_c();
     return ret;
 }
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 1aa75eabbd..99691f21bd 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -21,7 +21,7 @@
 #define AVCODEC_VERSION_H
 
 #define LIBAVCODEC_VERSION_MAJOR 54
-#define LIBAVCODEC_VERSION_MINOR  6
+#define LIBAVCODEC_VERSION_MINOR  7
 #define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
diff --git a/libavcodec/vorbis_parser.c b/libavcodec/vorbis_parser.c
new file mode 100644
index 0000000000..f8d92e04d0
--- /dev/null
+++ b/libavcodec/vorbis_parser.c
@@ -0,0 +1,270 @@
+/*
+ * Copyright (c) 2012 Justin Ruggles
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * Vorbis audio parser
+ *
+ * Determines the duration for each packet.
+ */
+
+#include "get_bits.h"
+#include "parser.h"
+#include "xiph.h"
+#include "vorbis_parser.h"
+
+static int parse_id_header(AVCodecContext *avctx, VorbisParseContext *s,
+                           const uint8_t *buf, int buf_size)
+{
+    /* Id header should be 30 bytes */
+    if (buf_size < 30) {
+        av_log(avctx, AV_LOG_ERROR, "Id header is too short\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* make sure this is the Id header */
+    if (buf[0] != 1) {
+        av_log(avctx, AV_LOG_ERROR, "Wrong packet type in Id header\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* check for header signature */
+    if (memcmp(&buf[1], "vorbis", 6)) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid packet signature in Id header\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (!(buf[29] & 0x1)) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid framing bit in Id header\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->blocksize[0] = 1 << (buf[28] & 0xF);
+    s->blocksize[1] = 1 << (buf[28] >>  4);
+
+    return 0;
+}
+
+static int parse_setup_header(AVCodecContext *avctx, VorbisParseContext *s,
+                              const uint8_t *buf, int buf_size)
+{
+    GetBitContext gb, gb0;
+    uint8_t *rev_buf;
+    int i, ret = 0;
+    int got_framing_bit, mode_count, got_mode_header, last_mode_count = 0;
+
+    /* avoid overread */
+    if (buf_size < 7) {
+        av_log(avctx, AV_LOG_ERROR, "Setup header is too short\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* make sure this is the Setup header */
+    if (buf[0] != 5) {
+        av_log(avctx, AV_LOG_ERROR, "Wrong packet type in Setup header\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* check for header signature */
+    if (memcmp(&buf[1], "vorbis", 6)) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid packet signature in Setup header\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* reverse bytes so we can easily read backwards with get_bits() */
+    if (!(rev_buf = av_malloc(buf_size))) {
+        av_log(avctx, AV_LOG_ERROR, "Out of memory\n");
+        return AVERROR(ENOMEM);
+    }
+    for (i = 0; i < buf_size; i++)
+        rev_buf[i] = buf[buf_size - 1 - i];
+    init_get_bits(&gb, rev_buf, buf_size * 8);
+
+    got_framing_bit = 0;
+    while (get_bits_left(&gb) > 97) {
+        if (get_bits1(&gb)) {
+            got_framing_bit = get_bits_count(&gb);
+            break;
+        }
+    }
+    if (!got_framing_bit) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid Setup header\n");
+        ret = AVERROR_INVALIDDATA;
+        goto bad_header;
+    }
+
+    /* Now we search backwards to find possible valid mode counts. This is not
+     * fool-proof because we could have false positive matches and read too
+     * far, but there isn't really any way to be sure without parsing through
+     * all the many variable-sized fields before the modes. This approach seems
+     * to work well in testing, and it is similar to how it is handled in
+     * liboggz. */
+    mode_count = 0;
+    got_mode_header = 0;
+    while (get_bits_left(&gb) >= 97) {
+        if (get_bits(&gb, 8) > 63 || get_bits(&gb, 16) || get_bits(&gb, 16))
+            break;
+        skip_bits(&gb, 1);
+        mode_count++;
+        if (mode_count > 64)
+            break;
+        gb0 = gb;
+        if (get_bits(&gb0, 6) + 1 == mode_count) {
+            got_mode_header = 1;
+            last_mode_count = mode_count;
+        }
+    }
+    if (!got_mode_header) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid Setup header\n");
+        ret = AVERROR_INVALIDDATA;
+        goto bad_header;
+    }
+    /* All samples I've seen use <= 2 modes, so ask for a sample if we find
+     * more than that, as it is most likely a false positive. If we get any
+     * we may need to approach this the long way and parse the whole Setup
+     * header, but I hope very much that it never comes to that. */
+    if (last_mode_count > 2) {
+        av_log_ask_for_sample(avctx, "%d modes found. This is either a false "
+                              "positive or a sample from an unknown encoder.\n",
+                              last_mode_count);
+    }
+    /* We're limiting the mode count to 63 so that we know that the previous
+     * block flag will be in the first packet byte. */
+    if (last_mode_count > 63) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported mode count: %d\n",
+               last_mode_count);
+        ret = AVERROR_INVALIDDATA;
+        goto bad_header;
+    }
+    s->mode_count = mode_count = last_mode_count;
+    /* Determine the number of bits required to code the mode and turn that
+     * into a bitmask to directly access the mode from the first frame byte. */
+    s->mode_mask = ((1 << (av_log2(mode_count - 1) + 1)) - 1) << 1;
+    /* The previous window flag is the next bit after the mode */
+    s->prev_mask = (s->mode_mask | 0x1) + 1;
+
+    init_get_bits(&gb, rev_buf, buf_size * 8);
+    skip_bits_long(&gb, got_framing_bit);
+    for (i = mode_count - 1; i >= 0; i--) {
+        skip_bits_long(&gb, 40);
+        s->mode_blocksize[i] = s->blocksize[get_bits1(&gb)];
+    }
+
+bad_header:
+    av_free(rev_buf);
+    return ret;
+}
+
+int avpriv_vorbis_parse_extradata(AVCodecContext *avctx, VorbisParseContext *s)
+{
+    uint8_t *header_start[3];
+    int header_len[3];
+    int ret;
+
+    s->avctx = avctx;
+    s->extradata_parsed = 1;
+
+    if ((ret = avpriv_split_xiph_headers(avctx->extradata,
+                                         avctx->extradata_size, 30,
+                                         header_start, header_len)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "Extradata corrupt.\n");
+        return ret;
+    }
+
+    if ((ret = parse_id_header(avctx, s, header_start[0], header_len[0])) < 0)
+        return ret;
+
+    if ((ret = parse_setup_header(avctx, s, header_start[2], header_len[2])) < 0)
+        return ret;
+
+    s->valid_extradata = 1;
+    s->previous_blocksize = s->mode_blocksize[0];
+
+    return 0;
+}
+
+int avpriv_vorbis_parse_frame(VorbisParseContext *s, const uint8_t *buf,
+                              int buf_size)
+{
+    int duration = 0;
+
+    if (s->valid_extradata && buf_size > 0) {
+        int mode, current_blocksize;
+        int previous_blocksize = s->previous_blocksize;
+
+        if (buf[0] & 1) {
+            av_log(s->avctx, AV_LOG_ERROR, "Invalid packet\n");
+            return AVERROR_INVALIDDATA;
+        }
+        if (s->mode_count == 1)
+            mode = 0;
+        else
+            mode = (buf[0] & s->mode_mask) >> 1;
+        if (mode >= s->mode_count) {
+            av_log(s->avctx, AV_LOG_ERROR, "Invalid mode in packet\n");
+            return AVERROR_INVALIDDATA;
+        }
+        if (mode) {
+            int flag = !!(buf[0] & s->prev_mask);
+            previous_blocksize = s->blocksize[flag];
+        }
+        current_blocksize     = s->mode_blocksize[mode];
+        duration              = (previous_blocksize + current_blocksize) >> 2;
+        s->previous_blocksize = current_blocksize;
+    }
+
+    return duration;
+}
+
+void avpriv_vorbis_parse_reset(VorbisParseContext *s)
+{
+    if (s->valid_extradata)
+        s->previous_blocksize = s->mode_blocksize[0];
+}
+
+#if CONFIG_VORBIS_PARSER
+static int vorbis_parse(AVCodecParserContext *s1, AVCodecContext *avctx,
+                        const uint8_t **poutbuf, int *poutbuf_size,
+                        const uint8_t *buf, int buf_size)
+{
+    VorbisParseContext *s = s1->priv_data;
+    int duration;
+
+    if (!s->extradata_parsed && avctx->extradata && avctx->extradata_size)
+        if (avpriv_vorbis_parse_extradata(avctx, s))
+            goto end;
+
+    if ((duration = avpriv_vorbis_parse_frame(s, buf, buf_size)) >= 0)
+        s1->duration = duration;
+
+end:
+    /* always return the full packet. this parser isn't doing any splitting or
+       combining, only packet analysis */
+    *poutbuf      = buf;
+    *poutbuf_size = buf_size;
+    return buf_size;
+}
+
+AVCodecParser ff_vorbis_parser = {
+    .codec_ids      = { CODEC_ID_VORBIS },
+    .priv_data_size = sizeof(VorbisParseContext),
+    .parser_parse   = vorbis_parse,
+};
+#endif /* CONFIG_VORBIS_PARSER */
diff --git a/libavcodec/vorbis_parser.h b/libavcodec/vorbis_parser.h
new file mode 100644
index 0000000000..480a918f7b
--- /dev/null
+++ b/libavcodec/vorbis_parser.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2012 Justin Ruggles
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * Vorbis audio parser
+ *
+ * Determines the duration for each packet.
+ */
+
+#ifndef AVCODEC_VORBIS_PARSER_H
+#define AVCODEC_VORBIS_PARSER_H
+
+#include "avcodec.h"
+
+typedef struct VorbisParseContext {
+    AVCodecContext *avctx;      ///< codec context
+    int extradata_parsed;       ///< we have attempted to parse extradata
+    int valid_extradata;        ///< extradata is valid, so we can calculate duration
+    int blocksize[2];           ///< short and long window sizes
+    int previous_blocksize;     ///< previous window size
+    int mode_blocksize[64];     ///< window size mapping for each mode
+    int mode_count;             ///< number of modes
+    int mode_mask;              ///< bitmask used to get the mode in each packet
+    int prev_mask;              ///< bitmask used to get the previous mode flag in each packet
+} VorbisParseContext;
+
+/**
+ * Initialize the Vorbis parser using headers in the extradata.
+ *
+ * @param avctx codec context
+ * @param s     Vorbis parser context
+ */
+int avpriv_vorbis_parse_extradata(AVCodecContext *avctx, VorbisParseContext *s);
+
+/**
+ * Get the duration for a Vorbis packet.
+ *
+ * avpriv_vorbis_parse_extradata() must have been successfully called prior to
+ * this in order for a correct duration to be returned.
+ *
+ * @param s        Vorbis parser context
+ * @param buf      buffer containing a Vorbis frame
+ * @param buf_size size of the buffer
+ */
+int avpriv_vorbis_parse_frame(VorbisParseContext *s, const uint8_t *buf,
+                              int buf_size);
+
+void avpriv_vorbis_parse_reset(VorbisParseContext *s);
+
+#endif /* AVCODEC_VORBIS_PARSER_H */
diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
index 54651f42a3..702c0d0429 100644
--- a/libavcodec/vorbisdec.c
+++ b/libavcodec/vorbisdec.c
@@ -1043,7 +1043,6 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
 
     avccontext->channels    = vc->audio_channels;
     avccontext->sample_rate = vc->audio_samplerate;
-    avccontext->frame_size  = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
 
     avcodec_get_frame_defaults(&vc->frame);
     avccontext->coded_frame = &vc->frame;
@@ -1522,8 +1521,10 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
     blockflag = vc->modes[mode_number].blockflag;
     blocksize = vc->blocksize[blockflag];
     vlen = blocksize / 2;
-    if (blockflag)
-        skip_bits(gb, 2); // previous_window, next_window
+    if (blockflag) {
+        previous_window = get_bits(gb, 1);
+        skip_bits1(gb); // next_window
+    }
 
     memset(ch_res_ptr,   0, sizeof(float) * vc->audio_channels * vlen); //FIXME can this be removed ?
     memset(ch_floor_ptr, 0, sizeof(float) * vc->audio_channels * vlen); //FIXME can this be removed ?
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;
diff --git a/tests/ref/fate/tiertex-seq b/tests/ref/fate/tiertex-seq
index 595309342e..74407e7674 100644
--- a/tests/ref/fate/tiertex-seq
+++ b/tests/ref/fate/tiertex-seq
@@ -1,100 +1,100 @@
 #tb 0: 1/25
 #tb 1: 1/22050
-0,          0,          0,        1,    98304, 0x2e5db4a4
 1,          0,          0,      882,     1764, 0x00000000
 1,        882,        882,      882,     1764, 0x80a253d9
+0,          2,          2,        1,    98304, 0x2e5db4a4
 1,       1764,       1764,      882,     1764, 0x95a16721
 1,       2646,       2646,      882,     1764, 0x0f0d4cb6
-0,          4,          4,        1,    98304, 0xb20c19d0
 1,       3528,       3528,      882,     1764, 0x75026779
 1,       4410,       4410,      882,     1764, 0xb4356e37
+0,          6,          6,        1,    98304, 0xb20c19d0
 1,       5292,       5292,      882,     1764, 0xfafa64cb
-0,          7,          7,        1,    98304, 0x6b8538c0
 1,       6174,       6174,      882,     1764, 0xe8fd7970
 1,       7056,       7056,      882,     1764, 0x666879b7
+0,          9,          9,        1,    98304, 0x6b8538c0
 1,       7938,       7938,      882,     1764, 0xf2cd7770
 1,       8820,       8820,      882,     1764, 0x54317a1c
-0,         11,         11,        1,    98304, 0x172207e3
 1,       9702,       9702,      882,     1764, 0x9c396930
 1,      10584,      10584,      882,     1764, 0x87115ec4
+0,         13,         13,        1,    98304, 0x172207e3
 1,      11466,      11466,      882,     1764, 0x0c9b69b6
-0,         14,         14,        1,    98304, 0x63fb7dc1
 1,      12348,      12348,      882,     1764, 0x8c3a758a
 1,      13230,      13230,      882,     1764, 0x605d776a
+0,         16,         16,        1,    98304, 0x63fb7dc1
 1,      14112,      14112,      882,     1764, 0x0556852d
 1,      14994,      14994,      882,     1764, 0x7d4363f8
-0,         18,         18,        1,    98304, 0x37cf1601
 1,      15876,      15876,      882,     1764, 0xc5cd75d0
 1,      16758,      16758,      882,     1764, 0x3ff3646d
+0,         20,         20,        1,    98304, 0x37cf1601
 1,      17640,      17640,      882,     1764, 0x10136d25
 1,      18522,      18522,      882,     1764, 0xeb1a6cd0
-0,         22,         22,        1,    98304, 0x82941990
 1,      19404,      19404,      882,     1764, 0xef937ed1
 1,      20286,      20286,      882,     1764, 0x2d2b6f79
+0,         24,         24,        1,    98304, 0x82941990
 1,      21168,      21168,      882,     1764, 0x6f457231
-0,         25,         25,        1,    98304, 0xe0a5309e
 1,      22050,      22050,      882,     1764, 0x56267c9d
 1,      22932,      22932,      882,     1764, 0xd49e79c8
+0,         27,         27,        1,    98304, 0xe0a5309e
 1,      23814,      23814,      882,     1764, 0xc726703d
 1,      24696,      24696,      882,     1764, 0x2abf8074
-0,         29,         29,        1,    98304, 0x164cb67d
 1,      25578,      25578,      882,     1764, 0xb50c556d
 1,      26460,      26460,      882,     1764, 0xc1f2523c
+0,         31,         31,        1,    98304, 0x164cb67d
 1,      27342,      27342,      882,     1764, 0x850a6f93
-0,         32,         32,        1,    98304, 0xed2189f8
 1,      28224,      28224,      882,     1764, 0x8da76c31
 1,      29106,      29106,      882,     1764, 0xfcccdf13
+0,         34,         34,        1,    98304, 0xed2189f8
 1,      29988,      29988,      882,     1764, 0x00000000
 1,      30870,      30870,      882,     1764, 0x00000000
-0,         36,         36,        1,    98304, 0x7215e529
 1,      31752,      31752,      882,     1764, 0x00000000
 1,      32634,      32634,      882,     1764, 0x00000000
+0,         38,         38,        1,    98304, 0x7215e529
 1,      33516,      33516,      882,     1764, 0x00000000
-0,         39,         39,        1,    98304, 0x170c783b
 1,      34398,      34398,      882,     1764, 0x00000000
 1,      35280,      35280,      882,     1764, 0x00000000
+0,         41,         41,        1,    98304, 0x170c783b
 1,      36162,      36162,      882,     1764, 0x00000000
 1,      37044,      37044,      882,     1764, 0x00000000
-0,         43,         43,        1,    98304, 0xf6bd74c7
 1,      37926,      37926,      882,     1764, 0x00000000
 1,      38808,      38808,      882,     1764, 0x00000000
+0,         45,         45,        1,    98304, 0xf6bd74c7
 1,      39690,      39690,      882,     1764, 0x00000000
 1,      40572,      40572,      882,     1764, 0x00000000
-0,         47,         47,        1,    98304, 0x1efd38c4
 1,      41454,      41454,      882,     1764, 0x00000000
 1,      42336,      42336,      882,     1764, 0x00000000
+0,         49,         49,        1,    98304, 0x1efd38c4
 1,      43218,      43218,      882,     1764, 0x00000000
-0,         50,         50,        1,    98304, 0x29c26bba
 1,      44100,      44100,      882,     1764, 0x00000000
 1,      44982,      44982,      882,     1764, 0x00000000
+0,         52,         52,        1,    98304, 0x29c26bba
 1,      45864,      45864,      882,     1764, 0x00000000
 1,      46746,      46746,      882,     1764, 0x00000000
-0,         54,         54,        1,    98304, 0x880a6313
 1,      47628,      47628,      882,     1764, 0x00000000
 1,      48510,      48510,      882,     1764, 0x00000000
+0,         56,         56,        1,    98304, 0x880a6313
 1,      49392,      49392,      882,     1764, 0x00000000
-0,         57,         57,        1,    98304, 0x73f5bb00
 1,      50274,      50274,      882,     1764, 0x00000000
 1,      51156,      51156,      882,     1764, 0x00000000
+0,         59,         59,        1,    98304, 0x73f5bb00
 1,      52038,      52038,      882,     1764, 0x00000000
 1,      52920,      52920,      882,     1764, 0x00000000
-0,         61,         61,        1,    98304, 0xc85b19ec
 1,      53802,      53802,      882,     1764, 0x00000000
 1,      54684,      54684,      882,     1764, 0x00000000
+0,         63,         63,        1,    98304, 0xc85b19ec
 1,      55566,      55566,      882,     1764, 0x00000000
-0,         64,         64,        1,    98304, 0x00000000
 1,      56448,      56448,      882,     1764, 0x00000000
 1,      57330,      57330,      882,     1764, 0x00000000
+0,         66,         66,        1,    98304, 0x00000000
 1,      58212,      58212,      882,     1764, 0x00000000
 1,      59094,      59094,      882,     1764, 0x00000000
-0,         68,         68,        1,    98304, 0x00000000
 1,      59976,      59976,      882,     1764, 0x00000000
 1,      60858,      60858,      882,     1764, 0x00000000
+0,         70,         70,        1,    98304, 0x00000000
 1,      61740,      61740,      882,     1764, 0x00000000
 1,      62622,      62622,      882,     1764, 0x00000000
-0,         72,         72,        1,    98304, 0x00000000
 1,      63504,      63504,      882,     1764, 0x00000000
 1,      64386,      64386,      882,     1764, 0x00000000
+0,         74,         74,        1,    98304, 0x00000000
 1,      65268,      65268,      882,     1764, 0x00000000
 1,      66150,      66150,      882,     1764, 0x00000000
 1,      67032,      67032,      882,     1764, 0x00000000
diff --git a/tests/ref/fate/vqf-demux b/tests/ref/fate/vqf-demux
index 3d4cd3dd78..3acae60f5a 100644
--- a/tests/ref/fate/vqf-demux
+++ b/tests/ref/fate/vqf-demux
@@ -1 +1 @@
-178a10705baabc5b82bd79240f38a700
+d72fb75fb22f4bcc94a1dc7af5356ec1
diff --git a/tests/ref/seek/lavf_voc b/tests/ref/seek/lavf_voc
index 5670d1a992..387aef6cb8 100644
--- a/tests/ref/seek/lavf_voc
+++ b/tests/ref/seek/lavf_voc
@@ -1,27 +1,27 @@
 ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     32 size:  1024
 ret:-1         st:-1 flags:0  ts:-1.000000
 ret:-1         st:-1 flags:1  ts: 1.894167
-ret:-1         st: 0 flags:0  ts: 0.788333
-ret:-1         st: 0 flags:1  ts:-0.317500
+ret:-1         st: 0 flags:0  ts: 0.788335
+ret:-1         st: 0 flags:1  ts:-0.317508
 ret:-1         st:-1 flags:0  ts: 2.576668
 ret:-1         st:-1 flags:1  ts: 1.470835
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 ret:-1         st:-1 flags:0  ts: 2.153336
 ret:-1         st:-1 flags:1  ts: 1.047503
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 ret:-1         st:-1 flags:0  ts: 1.730004
 ret:-1         st:-1 flags:1  ts: 0.624171
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 ret:-1         st:-1 flags:0  ts: 1.306672
 ret:-1         st:-1 flags:1  ts: 0.200839
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 ret:-1         st:-1 flags:0  ts: 0.883340
 ret:-1         st:-1 flags:1  ts:-0.222493
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 ret:-1         st:-1 flags:0  ts: 0.460008
 ret:-1         st:-1 flags:1  ts:-0.645825