falkTX
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FFmpeg
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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>
tags/n0.11
Michael Niedermayer 13 years ago
parent
f972193a15 b7beabab4b
commit
15c6be8c7d
24 changed files with 473 additions and 83 deletions
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  1. +1
    -0
      libavcodec/Makefile
  2. +1
    -0
      libavcodec/allcodecs.c
  3. +19
    -1
      libavcodec/avcodec.h
  4. +3
    -3
      libavcodec/mpegaudio_parser.c
  5. +10
    -3
      libavcodec/utils.c
  6. +1
    -1
      libavcodec/version.h
  7. +270
    -0
      libavcodec/vorbis_parser.c
  8. +68
    -0
      libavcodec/vorbis_parser.h
  9. +4
    -3
      libavcodec/vorbisdec.c
  10. +13
    -12
      libavformat/electronicarts.c
  11. +2
    -6
      libavformat/mpegts.c
  12. +1
    -0
      libavformat/oggparsevorbis.c
  13. +4
    -1
      libavformat/psxstr.c
  14. +3
    -0
      libavformat/rtpdec.c
  15. +7
    -4
      libavformat/segafilm.c
  16. +3
    -0
      libavformat/thp.c
  17. +2
    -1
      libavformat/tiertexseq.c
  18. +12
    -4
      libavformat/tta.c
  19. +1
    -5
      libavformat/utils.c
  20. +12
    -4
      libavformat/vocdec.c
  21. +2
    -1
      libavformat/vqf.c
  22. +21
    -21
      tests/ref/fate/tiertex-seq
  23. +1
    -1
      tests/ref/fate/vqf-demux
  24. +12
    -12
      tests/ref/seek/lavf_voc

+ 1
- 0
libavcodec/Makefile View File

@@ -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



+ 1
- 0
libavcodec/allcodecs.c View File

@@ -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);



+ 19
- 1
libavcodec/avcodec.h View File

@@ -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.


+ 3
- 3
libavcodec/mpegaudio_parser.c View File

@@ -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;


+ 10
- 3
libavcodec/utils.c View File

@@ -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;
}


+ 1
- 1
libavcodec/version.h View File

@@ -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, \


+ 270
- 0
libavcodec/vorbis_parser.c View File

@@ -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 */

+ 68
- 0
libavcodec/vorbis_parser.h View File

@@ -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 */

+ 4
- 3
libavcodec/vorbisdec.c View File

@@ -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 ?


+ 13
- 12
libavformat/electronicarts.c View File

@@ -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;


+ 2
- 6
libavformat/mpegts.c View File

@@ -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 ||


+ 1
- 0
libavformat/oggparsevorbis.c View File

@@ -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;


+ 4
- 1
libavformat/psxstr.c View File

@@ -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]);


+ 3
- 0
libavformat/rtpdec.c View File

@@ -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. */


+ 7
- 4
libavformat/segafilm.c View File

@@ -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 /


+ 3
- 0
libavformat/thp.c View File

@@ -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++;
}


+ 2
- 1
libavformat/tiertexseq.c View File

@@ -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;
}


+ 12
- 4
libavformat/tta.c View File

@@ -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;
}



+ 1
- 5
libavformat/utils.c View File

@@ -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 ||


+ 12
- 4
libavformat/vocdec.c View File

@@ -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);


+ 2
- 1
libavformat/vqf.c View File

@@ -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;


+ 21
- 21
tests/ref/fate/tiertex-seq View File

@@ -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

+ 1
- 1
tests/ref/fate/vqf-demux View File

@@ -1 +1 @@
178a10705baabc5b82bd79240f38a700
d72fb75fb22f4bcc94a1dc7af5356ec1

+ 12
- 12
tests/ref/seek/lavf_voc View File

@@ -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
ret:-1 st: 0 flags:0 ts: 0.365000
ret:-1 st: 0 flags:1 ts:-0.740833
ret:-1 st: 0 flags:0 ts: 0.365006
ret:-1 st: 0 flags:1 ts:-0.740837
ret:-1 st:-1 flags:0 ts: 2.153336
ret:-1 st:-1 flags:1 ts: 1.047503
ret:-1 st: 0 flags:0 ts:-0.058333
ret:-1 st: 0 flags:1 ts: 2.835833
ret:-1 st: 0 flags:0 ts:-0.058323
ret:-1 st: 0 flags:1 ts: 2.835834
ret:-1 st:-1 flags:0 ts: 1.730004
ret:-1 st:-1 flags:1 ts: 0.624171
ret:-1 st: 0 flags:0 ts:-0.481667
ret:-1 st: 0 flags:1 ts: 2.412500
ret:-1 st: 0 flags:0 ts:-0.481652
ret:-1 st: 0 flags:1 ts: 2.412505
ret:-1 st:-1 flags:0 ts: 1.306672
ret:-1 st:-1 flags:1 ts: 0.200839
ret:-1 st: 0 flags:0 ts:-0.904989
ret:-1 st: 0 flags:1 ts: 1.989178
ret:-1 st: 0 flags:0 ts:-0.905003
ret:-1 st: 0 flags:1 ts: 1.989176
ret:-1 st:-1 flags:0 ts: 0.883340
ret:-1 st:-1 flags:1 ts:-0.222493
ret:-1 st: 0 flags:0 ts: 2.671678
ret:-1 st: 0 flags:1 ts: 1.565844
ret:-1 st: 0 flags:0 ts: 2.671668
ret:-1 st: 0 flags:1 ts: 1.565847
ret:-1 st:-1 flags:0 ts: 0.460008
ret:-1 st:-1 flags:1 ts:-0.645825

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