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/*
* WAV demuxer
* Copyright (c) 2001, 2002 Fabrice Bellard
*
* Sony Wave64 demuxer
* RF64 demuxer
* Copyright (c) 2009 Daniel Verkamp
*
* BW64 demuxer
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "config.h"
#include "libavutil/avassert.h"
#include "libavutil/dict.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "avformat.h"
#include "avio.h"
#include "avio_internal.h"
#include "id3v2.h"
#include "internal.h"
#include "metadata.h"
#include "pcm.h"
#include "riff.h"
#include "w64.h"
#include "spdif.h"
typedef struct WAVDemuxContext {
const AVClass *class;
int64_t data_end;
int w64;
AVStream *vst;
int64_t smv_data_ofs;
int smv_block_size;
int smv_frames_per_jpeg;
int smv_block;
int smv_last_stream;
int smv_eof;
int audio_eof;
int ignore_length;
int max_size;
int spdif;
int smv_given_first;
int unaligned; // e.g. if an odd number of bytes ID3 tag was prepended
int rifx; // RIFX: integer byte order for parameters is big endian
} WAVDemuxContext;
#define OFFSET(x) offsetof(WAVDemuxContext, x)
#define DEC AV_OPT_FLAG_DECODING_PARAM
static const AVOption demux_options[] = {
#define W64_DEMUXER_OPTIONS_OFFSET (1 * CONFIG_WAV_DEMUXER)
#if CONFIG_WAV_DEMUXER
{ "ignore_length", "Ignore length", OFFSET(ignore_length), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, DEC },
#endif
{ "max_size", "max size of single packet", OFFSET(max_size), AV_OPT_TYPE_INT, { .i64 = 4096 }, 1024, 1 << 22, DEC },
{ NULL },
};
static void set_spdif(AVFormatContext *s, WAVDemuxContext *wav)
{
if (CONFIG_SPDIF_DEMUXER && s->streams[0]->codecpar->codec_tag == 1) {
enum AVCodecID codec;
int len = 1<<16;
int ret = ffio_ensure_seekback(s->pb, len);
if (ret >= 0) {
uint8_t *buf = av_malloc(len + AV_INPUT_BUFFER_PADDING_SIZE);
if (!buf) {
ret = AVERROR(ENOMEM);
} else {
int64_t pos = avio_tell(s->pb);
len = ret = avio_read(s->pb, buf, len);
if (len >= 0) {
ret = ff_spdif_probe(buf, len, &codec);
if (ret > AVPROBE_SCORE_EXTENSION) {
s->streams[0]->codecpar->codec_id = codec;
wav->spdif = 1;
}
}
avio_seek(s->pb, pos, SEEK_SET);
av_free(buf);
}
}
if (ret < 0)
av_log(s, AV_LOG_WARNING, "Cannot check for SPDIF\n");
}
}
#if CONFIG_WAV_DEMUXER
static int64_t next_tag(AVIOContext *pb, uint32_t *tag, int big_endian)
{
*tag = avio_rl32(pb);
if (!big_endian) {
return avio_rl32(pb);
} else {
return avio_rb32(pb);
}
}
/* RIFF chunks are always at even offsets relative to where they start. */
static int64_t wav_seek_tag(WAVDemuxContext * wav, AVIOContext *s, int64_t offset, int whence)
{
offset += offset < INT64_MAX && offset + wav->unaligned & 1;
return avio_seek(s, offset, whence);
}
/* return the size of the found tag */
static int64_t find_tag(WAVDemuxContext * wav, AVIOContext *pb, uint32_t tag1)
{
unsigned int tag;
int64_t size;
for (;;) {
if (avio_feof(pb))
return AVERROR_EOF;
size = next_tag(pb, &tag, wav->rifx);
if (tag == tag1)
break;
wav_seek_tag(wav, pb, size, SEEK_CUR);
}
return size;
}
static int wav_probe(const AVProbeData *p)
{
/* check file header */
if (p->buf_size <= 32)
return 0;
if (!memcmp(p->buf + 8, "WAVE", 4)) {
if (!memcmp(p->buf, "RIFF", 4) || !memcmp(p->buf, "RIFX", 4))
13 years ago
/* Since the ACT demuxer has a standard WAV header at the top of
* its own, the returned score is decreased to avoid a probe
* conflict between ACT and WAV. */
return AVPROBE_SCORE_MAX - 1;
else if ((!memcmp(p->buf, "RF64", 4) ||
!memcmp(p->buf, "BW64", 4)) &&
!memcmp(p->buf + 12, "ds64", 4))
return AVPROBE_SCORE_MAX;
}
return 0;
}
static void handle_stream_probing(AVStream *st)
{
if (st->codecpar->codec_id == AV_CODEC_ID_PCM_S16LE) {
FFStream *const sti = ffstream(st);
sti->request_probe = AVPROBE_SCORE_EXTENSION;
sti->probe_packets = FFMIN(sti->probe_packets, 32);
}
}
static int wav_parse_fmt_tag(AVFormatContext *s, int64_t size, AVStream *st)
{
AVIOContext *pb = s->pb;
WAVDemuxContext *wav = s->priv_data;
int ret;
/* parse fmt header */
ret = ff_get_wav_header(s, pb, st->codecpar, size, wav->rifx);
if (ret < 0)
return ret;
handle_stream_probing(st);
ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
return 0;
}
static int wav_parse_xma2_tag(AVFormatContext *s, int64_t size, AVStream *st)
{
AVIOContext *pb = s->pb;
int version, num_streams, i, channels = 0, ret;
if (size < 36)
return AVERROR_INVALIDDATA;
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = AV_CODEC_ID_XMA2;
ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
version = avio_r8(pb);
if (version != 3 && version != 4)
return AVERROR_INVALIDDATA;
num_streams = avio_r8(pb);
if (size != (32 + ((version==3)?0:8) + 4*num_streams))
return AVERROR_INVALIDDATA;
avio_skip(pb, 10);
st->codecpar->sample_rate = avio_rb32(pb);
if (version == 4)
avio_skip(pb, 8);
avio_skip(pb, 4);
st->duration = avio_rb32(pb);
avio_skip(pb, 8);
for (i = 0; i < num_streams; i++) {
channels += avio_r8(pb);
avio_skip(pb, 3);
}
st->codecpar->channels = channels;
if (st->codecpar->channels <= 0 || st->codecpar->sample_rate <= 0)
return AVERROR_INVALIDDATA;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
avio_seek(pb, -size, SEEK_CUR);
if ((ret = ff_get_extradata(s, st->codecpar, pb, size)) < 0)
return ret;
return 0;
}
static inline int wav_parse_bext_string(AVFormatContext *s, const char *key,
int length)
{
char temp[257];
int ret;
av_assert0(length < sizeof(temp));
if ((ret = ffio_read_size(s->pb, temp, length)) < 0)
return ret;
temp[length] = 0;
if (strlen(temp))
return av_dict_set(&s->metadata, key, temp, 0);
return 0;
}
static int wav_parse_bext_tag(AVFormatContext *s, int64_t size)
{
char temp[131], *coding_history;
int ret, x;
uint64_t time_reference;
int64_t umid_parts[8], umid_mask = 0;
if ((ret = wav_parse_bext_string(s, "description", 256)) < 0 ||
(ret = wav_parse_bext_string(s, "originator", 32)) < 0 ||
(ret = wav_parse_bext_string(s, "originator_reference", 32)) < 0 ||
(ret = wav_parse_bext_string(s, "origination_date", 10)) < 0 ||
(ret = wav_parse_bext_string(s, "origination_time", 8)) < 0)
return ret;
time_reference = avio_rl64(s->pb);
snprintf(temp, sizeof(temp), "%"PRIu64, time_reference);
if ((ret = av_dict_set(&s->metadata, "time_reference", temp, 0)) < 0)
return ret;
/* check if version is >= 1, in which case an UMID may be present */
if (avio_rl16(s->pb) >= 1) {
for (x = 0; x < 8; x++)
umid_mask |= umid_parts[x] = avio_rb64(s->pb);
if (umid_mask) {
/* the string formatting below is per SMPTE 330M-2004 Annex C */
if (umid_parts[4] == 0 && umid_parts[5] == 0 &&
umid_parts[6] == 0 && umid_parts[7] == 0) {
/* basic UMID */
snprintf(temp, sizeof(temp),
"0x%016"PRIX64"%016"PRIX64"%016"PRIX64"%016"PRIX64,
umid_parts[0], umid_parts[1],
umid_parts[2], umid_parts[3]);
} else {
/* extended UMID */
snprintf(temp, sizeof(temp),
"0x%016"PRIX64"%016"PRIX64"%016"PRIX64"%016"PRIX64
"%016"PRIX64"%016"PRIX64"%016"PRIX64"%016"PRIX64,
umid_parts[0], umid_parts[1],
umid_parts[2], umid_parts[3],
umid_parts[4], umid_parts[5],
umid_parts[6], umid_parts[7]);
}
if ((ret = av_dict_set(&s->metadata, "umid", temp, 0)) < 0)
return ret;
}
avio_skip(s->pb, 190);
} else
avio_skip(s->pb, 254);
if (size > 602) {
/* CodingHistory present */
size -= 602;
if (!(coding_history = av_malloc(size + 1)))
return AVERROR(ENOMEM);
if ((ret = ffio_read_size(s->pb, coding_history, size)) < 0) {
av_free(coding_history);
return ret;
}
coding_history[size] = 0;
if ((ret = av_dict_set(&s->metadata, "coding_history", coding_history,
AV_DICT_DONT_STRDUP_VAL)) < 0)
return ret;
}
return 0;
}
static const AVMetadataConv wav_metadata_conv[] = {
{ "description", "comment" },
{ "originator", "encoded_by" },
{ "origination_date", "date" },
{ "origination_time", "creation_time" },
{ 0 },
};
/* wav input */
static int wav_read_header(AVFormatContext *s)
{
int64_t size, av_uninit(data_size);
int64_t sample_count = 0;
int rf64 = 0, bw64 = 0;
uint32_t tag;
AVIOContext *pb = s->pb;
AVStream *st = NULL;
WAVDemuxContext *wav = s->priv_data;
int ret, got_fmt = 0, got_xma2 = 0;
int64_t next_tag_ofs, data_ofs = -1;
wav->unaligned = avio_tell(s->pb) & 1;
wav->smv_data_ofs = -1;
/* read chunk ID */
tag = avio_rl32(pb);
switch (tag) {
case MKTAG('R', 'I', 'F', 'F'):
break;
case MKTAG('R', 'I', 'F', 'X'):
wav->rifx = 1;
break;
case MKTAG('R', 'F', '6', '4'):
rf64 = 1;
break;
case MKTAG('B', 'W', '6', '4'):
bw64 = 1;
break;
default:
av_log(s, AV_LOG_ERROR, "invalid start code %s in RIFF header\n",
av_fourcc2str(tag));
return AVERROR_INVALIDDATA;
}
/* read chunk size */
avio_rl32(pb);
/* read format */
if (avio_rl32(pb) != MKTAG('W', 'A', 'V', 'E')) {
av_log(s, AV_LOG_ERROR, "invalid format in RIFF header\n");
return AVERROR_INVALIDDATA;
}
if (rf64 || bw64) {
if (avio_rl32(pb) != MKTAG('d', 's', '6', '4'))
return AVERROR_INVALIDDATA;
size = avio_rl32(pb);
if (size < 24)
return AVERROR_INVALIDDATA;
avio_rl64(pb); /* RIFF size */
data_size = avio_rl64(pb);
sample_count = avio_rl64(pb);
if (data_size < 0 || sample_count < 0) {
av_log(s, AV_LOG_ERROR, "negative data_size and/or sample_count in "
"ds64: data_size = %"PRId64", sample_count = %"PRId64"\n",
data_size, sample_count);
return AVERROR_INVALIDDATA;
}
avio_skip(pb, size - 24); /* skip rest of ds64 chunk */
}
/* Create the audio stream now so that its index is always zero */
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
for (;;) {
AVStream *vst;
size = next_tag(pb, &tag, wav->rifx);
next_tag_ofs = avio_tell(pb) + size;
if (avio_feof(pb))
break;
switch (tag) {
case MKTAG('f', 'm', 't', ' '):
/* only parse the first 'fmt ' tag found */
if (!got_xma2 && !got_fmt && (ret = wav_parse_fmt_tag(s, size, st)) < 0) {
return ret;
} else if (got_fmt)
av_log(s, AV_LOG_WARNING, "found more than one 'fmt ' tag\n");
got_fmt = 1;
break;
case MKTAG('X', 'M', 'A', '2'):
/* only parse the first 'XMA2' tag found */
if (!got_fmt && !got_xma2 && (ret = wav_parse_xma2_tag(s, size, st)) < 0) {
return ret;
} else if (got_xma2)
av_log(s, AV_LOG_WARNING, "found more than one 'XMA2' tag\n");
got_xma2 = 1;
break;
case MKTAG('d', 'a', 't', 'a'):
if (!(pb->seekable & AVIO_SEEKABLE_NORMAL) && !got_fmt && !got_xma2) {
av_log(s, AV_LOG_ERROR,
"found no 'fmt ' tag before the 'data' tag\n");
return AVERROR_INVALIDDATA;
}
if (rf64 || bw64) {
next_tag_ofs = wav->data_end = avio_tell(pb) + data_size;
} else if (size != 0xFFFFFFFF) {
data_size = size;
next_tag_ofs = wav->data_end = size ? next_tag_ofs : INT64_MAX;
} else {
av_log(s, AV_LOG_WARNING, "Ignoring maximum wav data size, "
"file may be invalid\n");
data_size = 0;
next_tag_ofs = wav->data_end = INT64_MAX;
}
data_ofs = avio_tell(pb);
/* don't look for footer metadata if we can't seek or if we don't
* know where the data tag ends
*/
if (!(pb->seekable & AVIO_SEEKABLE_NORMAL) || (!(rf64 && !bw64) && !size))
goto break_loop;
break;
case MKTAG('f', 'a', 'c', 't'):
if (!sample_count)
sample_count = (!wav->rifx ? avio_rl32(pb) : avio_rb32(pb));
break;
case MKTAG('b', 'e', 'x', 't'):
if ((ret = wav_parse_bext_tag(s, size)) < 0)
return ret;
break;
case MKTAG('S','M','V','0'):
if (!got_fmt) {
av_log(s, AV_LOG_ERROR, "found no 'fmt ' tag before the 'SMV0' tag\n");
return AVERROR_INVALIDDATA;
}
// SMV file, a wav file with video appended.
if (size != MKTAG('0','2','0','0')) {
av_log(s, AV_LOG_ERROR, "Unknown SMV version found\n");
goto break_loop;
}
av_log(s, AV_LOG_DEBUG, "Found SMV data\n");
wav->smv_given_first = 0;
vst = avformat_new_stream(s, NULL);
if (!vst)
return AVERROR(ENOMEM);
wav->vst = vst;
avio_r8(pb);
vst->id = 1;
vst->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
vst->codecpar->codec_id = AV_CODEC_ID_SMVJPEG;
vst->codecpar->width = avio_rl24(pb);
vst->codecpar->height = avio_rl24(pb);
if ((ret = ff_alloc_extradata(vst->codecpar, 4)) < 0) {
av_log(s, AV_LOG_ERROR, "Could not allocate extradata.\n");
return ret;
}
size = avio_rl24(pb);
wav->smv_data_ofs = avio_tell(pb) + (size - 5) * 3;
avio_rl24(pb);
wav->smv_block_size = avio_rl24(pb);
if (!wav->smv_block_size)
return AVERROR_INVALIDDATA;
avpriv_set_pts_info(vst, 32, 1, avio_rl24(pb));
vst->duration = avio_rl24(pb);
avio_rl24(pb);
avio_rl24(pb);
wav->smv_frames_per_jpeg = avio_rl24(pb);
if (wav->smv_frames_per_jpeg > 65536) {
av_log(s, AV_LOG_ERROR, "too many frames per jpeg\n");
return AVERROR_INVALIDDATA;
}
AV_WL32(vst->codecpar->extradata, wav->smv_frames_per_jpeg);
goto break_loop;
case MKTAG('L', 'I', 'S', 'T'):
case MKTAG('l', 'i', 's', 't'):
if (size < 4) {
av_log(s, AV_LOG_ERROR, "too short LIST tag\n");
return AVERROR_INVALIDDATA;
}
switch (avio_rl32(pb)) {
case MKTAG('I', 'N', 'F', 'O'):
ff_read_riff_info(s, size - 4);
break;
case MKTAG('a', 'd', 't', 'l'):
if (s->nb_chapters > 0) {
while (avio_tell(pb) < next_tag_ofs &&
!avio_feof(pb)) {
char cue_label[512];
unsigned id, sub_size;
if (avio_rl32(pb) != MKTAG('l', 'a', 'b', 'l'))
break;
sub_size = avio_rl32(pb);
if (sub_size < 5)
break;
id = avio_rl32(pb);
avio_get_str(pb, sub_size - 4, cue_label, sizeof(cue_label));
avio_skip(pb, avio_tell(pb) & 1);
for (int i = 0; i < s->nb_chapters; i++) {
if (s->chapters[i]->id == id) {
av_dict_set(&s->chapters[i]->metadata, "title", cue_label, 0);
break;
}
}
}
}
break;
}
break;
case MKTAG('I', 'D', '3', ' '):
case MKTAG('i', 'd', '3', ' '): {
avformat/id3v2: Don't reverse the order of id3v2 APICs When parsing ID3v2 tags, special (non-text) metadata is not applied directly and unconditionally; instead it is stored in a linked list in which elements are prepended. When traversing the list to add APICs (or private tags) at the end, the order is reversed. The same also happens for chapters and therefore the chapter parsing code already reverses the chapters. This commit changes this: By keeping pointers to both head and tail of the linked list one can preserve the order of the entries and remove the reordering code for chapters. Only the pointer to head will be exported: No current caller uses a nonempty list, so exporting both head and tail is unnecessary. This removes the functionality to combine the lists of special metadata read from different ID3v2 tags, but that doesn't make really much sense anyway (and would be trivial to implement if desired) and allows to remove the now unnecessary initializations performed by the callers. The FATE-reference for the id3v2-priv test had to be updated because the order of the tags read into the dict is reversed; for id3v2-priv-remux only the md5 and not the ffprobe output of the remuxed file changes because the order of the private tags has up until now been reversed twice. The references for the aiff/mp3 cover-art tests needed to be updated, because the order of the attached pics is reversed upon reading. It is still not correct, because the muxers write the pics in the order in which they arrive at the muxer instead of the order given by pkt->stream_index. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
4 years ago
ID3v2ExtraMeta *id3v2_extra_meta;
ff_id3v2_read_dict(pb, &ffformatcontext(s)->id3v2_meta, ID3v2_DEFAULT_MAGIC, &id3v2_extra_meta);
if (id3v2_extra_meta) {
ff_id3v2_parse_apic(s, id3v2_extra_meta);
ff_id3v2_parse_chapters(s, id3v2_extra_meta);
ff_id3v2_parse_priv(s, id3v2_extra_meta);
}
ff_id3v2_free_extra_meta(&id3v2_extra_meta);
}
break;
case MKTAG('c', 'u', 'e', ' '):
if (size >= 4 && got_fmt && st->codecpar->sample_rate > 0) {
AVRational tb = {1, st->codecpar->sample_rate};
unsigned nb_cues = avio_rl32(pb);
if (size >= nb_cues * 24LL + 4LL) {
for (int i = 0; i < nb_cues; i++) {
unsigned offset, id = avio_rl32(pb);
if (avio_feof(pb))
return AVERROR_INVALIDDATA;
avio_skip(pb, 16);
offset = avio_rl32(pb);
if (!avpriv_new_chapter(s, id, tb, offset, AV_NOPTS_VALUE, NULL))
return AVERROR(ENOMEM);
}
}
}
break;
}
/* seek to next tag unless we know that we'll run into EOF */
if ((avio_size(pb) > 0 && next_tag_ofs >= avio_size(pb)) ||
wav_seek_tag(wav, pb, next_tag_ofs, SEEK_SET) < 0) {
break;
}
}
break_loop:
if (!got_fmt && !got_xma2) {
av_log(s, AV_LOG_ERROR, "no 'fmt ' or 'XMA2' tag found\n");
return AVERROR_INVALIDDATA;
}
if (data_ofs < 0) {
av_log(s, AV_LOG_ERROR, "no 'data' tag found\n");
return AVERROR_INVALIDDATA;
}
avio_seek(pb, data_ofs, SEEK_SET);
if (data_size > (INT64_MAX>>3)) {
av_log(s, AV_LOG_WARNING, "Data size %"PRId64" is too large\n", data_size);
data_size = 0;
}
if ( st->codecpar->bit_rate > 0 && data_size > 0
&& st->codecpar->sample_rate > 0
&& sample_count > 0 && st->codecpar->channels > 1
&& sample_count % st->codecpar->channels == 0) {
if (fabs(8.0 * data_size * st->codecpar->channels * st->codecpar->sample_rate /
sample_count /st->codecpar->bit_rate - 1.0) < 0.3)
sample_count /= st->codecpar->channels;
}
if ( data_size > 0 && sample_count && st->codecpar->channels
&& (data_size << 3) / sample_count / st->codecpar->channels > st->codecpar->bits_per_coded_sample + 1) {
av_log(s, AV_LOG_WARNING, "ignoring wrong sample_count %"PRId64"\n", sample_count);
sample_count = 0;
}
/* G.729 hack (for Ticket4577)
* FIXME: Come up with cleaner, more general solution */
if (st->codecpar->codec_id == AV_CODEC_ID_G729 && sample_count && (data_size << 3) > sample_count) {
av_log(s, AV_LOG_WARNING, "ignoring wrong sample_count %"PRId64"\n", sample_count);
sample_count = 0;
}
if (!sample_count || av_get_exact_bits_per_sample(st->codecpar->codec_id) > 0)
if ( st->codecpar->channels
&& data_size
&& av_get_bits_per_sample(st->codecpar->codec_id)
&& wav->data_end <= avio_size(pb))
sample_count = (data_size << 3)
/
(st->codecpar->channels * (uint64_t)av_get_bits_per_sample(st->codecpar->codec_id));
if (sample_count)
st->duration = sample_count;
if (st->codecpar->codec_id == AV_CODEC_ID_PCM_S32LE &&
st->codecpar->block_align == st->codecpar->channels * 4 &&
st->codecpar->bits_per_coded_sample == 32 &&
st->codecpar->extradata_size == 2 &&
AV_RL16(st->codecpar->extradata) == 1) {
st->codecpar->codec_id = AV_CODEC_ID_PCM_F16LE;
st->codecpar->bits_per_coded_sample = 16;
} else if (st->codecpar->codec_id == AV_CODEC_ID_PCM_S24LE &&
st->codecpar->block_align == st->codecpar->channels * 4 &&
st->codecpar->bits_per_coded_sample == 24) {
st->codecpar->codec_id = AV_CODEC_ID_PCM_F24LE;
} else if (st->codecpar->codec_id == AV_CODEC_ID_XMA1 ||
st->codecpar->codec_id == AV_CODEC_ID_XMA2) {
st->codecpar->block_align = 2048;
} else if (st->codecpar->codec_id == AV_CODEC_ID_ADPCM_MS && st->codecpar->channels > 2 &&
st->codecpar->block_align < INT_MAX / st->codecpar->channels) {
st->codecpar->block_align *= st->codecpar->channels;
}
ff_metadata_conv_ctx(s, NULL, wav_metadata_conv);
ff_metadata_conv_ctx(s, NULL, ff_riff_info_conv);
set_spdif(s, wav);
return 0;
}
/**
* Find chunk with w64 GUID by skipping over other chunks.
* @return the size of the found chunk
*/
static int64_t find_guid(AVIOContext *pb, const uint8_t guid1[16])
{
uint8_t guid[16];
int64_t size;
while (!avio_feof(pb)) {
avio_read(pb, guid, 16);
size = avio_rl64(pb);
if (size <= 24 || size > INT64_MAX - 8)
return AVERROR_INVALIDDATA;
if (!memcmp(guid, guid1, 16))
return size;
avio_skip(pb, FFALIGN(size, INT64_C(8)) - 24);
}
return AVERROR_EOF;
}
static int wav_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, size;
int64_t left;
WAVDemuxContext *wav = s->priv_data;
AVStream *st = s->streams[0];
if (CONFIG_SPDIF_DEMUXER && wav->spdif == 1)
return ff_spdif_read_packet(s, pkt);
if (wav->smv_data_ofs > 0) {
int64_t audio_dts, video_dts;
AVStream *vst = wav->vst;
smv_retry:
audio_dts = (int32_t)ffstream( st)->cur_dts;
video_dts = (int32_t)ffstream(vst)->cur_dts;
if (audio_dts != AV_NOPTS_VALUE && video_dts != AV_NOPTS_VALUE) {
/*We always return a video frame first to get the pixel format first*/
wav->smv_last_stream = wav->smv_given_first ?
av_compare_ts(video_dts, vst->time_base,
audio_dts, st->time_base) > 0 : 0;
wav->smv_given_first = 1;
}
wav->smv_last_stream = !wav->smv_last_stream;
wav->smv_last_stream |= wav->audio_eof;
wav->smv_last_stream &= !wav->smv_eof;
if (wav->smv_last_stream) {
uint64_t old_pos = avio_tell(s->pb);
uint64_t new_pos = wav->smv_data_ofs +
wav->smv_block * (int64_t)wav->smv_block_size;
if (avio_seek(s->pb, new_pos, SEEK_SET) < 0) {
ret = AVERROR_EOF;
goto smv_out;
}
size = avio_rl24(s->pb);
ret = av_get_packet(s->pb, pkt, size);
if (ret < 0)
goto smv_out;
pkt->pos -= 3;
pkt->pts = wav->smv_block * wav->smv_frames_per_jpeg;
pkt->duration = wav->smv_frames_per_jpeg;
wav->smv_block++;
pkt->stream_index = vst->index;
smv_out:
avio_seek(s->pb, old_pos, SEEK_SET);
if (ret == AVERROR_EOF) {
wav->smv_eof = 1;
goto smv_retry;
}
return ret;
}
}
left = wav->data_end - avio_tell(s->pb);
if (wav->ignore_length)
left = INT_MAX;
if (left <= 0) {
if (CONFIG_W64_DEMUXER && wav->w64)
left = find_guid(s->pb, ff_w64_guid_data) - 24;
else
left = find_tag(wav, s->pb, MKTAG('d', 'a', 't', 'a'));
if (left < 0) {
wav->audio_eof = 1;
if (wav->smv_data_ofs > 0 && !wav->smv_eof)
goto smv_retry;
return AVERROR_EOF;
}
wav->data_end = avio_tell(s->pb) + left;
}
size = wav->max_size;
lavf: replace AVStream.codec with AVStream.codecpar Currently, AVStream contains an embedded AVCodecContext instance, which is used by demuxers to export stream parameters to the caller and by muxers to receive stream parameters from the caller. It is also used internally as the codec context that is passed to parsers. In addition, it is also widely used by the callers as the decoding (when demuxer) or encoding (when muxing) context, though this has been officially discouraged since Libav 11. There are multiple important problems with this approach: - the fields in AVCodecContext are in general one of * stream parameters * codec options * codec state However, it's not clear which ones are which. It is consequently unclear which fields are a demuxer allowed to set or a muxer allowed to read. This leads to erratic behaviour depending on whether decoding or encoding is being performed or not (and whether it uses the AVStream embedded codec context). - various synchronization issues arising from the fact that the same context is used by several different APIs (muxers/demuxers, parsers, bitstream filters and encoders/decoders) simultaneously, with there being no clear rules for who can modify what and the different processes being typically delayed with respect to each other. - avformat_find_stream_info() making it necessary to support opening and closing a single codec context multiple times, thus complicating the semantics of freeing various allocated objects in the codec context. Those problems are resolved by replacing the AVStream embedded codec context with a newly added AVCodecParameters instance, which stores only the stream parameters exported by the demuxers or read by the muxers.
11 years ago
if (st->codecpar->block_align > 1) {
if (size < st->codecpar->block_align)
size = st->codecpar->block_align;
size = (size / st->codecpar->block_align) * st->codecpar->block_align;
}
size = FFMIN(size, left);
ret = av_get_packet(s->pb, pkt, size);
if (ret < 0)
return ret;
pkt->stream_index = 0;
return ret;
}
static int wav_read_seek(AVFormatContext *s,
int stream_index, int64_t timestamp, int flags)
{
WAVDemuxContext *wav = s->priv_data;
AVStream *ast = s->streams[0], *vst = wav->vst;
wav->smv_eof = 0;
wav->audio_eof = 0;
if (stream_index != 0 && (!vst || stream_index != vst->index))
return AVERROR(EINVAL);
if (wav->smv_data_ofs > 0) {
int64_t smv_timestamp = timestamp;
if (stream_index == 0)
smv_timestamp = av_rescale_q(timestamp, ast->time_base, vst->time_base);
else
timestamp = av_rescale_q(smv_timestamp, vst->time_base, ast->time_base);
if (wav->smv_frames_per_jpeg > 0) {
wav->smv_block = smv_timestamp / wav->smv_frames_per_jpeg;
}
}
switch (ast->codecpar->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
case AV_CODEC_ID_AC3:
case AV_CODEC_ID_DTS:
case AV_CODEC_ID_XMA2:
/* use generic seeking with dynamically generated indexes */
return -1;
default:
break;
}
return ff_pcm_read_seek(s, 0, timestamp, flags);
}
static const AVClass wav_demuxer_class = {
.class_name = "WAV demuxer",
.item_name = av_default_item_name,
.option = demux_options,
.version = LIBAVUTIL_VERSION_INT,
};
const AVInputFormat ff_wav_demuxer = {
.name = "wav",
.long_name = NULL_IF_CONFIG_SMALL("WAV / WAVE (Waveform Audio)"),
.priv_data_size = sizeof(WAVDemuxContext),
.read_probe = wav_probe,
.read_header = wav_read_header,
.read_packet = wav_read_packet,
.read_seek = wav_read_seek,
.flags = AVFMT_GENERIC_INDEX,
.codec_tag = ff_wav_codec_tags_list,
.priv_class = &wav_demuxer_class,
};
#endif /* CONFIG_WAV_DEMUXER */
#if CONFIG_W64_DEMUXER
static int w64_probe(const AVProbeData *p)
{
if (p->buf_size <= 40)
return 0;
if (!memcmp(p->buf, ff_w64_guid_riff, 16) &&
!memcmp(p->buf + 24, ff_w64_guid_wave, 16))
return AVPROBE_SCORE_MAX;
else
return 0;
}
static int w64_read_header(AVFormatContext *s)
{
int64_t size, data_ofs = 0;
AVIOContext *pb = s->pb;
WAVDemuxContext *wav = s->priv_data;
AVStream *st;
uint8_t guid[16];
int ret;
avio_read(pb, guid, 16);
if (memcmp(guid, ff_w64_guid_riff, 16))
return AVERROR_INVALIDDATA;
/* riff + wave + fmt + sizes */
if (avio_rl64(pb) < 16 + 8 + 16 + 8 + 16 + 8)
return AVERROR_INVALIDDATA;
avio_read(pb, guid, 16);
if (memcmp(guid, ff_w64_guid_wave, 16)) {
av_log(s, AV_LOG_ERROR, "could not find wave guid\n");
return AVERROR_INVALIDDATA;
}
wav->w64 = 1;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
while (!avio_feof(pb)) {
if (avio_read(pb, guid, 16) != 16)
break;
size = avio_rl64(pb);
if (size <= 24 || INT64_MAX - size < avio_tell(pb))
return AVERROR_INVALIDDATA;
if (!memcmp(guid, ff_w64_guid_fmt, 16)) {
/* subtract chunk header size - normal wav file doesn't count it */
ret = ff_get_wav_header(s, pb, st->codecpar, size - 24, 0);
if (ret < 0)
return ret;
avio_skip(pb, FFALIGN(size, INT64_C(8)) - size);
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
} else if (!memcmp(guid, ff_w64_guid_fact, 16)) {
int64_t samples;
samples = avio_rl64(pb);
if (samples > 0)
st->duration = samples;
avio_skip(pb, FFALIGN(size, INT64_C(8)) - 32);
} else if (!memcmp(guid, ff_w64_guid_data, 16)) {
wav->data_end = avio_tell(pb) + size - 24;
data_ofs = avio_tell(pb);
if (!(pb->seekable & AVIO_SEEKABLE_NORMAL))
break;
avio_skip(pb, size - 24);
} else if (!memcmp(guid, ff_w64_guid_summarylist, 16)) {
int64_t start, end, cur;
uint32_t count, chunk_size, i;
int64_t filesize = avio_size(s->pb);
start = avio_tell(pb);
end = start + FFALIGN(size, INT64_C(8)) - 24;
count = avio_rl32(pb);
for (i = 0; i < count; i++) {
char chunk_key[5], *value;
if (avio_feof(pb) || (cur = avio_tell(pb)) < 0 || cur > end - 8 /* = tag + size */)
break;
chunk_key[4] = 0;
avio_read(pb, chunk_key, 4);
chunk_size = avio_rl32(pb);
if (chunk_size == UINT32_MAX || (filesize >= 0 && chunk_size > filesize))
return AVERROR_INVALIDDATA;
value = av_malloc(chunk_size + 1);
if (!value)
return AVERROR(ENOMEM);
ret = avio_get_str16le(pb, chunk_size, value, chunk_size);
if (ret < 0) {
av_free(value);
return ret;
}
avio_skip(pb, chunk_size - ret);
av_dict_set(&s->metadata, chunk_key, value, AV_DICT_DONT_STRDUP_VAL);
}
avio_skip(pb, end - avio_tell(pb));
} else {
av_log(s, AV_LOG_DEBUG, "unknown guid: "FF_PRI_GUID"\n", FF_ARG_GUID(guid));
avio_skip(pb, FFALIGN(size, INT64_C(8)) - 24);
}
}
if (!data_ofs)
return AVERROR_EOF;
ff_metadata_conv_ctx(s, NULL, wav_metadata_conv);
ff_metadata_conv_ctx(s, NULL, ff_riff_info_conv);
handle_stream_probing(st);
ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
avio_seek(pb, data_ofs, SEEK_SET);
set_spdif(s, wav);
return 0;
}
static const AVClass w64_demuxer_class = {
.class_name = "W64 demuxer",
.item_name = av_default_item_name,
.option = &demux_options[W64_DEMUXER_OPTIONS_OFFSET],
.version = LIBAVUTIL_VERSION_INT,
};
const AVInputFormat ff_w64_demuxer = {
.name = "w64",
.long_name = NULL_IF_CONFIG_SMALL("Sony Wave64"),
.priv_data_size = sizeof(WAVDemuxContext),
.read_probe = w64_probe,
.read_header = w64_read_header,
.read_packet = wav_read_packet,
.read_seek = wav_read_seek,
.flags = AVFMT_GENERIC_INDEX,
.codec_tag = ff_wav_codec_tags_list,
.priv_class = &w64_demuxer_class,
};
#endif /* CONFIG_W64_DEMUXER */