riff: Move muxing code to a separate file

This commit is contained in:
Diego Biurrun 2013-08-04 14:10:54 +02:00
parent fcc455ff2e
commit 3dd5c95dee
4 changed files with 307 additions and 270 deletions

10
configure vendored
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@ -1390,6 +1390,7 @@ CONFIG_EXTRA="
mpegvideoenc
nettle
rangecoder
riffenc
rtpdec
rtpenc_chain
sinewin
@ -1791,7 +1792,9 @@ libxvid_encoder_deps="libxvid"
# demuxers / muxers
ac3_demuxer_select="ac3_parser"
asf_muxer_select="riffenc"
asf_stream_muxer_select="asf_muxer"
avi_muxer_select="riffenc"
avisynth_demuxer_deps="avisynth"
dirac_demuxer_select="dirac_parser"
eac3_demuxer_select="ac3_parser"
@ -1800,13 +1803,16 @@ ipod_muxer_select="mov_muxer"
ismv_muxer_select="mov_muxer"
matroska_audio_muxer_select="matroska_muxer"
matroska_demuxer_suggest="bzlib lzo zlib"
matroska_muxer_select="riffenc"
mmf_muxer_select="riffenc"
mov_demuxer_suggest="zlib"
mov_muxer_select="rtpenc_chain"
mov_muxer_select="riffenc rtpenc_chain"
mp3_demuxer_select="mpegaudio_parser"
mp4_muxer_select="mov_muxer"
mpegts_muxer_select="adts_muxer latm_muxer"
mpegtsraw_demuxer_select="mpegts_demuxer"
mxf_d10_muxer_select="mxf_muxer"
nut_muxer_select="riffenc"
ogg_demuxer_select="golomb"
psp_muxer_select="mov_muxer"
rtp_demuxer_select="sdp_demuxer"
@ -1822,6 +1828,8 @@ tak_demuxer_select="tak_parser"
tg2_muxer_select="mov_muxer"
tgp_muxer_select="mov_muxer"
w64_demuxer_deps="wav_demuxer"
wav_muxer_select="riffenc"
webm_muxer_select="riffenc"
# indevs / outdevs
alsa_indev_deps="alsa_asoundlib_h snd_pcm_htimestamp"

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@ -24,6 +24,7 @@ OBJS = allformats.o \
utils.o \
OBJS-$(CONFIG_NETWORK) += network.o
OBJS-$(CONFIG_RIFFENC) += riffenc.o
OBJS-$(CONFIG_RTPDEC) += rdt.o \
rtp.o \
rtpdec.o \

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@ -19,7 +19,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/mathematics.h"
#include "libavutil/error.h"
#include "libavutil/log.h"
#include "libavcodec/avcodec.h"
#include "avformat.h"
#include "avio_internal.h"
@ -421,274 +422,6 @@ const struct AVCodecTag *avformat_get_riff_audio_tags(void)
return ff_codec_wav_tags;
}
#if CONFIG_MUXERS
int64_t ff_start_tag(AVIOContext *pb, const char *tag)
{
ffio_wfourcc(pb, tag);
avio_wl32(pb, 0);
return avio_tell(pb);
}
void ff_end_tag(AVIOContext *pb, int64_t start)
{
int64_t pos;
pos = avio_tell(pb);
avio_seek(pb, start - 4, SEEK_SET);
avio_wl32(pb, (uint32_t)(pos - start));
avio_seek(pb, pos, SEEK_SET);
}
/* WAVEFORMATEX header */
/* returns the size or -1 on error */
int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
{
int bps, blkalign, bytespersec, frame_size;
int hdrsize = 18;
int waveformatextensible;
uint8_t temp[256];
uint8_t *riff_extradata = temp;
uint8_t *riff_extradata_start = temp;
if (!enc->codec_tag || enc->codec_tag > 0xffff)
return -1;
/* We use the known constant frame size for the codec if known, otherwise
* fall back on using AVCodecContext.frame_size, which is not as reliable
* for indicating packet duration. */
frame_size = av_get_audio_frame_duration(enc, 0);
if (!frame_size)
frame_size = enc->frame_size;
waveformatextensible = (enc->channels > 2 && enc->channel_layout) ||
enc->sample_rate > 48000 ||
av_get_bits_per_sample(enc->codec_id) > 16;
if (waveformatextensible)
avio_wl16(pb, 0xfffe);
else
avio_wl16(pb, enc->codec_tag);
avio_wl16(pb, enc->channels);
avio_wl32(pb, enc->sample_rate);
if (enc->codec_id == AV_CODEC_ID_MP2 ||
enc->codec_id == AV_CODEC_ID_MP3 ||
enc->codec_id == AV_CODEC_ID_GSM_MS) {
bps = 0;
} else {
if (!(bps = av_get_bits_per_sample(enc->codec_id))) {
if (enc->bits_per_coded_sample)
bps = enc->bits_per_coded_sample;
else
bps = 16; // default to 16
}
}
if (bps != enc->bits_per_coded_sample && enc->bits_per_coded_sample) {
av_log(enc, AV_LOG_WARNING,
"requested bits_per_coded_sample (%d) "
"and actually stored (%d) differ\n",
enc->bits_per_coded_sample, bps);
}
if (enc->codec_id == AV_CODEC_ID_MP2 ||
enc->codec_id == AV_CODEC_ID_MP3) {
/* This is wrong, but it seems many demuxers do not work if this
* is set correctly. */
blkalign = frame_size;
// blkalign = 144 * enc->bit_rate/enc->sample_rate;
} else if (enc->codec_id == AV_CODEC_ID_AC3) {
blkalign = 3840; /* maximum bytes per frame */
} else if (enc->block_align != 0) { /* specified by the codec */
blkalign = enc->block_align;
} else
blkalign = bps * enc->channels / av_gcd(8, bps);
if (enc->codec_id == AV_CODEC_ID_PCM_U8 ||
enc->codec_id == AV_CODEC_ID_PCM_S24LE ||
enc->codec_id == AV_CODEC_ID_PCM_S32LE ||
enc->codec_id == AV_CODEC_ID_PCM_F32LE ||
enc->codec_id == AV_CODEC_ID_PCM_F64LE ||
enc->codec_id == AV_CODEC_ID_PCM_S16LE) {
bytespersec = enc->sample_rate * blkalign;
} else {
bytespersec = enc->bit_rate / 8;
}
avio_wl32(pb, bytespersec); /* bytes per second */
avio_wl16(pb, blkalign); /* block align */
avio_wl16(pb, bps); /* bits per sample */
if (enc->codec_id == AV_CODEC_ID_MP3) {
hdrsize += 12;
bytestream_put_le16(&riff_extradata, 1); /* wID */
bytestream_put_le32(&riff_extradata, 2); /* fdwFlags */
bytestream_put_le16(&riff_extradata, 1152); /* nBlockSize */
bytestream_put_le16(&riff_extradata, 1); /* nFramesPerBlock */
bytestream_put_le16(&riff_extradata, 1393); /* nCodecDelay */
} else if (enc->codec_id == AV_CODEC_ID_MP2) {
hdrsize += 22;
/* fwHeadLayer */
bytestream_put_le16(&riff_extradata, 2);
/* dwHeadBitrate */
bytestream_put_le32(&riff_extradata, enc->bit_rate);
/* fwHeadMode */
bytestream_put_le16(&riff_extradata, enc->channels == 2 ? 1 : 8);
/* fwHeadModeExt */
bytestream_put_le16(&riff_extradata, 0);
/* wHeadEmphasis */
bytestream_put_le16(&riff_extradata, 1);
/* fwHeadFlags */
bytestream_put_le16(&riff_extradata, 16);
/* dwPTSLow */
bytestream_put_le32(&riff_extradata, 0);
/* dwPTSHigh */
bytestream_put_le32(&riff_extradata, 0);
} else if (enc->codec_id == AV_CODEC_ID_GSM_MS ||
enc->codec_id == AV_CODEC_ID_ADPCM_IMA_WAV) {
hdrsize += 2;
/* wSamplesPerBlock */
bytestream_put_le16(&riff_extradata, frame_size);
} else if (enc->extradata_size) {
riff_extradata_start = enc->extradata;
riff_extradata = enc->extradata + enc->extradata_size;
hdrsize += enc->extradata_size;
}
/* write WAVEFORMATEXTENSIBLE extensions */
if (waveformatextensible) {
hdrsize += 22;
/* 22 is WAVEFORMATEXTENSIBLE size */
avio_wl16(pb, riff_extradata - riff_extradata_start + 22);
/* ValidBitsPerSample || SamplesPerBlock || Reserved */
avio_wl16(pb, bps);
/* dwChannelMask */
avio_wl32(pb, enc->channel_layout);
/* GUID + next 3 */
avio_wl32(pb, enc->codec_tag);
avio_wl32(pb, 0x00100000);
avio_wl32(pb, 0xAA000080);
avio_wl32(pb, 0x719B3800);
} else {
avio_wl16(pb, riff_extradata - riff_extradata_start); /* cbSize */
}
avio_write(pb, riff_extradata_start, riff_extradata - riff_extradata_start);
if (hdrsize & 1) {
hdrsize++;
avio_w8(pb, 0);
}
return hdrsize;
}
/* BITMAPINFOHEADER header */
void ff_put_bmp_header(AVIOContext *pb, AVCodecContext *enc,
const AVCodecTag *tags, int for_asf)
{
/* size */
avio_wl32(pb, 40 + enc->extradata_size);
avio_wl32(pb, enc->width);
//We always store RGB TopDown
avio_wl32(pb, enc->codec_tag ? enc->height : -enc->height);
/* planes */
avio_wl16(pb, 1);
/* depth */
avio_wl16(pb, enc->bits_per_coded_sample ? enc->bits_per_coded_sample : 24);
/* compression type */
avio_wl32(pb, enc->codec_tag);
avio_wl32(pb, enc->width * enc->height * 3);
avio_wl32(pb, 0);
avio_wl32(pb, 0);
avio_wl32(pb, 0);
avio_wl32(pb, 0);
avio_write(pb, enc->extradata, enc->extradata_size);
if (!for_asf && enc->extradata_size & 1)
avio_w8(pb, 0);
}
void ff_parse_specific_params(AVCodecContext *stream, int *au_rate,
int *au_ssize, int *au_scale)
{
int gcd;
int audio_frame_size;
/* We use the known constant frame size for the codec if known, otherwise
* fall back on using AVCodecContext.frame_size, which is not as reliable
* for indicating packet duration. */
audio_frame_size = av_get_audio_frame_duration(stream, 0);
if (!audio_frame_size)
audio_frame_size = stream->frame_size;
*au_ssize = stream->block_align;
if (audio_frame_size && stream->sample_rate) {
*au_scale = audio_frame_size;
*au_rate = stream->sample_rate;
} else if (stream->codec_type == AVMEDIA_TYPE_VIDEO ||
stream->codec_type == AVMEDIA_TYPE_DATA ||
stream->codec_type == AVMEDIA_TYPE_SUBTITLE) {
*au_scale = stream->time_base.num;
*au_rate = stream->time_base.den;
} else {
*au_scale = stream->block_align ? stream->block_align * 8 : 8;
*au_rate = stream->bit_rate ? stream->bit_rate :
8 * stream->sample_rate;
}
gcd = av_gcd(*au_scale, *au_rate);
*au_scale /= gcd;
*au_rate /= gcd;
}
void ff_riff_write_info_tag(AVIOContext *pb, const char *tag, const char *str)
{
int len = strlen(str);
if (len > 0) {
len++;
ffio_wfourcc(pb, tag);
avio_wl32(pb, len);
avio_put_str(pb, str);
if (len & 1)
avio_w8(pb, 0);
}
}
static const char riff_tags[][5] = {
"IARL", "IART", "ICMS", "ICMT", "ICOP", "ICRD", "ICRP", "IDIM", "IDPI",
"IENG", "IGNR", "IKEY", "ILGT", "ILNG", "IMED", "INAM", "IPLT", "IPRD",
"IPRT", "ISBJ", "ISFT", "ISHP", "ISRC", "ISRF", "ITCH",
{ 0 }
};
static int riff_has_valid_tags(AVFormatContext *s)
{
int i;
for (i = 0; *riff_tags[i]; i++)
if (av_dict_get(s->metadata, riff_tags[i], NULL, AV_DICT_MATCH_CASE))
return 1;
return 0;
}
void ff_riff_write_info(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
int i;
int64_t list_pos;
AVDictionaryEntry *t = NULL;
ff_metadata_conv(&s->metadata, ff_riff_info_conv, NULL);
/* writing empty LIST is not nice and may cause problems */
if (!riff_has_valid_tags(s))
return;
list_pos = ff_start_tag(pb, "LIST");
ffio_wfourcc(pb, "INFO");
for (i = 0; *riff_tags[i]; i++)
if ((t = av_dict_get(s->metadata, riff_tags[i],
NULL, AV_DICT_MATCH_CASE)))
ff_riff_write_info_tag(s->pb, t->key, t->value);
ff_end_tag(pb, list_pos);
}
#endif /* CONFIG_MUXERS */
#if CONFIG_DEMUXERS
/* We could be given one of the three possible structures here:
* WAVEFORMAT, PCMWAVEFORMAT or WAVEFORMATEX. Each structure

295
libavformat/riffenc.c Normal file
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@ -0,0 +1,295 @@
/*
* RIFF muxing functions
* Copyright (c) 2000 Fabrice Bellard
*
* 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
*/
#include "libavutil/dict.h"
#include "libavutil/log.h"
#include "libavutil/mathematics.h"
#include "libavcodec/avcodec.h"
#include "libavcodec/bytestream.h"
#include "avformat.h"
#include "avio_internal.h"
#include "riff.h"
int64_t ff_start_tag(AVIOContext *pb, const char *tag)
{
ffio_wfourcc(pb, tag);
avio_wl32(pb, 0);
return avio_tell(pb);
}
void ff_end_tag(AVIOContext *pb, int64_t start)
{
int64_t pos;
pos = avio_tell(pb);
avio_seek(pb, start - 4, SEEK_SET);
avio_wl32(pb, (uint32_t)(pos - start));
avio_seek(pb, pos, SEEK_SET);
}
/* WAVEFORMATEX header */
/* returns the size or -1 on error */
int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
{
int bps, blkalign, bytespersec, frame_size;
int hdrsize = 18;
int waveformatextensible;
uint8_t temp[256];
uint8_t *riff_extradata = temp;
uint8_t *riff_extradata_start = temp;
if (!enc->codec_tag || enc->codec_tag > 0xffff)
return -1;
/* We use the known constant frame size for the codec if known, otherwise
* fall back on using AVCodecContext.frame_size, which is not as reliable
* for indicating packet duration. */
frame_size = av_get_audio_frame_duration(enc, 0);
if (!frame_size)
frame_size = enc->frame_size;
waveformatextensible = (enc->channels > 2 && enc->channel_layout) ||
enc->sample_rate > 48000 ||
av_get_bits_per_sample(enc->codec_id) > 16;
if (waveformatextensible)
avio_wl16(pb, 0xfffe);
else
avio_wl16(pb, enc->codec_tag);
avio_wl16(pb, enc->channels);
avio_wl32(pb, enc->sample_rate);
if (enc->codec_id == AV_CODEC_ID_MP2 ||
enc->codec_id == AV_CODEC_ID_MP3 ||
enc->codec_id == AV_CODEC_ID_GSM_MS) {
bps = 0;
} else {
if (!(bps = av_get_bits_per_sample(enc->codec_id))) {
if (enc->bits_per_coded_sample)
bps = enc->bits_per_coded_sample;
else
bps = 16; // default to 16
}
}
if (bps != enc->bits_per_coded_sample && enc->bits_per_coded_sample) {
av_log(enc, AV_LOG_WARNING,
"requested bits_per_coded_sample (%d) "
"and actually stored (%d) differ\n",
enc->bits_per_coded_sample, bps);
}
if (enc->codec_id == AV_CODEC_ID_MP2 ||
enc->codec_id == AV_CODEC_ID_MP3) {
/* This is wrong, but it seems many demuxers do not work if this
* is set correctly. */
blkalign = frame_size;
// blkalign = 144 * enc->bit_rate/enc->sample_rate;
} else if (enc->codec_id == AV_CODEC_ID_AC3) {
blkalign = 3840; /* maximum bytes per frame */
} else if (enc->block_align != 0) { /* specified by the codec */
blkalign = enc->block_align;
} else
blkalign = bps * enc->channels / av_gcd(8, bps);
if (enc->codec_id == AV_CODEC_ID_PCM_U8 ||
enc->codec_id == AV_CODEC_ID_PCM_S24LE ||
enc->codec_id == AV_CODEC_ID_PCM_S32LE ||
enc->codec_id == AV_CODEC_ID_PCM_F32LE ||
enc->codec_id == AV_CODEC_ID_PCM_F64LE ||
enc->codec_id == AV_CODEC_ID_PCM_S16LE) {
bytespersec = enc->sample_rate * blkalign;
} else {
bytespersec = enc->bit_rate / 8;
}
avio_wl32(pb, bytespersec); /* bytes per second */
avio_wl16(pb, blkalign); /* block align */
avio_wl16(pb, bps); /* bits per sample */
if (enc->codec_id == AV_CODEC_ID_MP3) {
hdrsize += 12;
bytestream_put_le16(&riff_extradata, 1); /* wID */
bytestream_put_le32(&riff_extradata, 2); /* fdwFlags */
bytestream_put_le16(&riff_extradata, 1152); /* nBlockSize */
bytestream_put_le16(&riff_extradata, 1); /* nFramesPerBlock */
bytestream_put_le16(&riff_extradata, 1393); /* nCodecDelay */
} else if (enc->codec_id == AV_CODEC_ID_MP2) {
hdrsize += 22;
/* fwHeadLayer */
bytestream_put_le16(&riff_extradata, 2);
/* dwHeadBitrate */
bytestream_put_le32(&riff_extradata, enc->bit_rate);
/* fwHeadMode */
bytestream_put_le16(&riff_extradata, enc->channels == 2 ? 1 : 8);
/* fwHeadModeExt */
bytestream_put_le16(&riff_extradata, 0);
/* wHeadEmphasis */
bytestream_put_le16(&riff_extradata, 1);
/* fwHeadFlags */
bytestream_put_le16(&riff_extradata, 16);
/* dwPTSLow */
bytestream_put_le32(&riff_extradata, 0);
/* dwPTSHigh */
bytestream_put_le32(&riff_extradata, 0);
} else if (enc->codec_id == AV_CODEC_ID_GSM_MS ||
enc->codec_id == AV_CODEC_ID_ADPCM_IMA_WAV) {
hdrsize += 2;
/* wSamplesPerBlock */
bytestream_put_le16(&riff_extradata, frame_size);
} else if (enc->extradata_size) {
riff_extradata_start = enc->extradata;
riff_extradata = enc->extradata + enc->extradata_size;
hdrsize += enc->extradata_size;
}
/* write WAVEFORMATEXTENSIBLE extensions */
if (waveformatextensible) {
hdrsize += 22;
/* 22 is WAVEFORMATEXTENSIBLE size */
avio_wl16(pb, riff_extradata - riff_extradata_start + 22);
/* ValidBitsPerSample || SamplesPerBlock || Reserved */
avio_wl16(pb, bps);
/* dwChannelMask */
avio_wl32(pb, enc->channel_layout);
/* GUID + next 3 */
avio_wl32(pb, enc->codec_tag);
avio_wl32(pb, 0x00100000);
avio_wl32(pb, 0xAA000080);
avio_wl32(pb, 0x719B3800);
} else {
avio_wl16(pb, riff_extradata - riff_extradata_start); /* cbSize */
}
avio_write(pb, riff_extradata_start, riff_extradata - riff_extradata_start);
if (hdrsize & 1) {
hdrsize++;
avio_w8(pb, 0);
}
return hdrsize;
}
/* BITMAPINFOHEADER header */
void ff_put_bmp_header(AVIOContext *pb, AVCodecContext *enc,
const AVCodecTag *tags, int for_asf)
{
/* size */
avio_wl32(pb, 40 + enc->extradata_size);
avio_wl32(pb, enc->width);
//We always store RGB TopDown
avio_wl32(pb, enc->codec_tag ? enc->height : -enc->height);
/* planes */
avio_wl16(pb, 1);
/* depth */
avio_wl16(pb, enc->bits_per_coded_sample ? enc->bits_per_coded_sample : 24);
/* compression type */
avio_wl32(pb, enc->codec_tag);
avio_wl32(pb, enc->width * enc->height * 3);
avio_wl32(pb, 0);
avio_wl32(pb, 0);
avio_wl32(pb, 0);
avio_wl32(pb, 0);
avio_write(pb, enc->extradata, enc->extradata_size);
if (!for_asf && enc->extradata_size & 1)
avio_w8(pb, 0);
}
void ff_parse_specific_params(AVCodecContext *stream, int *au_rate,
int *au_ssize, int *au_scale)
{
int gcd;
int audio_frame_size;
/* We use the known constant frame size for the codec if known, otherwise
* fall back on using AVCodecContext.frame_size, which is not as reliable
* for indicating packet duration. */
audio_frame_size = av_get_audio_frame_duration(stream, 0);
if (!audio_frame_size)
audio_frame_size = stream->frame_size;
*au_ssize = stream->block_align;
if (audio_frame_size && stream->sample_rate) {
*au_scale = audio_frame_size;
*au_rate = stream->sample_rate;
} else if (stream->codec_type == AVMEDIA_TYPE_VIDEO ||
stream->codec_type == AVMEDIA_TYPE_DATA ||
stream->codec_type == AVMEDIA_TYPE_SUBTITLE) {
*au_scale = stream->time_base.num;
*au_rate = stream->time_base.den;
} else {
*au_scale = stream->block_align ? stream->block_align * 8 : 8;
*au_rate = stream->bit_rate ? stream->bit_rate :
8 * stream->sample_rate;
}
gcd = av_gcd(*au_scale, *au_rate);
*au_scale /= gcd;
*au_rate /= gcd;
}
void ff_riff_write_info_tag(AVIOContext *pb, const char *tag, const char *str)
{
int len = strlen(str);
if (len > 0) {
len++;
ffio_wfourcc(pb, tag);
avio_wl32(pb, len);
avio_put_str(pb, str);
if (len & 1)
avio_w8(pb, 0);
}
}
static const char riff_tags[][5] = {
"IARL", "IART", "ICMS", "ICMT", "ICOP", "ICRD", "ICRP", "IDIM", "IDPI",
"IENG", "IGNR", "IKEY", "ILGT", "ILNG", "IMED", "INAM", "IPLT", "IPRD",
"IPRT", "ISBJ", "ISFT", "ISHP", "ISRC", "ISRF", "ITCH",
{ 0 }
};
static int riff_has_valid_tags(AVFormatContext *s)
{
int i;
for (i = 0; *riff_tags[i]; i++)
if (av_dict_get(s->metadata, riff_tags[i], NULL, AV_DICT_MATCH_CASE))
return 1;
return 0;
}
void ff_riff_write_info(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
int i;
int64_t list_pos;
AVDictionaryEntry *t = NULL;
ff_metadata_conv(&s->metadata, ff_riff_info_conv, NULL);
/* writing empty LIST is not nice and may cause problems */
if (!riff_has_valid_tags(s))
return;
list_pos = ff_start_tag(pb, "LIST");
ffio_wfourcc(pb, "INFO");
for (i = 0; *riff_tags[i]; i++)
if ((t = av_dict_get(s->metadata, riff_tags[i],
NULL, AV_DICT_MATCH_CASE)))
ff_riff_write_info_tag(s->pb, t->key, t->value);
ff_end_tag(pb, list_pos);
}