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FFmpeg/libavformat/rtpdec_mpeg4.c

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/*
* Common code for the RTP depacketization of MPEG-4 formats.
* Copyright (c) 2010 Fabrice Bellard
* Romain Degez
*
* 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
*/
/**
* @file
* @brief MPEG-4 / RTP Code
* @author Fabrice Bellard
* @author Romain Degez
*/
#include "rtpdec_formats.h"
#include "internal.h"
#include "libavutil/attributes.h"
#include "libavutil/avstring.h"
#include "libavcodec/get_bits.h"
#define MAX_AAC_HBR_FRAME_SIZE 8191
/** Structure listing useful vars to parse RTP packet payload */
struct PayloadContext {
int sizelength;
int indexlength;
int indexdeltalength;
int profile_level_id;
int streamtype;
int objecttype;
char *mode;
/** mpeg 4 AU headers */
struct AUHeaders {
int size;
int index;
int cts_flag;
int cts;
int dts_flag;
int dts;
int rap_flag;
int streamstate;
} *au_headers;
int au_headers_allocated;
int nb_au_headers;
int au_headers_length_bytes;
int cur_au_index;
uint8_t buf[FFMAX(RTP_MAX_PACKET_LENGTH, MAX_AAC_HBR_FRAME_SIZE)];
int buf_pos, buf_size;
uint32_t timestamp;
};
typedef struct AttrNameMap {
const char *str;
uint16_t type;
uint32_t offset;
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
/** Range for integer values */
struct Range {
int min;
int max;
} range;
} AttrNameMap;
/* All known fmtp parameters and the corresponding RTPAttrTypeEnum */
#define ATTR_NAME_TYPE_INT 0
#define ATTR_NAME_TYPE_STR 1
static const AttrNameMap attr_names[] = {
{ "SizeLength", ATTR_NAME_TYPE_INT,
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
offsetof(PayloadContext, sizelength),
{0, 32} }, // SizeLength number of bits used to encode AU-size integer value
{ "IndexLength", ATTR_NAME_TYPE_INT,
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
offsetof(PayloadContext, indexlength),
{0, 32} }, // IndexLength number of bits used to encode AU-Index integer value
{ "IndexDeltaLength", ATTR_NAME_TYPE_INT,
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
offsetof(PayloadContext, indexdeltalength),
{0, 32} }, // IndexDeltaLength number of bits to encode AU-Index-delta integer value
{ "profile-level-id", ATTR_NAME_TYPE_INT,
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
offsetof(PayloadContext, profile_level_id),
{INT32_MIN, INT32_MAX} }, // It differs depending on StreamType
{ "StreamType", ATTR_NAME_TYPE_INT,
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
offsetof(PayloadContext, streamtype),
{0x00, 0x3F} }, // Values from ISO/IEC 14496-1, 'StreamType Values' table
{ "mode", ATTR_NAME_TYPE_STR,
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
offsetof(PayloadContext, mode),
{0} },
{ NULL, -1, -1, {0} },
};
static void close_context(PayloadContext *data)
{
av_freep(&data->au_headers);
av_freep(&data->mode);
}
static int parse_fmtp_config(AVCodecParameters *par, const char *value)
{
/* decode the hexa encoded parameter */
int len = ff_hex_to_data(NULL, value), ret;
if ((ret = ff_alloc_extradata(par, len)) < 0)
return ret;
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.
2014-06-18 21:42:52 +03:00
ff_hex_to_data(par->extradata, value);
return 0;
}
static int rtp_parse_mp4_au(PayloadContext *data, const uint8_t *buf, int len)
{
int au_headers_length, au_header_size, i;
GetBitContext getbitcontext;
int ret;
if (len < 2)
return AVERROR_INVALIDDATA;
/* decode the first 2 bytes where the AUHeader sections are stored
length in bits */
au_headers_length = AV_RB16(buf);
if (au_headers_length > RTP_MAX_PACKET_LENGTH)
return -1;
data->au_headers_length_bytes = (au_headers_length + 7) / 8;
/* skip AU headers length section (2 bytes) */
buf += 2;
len -= 2;
if (len < data->au_headers_length_bytes)
return AVERROR_INVALIDDATA;
ret = init_get_bits(&getbitcontext, buf, data->au_headers_length_bytes * 8);
if (ret < 0)
return ret;
/* XXX: Wrong if optional additional sections are present (cts, dts etc...) */
au_header_size = data->sizelength + data->indexlength;
if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
return -1;
data->nb_au_headers = au_headers_length / au_header_size;
if (!data->au_headers || data->au_headers_allocated < data->nb_au_headers) {
av_free(data->au_headers);
data->au_headers = av_malloc(sizeof(struct AUHeaders) * data->nb_au_headers);
if (!data->au_headers)
return AVERROR(ENOMEM);
data->au_headers_allocated = data->nb_au_headers;
}
for (i = 0; i < data->nb_au_headers; ++i) {
data->au_headers[i].size = get_bits_long(&getbitcontext, data->sizelength);
data->au_headers[i].index = get_bits_long(&getbitcontext, data->indexlength);
}
return 0;
}
/* Follows RFC 3640 */
static int aac_parse_packet(AVFormatContext *ctx, PayloadContext *data,
AVStream *st, AVPacket *pkt, uint32_t *timestamp,
const uint8_t *buf, int len, uint16_t seq,
int flags)
{
int ret;
if (!buf) {
if (data->cur_au_index > data->nb_au_headers) {
av_log(ctx, AV_LOG_ERROR, "Invalid parser state\n");
return AVERROR_INVALIDDATA;
}
if (data->buf_size - data->buf_pos < data->au_headers[data->cur_au_index].size) {
av_log(ctx, AV_LOG_ERROR, "Invalid AU size\n");
return AVERROR_INVALIDDATA;
}
if ((ret = av_new_packet(pkt, data->au_headers[data->cur_au_index].size)) < 0) {
av_log(ctx, AV_LOG_ERROR, "Out of memory\n");
return ret;
}
memcpy(pkt->data, &data->buf[data->buf_pos], data->au_headers[data->cur_au_index].size);
data->buf_pos += data->au_headers[data->cur_au_index].size;
pkt->stream_index = st->index;
data->cur_au_index++;
if (data->cur_au_index == data->nb_au_headers) {
data->buf_pos = 0;
return 0;
}
return 1;
}
if (rtp_parse_mp4_au(data, buf, len)) {
av_log(ctx, AV_LOG_ERROR, "Error parsing AU headers\n");
return -1;
}
buf += data->au_headers_length_bytes + 2;
len -= data->au_headers_length_bytes + 2;
if (data->nb_au_headers == 1 && len < data->au_headers[0].size) {
/* Packet is fragmented */
if (!data->buf_pos) {
if (data->au_headers[0].size > MAX_AAC_HBR_FRAME_SIZE) {
av_log(ctx, AV_LOG_ERROR, "Invalid AU size\n");
return AVERROR_INVALIDDATA;
}
data->buf_size = data->au_headers[0].size;
data->timestamp = *timestamp;
}
if (data->timestamp != *timestamp ||
data->au_headers[0].size != data->buf_size ||
data->buf_pos + len > MAX_AAC_HBR_FRAME_SIZE) {
data->buf_pos = 0;
data->buf_size = 0;
av_log(ctx, AV_LOG_ERROR, "Invalid packet received\n");
return AVERROR_INVALIDDATA;
}
memcpy(&data->buf[data->buf_pos], buf, len);
data->buf_pos += len;
if (!(flags & RTP_FLAG_MARKER))
return AVERROR(EAGAIN);
if (data->buf_pos != data->buf_size) {
data->buf_pos = 0;
av_log(ctx, AV_LOG_ERROR, "Missed some packets, discarding frame\n");
return AVERROR_INVALIDDATA;
}
data->buf_pos = 0;
ret = av_new_packet(pkt, data->buf_size);
if (ret < 0) {
av_log(ctx, AV_LOG_ERROR, "Out of memory\n");
return ret;
}
pkt->stream_index = st->index;
memcpy(pkt->data, data->buf, data->buf_size);
return 0;
}
if (len < data->au_headers[0].size) {
av_log(ctx, AV_LOG_ERROR, "First AU larger than packet size\n");
return AVERROR_INVALIDDATA;
}
if ((ret = av_new_packet(pkt, data->au_headers[0].size)) < 0) {
av_log(ctx, AV_LOG_ERROR, "Out of memory\n");
return ret;
}
memcpy(pkt->data, buf, data->au_headers[0].size);
len -= data->au_headers[0].size;
buf += data->au_headers[0].size;
pkt->stream_index = st->index;
if (len > 0 && data->nb_au_headers > 1) {
data->buf_size = FFMIN(len, sizeof(data->buf));
memcpy(data->buf, buf, data->buf_size);
data->cur_au_index = 1;
data->buf_pos = 0;
return 1;
}
return 0;
}
static int parse_fmtp(AVFormatContext *s,
AVStream *stream, PayloadContext *data,
const char *attr, const char *value)
{
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.
2014-06-18 21:42:52 +03:00
AVCodecParameters *par = stream->codecpar;
int res, i;
if (!strcmp(attr, "config")) {
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.
2014-06-18 21:42:52 +03:00
res = parse_fmtp_config(par, value);
if (res < 0)
return res;
}
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.
2014-06-18 21:42:52 +03:00
if (par->codec_id == AV_CODEC_ID_AAC) {
/* Looking for a known attribute */
for (i = 0; attr_names[i].str; ++i) {
if (!av_strcasecmp(attr, attr_names[i].str)) {
if (attr_names[i].type == ATTR_NAME_TYPE_INT) {
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
char *end_ptr = NULL;
long long int val = strtoll(value, &end_ptr, 10);
if (end_ptr == value || end_ptr[0] != '\0') {
2016-08-19 18:35:33 +02:00
av_log(s, AV_LOG_ERROR,
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
"The %s field value is not a valid number: %s\n",
attr, value);
2016-08-19 18:35:33 +02:00
return AVERROR_INVALIDDATA;
}
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
if (val < attr_names[i].range.min ||
val > attr_names[i].range.max) {
av_log(s, AV_LOG_ERROR,
"fmtp field %s should be in range [%d,%d] (provided value: %lld)",
attr, attr_names[i].range.min, attr_names[i].range.max, val);
return AVERROR_INVALIDDATA;
}
*(int *)((char *)data+
avformat/rtpdec_mpeg4: Fix integer parameters size check in SDP fmtp line === PROBLEM === I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line: ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4 FFmpeg then fail to record audio and output this logs: [rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40) [rtsp @ 0xcda1f0] Error parsing AU headers ... [rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format In SDP provided by my RTSP server I had this fmtp line: a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3; In FFmpeg code, I found a check introduced by commit 24130234cd9dd733116d17b724ea4c8e12ce097a. It disallows values greater than 32 for fmtp line parameters. RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line. However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55. === FIX === As each parameter may have its own min and max values I propose to introduce a range for each parameter. For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges. This patch fix my problem and I now can record my RTSP AAC stream to mp4. It has passed the full fate tests suite sucessfully. Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-07-24 10:20:14 +02:00
attr_names[i].offset) = (int) val;
2016-08-19 18:35:33 +02:00
} else if (attr_names[i].type == ATTR_NAME_TYPE_STR) {
char *val = av_strdup(value);
if (!val)
return AVERROR(ENOMEM);
*(char **)((char *)data+
2016-08-19 18:35:33 +02:00
attr_names[i].offset) = val;
}
}
}
}
return 0;
}
static int parse_sdp_line(AVFormatContext *s, int st_index,
PayloadContext *data, const char *line)
{
const char *p;
if (st_index < 0)
return 0;
if (av_strstart(line, "fmtp:", &p))
return ff_parse_fmtp(s, s->streams[st_index], data, p, parse_fmtp);
return 0;
}
const RTPDynamicProtocolHandler ff_mp4v_es_dynamic_handler = {
.enc_name = "MP4V-ES",
.codec_type = AVMEDIA_TYPE_VIDEO,
.codec_id = AV_CODEC_ID_MPEG4,
.need_parsing = AVSTREAM_PARSE_FULL,
.priv_data_size = sizeof(PayloadContext),
.parse_sdp_a_line = parse_sdp_line,
};
const RTPDynamicProtocolHandler ff_mpeg4_generic_dynamic_handler = {
.enc_name = "mpeg4-generic",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_AAC,
.priv_data_size = sizeof(PayloadContext),
.parse_sdp_a_line = parse_sdp_line,
.close = close_context,
.parse_packet = aac_parse_packet,
};