FFmpeg  4.3
spdifdec.c
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1 /*
2  * IEC 61937 demuxer
3  * Copyright (c) 2010 Anssi Hannula <anssi.hannula at iki.fi>
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * IEC 61937 demuxer, used for compressed data in S/PDIF
25  * @author Anssi Hannula
26  */
27 
28 #include "libavutil/bswap.h"
29 
30 #include "libavcodec/ac3.h"
31 #include "libavcodec/adts_parser.h"
32 
33 #include "avformat.h"
34 #include "spdif.h"
35 
37  enum IEC61937DataType data_type,
38  const char *buf, int *offset,
39  enum AVCodecID *codec)
40 {
41  uint32_t samples;
43  int ret;
44 
45  switch (data_type & 0xff) {
46  case IEC61937_AC3:
47  *offset = AC3_FRAME_SIZE << 2;
48  *codec = AV_CODEC_ID_AC3;
49  break;
52  *codec = AV_CODEC_ID_MP1;
53  break;
56  *codec = AV_CODEC_ID_MP3;
57  break;
58  case IEC61937_MPEG2_EXT:
59  *offset = 4608;
60  *codec = AV_CODEC_ID_MP3;
61  break;
62  case IEC61937_MPEG2_AAC:
64  if (ret < 0) {
65  if (s) /* be silent during a probe */
66  av_log(s, AV_LOG_ERROR, "Invalid AAC packet in IEC 61937\n");
67  return ret;
68  }
69  *offset = samples << 2;
70  *codec = AV_CODEC_ID_AAC;
71  break;
74  *codec = AV_CODEC_ID_MP1;
75  break;
78  *codec = AV_CODEC_ID_MP2;
79  break;
82  *codec = AV_CODEC_ID_MP3;
83  break;
84  case IEC61937_DTS1:
85  *offset = 2048;
86  *codec = AV_CODEC_ID_DTS;
87  break;
88  case IEC61937_DTS2:
89  *offset = 4096;
90  *codec = AV_CODEC_ID_DTS;
91  break;
92  case IEC61937_DTS3:
93  *offset = 8192;
94  *codec = AV_CODEC_ID_DTS;
95  break;
96  default:
97  if (s) { /* be silent during a probe */
98  avpriv_request_sample(s, "Data type 0x%04x in IEC 61937",
99  data_type);
100  }
101  return AVERROR_PATCHWELCOME;
102  }
103  return 0;
104 }
105 
106 /* Largest offset between bursts we currently handle, i.e. AAC with
107  samples = 4096 */
108 #define SPDIF_MAX_OFFSET 16384
109 
110 static int spdif_probe(const AVProbeData *p)
111 {
112  enum AVCodecID codec;
113  return ff_spdif_probe (p->buf, p->buf_size, &codec);
114 }
115 
116 int ff_spdif_probe(const uint8_t *p_buf, int buf_size, enum AVCodecID *codec)
117 {
118  const uint8_t *buf = p_buf;
119  const uint8_t *probe_end = p_buf + FFMIN(2 * SPDIF_MAX_OFFSET, buf_size - 1);
120  const uint8_t *expected_code = buf + 7;
121  uint32_t state = 0;
122  int sync_codes = 0;
123  int consecutive_codes = 0;
124  int offset;
125 
126  for (; buf < probe_end; buf++) {
127  state = (state << 8) | *buf;
128 
129  if (state == (AV_BSWAP16C(SYNCWORD1) << 16 | AV_BSWAP16C(SYNCWORD2))
130  && buf[1] < 0x37) {
131  sync_codes++;
132 
133  if (buf == expected_code) {
134  if (++consecutive_codes >= 2)
135  return AVPROBE_SCORE_MAX;
136  } else
137  consecutive_codes = 0;
138 
139  if (buf + 4 + AV_AAC_ADTS_HEADER_SIZE > p_buf + buf_size)
140  break;
141 
142  /* continue probing to find more sync codes */
143  probe_end = FFMIN(buf + SPDIF_MAX_OFFSET, p_buf + buf_size - 1);
144 
145  /* skip directly to the next sync code */
146  if (!spdif_get_offset_and_codec(NULL, (buf[2] << 8) | buf[1],
147  &buf[5], &offset, codec)) {
148  if (buf + offset >= p_buf + buf_size)
149  break;
150  expected_code = buf + offset;
151  buf = expected_code - 7;
152  }
153  }
154  }
155 
156  if (!sync_codes)
157  return 0;
158 
159  if (sync_codes >= 6)
160  /* good amount of sync codes but with unexpected offsets */
162 
163  /* some sync codes were found */
164  return AVPROBE_SCORE_EXTENSION / 4;
165 }
166 
168 {
169  s->ctx_flags |= AVFMTCTX_NOHEADER;
170  return 0;
171 }
172 
174 {
175  AVIOContext *pb = s->pb;
176  enum IEC61937DataType data_type;
177  enum AVCodecID codec_id;
178  uint32_t state = 0;
179  int pkt_size_bits, offset, ret;
180 
181  while (state != (AV_BSWAP16C(SYNCWORD1) << 16 | AV_BSWAP16C(SYNCWORD2))) {
182  state = (state << 8) | avio_r8(pb);
183  if (avio_feof(pb))
184  return AVERROR_EOF;
185  }
186 
187  data_type = avio_rl16(pb);
188  pkt_size_bits = avio_rl16(pb);
189 
190  if (pkt_size_bits % 16)
191  avpriv_request_sample(s, "Packet not ending at a 16-bit boundary");
192 
193  ret = av_new_packet(pkt, FFALIGN(pkt_size_bits, 16) >> 3);
194  if (ret)
195  return ret;
196 
198 
199  if (avio_read(pb, pkt->data, pkt->size) < pkt->size) {
200  return AVERROR_EOF;
201  }
202  ff_spdif_bswap_buf16((uint16_t *)pkt->data, (uint16_t *)pkt->data, pkt->size >> 1);
203 
204  ret = spdif_get_offset_and_codec(s, data_type, pkt->data,
205  &offset, &codec_id);
206  if (ret < 0) {
207  return ret;
208  }
209 
210  /* skip over the padding to the beginning of the next frame */
212 
213  if (!s->nb_streams) {
214  /* first packet, create a stream */
216  if (!st) {
217  return AVERROR(ENOMEM);
218  }
220  st->codecpar->codec_id = codec_id;
221  } else if (codec_id != s->streams[0]->codecpar->codec_id) {
222  avpriv_report_missing_feature(s, "Codec change in IEC 61937");
223  return AVERROR_PATCHWELCOME;
224  }
225 
226  if (!s->bit_rate && s->streams[0]->codecpar->sample_rate)
227  /* stream bitrate matches 16-bit stereo PCM bitrate for currently
228  supported codecs */
229  s->bit_rate = 2 * 16 * s->streams[0]->codecpar->sample_rate;
230 
231  return 0;
232 }
233 
235  .name = "spdif",
236  .long_name = NULL_IF_CONFIG_SMALL("IEC 61937 (compressed data in S/PDIF)"),
237  .read_probe = spdif_probe,
238  .read_header = spdif_read_header,
239  .read_packet = ff_spdif_read_packet,
240  .flags = AVFMT_GENERIC_INDEX,
241 };
AV_CODEC_ID_AC3
@ AV_CODEC_ID_AC3
Definition: codec_id.h:413
IEC61937DataType
IEC61937DataType
Definition: spdif.h:32
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
avformat_new_stream
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4519
ff_spdif_demuxer
AVInputFormat ff_spdif_demuxer
Definition: spdifdec.c:234
AVCodecParameters::codec_type
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:56
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:55
AV_AAC_ADTS_HEADER_SIZE
#define AV_AAC_ADTS_HEADER_SIZE
Definition: adts_parser.h:25
AVPacket::data
uint8_t * data
Definition: packet.h:355
IEC61937_MPEG2_LAYER1_LSF
@ IEC61937_MPEG2_LAYER1_LSF
MPEG-2, layer-1 low sampling frequency.
Definition: spdif.h:38
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
AVProbeData::buf_size
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:444
av_adts_header_parse
int av_adts_header_parse(const uint8_t *buf, uint32_t *samples, uint8_t *frames)
Extract the number of samples and frames from AAC data.
Definition: adts_parser.c:27
AVPROBE_SCORE_MAX
#define AVPROBE_SCORE_MAX
maximum score
Definition: avformat.h:453
ff_spdif_read_packet
int ff_spdif_read_packet(AVFormatContext *s, AVPacket *pkt)
Definition: spdifdec.c:173
frames
if it could not because there are no more frames
Definition: filter_design.txt:266
SPDIF_MAX_OFFSET
#define SPDIF_MAX_OFFSET
Definition: spdifdec.c:108
avio_tell
static av_always_inline int64_t avio_tell(AVIOContext *s)
ftell() equivalent for AVIOContext.
Definition: avio.h:557
AVFMT_GENERIC_INDEX
#define AVFMT_GENERIC_INDEX
Use generic index building code.
Definition: avformat.h:463
AV_CODEC_ID_MP3
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: codec_id.h:411
avio_rl16
unsigned int avio_rl16(AVIOContext *s)
Definition: aviobuf.c:731
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVInputFormat
Definition: avformat.h:636
ff_spdif_bswap_buf16
void ff_spdif_bswap_buf16(uint16_t *dst, const uint16_t *src, int w)
Definition: spdif.c:26
IEC61937_MPEG2_EXT
@ IEC61937_MPEG2_EXT
MPEG-2 data with extension.
Definition: spdif.h:36
s
#define s(width, name)
Definition: cbs_vp9.c:257
av_new_packet
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:88
AVInputFormat::name
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:641
AVProbeData::buf
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:443
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AV_CODEC_ID_MP2
@ AV_CODEC_ID_MP2
Definition: codec_id.h:410
AV_BSWAP16C
#define AV_BSWAP16C(x)
Definition: bswap.h:51
codec_id
enum AVCodecID codec_id
Definition: vaapi_decode.c:369
AVFormatContext
Format I/O context.
Definition: avformat.h:1335
IEC61937_AC3
@ IEC61937_AC3
AC-3 data.
Definition: spdif.h:33
AVStream::codecpar
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1012
NULL
#define NULL
Definition: coverity.c:32
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
AVFMTCTX_NOHEADER
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1284
AVProbeData
This structure contains the data a format has to probe a file.
Definition: avformat.h:441
AVPROBE_SCORE_EXTENSION
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:451
AVCodecID
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: codec_id.h:46
IEC61937_DTS3
@ IEC61937_DTS3
DTS type III (2048 samples)
Definition: spdif.h:43
AV_CODEC_ID_AAC
@ AV_CODEC_ID_AAC
Definition: codec_id.h:412
spdif_get_offset_and_codec
static int spdif_get_offset_and_codec(AVFormatContext *s, enum IEC61937DataType data_type, const char *buf, int *offset, enum AVCodecID *codec)
Definition: spdifdec.c:36
IEC61937_MPEG1_LAYER23
@ IEC61937_MPEG1_LAYER23
MPEG-1 layer 2 or 3 data or MPEG-2 without extension.
Definition: spdif.h:35
AVIOContext
Bytestream IO Context.
Definition: avio.h:161
AVPacket::size
int size
Definition: packet.h:356
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:186
IEC61937_DTS2
@ IEC61937_DTS2
DTS type II (1024 samples)
Definition: spdif.h:42
AV_CODEC_ID_DTS
@ AV_CODEC_ID_DTS
Definition: codec_id.h:414
state
static struct @314 state
avpriv_report_missing_feature
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
avio_r8
int avio_r8(AVIOContext *s)
Definition: aviobuf.c:616
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
offset
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
Definition: writing_filters.txt:86
IEC61937_DTS1
@ IEC61937_DTS1
DTS type I (512 samples)
Definition: spdif.h:41
ff_spdif_probe
int ff_spdif_probe(const uint8_t *p_buf, int buf_size, enum AVCodecID *codec)
Definition: spdifdec.c:116
IEC61937_MPEG1_LAYER1
@ IEC61937_MPEG1_LAYER1
MPEG-1 layer 1.
Definition: spdif.h:34
uint8_t
uint8_t
Definition: audio_convert.c:194
ret
ret
Definition: filter_design.txt:187
AVStream
Stream structure.
Definition: avformat.h:865
bswap.h
adts_parser.h
avformat.h
SYNCWORD2
#define SYNCWORD2
Definition: spdif.h:29
spdif_probe
static int spdif_probe(const AVProbeData *p)
Definition: spdifdec.c:110
BURST_HEADER_SIZE
#define BURST_HEADER_SIZE
Definition: spdif.h:30
pkt
static AVPacket pkt
Definition: demuxing_decoding.c:54
avio_read
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:625
avio_skip
int64_t avio_skip(AVIOContext *s, int64_t offset)
Skip given number of bytes forward.
Definition: aviobuf.c:329
AC3_FRAME_SIZE
#define AC3_FRAME_SIZE
Definition: ac3.h:38
IEC61937_MPEG2_LAYER2_LSF
@ IEC61937_MPEG2_LAYER2_LSF
MPEG-2, layer-2 low sampling frequency.
Definition: spdif.h:39
avpriv_request_sample
#define avpriv_request_sample(...)
Definition: tableprint_vlc.h:39
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:48
AVCodecParameters::codec_id
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:60
AVPacket
This structure stores compressed data.
Definition: packet.h:332
ac3.h
AVPacket::pos
int64_t pos
byte position in stream, -1 if unknown
Definition: packet.h:375
spdif_read_header
static int spdif_read_header(AVFormatContext *s)
Definition: spdifdec.c:167
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
IEC61937_MPEG2_AAC
@ IEC61937_MPEG2_AAC
MPEG-2 AAC ADTS.
Definition: spdif.h:37
AV_CODEC_ID_MP1
@ AV_CODEC_ID_MP1
Definition: codec_id.h:452
spdif_mpeg_pkt_offset
static const uint16_t spdif_mpeg_pkt_offset[2][3]
Definition: spdif.h:55
IEC61937_MPEG2_LAYER3_LSF
@ IEC61937_MPEG2_LAYER3_LSF
MPEG-2, layer-3 low sampling frequency.
Definition: spdif.h:40
spdif.h
SYNCWORD1
#define SYNCWORD1
Definition: spdif.h:28
avio_feof
int avio_feof(AVIOContext *s)
Similar to feof() but also returns nonzero on read errors.
Definition: aviobuf.c:356