FFmpeg
s302m.c
Go to the documentation of this file.
1 /*
2  * SMPTE 302M decoder
3  * Copyright (c) 2008 Laurent Aimar <fenrir@videolan.org>
4  * Copyright (c) 2009 Baptiste Coudurier <baptiste.coudurier@gmail.com>
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #include "libavutil/intreadwrite.h"
24 #include "libavutil/opt.h"
25 #include "libavutil/log.h"
26 #include "avcodec.h"
27 #include "internal.h"
28 #include "mathops.h"
29 
30 #define AES3_HEADER_LEN 4
31 
32 typedef struct S302Context {
33  AVClass *class;
35 } S302Context;
36 
38  int buf_size)
39 {
40  uint32_t h;
41  int frame_size, channels, bits;
42 
43  if (buf_size <= AES3_HEADER_LEN) {
44  av_log(avctx, AV_LOG_ERROR, "frame is too short\n");
45  return AVERROR_INVALIDDATA;
46  }
47 
48  /*
49  * AES3 header :
50  * size: 16
51  * number channels 2
52  * channel_id 8
53  * bits per samples 2
54  * alignments 4
55  */
56 
57  h = AV_RB32(buf);
58  frame_size = (h >> 16) & 0xffff;
59  channels = ((h >> 14) & 0x0003) * 2 + 2;
60  bits = ((h >> 4) & 0x0003) * 4 + 16;
61 
62  if (AES3_HEADER_LEN + frame_size != buf_size || bits > 24) {
63  av_log(avctx, AV_LOG_ERROR, "frame has invalid header\n");
64  return AVERROR_INVALIDDATA;
65  }
66 
67  /* Set output properties */
68  avctx->bits_per_raw_sample = bits;
69  if (bits > 16)
71  else
73 
74  avctx->channels = channels;
75  switch(channels) {
76  case 2:
78  break;
79  case 4:
81  break;
82  case 6:
84  break;
85  case 8:
87  }
88 
89  return frame_size;
90 }
91 
92 static int s302m_decode_frame(AVCodecContext *avctx, void *data,
93  int *got_frame_ptr, AVPacket *avpkt)
94 {
95  S302Context *s = avctx->priv_data;
96  AVFrame *frame = data;
97  const uint8_t *buf = avpkt->data;
98  int buf_size = avpkt->size;
99  int block_size, ret;
100  int i;
101  int non_pcm_data_type = -1;
102 
103  int frame_size = s302m_parse_frame_header(avctx, buf, buf_size);
104  if (frame_size < 0)
105  return frame_size;
106 
107  buf_size -= AES3_HEADER_LEN;
108  buf += AES3_HEADER_LEN;
109 
110  /* get output buffer */
111  block_size = (avctx->bits_per_raw_sample + 4) / 4;
112  frame->nb_samples = 2 * (buf_size / block_size) / avctx->channels;
113  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
114  return ret;
115 
116  avctx->bit_rate = 48000 * avctx->channels * (avctx->bits_per_raw_sample + 4) +
117  32 * 48000 / frame->nb_samples;
118  buf_size = (frame->nb_samples * avctx->channels / 2) * block_size;
119 
120  if (avctx->bits_per_raw_sample == 24) {
121  uint32_t *o = (uint32_t *)frame->data[0];
122  for (; buf_size > 6; buf_size -= 7) {
123  *o++ = ((unsigned)ff_reverse[buf[2]] << 24) |
124  (ff_reverse[buf[1]] << 16) |
125  (ff_reverse[buf[0]] << 8);
126  *o++ = ((unsigned)ff_reverse[buf[6] & 0xf0] << 28) |
127  (ff_reverse[buf[5]] << 20) |
128  (ff_reverse[buf[4]] << 12) |
129  (ff_reverse[buf[3] & 0x0f] << 4);
130  buf += 7;
131  }
132  o = (uint32_t *)frame->data[0];
133  if (avctx->channels == 2)
134  for (i=0; i<frame->nb_samples * 2 - 6; i+=2) {
135  if (o[i] || o[i+1] || o[i+2] || o[i+3])
136  break;
137  if (o[i+4] == 0x96F87200U && o[i+5] == 0xA54E1F00) {
138  non_pcm_data_type = (o[i+6] >> 16) & 0x1F;
139  break;
140  }
141  }
142  } else if (avctx->bits_per_raw_sample == 20) {
143  uint32_t *o = (uint32_t *)frame->data[0];
144  for (; buf_size > 5; buf_size -= 6) {
145  *o++ = ((unsigned)ff_reverse[buf[2] & 0xf0] << 28) |
146  (ff_reverse[buf[1]] << 20) |
147  (ff_reverse[buf[0]] << 12);
148  *o++ = ((unsigned)ff_reverse[buf[5] & 0xf0] << 28) |
149  (ff_reverse[buf[4]] << 20) |
150  (ff_reverse[buf[3]] << 12);
151  buf += 6;
152  }
153  o = (uint32_t *)frame->data[0];
154  if (avctx->channels == 2)
155  for (i=0; i<frame->nb_samples * 2 - 6; i+=2) {
156  if (o[i] || o[i+1] || o[i+2] || o[i+3])
157  break;
158  if (o[i+4] == 0x6F872000U && o[i+5] == 0x54E1F000) {
159  non_pcm_data_type = (o[i+6] >> 16) & 0x1F;
160  break;
161  }
162  }
163  } else {
164  uint16_t *o = (uint16_t *)frame->data[0];
165  for (; buf_size > 4; buf_size -= 5) {
166  *o++ = (ff_reverse[buf[1]] << 8) |
167  ff_reverse[buf[0]];
168  *o++ = (ff_reverse[buf[4] & 0xf0] << 12) |
169  (ff_reverse[buf[3]] << 4) |
170  (ff_reverse[buf[2]] >> 4);
171  buf += 5;
172  }
173  o = (uint16_t *)frame->data[0];
174  if (avctx->channels == 2)
175  for (i=0; i<frame->nb_samples * 2 - 6; i+=2) {
176  if (o[i] || o[i+1] || o[i+2] || o[i+3])
177  break;
178  if (o[i+4] == 0xF872U && o[i+5] == 0x4E1F) {
179  non_pcm_data_type = (o[i+6] & 0x1F);
180  break;
181  }
182  }
183  }
184 
185  if (non_pcm_data_type != -1) {
186  if (s->non_pcm_mode == 3) {
187  av_log(avctx, AV_LOG_ERROR,
188  "S302 non PCM mode with data type %d not supported\n",
189  non_pcm_data_type);
190  return AVERROR_PATCHWELCOME;
191  }
192  if (s->non_pcm_mode & 1) {
193  return avpkt->size;
194  }
195  }
196 
197  avctx->sample_rate = 48000;
198 
199  *got_frame_ptr = 1;
200 
201  return avpkt->size;
202 }
203 
204 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_DECODING_PARAM
205 static const AVOption s302m_options[] = {
206  {"non_pcm_mode", "Chooses what to do with NON-PCM", offsetof(S302Context, non_pcm_mode), AV_OPT_TYPE_INT, {.i64 = 3}, 0, 3, FLAGS, "non_pcm_mode"},
207  {"copy" , "Pass NON-PCM through unchanged" , 0, AV_OPT_TYPE_CONST, {.i64 = 0}, 0, 3, FLAGS, "non_pcm_mode"},
208  {"drop" , "Drop NON-PCM" , 0, AV_OPT_TYPE_CONST, {.i64 = 1}, 0, 3, FLAGS, "non_pcm_mode"},
209  {"decode_copy" , "Decode if possible else passthrough", 0, AV_OPT_TYPE_CONST, {.i64 = 2}, 0, 3, FLAGS, "non_pcm_mode"},
210  {"decode_drop" , "Decode if possible else drop" , 0, AV_OPT_TYPE_CONST, {.i64 = 3}, 0, 3, FLAGS, "non_pcm_mode"},
211  {NULL}
212 };
213 
214 static const AVClass s302m_class = {
215  .class_name = "SMPTE 302M Decoder",
216  .item_name = av_default_item_name,
217  .option = s302m_options,
218  .version = LIBAVUTIL_VERSION_INT,
219 };
220 
222  .name = "s302m",
223  .long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
224  .type = AVMEDIA_TYPE_AUDIO,
225  .id = AV_CODEC_ID_S302M,
226  .priv_data_size = sizeof(S302Context),
228  .capabilities = AV_CODEC_CAP_DR1,
229  .priv_class = &s302m_class,
230 };
#define NULL
Definition: coverity.c:32
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1618
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf, int buf_size)
Definition: s302m.c:37
const uint8_t ff_reverse[256]
Definition: reverse.c:23
channels
Definition: aptx.c:30
int size
Definition: avcodec.h:1481
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
static const AVOption s302m_options[]
Definition: s302m.c:205
#define AV_CH_LAYOUT_STEREO
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:2799
AVCodec.
Definition: avcodec.h:3492
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2236
uint8_t
AVOptions.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
uint8_t * data
Definition: avcodec.h:1480
signed 32 bits
Definition: samplefmt.h:62
#define av_log(a,...)
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVCodec ff_s302m_decoder
Definition: s302m.c:221
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
#define AV_CH_LAYOUT_QUAD
const char * name
Name of the codec implementation.
Definition: avcodec.h:3499
uint8_t bits
Definition: vp3data.h:202
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2279
#define FLAGS
Definition: s302m.c:204
static const AVClass s302m_class
Definition: s302m.c:214
#define AV_CH_LAYOUT_STEREO_DOWNMIX
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
#define s(width, name)
Definition: cbs_vp9.c:257
#define AV_CH_LAYOUT_5POINT1_BACK
int non_pcm_mode
Definition: s302m.c:34
if(ret)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
int frame_size
Definition: mxfenc.c:2214
Libavcodec external API header.
int sample_rate
samples per second
Definition: avcodec.h:2228
main external API structure.
Definition: avcodec.h:1568
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1968
void * buf
Definition: avisynth_c.h:766
Describe the class of an AVClass context structure.
Definition: log.h:67
#define AES3_HEADER_LEN
Definition: s302m.c:30
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
void * priv_data
Definition: avcodec.h:1595
int channels
number of audio channels
Definition: avcodec.h:2229
static int s302m_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: s302m.c:92
This structure stores compressed data.
Definition: avcodec.h:1457
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:984
for(j=16;j >0;--j)