FFmpeg
s302menc.c
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1 /*
2  * SMPTE 302M encoder
3  * Copyright (c) 2010 Google, Inc.
4  * Copyright (c) 2013 Darryl Wallace <wallacdj@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 "avcodec.h"
24 #include "internal.h"
25 #include "mathops.h"
26 #include "put_bits.h"
27 
28 #define AES3_HEADER_LEN 4
29 
30 typedef struct S302MEncContext {
31  uint8_t framing_index; /* Set for even channels on multiple of 192 samples */
33 
35 {
36  S302MEncContext *s = avctx->priv_data;
37 
38  if (avctx->channels & 1 || avctx->channels > 8) {
39  av_log(avctx, AV_LOG_ERROR,
40  "Encoding %d channel(s) is not allowed. Only 2, 4, 6 and 8 channels are supported.\n",
41  avctx->channels);
42  return AVERROR(EINVAL);
43  }
44 
45  switch (avctx->sample_fmt) {
46  case AV_SAMPLE_FMT_S16:
47  avctx->bits_per_raw_sample = 16;
48  break;
49  case AV_SAMPLE_FMT_S32:
50  if (avctx->bits_per_raw_sample > 20) {
51  if (avctx->bits_per_raw_sample > 24)
52  av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
53  avctx->bits_per_raw_sample = 24;
54  } else if (!avctx->bits_per_raw_sample) {
55  avctx->bits_per_raw_sample = 24;
56  } else if (avctx->bits_per_raw_sample <= 20) {
57  avctx->bits_per_raw_sample = 20;
58  }
59  }
60 
61  avctx->frame_size = 0;
62  avctx->bit_rate = 48000 * avctx->channels *
63  (avctx->bits_per_raw_sample + 4);
64  s->framing_index = 0;
65 
66  return 0;
67 }
68 
69 static int s302m_encode2_frame(AVCodecContext *avctx, AVPacket *avpkt,
70  const AVFrame *frame, int *got_packet_ptr)
71 {
72  S302MEncContext *s = avctx->priv_data;
73  const int buf_size = AES3_HEADER_LEN +
74  (frame->nb_samples *
75  avctx->channels *
76  (avctx->bits_per_raw_sample + 4)) / 8;
77  int ret, c, channels;
78  uint8_t *o;
79  PutBitContext pb;
80 
81  if (buf_size - AES3_HEADER_LEN > UINT16_MAX) {
82  av_log(avctx, AV_LOG_ERROR, "number of samples in frame too big\n");
83  return AVERROR(EINVAL);
84  }
85 
86  if ((ret = ff_alloc_packet2(avctx, avpkt, buf_size, 0)) < 0)
87  return ret;
88 
89  o = avpkt->data;
90  init_put_bits(&pb, o, buf_size);
91  put_bits(&pb, 16, buf_size - AES3_HEADER_LEN);
92  put_bits(&pb, 2, (avctx->channels - 2) >> 1); // number of channels
93  put_bits(&pb, 8, 0); // channel ID
94  put_bits(&pb, 2, (avctx->bits_per_raw_sample - 16) / 4); // bits per samples (0 = 16bit, 1 = 20bit, 2 = 24bit)
95  put_bits(&pb, 4, 0); // alignments
96  flush_put_bits(&pb);
97  o += AES3_HEADER_LEN;
98 
99  if (avctx->bits_per_raw_sample == 24) {
100  const uint32_t *samples = (uint32_t *)frame->data[0];
101 
102  for (c = 0; c < frame->nb_samples; c++) {
103  uint8_t vucf = s->framing_index == 0 ? 0x10: 0;
104 
105  for (channels = 0; channels < avctx->channels; channels += 2) {
106  o[0] = ff_reverse[(samples[0] & 0x0000FF00) >> 8];
107  o[1] = ff_reverse[(samples[0] & 0x00FF0000) >> 16];
108  o[2] = ff_reverse[(samples[0] & 0xFF000000) >> 24];
109  o[3] = ff_reverse[(samples[1] & 0x00000F00) >> 4] | vucf;
110  o[4] = ff_reverse[(samples[1] & 0x000FF000) >> 12];
111  o[5] = ff_reverse[(samples[1] & 0x0FF00000) >> 20];
112  o[6] = ff_reverse[(samples[1] & 0xF0000000) >> 28];
113  o += 7;
114  samples += 2;
115  }
116 
117  s->framing_index++;
118  if (s->framing_index >= 192)
119  s->framing_index = 0;
120  }
121  } else if (avctx->bits_per_raw_sample == 20) {
122  const uint32_t *samples = (uint32_t *)frame->data[0];
123 
124  for (c = 0; c < frame->nb_samples; c++) {
125  uint8_t vucf = s->framing_index == 0 ? 0x80: 0;
126 
127  for (channels = 0; channels < avctx->channels; channels += 2) {
128  o[0] = ff_reverse[ (samples[0] & 0x000FF000) >> 12];
129  o[1] = ff_reverse[ (samples[0] & 0x0FF00000) >> 20];
130  o[2] = ff_reverse[((samples[0] & 0xF0000000) >> 28) | vucf];
131  o[3] = ff_reverse[ (samples[1] & 0x000FF000) >> 12];
132  o[4] = ff_reverse[ (samples[1] & 0x0FF00000) >> 20];
133  o[5] = ff_reverse[ (samples[1] & 0xF0000000) >> 28];
134  o += 6;
135  samples += 2;
136  }
137 
138  s->framing_index++;
139  if (s->framing_index >= 192)
140  s->framing_index = 0;
141  }
142  } else if (avctx->bits_per_raw_sample == 16) {
143  const uint16_t *samples = (uint16_t *)frame->data[0];
144 
145  for (c = 0; c < frame->nb_samples; c++) {
146  uint8_t vucf = s->framing_index == 0 ? 0x10 : 0;
147 
148  for (channels = 0; channels < avctx->channels; channels += 2) {
149  o[0] = ff_reverse[ samples[0] & 0xFF];
150  o[1] = ff_reverse[(samples[0] & 0xFF00) >> 8];
151  o[2] = ff_reverse[(samples[1] & 0x0F) << 4] | vucf;
152  o[3] = ff_reverse[(samples[1] & 0x0FF0) >> 4];
153  o[4] = ff_reverse[(samples[1] & 0xF000) >> 12];
154  o += 5;
155  samples += 2;
156 
157  }
158 
159  s->framing_index++;
160  if (s->framing_index >= 192)
161  s->framing_index = 0;
162  }
163  }
164 
165  *got_packet_ptr = 1;
166 
167  return 0;
168 }
169 
171  .name = "s302m",
172  .long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
173  .type = AVMEDIA_TYPE_AUDIO,
174  .id = AV_CODEC_ID_S302M,
175  .priv_data_size = sizeof(S302MEncContext),
177  .encode2 = s302m_encode2_frame,
178  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
182  .supported_samplerates = (const int[]) { 48000, 0 },
183  /* .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO,
184  AV_CH_LAYOUT_QUAD,
185  AV_CH_LAYOUT_5POINT1_BACK,
186  AV_CH_LAYOUT_5POINT1_BACK | AV_CH_LAYOUT_STEREO_DOWNMIX,
187  0 }, */
188 };
This structure describes decoded (raw) audio or video data.
Definition: frame.h:268
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:208
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1615
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
const uint8_t ff_reverse[256]
Definition: reverse.c:23
channels
Definition: aptx.c:30
#define AV_CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
Definition: avcodec.h:1029
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:2792
AVCodec.
Definition: avcodec.h:3477
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:32
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2229
uint8_t
#define av_cold
Definition: attributes.h:82
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
uint8_t * data
Definition: avcodec.h:1477
signed 32 bits
Definition: samplefmt.h:62
#define av_log(a,...)
static int s302m_encode2_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: s302menc.c:69
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
const char * name
Name of the codec implementation.
Definition: avcodec.h:3484
#define AV_CODEC_CAP_VARIABLE_FRAME_SIZE
Audio encoder supports receiving a different number of samples in each call.
Definition: avcodec.h:1053
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
AVCodec ff_s302m_encoder
Definition: s302menc.c:170
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2241
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
main external API structure.
Definition: avcodec.h:1565
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:282
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:101
#define AES3_HEADER_LEN
Definition: s302menc.c:28
signed 16 bits
Definition: samplefmt.h:61
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
void * priv_data
Definition: avcodec.h:1592
int channels
number of audio channels
Definition: avcodec.h:2222
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
Filter the word “frame” indicates either a video frame or a group of audio samples
uint8_t framing_index
Definition: s302menc.c:31
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
static av_cold int s302m_encode_init(AVCodecContext *avctx)
Definition: s302menc.c:34
This structure stores compressed data.
Definition: avcodec.h:1454
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:334
for(j=16;j >0;--j)
bitstream writer API