Libav
pcm.c
Go to the documentation of this file.
1 /*
2  * PCM codecs
3  * Copyright (c) 2001 Fabrice Bellard
4  *
5  * This file is part of Libav.
6  *
7  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
27 #include "libavutil/attributes.h"
28 #include "avcodec.h"
29 #include "bytestream.h"
30 #include "internal.h"
31 #include "mathops.h"
32 #include "pcm_tablegen.h"
33 
35 {
36  avctx->frame_size = 0;
37  switch (avctx->codec->id) {
40  break;
43  break;
44  default:
45  break;
46  }
47 
49  avctx->block_align = avctx->channels * avctx->bits_per_coded_sample / 8;
50  avctx->bit_rate = avctx->block_align * avctx->sample_rate * 8;
51  avctx->coded_frame = av_frame_alloc();
52  if (!avctx->coded_frame)
53  return AVERROR(ENOMEM);
54 
55  return 0;
56 }
57 
59 {
60  av_freep(&avctx->coded_frame);
61 
62  return 0;
63 }
64 
75 #define ENCODE(type, endian, src, dst, n, shift, offset) \
76  samples_ ## type = (const type *) src; \
77  for (; n > 0; n--) { \
78  register type v = (*samples_ ## type++ >> shift) + offset; \
79  bytestream_put_ ## endian(&dst, v); \
80  }
81 
82 static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
83  const AVFrame *frame, int *got_packet_ptr)
84 {
85  int n, sample_size, v, ret;
86  const short *samples;
87  unsigned char *dst;
88  const uint8_t *srcu8;
89  const int16_t *samples_int16_t;
90  const int32_t *samples_int32_t;
91  const int64_t *samples_int64_t;
92  const uint16_t *samples_uint16_t;
93  const uint32_t *samples_uint32_t;
94 
95  sample_size = av_get_bits_per_sample(avctx->codec->id) / 8;
96  n = frame->nb_samples * avctx->channels;
97  samples = (const short *)frame->data[0];
98 
99  if ((ret = ff_alloc_packet(avpkt, n * sample_size))) {
100  av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
101  return ret;
102  }
103  dst = avpkt->data;
104 
105  switch (avctx->codec->id) {
107  ENCODE(uint32_t, le32, samples, dst, n, 0, 0x80000000)
108  break;
110  ENCODE(uint32_t, be32, samples, dst, n, 0, 0x80000000)
111  break;
113  ENCODE(int32_t, le24, samples, dst, n, 8, 0)
114  break;
116  ENCODE(int32_t, be24, samples, dst, n, 8, 0)
117  break;
119  ENCODE(uint32_t, le24, samples, dst, n, 8, 0x800000)
120  break;
122  ENCODE(uint32_t, be24, samples, dst, n, 8, 0x800000)
123  break;
125  for (; n > 0; n--) {
126  uint32_t tmp = ff_reverse[(*samples >> 8) & 0xff] +
127  (ff_reverse[*samples & 0xff] << 8);
128  tmp <<= 4; // sync flags would go here
129  bytestream_put_be24(&dst, tmp);
130  samples++;
131  }
132  break;
134  ENCODE(uint16_t, le16, samples, dst, n, 0, 0x8000)
135  break;
137  ENCODE(uint16_t, be16, samples, dst, n, 0, 0x8000)
138  break;
139  case AV_CODEC_ID_PCM_S8:
140  srcu8 = frame->data[0];
141  for (; n > 0; n--) {
142  v = *srcu8++;
143  *dst++ = v - 128;
144  }
145  break;
146 #if HAVE_BIGENDIAN
148  ENCODE(int64_t, le64, samples, dst, n, 0, 0)
149  break;
152  ENCODE(int32_t, le32, samples, dst, n, 0, 0)
153  break;
155  ENCODE(int16_t, le16, samples, dst, n, 0, 0)
156  break;
161 #else
163  ENCODE(int64_t, be64, samples, dst, n, 0, 0)
164  break;
167  ENCODE(int32_t, be32, samples, dst, n, 0, 0)
168  break;
170  ENCODE(int16_t, be16, samples, dst, n, 0, 0)
171  break;
176 #endif /* HAVE_BIGENDIAN */
177  case AV_CODEC_ID_PCM_U8:
178  memcpy(dst, samples, n * sample_size);
179  dst += n * sample_size;
180  break;
182  for (; n > 0; n--) {
183  v = *samples++;
184  *dst++ = linear_to_alaw[(v + 32768) >> 2];
185  }
186  break;
188  for (; n > 0; n--) {
189  v = *samples++;
190  *dst++ = linear_to_ulaw[(v + 32768) >> 2];
191  }
192  break;
193  default:
194  return -1;
195  }
196 
197  *got_packet_ptr = 1;
198  return 0;
199 }
200 
201 typedef struct PCMDecode {
202  short table[256];
203 } PCMDecode;
204 
206 {
207  PCMDecode *s = avctx->priv_data;
208  int i;
209 
210  if (avctx->channels <= 0) {
211  av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n");
212  return AVERROR(EINVAL);
213  }
214 
215  switch (avctx->codec->id) {
217  for (i = 0; i < 256; i++)
218  s->table[i] = alaw2linear(i);
219  break;
221  for (i = 0; i < 256; i++)
222  s->table[i] = ulaw2linear(i);
223  break;
224  default:
225  break;
226  }
227 
228  avctx->sample_fmt = avctx->codec->sample_fmts[0];
229 
230  if (avctx->sample_fmt == AV_SAMPLE_FMT_S32)
232 
233  return 0;
234 }
235 
246 #define DECODE(size, endian, src, dst, n, shift, offset) \
247  for (; n > 0; n--) { \
248  uint ## size ## _t v = bytestream_get_ ## endian(&src); \
249  AV_WN ## size ## A(dst, (v - offset) << shift); \
250  dst += size / 8; \
251  }
252 
253 #if HAVE_BIGENDIAN
254 #define DECODE_PLANAR(size, endian, src, dst, n, shift, offset) \
255  { \
256  int av_unused n2; \
257  n /= avctx->channels; \
258  for (c = 0; c < avctx->channels; c++) { \
259  samples = frame->extended_data[c]; \
260  n2 = n; \
261  DECODE(size, endian, src, samples, n2, 0, 0) \
262  } \
263  }
264 #else
265 #define DECODE_PLANAR(size, endian, src, dst, n, shift, offset) \
266  { \
267  int av_unused n2; \
268  n /= avctx->channels; \
269  for (c = 0; c < avctx->channels; c++) { \
270  samples = frame->extended_data[c]; \
271  memcpy(samples, src, n * size / 8); \
272  src += n * size / 8; \
273  } \
274  }
275 #endif /* HAVE_BIGENDIAN */
276 
277 static int pcm_decode_frame(AVCodecContext *avctx, void *data,
278  int *got_frame_ptr, AVPacket *avpkt)
279 {
280  const uint8_t *src = avpkt->data;
281  int buf_size = avpkt->size;
282  PCMDecode *s = avctx->priv_data;
283  AVFrame *frame = data;
284  int sample_size, c, n, ret, samples_per_block;
285  uint8_t *samples;
286  int32_t *dst_int32_t;
287 
288  sample_size = av_get_bits_per_sample(avctx->codec_id) / 8;
289 
290  /* av_get_bits_per_sample returns 0 for AV_CODEC_ID_PCM_DVD */
291  samples_per_block = 1;
292  if (avctx->codec_id == AV_CODEC_ID_PCM_LXF) {
293  /* we process 40-bit blocks per channel for LXF */
294  samples_per_block = 2;
295  sample_size = 5;
296  }
297 
298  if (sample_size == 0) {
299  av_log(avctx, AV_LOG_ERROR, "Invalid sample_size\n");
300  return AVERROR(EINVAL);
301  }
302 
303  n = avctx->channels * sample_size;
304 
305  if (n && buf_size % n) {
306  if (buf_size < n) {
307  av_log(avctx, AV_LOG_ERROR, "invalid PCM packet\n");
308  return -1;
309  } else
310  buf_size -= buf_size % n;
311  }
312 
313  n = buf_size / sample_size;
314 
315  /* get output buffer */
316  frame->nb_samples = n * samples_per_block / avctx->channels;
317  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
318  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
319  return ret;
320  }
321  samples = frame->data[0];
322 
323  switch (avctx->codec->id) {
325  DECODE(32, le32, src, samples, n, 0, 0x80000000)
326  break;
328  DECODE(32, be32, src, samples, n, 0, 0x80000000)
329  break;
331  DECODE(32, le24, src, samples, n, 8, 0)
332  break;
334  DECODE(32, be24, src, samples, n, 8, 0)
335  break;
337  DECODE(32, le24, src, samples, n, 8, 0x800000)
338  break;
340  DECODE(32, be24, src, samples, n, 8, 0x800000)
341  break;
343  for (; n > 0; n--) {
344  uint32_t v = bytestream_get_be24(&src);
345  v >>= 4; // sync flags are here
346  AV_WN16A(samples, ff_reverse[(v >> 8) & 0xff] +
347  (ff_reverse[v & 0xff] << 8));
348  samples += 2;
349  }
350  break;
352  DECODE_PLANAR(16, le16, src, samples, n, 0, 0);
353  break;
355  DECODE_PLANAR(32, le24, src, samples, n, 8, 0);
356  break;
358  DECODE_PLANAR(32, le32, src, samples, n, 0, 0);
359  break;
361  DECODE(16, le16, src, samples, n, 0, 0x8000)
362  break;
364  DECODE(16, be16, src, samples, n, 0, 0x8000)
365  break;
366  case AV_CODEC_ID_PCM_S8:
367  for (; n > 0; n--)
368  *samples++ = *src++ + 128;
369  break;
370 #if HAVE_BIGENDIAN
372  DECODE(64, le64, src, samples, n, 0, 0)
373  break;
376  DECODE(32, le32, src, samples, n, 0, 0)
377  break;
379  DECODE(16, le16, src, samples, n, 0, 0)
380  break;
385 #else
387  DECODE(64, be64, src, samples, n, 0, 0)
388  break;
391  DECODE(32, be32, src, samples, n, 0, 0)
392  break;
394  DECODE(16, be16, src, samples, n, 0, 0)
395  break;
400 #endif /* HAVE_BIGENDIAN */
401  case AV_CODEC_ID_PCM_U8:
402  memcpy(samples, src, n * sample_size);
403  break;
405  for (; n > 0; n--) {
406  int v = *src++;
407  if (v < 128)
408  v = 128 - v;
409  *samples++ = v;
410  }
411  break;
414  for (; n > 0; n--) {
415  AV_WN16A(samples, s->table[*src++]);
416  samples += 2;
417  }
418  break;
419  case AV_CODEC_ID_PCM_LXF:
420  {
421  int i;
422  n /= avctx->channels;
423  for (c = 0; c < avctx->channels; c++) {
424  dst_int32_t = (int32_t *)frame->extended_data[c];
425  for (i = 0; i < n; i++) {
426  // extract low 20 bits and expand to 32 bits
427  *dst_int32_t++ = (src[2] << 28) |
428  (src[1] << 20) |
429  (src[0] << 12) |
430  ((src[2] & 0x0F) << 8) |
431  src[1];
432  // extract high 20 bits and expand to 32 bits
433  *dst_int32_t++ = (src[4] << 24) |
434  (src[3] << 16) |
435  ((src[2] & 0xF0) << 8) |
436  (src[4] << 4) |
437  (src[3] >> 4);
438  src += 5;
439  }
440  }
441  break;
442  }
443  default:
444  return -1;
445  }
446 
447  *got_frame_ptr = 1;
448 
449  return buf_size;
450 }
451 
452 #define PCM_ENCODER_0(id_, sample_fmt_, name_, long_name_)
453 #define PCM_ENCODER_1(id_, sample_fmt_, name_, long_name_) \
454 AVCodec ff_ ## name_ ## _encoder = { \
455  .name = #name_, \
456  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
457  .type = AVMEDIA_TYPE_AUDIO, \
458  .id = AV_CODEC_ID_ ## id_, \
459  .init = pcm_encode_init, \
460  .encode2 = pcm_encode_frame, \
461  .close = pcm_encode_close, \
462  .capabilities = CODEC_CAP_VARIABLE_FRAME_SIZE, \
463  .sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
464  AV_SAMPLE_FMT_NONE }, \
465 }
466 
467 #define PCM_ENCODER_2(cf, id, sample_fmt, name, long_name) \
468  PCM_ENCODER_ ## cf(id, sample_fmt, name, long_name)
469 #define PCM_ENCODER_3(cf, id, sample_fmt, name, long_name) \
470  PCM_ENCODER_2(cf, id, sample_fmt, name, long_name)
471 #define PCM_ENCODER(id, sample_fmt, name, long_name) \
472  PCM_ENCODER_3(CONFIG_ ## id ## _ENCODER, id, sample_fmt, name, long_name)
473 
474 #define PCM_DECODER_0(id, sample_fmt, name, long_name)
475 #define PCM_DECODER_1(id_, sample_fmt_, name_, long_name_) \
476 AVCodec ff_ ## name_ ## _decoder = { \
477  .name = #name_, \
478  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
479  .type = AVMEDIA_TYPE_AUDIO, \
480  .id = AV_CODEC_ID_ ## id_, \
481  .priv_data_size = sizeof(PCMDecode), \
482  .init = pcm_decode_init, \
483  .decode = pcm_decode_frame, \
484  .capabilities = CODEC_CAP_DR1, \
485  .sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
486  AV_SAMPLE_FMT_NONE }, \
487 }
488 
489 #define PCM_DECODER_2(cf, id, sample_fmt, name, long_name) \
490  PCM_DECODER_ ## cf(id, sample_fmt, name, long_name)
491 #define PCM_DECODER_3(cf, id, sample_fmt, name, long_name) \
492  PCM_DECODER_2(cf, id, sample_fmt, name, long_name)
493 #define PCM_DECODER(id, sample_fmt, name, long_name) \
494  PCM_DECODER_3(CONFIG_ ## id ## _DECODER, id, sample_fmt, name, long_name)
495 
496 #define PCM_CODEC(id, sample_fmt_, name, long_name_) \
497  PCM_ENCODER(id, sample_fmt_, name, long_name_); \
498  PCM_DECODER(id, sample_fmt_, name, long_name_)
499 
500 /* Note: Do not forget to add new entries to the Makefile as well. */
501 PCM_CODEC (PCM_ALAW, AV_SAMPLE_FMT_S16, pcm_alaw, "PCM A-law");
502 PCM_CODEC (PCM_F32BE, AV_SAMPLE_FMT_FLT, pcm_f32be, "PCM 32-bit floating point big-endian");
503 PCM_CODEC (PCM_F32LE, AV_SAMPLE_FMT_FLT, pcm_f32le, "PCM 32-bit floating point little-endian");
504 PCM_CODEC (PCM_F64BE, AV_SAMPLE_FMT_DBL, pcm_f64be, "PCM 64-bit floating point big-endian");
505 PCM_CODEC (PCM_F64LE, AV_SAMPLE_FMT_DBL, pcm_f64le, "PCM 64-bit floating point little-endian");
506 PCM_DECODER(PCM_LXF, AV_SAMPLE_FMT_S32P, pcm_lxf, "PCM signed 20-bit little-endian planar");
507 PCM_CODEC (PCM_MULAW, AV_SAMPLE_FMT_S16, pcm_mulaw, "PCM mu-law");
508 PCM_CODEC (PCM_S8, AV_SAMPLE_FMT_U8, pcm_s8, "PCM signed 8-bit");
509 PCM_CODEC (PCM_S16BE, AV_SAMPLE_FMT_S16, pcm_s16be, "PCM signed 16-bit big-endian");
510 PCM_CODEC (PCM_S16LE, AV_SAMPLE_FMT_S16, pcm_s16le, "PCM signed 16-bit little-endian");
511 PCM_DECODER(PCM_S16LE_PLANAR, AV_SAMPLE_FMT_S16P, pcm_s16le_planar, "PCM 16-bit little-endian planar");
512 PCM_CODEC (PCM_S24BE, AV_SAMPLE_FMT_S32, pcm_s24be, "PCM signed 24-bit big-endian");
513 PCM_CODEC (PCM_S24DAUD, AV_SAMPLE_FMT_S16, pcm_s24daud, "PCM D-Cinema audio signed 24-bit");
514 PCM_CODEC (PCM_S24LE, AV_SAMPLE_FMT_S32, pcm_s24le, "PCM signed 24-bit little-endian");
515 PCM_DECODER(PCM_S24LE_PLANAR, AV_SAMPLE_FMT_S32P,pcm_s24le_planar, "PCM signed 24-bit little-endian planar");
516 PCM_CODEC (PCM_S32BE, AV_SAMPLE_FMT_S32, pcm_s32be, "PCM signed 32-bit big-endian");
517 PCM_CODEC (PCM_S32LE, AV_SAMPLE_FMT_S32, pcm_s32le, "PCM signed 32-bit little-endian");
518 PCM_DECODER(PCM_S32LE_PLANAR, AV_SAMPLE_FMT_S32P,pcm_s32le_planar, "PCM signed 32-bit little-endian planar");
519 PCM_CODEC (PCM_U8, AV_SAMPLE_FMT_U8, pcm_u8, "PCM unsigned 8-bit");
520 PCM_CODEC (PCM_U16BE, AV_SAMPLE_FMT_S16, pcm_u16be, "PCM unsigned 16-bit big-endian");
521 PCM_CODEC (PCM_U16LE, AV_SAMPLE_FMT_S16, pcm_u16le, "PCM unsigned 16-bit little-endian");
522 PCM_CODEC (PCM_U24BE, AV_SAMPLE_FMT_S32, pcm_u24be, "PCM unsigned 24-bit big-endian");
523 PCM_CODEC (PCM_U24LE, AV_SAMPLE_FMT_S32, pcm_u24le, "PCM unsigned 24-bit little-endian");
524 PCM_CODEC (PCM_U32BE, AV_SAMPLE_FMT_S32, pcm_u32be, "PCM unsigned 32-bit big-endian");
525 PCM_CODEC (PCM_U32LE, AV_SAMPLE_FMT_S32, pcm_u32le, "PCM unsigned 32-bit little-endian");
526 PCM_DECODER(PCM_ZORK, AV_SAMPLE_FMT_U8, pcm_zork, "PCM Zork");
const struct AVCodec * codec
Definition: avcodec.h:1063
This structure describes decoded (raw) audio or video data.
Definition: frame.h:107
static void pcm_alaw_tableinit(void)
Definition: pcm_tablegen.h:108
AVFrame * coded_frame
the picture in the bitstream
Definition: avcodec.h:2506
static void pcm_ulaw_tableinit(void)
Definition: pcm_tablegen.h:113
#define DECODE_PLANAR(size, endian, src, dst, n, shift, offset)
Definition: pcm.c:265
static av_cold int ulaw2linear(unsigned char u_val)
Definition: pcm_tablegen.h:58
static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: pcm.c:82
int size
Definition: avcodec.h:974
static int16_t * samples
Definition: output.c:53
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:2488
signed 16 bits
Definition: samplefmt.h:52
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1816
Macro definitions for various function/variable attributes.
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:198
Definition: pcm.c:201
short table[256]
Definition: pcm.c:202
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1787
uint8_t
#define av_cold
Definition: attributes.h:66
AV_SAMPLE_FMT_U8
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:43
const char data[16]
Definition: mxf.c:66
uint8_t * data
Definition: avcodec.h:973
static uint8_t linear_to_ulaw[16384]
Definition: pcm_tablegen.h:82
signed 32 bits, planar
Definition: samplefmt.h:59
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:2481
static int pcm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: pcm.c:277
enum AVCodecID id
Definition: avcodec.h:2769
#define PCM_CODEC(id, sample_fmt_, name, long_name_)
Definition: pcm.c:496
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:1938
#define AVERROR(e)
Definition: error.h:43
#define DECODE(size, endian, src, dst, n, shift, offset)
Read PCM samples macro.
Definition: pcm.c:246
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:148
static av_cold int pcm_encode_close(AVCodecContext *avctx)
Definition: pcm.c:58
signed 32 bits
Definition: samplefmt.h:53
#define ENCODE(type, endian, src, dst, n, shift, offset)
Write PCM samples macro.
Definition: pcm.c:75
static av_cold int pcm_encode_init(AVCodecContext *avctx)
Definition: pcm.c:34
int bit_rate
the average bitrate
Definition: avcodec.h:1112
static av_cold int alaw2linear(unsigned char a_val)
Definition: pcm_tablegen.h:43
int32_t
int ff_alloc_packet(AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1125
#define AV_WN16A(p, v)
Definition: intreadwrite.h:454
if(ac->has_optimized_func)
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1799
Libavcodec external API header.
enum AVCodecID codec_id
Definition: avcodec.h:1065
int sample_rate
samples per second
Definition: avcodec.h:1779
main external API structure.
Definition: avcodec.h:1054
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:575
static av_cold int pcm_decode_init(AVCodecContext *avctx)
Definition: pcm.c:205
#define PCM_DECODER(id, sample_fmt, name, long_name)
Definition: pcm.c:493
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:113
common internal api header.
static uint8_t linear_to_alaw[16384]
Definition: pcm_tablegen.h:81
void * priv_data
Definition: avcodec.h:1090
int channels
number of audio channels
Definition: avcodec.h:1780
signed 16 bits, planar
Definition: samplefmt.h:58
const uint8_t ff_reverse[256]
Definition: mathtables.c:70
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: avcodec.h:2778
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:141
This structure stores compressed data.
Definition: avcodec.h:950
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:151
for(j=16;j >0;--j)