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libmp3lame.c
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1 /*
2  * Interface to libmp3lame for mp3 encoding
3  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
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  * Interface to libmp3lame for mp3 encoding.
25  */
26 
27 #include <lame/lame.h>
28 
30 #include "libavutil/common.h"
31 #include "libavutil/float_dsp.h"
32 #include "libavutil/intreadwrite.h"
33 #include "libavutil/log.h"
34 #include "libavutil/opt.h"
35 #include "avcodec.h"
36 #include "audio_frame_queue.h"
37 #include "internal.h"
38 #include "mpegaudio.h"
39 #include "mpegaudiodecheader.h"
40 
41 #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
42 
43 typedef struct LAMEContext {
44  AVClass *class;
46  lame_global_flags *gfp;
50  int reservoir;
52  int abr;
54  float *samples_flt[2];
57 } LAMEContext;
58 
59 
61 {
62  if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
63  int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
64 
65  ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
66  new_size);
67  if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
68  s->buffer_size = s->buffer_index = 0;
69  return err;
70  }
71  s->buffer_size = new_size;
72  }
73  return 0;
74 }
75 
77 {
78  LAMEContext *s = avctx->priv_data;
79 
80  av_freep(&s->samples_flt[0]);
81  av_freep(&s->samples_flt[1]);
82  av_freep(&s->buffer);
83  av_freep(&s->fdsp);
84 
86 
87  lame_close(s->gfp);
88  return 0;
89 }
90 
92 {
93  LAMEContext *s = avctx->priv_data;
94  int ret;
95 
96  s->avctx = avctx;
97 
98  /* initialize LAME and get defaults */
99  if (!(s->gfp = lame_init()))
100  return AVERROR(ENOMEM);
101 
102 
103  lame_set_num_channels(s->gfp, avctx->channels);
104  lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
105 
106  /* sample rate */
107  lame_set_in_samplerate (s->gfp, avctx->sample_rate);
108  lame_set_out_samplerate(s->gfp, avctx->sample_rate);
109 
110  /* algorithmic quality */
112  lame_set_quality(s->gfp, avctx->compression_level);
113 
114  /* rate control */
115  if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
116  lame_set_VBR(s->gfp, vbr_default);
117  lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
118  } else {
119  if (avctx->bit_rate) {
120  if (s->abr) { // ABR
121  lame_set_VBR(s->gfp, vbr_abr);
122  lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
123  } else // CBR
124  lame_set_brate(s->gfp, avctx->bit_rate / 1000);
125  }
126  }
127 
128  /* do not get a Xing VBR header frame from LAME */
129  lame_set_bWriteVbrTag(s->gfp,0);
130 
131  /* bit reservoir usage */
132  lame_set_disable_reservoir(s->gfp, !s->reservoir);
133 
134  /* set specified parameters */
135  if (lame_init_params(s->gfp) < 0) {
136  ret = -1;
137  goto error;
138  }
139 
140  /* get encoder delay */
141  avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
142  ff_af_queue_init(avctx, &s->afq);
143 
144  avctx->frame_size = lame_get_framesize(s->gfp);
145 
146  /* allocate float sample buffers */
147  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
148  int ch;
149  for (ch = 0; ch < avctx->channels; ch++) {
151  sizeof(*s->samples_flt[ch]));
152  if (!s->samples_flt[ch]) {
153  ret = AVERROR(ENOMEM);
154  goto error;
155  }
156  }
157  }
158 
159  ret = realloc_buffer(s);
160  if (ret < 0)
161  goto error;
162 
164  if (!s->fdsp) {
165  ret = AVERROR(ENOMEM);
166  goto error;
167  }
168 
169 
170  return 0;
171 error:
172  mp3lame_encode_close(avctx);
173  return ret;
174 }
175 
176 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
177  lame_result = func(s->gfp, \
178  (const buf_type *)buf_name[0], \
179  (const buf_type *)buf_name[1], frame->nb_samples, \
180  s->buffer + s->buffer_index, \
181  s->buffer_size - s->buffer_index); \
182 } while (0)
183 
184 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
185  const AVFrame *frame, int *got_packet_ptr)
186 {
187  LAMEContext *s = avctx->priv_data;
188  MPADecodeHeader hdr;
189  int len, ret, ch, discard_padding;
190  int lame_result;
191  uint32_t h;
192 
193  if (frame) {
194  switch (avctx->sample_fmt) {
195  case AV_SAMPLE_FMT_S16P:
196  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
197  break;
198  case AV_SAMPLE_FMT_S32P:
199  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
200  break;
201  case AV_SAMPLE_FMT_FLTP:
202  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
203  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
204  return AVERROR(EINVAL);
205  }
206  for (ch = 0; ch < avctx->channels; ch++) {
208  (const float *)frame->data[ch],
209  32768.0f,
210  FFALIGN(frame->nb_samples, 8));
211  }
212  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
213  break;
214  default:
215  return AVERROR_BUG;
216  }
217  } else if (!s->afq.frame_alloc) {
218  lame_result = 0;
219  } else {
220  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
221  s->buffer_size - s->buffer_index);
222  }
223  if (lame_result < 0) {
224  if (lame_result == -1) {
225  av_log(avctx, AV_LOG_ERROR,
226  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
228  }
229  return -1;
230  }
231  s->buffer_index += lame_result;
232  ret = realloc_buffer(s);
233  if (ret < 0) {
234  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
235  return ret;
236  }
237 
238  /* add current frame to the queue */
239  if (frame) {
240  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
241  return ret;
242  }
243 
244  /* Move 1 frame from the LAME buffer to the output packet, if available.
245  We have to parse the first frame header in the output buffer to
246  determine the frame size. */
247  if (s->buffer_index < 4)
248  return 0;
249  h = AV_RB32(s->buffer);
250 
251  ret = avpriv_mpegaudio_decode_header(&hdr, h);
252  if (ret < 0) {
253  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
254  return AVERROR_BUG;
255  } else if (ret) {
256  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
257  return -1;
258  }
259  len = hdr.frame_size;
260  ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
261  s->buffer_index);
262  if (len <= s->buffer_index) {
263  if ((ret = ff_alloc_packet2(avctx, avpkt, len, 0)) < 0)
264  return ret;
265  memcpy(avpkt->data, s->buffer, len);
266  s->buffer_index -= len;
267  memmove(s->buffer, s->buffer + len, s->buffer_index);
268 
269  /* Get the next frame pts/duration */
270  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
271  &avpkt->duration);
272 
273  discard_padding = avctx->frame_size - avpkt->duration;
274  // Check if subtraction resulted in an overflow
275  if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
276  av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
277  av_packet_unref(avpkt);
278  av_free(avpkt);
279  return AVERROR(EINVAL);
280  }
281  if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
282  uint8_t* side_data = av_packet_new_side_data(avpkt,
284  10);
285  if(!side_data) {
286  av_packet_unref(avpkt);
287  av_free(avpkt);
288  return AVERROR(ENOMEM);
289  }
290  if (!s->delay_sent) {
291  AV_WL32(side_data, avctx->initial_padding);
292  s->delay_sent = 1;
293  }
294  AV_WL32(side_data + 4, discard_padding);
295  }
296 
297  avpkt->size = len;
298  *got_packet_ptr = 1;
299  }
300  return 0;
301 }
302 
303 #define OFFSET(x) offsetof(LAMEContext, x)
304 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
305 static const AVOption options[] = {
306  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
307  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
308  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
309  { NULL },
310 };
311 
312 static const AVClass libmp3lame_class = {
313  .class_name = "libmp3lame encoder",
314  .item_name = av_default_item_name,
315  .option = options,
316  .version = LIBAVUTIL_VERSION_INT,
317 };
318 
320  { "b", "0" },
321  { NULL },
322 };
323 
324 static const int libmp3lame_sample_rates[] = {
325  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
326 };
327 
329  .name = "libmp3lame",
330  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
331  .type = AVMEDIA_TYPE_AUDIO,
332  .id = AV_CODEC_ID_MP3,
333  .priv_data_size = sizeof(LAMEContext),
335  .encode2 = mp3lame_encode_frame,
336  .close = mp3lame_encode_close,
338  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
342  .supported_samplerates = libmp3lame_sample_rates,
343  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
345  0 },
346  .priv_class = &libmp3lame_class,
347  .defaults = libmp3lame_defaults,
348 };
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:184
float, planar
Definition: samplefmt.h:69
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
static const AVClass libmp3lame_class
Definition: libmp3lame.c:312
#define NULL
Definition: coverity.c:32
AVFloatDSPContext * fdsp
Definition: libmp3lame.c:56
const char * s
Definition: avisynth_c.h:768
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:1764
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
AVOption.
Definition: opt.h:246
#define JOINT_STEREO
Definition: atrac3.c:51
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1741
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
Definition: libmp3lame.c:91
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AudioFrameQueue afq
Definition: libmp3lame.c:55
int size
Definition: avcodec.h:1602
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:324
AVCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:328
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:3601
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
Definition: libmp3lame.c:76
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
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:984
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2446
uint8_t
#define av_cold
Definition: attributes.h:82
AVOptions.
int buffer_size
Definition: libmp3lame.c:49
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1619
#define BUFFER_SIZE
Definition: libmp3lame.c:41
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
#define AE
Definition: libmp3lame.c:304
static AVFrame * frame
int reservoir
Definition: libmp3lame.c:50
uint8_t * data
Definition: avcodec.h:1601
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
#define ff_dlog(a,...)
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
uint8_t * buffer
Definition: libmp3lame.c:47
av_default_item_name
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
int initial_padding
Audio only.
Definition: avcodec.h:3367
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:517
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1771
const char * name
Name of the codec implementation.
Definition: avcodec.h:3608
static const AVOption options[]
Definition: libmp3lame.c:305
static const AVCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:319
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
static void error(const char *err)
#define STEREO
Definition: atrac3.c:52
static int realloc_buffer(LAMEContext *s)
Definition: libmp3lame.c:60
audio channel layout utility functions
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:886
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:833
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:989
signed 32 bits, planar
Definition: samplefmt.h:68
int32_t
int joint_stereo
Definition: libmp3lame.c:51
int buffer_index
Definition: libmp3lame.c:48
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:69
AVCodecContext * avctx
Definition: libmp3lame.c:45
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2458
int frame_size
Definition: mxfenc.c:1820
int av_reallocp(void *ptr, size_t size)
Allocate, reallocate, or free a block of memory through a pointer to a pointer.
Definition: mem.c:187
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int compression_level
Definition: avcodec.h:1763
int sample_rate
samples per second
Definition: avcodec.h:2438
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:215
main external API structure.
Definition: avcodec.h:1676
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:568
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
Describe the class of an AVClass context structure.
Definition: log.h:67
#define MONO
Definition: cook.c:60
int delay_sent
Definition: libmp3lame.c:53
Recommmends skipping the specified number of samples.
Definition: avcodec.h:1473
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:119
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1722
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1757
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:198
MPEG Audio header decoder.
common internal api header.
common internal and external API header
mpeg audio declarations for both encoder and decoder.
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:176
void * priv_data
Definition: avcodec.h:1718
#define av_free(p)
float * samples_flt[2]
Definition: libmp3lame.c:54
int len
int channels
number of audio channels
Definition: avcodec.h:2439
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:221
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define OFFSET(x)
Definition: libmp3lame.c:303
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
#define av_malloc_array(a, b)
lame_global_flags * gfp
Definition: libmp3lame.c:46
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
Definition: avpacket.c:318
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1578
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1594
#define AV_WL32(p, v)
Definition: intreadwrite.h:426