FFmpeg
libmp3lame.c
Go to the documentation of this file.
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  /* lowpass cutoff frequency */
129  if (avctx->cutoff)
130  lame_set_lowpassfreq(s->gfp, avctx->cutoff);
131 
132  /* do not get a Xing VBR header frame from LAME */
133  lame_set_bWriteVbrTag(s->gfp,0);
134 
135  /* bit reservoir usage */
136  lame_set_disable_reservoir(s->gfp, !s->reservoir);
137 
138  /* set specified parameters */
139  if (lame_init_params(s->gfp) < 0) {
140  ret = -1;
141  goto error;
142  }
143 
144  /* get encoder delay */
145  avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
146  ff_af_queue_init(avctx, &s->afq);
147 
148  avctx->frame_size = lame_get_framesize(s->gfp);
149 
150  /* allocate float sample buffers */
151  if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
152  int ch;
153  for (ch = 0; ch < avctx->channels; ch++) {
155  sizeof(*s->samples_flt[ch]));
156  if (!s->samples_flt[ch]) {
157  ret = AVERROR(ENOMEM);
158  goto error;
159  }
160  }
161  }
162 
163  ret = realloc_buffer(s);
164  if (ret < 0)
165  goto error;
166 
168  if (!s->fdsp) {
169  ret = AVERROR(ENOMEM);
170  goto error;
171  }
172 
173 
174  return 0;
175 error:
176  mp3lame_encode_close(avctx);
177  return ret;
178 }
179 
180 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
181  lame_result = func(s->gfp, \
182  (const buf_type *)buf_name[0], \
183  (const buf_type *)buf_name[1], frame->nb_samples, \
184  s->buffer + s->buffer_index, \
185  s->buffer_size - s->buffer_index); \
186 } while (0)
187 
189  const AVFrame *frame, int *got_packet_ptr)
190 {
191  LAMEContext *s = avctx->priv_data;
192  MPADecodeHeader hdr;
193  int len, ret, ch, discard_padding;
194  int lame_result;
195  uint32_t h;
196 
197  if (frame) {
198  switch (avctx->sample_fmt) {
199  case AV_SAMPLE_FMT_S16P:
200  ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
201  break;
202  case AV_SAMPLE_FMT_S32P:
203  ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
204  break;
205  case AV_SAMPLE_FMT_FLTP:
206  if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
207  av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
208  return AVERROR(EINVAL);
209  }
210  for (ch = 0; ch < avctx->channels; ch++) {
212  (const float *)frame->data[ch],
213  32768.0f,
214  FFALIGN(frame->nb_samples, 8));
215  }
216  ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
217  break;
218  default:
219  return AVERROR_BUG;
220  }
221  } else if (!s->afq.frame_alloc) {
222  lame_result = 0;
223  } else {
224  lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
225  s->buffer_size - s->buffer_index);
226  }
227  if (lame_result < 0) {
228  if (lame_result == -1) {
229  av_log(avctx, AV_LOG_ERROR,
230  "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
232  }
233  return -1;
234  }
235  s->buffer_index += lame_result;
236  ret = realloc_buffer(s);
237  if (ret < 0) {
238  av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
239  return ret;
240  }
241 
242  /* add current frame to the queue */
243  if (frame) {
244  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
245  return ret;
246  }
247 
248  /* Move 1 frame from the LAME buffer to the output packet, if available.
249  We have to parse the first frame header in the output buffer to
250  determine the frame size. */
251  if (s->buffer_index < 4)
252  return 0;
253  h = AV_RB32(s->buffer);
254 
255  ret = avpriv_mpegaudio_decode_header(&hdr, h);
256  if (ret < 0) {
257  av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
258  return AVERROR_BUG;
259  } else if (ret) {
260  av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
261  return -1;
262  }
263  len = hdr.frame_size;
264  ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
265  s->buffer_index);
266  if (len <= s->buffer_index) {
267  if ((ret = ff_alloc_packet2(avctx, avpkt, len, 0)) < 0)
268  return ret;
269  memcpy(avpkt->data, s->buffer, len);
270  s->buffer_index -= len;
271  memmove(s->buffer, s->buffer + len, s->buffer_index);
272 
273  /* Get the next frame pts/duration */
274  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
275  &avpkt->duration);
276 
277  discard_padding = avctx->frame_size - avpkt->duration;
278  // Check if subtraction resulted in an overflow
279  if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
280  av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
281  av_packet_unref(avpkt);
282  av_free(avpkt);
283  return AVERROR(EINVAL);
284  }
285  if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
286  uint8_t* side_data = av_packet_new_side_data(avpkt,
288  10);
289  if(!side_data) {
290  av_packet_unref(avpkt);
291  av_free(avpkt);
292  return AVERROR(ENOMEM);
293  }
294  if (!s->delay_sent) {
295  AV_WL32(side_data, avctx->initial_padding);
296  s->delay_sent = 1;
297  }
298  AV_WL32(side_data + 4, discard_padding);
299  }
300 
301  avpkt->size = len;
302  *got_packet_ptr = 1;
303  }
304  return 0;
305 }
306 
307 #define OFFSET(x) offsetof(LAMEContext, x)
308 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
309 static const AVOption options[] = {
310  { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
311  { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
312  { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
313  { NULL },
314 };
315 
316 static const AVClass libmp3lame_class = {
317  .class_name = "libmp3lame encoder",
318  .item_name = av_default_item_name,
319  .option = options,
320  .version = LIBAVUTIL_VERSION_INT,
321 };
322 
324  { "b", "0" },
325  { NULL },
326 };
327 
328 static const int libmp3lame_sample_rates[] = {
329  44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
330 };
331 
333  .name = "libmp3lame",
334  .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
335  .type = AVMEDIA_TYPE_AUDIO,
336  .id = AV_CODEC_ID_MP3,
337  .priv_data_size = sizeof(LAMEContext),
339  .encode2 = mp3lame_encode_frame,
340  .close = mp3lame_encode_close,
342  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
346  .supported_samplerates = libmp3lame_sample_rates,
347  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
349  0 },
350  .priv_class = &libmp3lame_class,
351  .defaults = libmp3lame_defaults,
352  .wrapper_name = "libmp3lame",
353 };
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libmp3lame.c:188
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:316
#define NULL
Definition: coverity.c:32
AVFloatDSPContext * fdsp
Definition: libmp3lame.c:56
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:1638
This structure describes decoded (raw) audio or video data.
Definition: frame.h:268
AVOption.
Definition: opt.h:246
#define JOINT_STEREO
Definition: atrac3.c:55
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1615
#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:1478
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&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);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=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){int planes=out->planar?out->ch_count:1;unsigned m=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){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
static const int libmp3lame_sample_rates[]
Definition: libmp3lame.c:328
AVCodec ff_libmp3lame_encoder
Definition: libmp3lame.c:332
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:3477
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:1006
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
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:1495
#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:308
int reservoir
Definition: libmp3lame.c:50
uint8_t * data
Definition: avcodec.h:1477
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
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
uint8_t * buffer
Definition: libmp3lame.c:47
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
int initial_padding
Audio only.
Definition: avcodec.h:3092
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:565
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1645
const char * name
Name of the codec implementation.
Definition: avcodec.h:3484
static const AVOption options[]
Definition: libmp3lame.c:309
static const AVCodecDefault libmp3lame_defaults[]
Definition: libmp3lame.c:323
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
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:908
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:850
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:1011
signed 32 bits, planar
Definition: samplefmt.h:68
int32_t
int joint_stereo
Definition: libmp3lame.c:51
int buffer_index
Definition: libmp3lame.c:48
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
AVCodecContext * avctx
Definition: libmp3lame.c:45
static void error(const char *err)
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2241
int frame_size
Definition: mxfenc.c:2216
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:163
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int compression_level
Definition: avcodec.h:1637
int sample_rate
samples per second
Definition: avcodec.h:2221
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:299
main external API structure.
Definition: avcodec.h:1565
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:599
#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:1300
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:85
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1631
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:282
MPEG Audio header decoder.
common internal api header.
#define STEREO
Definition: cook.c:61
common internal and external API header
mpeg audio declarations for both encoder and decoder.
#define ENCODE_BUFFER(func, buf_type, buf_name)
Definition: libmp3lame.c:180
void * priv_data
Definition: avcodec.h:1592
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:2265
#define av_free(p)
float * samples_flt[2]
Definition: libmp3lame.c:54
int len
int channels
number of audio channels
Definition: avcodec.h:2222
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:227
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define OFFSET(x)
Definition: libmp3lame.c:307
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:329
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
#define AV_CH_LAYOUT_MONO
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
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1470
#define AV_WL32(p, v)
Definition: intreadwrite.h:426