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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  /* 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 
188 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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:1606
This structure describes decoded (raw) audio or video data.
Definition: frame.h:226
AVOption.
Definition: opt.h:246
#define JOINT_STEREO
Definition: atrac3.c:55
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1583
#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:1446
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:3424
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:993
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:2197
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:1463
#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
static AVFrame * frame
int reservoir
Definition: libmp3lame.c:50
uint8_t * data
Definition: avcodec.h:1445
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 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:186
int initial_padding
Audio only.
Definition: avcodec.h:3047
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:559
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1613
const char * name
Name of the codec implementation.
Definition: avcodec.h:3431
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:895
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:842
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:998
signed 32 bits, planar
Definition: samplefmt.h:68
int32_t
int joint_stereo
Definition: libmp3lame.c:51
int buffer_index
Definition: libmp3lame.c:48
#define s(width, name)
Definition: cbs_vp9.c:257
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
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:2209
int frame_size
Definition: mxfenc.c:2092
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:1605
int sample_rate
samples per second
Definition: avcodec.h:2189
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:257
main external API structure.
Definition: avcodec.h:1533
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:598
#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:1268
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1599
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:240
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:1560
int cutoff
Audio cutoff bandwidth (0 means "automatic")
Definition: avcodec.h:2233
#define av_free(p)
float * samples_flt[2]
Definition: libmp3lame.c:54
int len
int channels
number of audio channels
Definition: avcodec.h:2190
#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
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1422
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:292
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1438
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
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_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);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