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af_amerge.c
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1 /*
2  * Copyright (c) 2011 Nicolas George <nicolas.george@normalesup.org>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14  * GNU Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 /**
22  * @file
23  * Audio merging filter
24  */
25 
26 #include "libavutil/avstring.h"
27 #include "libavutil/bprint.h"
29 #include "libavutil/opt.h"
30 #include "avfilter.h"
31 #include "audio.h"
32 #include "bufferqueue.h"
33 #include "internal.h"
34 
35 #define SWR_CH_MAX 64
36 
37 typedef struct {
38  const AVClass *class;
39  int nb_inputs;
40  int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
41  int bps;
42  struct amerge_input {
43  struct FFBufQueue queue;
44  int nb_ch; /**< number of channels for the input */
46  int pos;
47  } *in;
49 
50 #define OFFSET(x) offsetof(AMergeContext, x)
51 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
52 
53 static const AVOption amerge_options[] = {
54  { "inputs", "specify the number of inputs", OFFSET(nb_inputs),
55  AV_OPT_TYPE_INT, { .i64 = 2 }, 1, SWR_CH_MAX, FLAGS },
56  { NULL }
57 };
58 
59 AVFILTER_DEFINE_CLASS(amerge);
60 
62 {
63  AMergeContext *s = ctx->priv;
64  int i;
65 
66  for (i = 0; i < s->nb_inputs; i++) {
67  if (s->in)
69  if (ctx->input_pads)
70  av_freep(&ctx->input_pads[i].name);
71  }
72  av_freep(&s->in);
73 }
74 
76 {
77  AMergeContext *s = ctx->priv;
78  int64_t inlayout[SWR_CH_MAX], outlayout = 0;
81  int i, ret, overlap = 0, nb_ch = 0;
82 
83  for (i = 0; i < s->nb_inputs; i++) {
84  if (!ctx->inputs[i]->in_channel_layouts ||
87  "No channel layout for input %d\n", i + 1);
88  return AVERROR(EAGAIN);
89  }
90  inlayout[i] = ctx->inputs[i]->in_channel_layouts->channel_layouts[0];
91  if (ctx->inputs[i]->in_channel_layouts->nb_channel_layouts > 1) {
92  char buf[256];
93  av_get_channel_layout_string(buf, sizeof(buf), 0, inlayout[i]);
94  av_log(ctx, AV_LOG_INFO, "Using \"%s\" for input %d\n", buf, i + 1);
95  }
96  s->in[i].nb_ch = av_get_channel_layout_nb_channels(inlayout[i]);
97  if (outlayout & inlayout[i])
98  overlap++;
99  outlayout |= inlayout[i];
100  nb_ch += s->in[i].nb_ch;
101  }
102  if (nb_ch > SWR_CH_MAX) {
103  av_log(ctx, AV_LOG_ERROR, "Too many channels (max %d)\n", SWR_CH_MAX);
104  return AVERROR(EINVAL);
105  }
106  if (overlap) {
107  av_log(ctx, AV_LOG_WARNING,
108  "Input channel layouts overlap: "
109  "output layout will be determined by the number of distinct input channels\n");
110  for (i = 0; i < nb_ch; i++)
111  s->route[i] = i;
112  outlayout = av_get_default_channel_layout(nb_ch);
113  if (!outlayout && nb_ch)
114  outlayout = 0xFFFFFFFFFFFFFFFFULL >> (64 - nb_ch);
115  } else {
116  int *route[SWR_CH_MAX];
117  int c, out_ch_number = 0;
118 
119  route[0] = s->route;
120  for (i = 1; i < s->nb_inputs; i++)
121  route[i] = route[i - 1] + s->in[i - 1].nb_ch;
122  for (c = 0; c < 64; c++)
123  for (i = 0; i < s->nb_inputs; i++)
124  if ((inlayout[i] >> c) & 1)
125  *(route[i]++) = out_ch_number++;
126  }
128  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
129  return ret;
130  for (i = 0; i < s->nb_inputs; i++) {
131  layouts = NULL;
132  if ((ret = ff_add_channel_layout(&layouts, inlayout[i])) < 0)
133  return ret;
134  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
135  return ret;
136  }
137  layouts = NULL;
138  if ((ret = ff_add_channel_layout(&layouts, outlayout)) < 0)
139  return ret;
140  if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
141  return ret;
142 
144 }
145 
146 static int config_output(AVFilterLink *outlink)
147 {
148  AVFilterContext *ctx = outlink->src;
149  AMergeContext *s = ctx->priv;
150  AVBPrint bp;
151  int i;
152 
153  for (i = 1; i < s->nb_inputs; i++) {
154  if (ctx->inputs[i]->sample_rate != ctx->inputs[0]->sample_rate) {
155  av_log(ctx, AV_LOG_ERROR,
156  "Inputs must have the same sample rate "
157  "%d for in%d vs %d\n",
158  ctx->inputs[i]->sample_rate, i, ctx->inputs[0]->sample_rate);
159  return AVERROR(EINVAL);
160  }
161  }
162  s->bps = av_get_bytes_per_sample(ctx->outputs[0]->format);
163  outlink->sample_rate = ctx->inputs[0]->sample_rate;
164  outlink->time_base = ctx->inputs[0]->time_base;
165 
166  av_bprint_init(&bp, 0, 1);
167  for (i = 0; i < s->nb_inputs; i++) {
168  av_bprintf(&bp, "%sin%d:", i ? " + " : "", i);
170  }
171  av_bprintf(&bp, " -> out:");
173  av_log(ctx, AV_LOG_VERBOSE, "%s\n", bp.str);
174 
175  return 0;
176 }
177 
178 static int request_frame(AVFilterLink *outlink)
179 {
180  AVFilterContext *ctx = outlink->src;
181  AMergeContext *s = ctx->priv;
182  int i, ret;
183 
184  for (i = 0; i < s->nb_inputs; i++)
185  if (!s->in[i].nb_samples)
186  if ((ret = ff_request_frame(ctx->inputs[i])) < 0)
187  return ret;
188  return 0;
189 }
190 
191 /**
192  * Copy samples from several input streams to one output stream.
193  * @param nb_inputs number of inputs
194  * @param in inputs; used only for the nb_ch field;
195  * @param route routing values;
196  * input channel i goes to output channel route[i];
197  * i < in[0].nb_ch are the channels from the first output;
198  * i >= in[0].nb_ch are the channels from the second output
199  * @param ins pointer to the samples of each inputs, in packed format;
200  * will be left at the end of the copied samples
201  * @param outs pointer to the samples of the output, in packet format;
202  * must point to a buffer big enough;
203  * will be left at the end of the copied samples
204  * @param ns number of samples to copy
205  * @param bps bytes per sample
206  */
207 static inline void copy_samples(int nb_inputs, struct amerge_input in[],
208  int *route, uint8_t *ins[],
209  uint8_t **outs, int ns, int bps)
210 {
211  int *route_cur;
212  int i, c, nb_ch = 0;
213 
214  for (i = 0; i < nb_inputs; i++)
215  nb_ch += in[i].nb_ch;
216  while (ns--) {
217  route_cur = route;
218  for (i = 0; i < nb_inputs; i++) {
219  for (c = 0; c < in[i].nb_ch; c++) {
220  memcpy((*outs) + bps * *(route_cur++), ins[i], bps);
221  ins[i] += bps;
222  }
223  }
224  *outs += nb_ch * bps;
225  }
226 }
227 
228 static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
229 {
230  AVFilterContext *ctx = inlink->dst;
231  AMergeContext *s = ctx->priv;
232  AVFilterLink *const outlink = ctx->outputs[0];
233  int input_number;
234  int nb_samples, ns, i;
235  AVFrame *outbuf, *inbuf[SWR_CH_MAX];
236  uint8_t *ins[SWR_CH_MAX], *outs;
237 
238  for (input_number = 0; input_number < s->nb_inputs; input_number++)
239  if (inlink == ctx->inputs[input_number])
240  break;
241  av_assert1(input_number < s->nb_inputs);
242  if (ff_bufqueue_is_full(&s->in[input_number].queue)) {
243  av_frame_free(&insamples);
244  return AVERROR(ENOMEM);
245  }
246  ff_bufqueue_add(ctx, &s->in[input_number].queue, av_frame_clone(insamples));
247  s->in[input_number].nb_samples += insamples->nb_samples;
248  av_frame_free(&insamples);
249  nb_samples = s->in[0].nb_samples;
250  for (i = 1; i < s->nb_inputs; i++)
251  nb_samples = FFMIN(nb_samples, s->in[i].nb_samples);
252  if (!nb_samples)
253  return 0;
254 
255  outbuf = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
256  if (!outbuf)
257  return AVERROR(ENOMEM);
258  outs = outbuf->data[0];
259  for (i = 0; i < s->nb_inputs; i++) {
260  inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0);
261  ins[i] = inbuf[i]->data[0] +
262  s->in[i].pos * s->in[i].nb_ch * s->bps;
263  }
264  av_frame_copy_props(outbuf, inbuf[0]);
265  outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
266  inbuf[0]->pts +
267  av_rescale_q(s->in[0].pos,
268  av_make_q(1, ctx->inputs[0]->sample_rate),
269  ctx->outputs[0]->time_base);
270 
271  outbuf->nb_samples = nb_samples;
272  outbuf->channel_layout = outlink->channel_layout;
273  av_frame_set_channels(outbuf, outlink->channels);
274 
275  while (nb_samples) {
276  ns = nb_samples;
277  for (i = 0; i < s->nb_inputs; i++)
278  ns = FFMIN(ns, inbuf[i]->nb_samples - s->in[i].pos);
279  /* Unroll the most common sample formats: speed +~350% for the loop,
280  +~13% overall (including two common decoders) */
281  switch (s->bps) {
282  case 1:
283  copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 1);
284  break;
285  case 2:
286  copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 2);
287  break;
288  case 4:
289  copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 4);
290  break;
291  default:
292  copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, s->bps);
293  break;
294  }
295 
296  nb_samples -= ns;
297  for (i = 0; i < s->nb_inputs; i++) {
298  s->in[i].nb_samples -= ns;
299  s->in[i].pos += ns;
300  if (s->in[i].pos == inbuf[i]->nb_samples) {
301  s->in[i].pos = 0;
302  av_frame_free(&inbuf[i]);
303  ff_bufqueue_get(&s->in[i].queue);
304  inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0);
305  ins[i] = inbuf[i] ? inbuf[i]->data[0] : NULL;
306  }
307  }
308  }
309  return ff_filter_frame(ctx->outputs[0], outbuf);
310 }
311 
313 {
314  AMergeContext *s = ctx->priv;
315  int i;
316 
317  s->in = av_calloc(s->nb_inputs, sizeof(*s->in));
318  if (!s->in)
319  return AVERROR(ENOMEM);
320  for (i = 0; i < s->nb_inputs; i++) {
321  char *name = av_asprintf("in%d", i);
322  AVFilterPad pad = {
323  .name = name,
324  .type = AVMEDIA_TYPE_AUDIO,
325  .filter_frame = filter_frame,
326  };
327  if (!name)
328  return AVERROR(ENOMEM);
329  ff_insert_inpad(ctx, i, &pad);
330  }
331  return 0;
332 }
333 
334 static const AVFilterPad amerge_outputs[] = {
335  {
336  .name = "default",
337  .type = AVMEDIA_TYPE_AUDIO,
338  .config_props = config_output,
339  .request_frame = request_frame,
340  },
341  { NULL }
342 };
343 
345  .name = "amerge",
346  .description = NULL_IF_CONFIG_SMALL("Merge two or more audio streams into "
347  "a single multi-channel stream."),
348  .priv_size = sizeof(AMergeContext),
349  .init = init,
350  .uninit = uninit,
352  .inputs = NULL,
353  .outputs = amerge_outputs,
354  .priv_class = &amerge_class,
356 };
void av_frame_set_channels(AVFrame *frame, int val)
static AVFrame * ff_bufqueue_get(struct FFBufQueue *queue)
Get the first buffer from the queue and remove it.
Definition: bufferqueue.h:98
struct AMergeContext::amerge_input * in
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:768
void av_bprintf(AVBPrint *buf, const char *fmt,...)
Definition: bprint.c:94
int nb_ch
number of channels for the input
Definition: af_amerge.c:44
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
AVOption.
Definition: opt.h:245
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
Main libavfilter public API header.
#define SWR_CH_MAX
Definition: af_amerge.c:35
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
Definition: avfilter.h:105
struct FFBufQueue queue
Definition: af_amerge.c:43
static enum AVSampleFormat formats[]
Definition: avresample.c:163
AVFILTER_DEFINE_CLASS(amerge)
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:260
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
Structure holding the queue.
Definition: bufferqueue.h:49
const char * name
Pad name.
Definition: internal.h:59
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:315
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:435
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1189
uint8_t
#define av_cold
Definition: attributes.h:82
AVOptions.
static enum AVSampleFormat ff_packed_sample_fmts_array[]
Definition: audio.h:28
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:268
static av_cold int init(AVFilterContext *ctx)
Definition: af_amerge.c:312
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
#define av_log(a,...)
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
Definition: af_amerge.c:228
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amerge.c:61
A filter pad used for either input or output.
Definition: internal.h:53
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
static const AVFilterPad amerge_outputs[]
Definition: af_amerge.c:334
AVFilterPad * input_pads
array of input pads
Definition: avfilter.h:314
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:568
static const AVOption amerge_options[]
Definition: af_amerge.c:53
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:343
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:65
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:154
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
void av_bprint_init(AVBPrint *buf, unsigned size_init, unsigned size_max)
Definition: bprint.c:69
void * priv
private data for use by the filter
Definition: avfilter.h:322
uint64_t * channel_layouts
list of channel layouts
Definition: formats.h:86
uint64_t channel_layout
Channel layout of the audio data.
Definition: frame.h:349
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:113
AVFrame * queue[FF_BUFQUEUE_SIZE]
Definition: bufferqueue.h:50
static int request_frame(AVFilterLink *outlink)
Definition: af_amerge.c:178
static int config_output(AVFilterLink *outlink)
Definition: af_amerge.c:146
audio channel layout utility functions
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
#define FFMIN(a, b)
Definition: common.h:96
static int ff_bufqueue_is_full(struct FFBufQueue *queue)
Test if a buffer queue is full.
Definition: bufferqueue.h:60
AVFormatContext * ctx
Definition: movenc.c:48
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:386
static void ff_bufqueue_discard_all(struct FFBufQueue *queue)
Unref and remove all buffers from the queue.
Definition: bufferqueue.h:111
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
Definition: frame.c:472
A list of supported channel layouts.
Definition: formats.h:85
void av_bprint_channel_layout(struct AVBPrint *bp, int nb_channels, uint64_t channel_layout)
Append a description of a channel layout to a bprint buffer.
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
#define FLAGS
Definition: af_amerge.c:51
static int query_formats(AVFilterContext *ctx)
Definition: af_amerge.c:75
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:376
static AVRational av_make_q(int num, int den)
Create an AVRational.
Definition: rational.h:71
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
AVFilter ff_af_amerge
Definition: af_amerge.c:344
void * buf
Definition: avisynth_c.h:690
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:319
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
static int flags
Definition: cpu.c:47
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:198
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
int nb_channel_layouts
number of channel layouts
Definition: formats.h:87
#define OFFSET(x)
Definition: af_amerge.c:50
static double c[64]
unsigned bps
Definition: movenc.c:1379
int route[SWR_CH_MAX]
channels routing, see copy_samples
Definition: af_amerge.c:40
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:307
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
static void copy_samples(int nb_inputs, struct amerge_input in[], int *route, uint8_t *ins[], uint8_t **outs, int ns, int bps)
Copy samples from several input streams to one output stream.
Definition: af_amerge.c:207
#define av_freep(p)
static void ff_bufqueue_add(void *log, struct FFBufQueue *queue, AVFrame *buf)
Add a buffer to the queue.
Definition: bufferqueue.h:71
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:369
internal API functions
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:581
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:242
const char * name
Definition: opengl_enc.c:103
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
Definition: internal.h:283
static AVFrame * ff_bufqueue_peek(struct FFBufQueue *queue, unsigned index)
Get a buffer from the queue without altering it.
Definition: bufferqueue.h:87