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1 /*
2  * Copyright (c) 2011 Stefano Sabatini
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
14  * 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  */
21 /**
22  * @file
23  * filter for showing textual audio frame information
24  */
26 #include <inttypes.h>
27 #include <stddef.h>
29 #include "libavutil/adler32.h"
30 #include "libavutil/attributes.h"
32 #include "libavutil/common.h"
33 #include "libavutil/downmix_info.h"
34 #include "libavutil/intreadwrite.h"
35 #include "libavutil/mem.h"
36 #include "libavutil/replaygain.h"
37 #include "libavutil/timestamp.h"
38 #include "libavutil/samplefmt.h"
40 #include "libavcodec/avcodec.h"
42 #include "audio.h"
43 #include "avfilter.h"
44 #include "internal.h"
46 typedef struct AShowInfoContext {
47  /**
48  * Scratch space for individual plane checksums for planar audio
49  */
50  uint32_t *plane_checksums;
54 {
55  AShowInfoContext *s = ctx->priv;
57 }
60 {
61  enum AVMatrixEncoding enc;
63  av_log(ctx, AV_LOG_INFO, "matrix encoding: ");
65  if (sd->size < sizeof(enum AVMatrixEncoding)) {
66  av_log(ctx, AV_LOG_INFO, "invalid data");
67  return;
68  }
70  enc = *(enum AVMatrixEncoding *)sd->data;
71  switch (enc) {
72  case AV_MATRIX_ENCODING_NONE: av_log(ctx, AV_LOG_INFO, "none"); break;
73  case AV_MATRIX_ENCODING_DOLBY: av_log(ctx, AV_LOG_INFO, "Dolby Surround"); break;
74  case AV_MATRIX_ENCODING_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
75  case AV_MATRIX_ENCODING_DPLIIX: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIx"); break;
76  case AV_MATRIX_ENCODING_DPLIIZ: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIz"); break;
77  case AV_MATRIX_ENCODING_DOLBYEX: av_log(ctx, AV_LOG_INFO, "Dolby EX"); break;
78  case AV_MATRIX_ENCODING_DOLBYHEADPHONE: av_log(ctx, AV_LOG_INFO, "Dolby Headphone"); break;
79  default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
80  }
81 }
84 {
85  AVDownmixInfo *di;
87  av_log(ctx, AV_LOG_INFO, "downmix: ");
88  if (sd->size < sizeof(*di)) {
89  av_log(ctx, AV_LOG_INFO, "invalid data");
90  return;
91  }
93  di = (AVDownmixInfo *)sd->data;
95  av_log(ctx, AV_LOG_INFO, "preferred downmix type - ");
96  switch (di->preferred_downmix_type) {
97  case AV_DOWNMIX_TYPE_LORO: av_log(ctx, AV_LOG_INFO, "Lo/Ro"); break;
98  case AV_DOWNMIX_TYPE_LTRT: av_log(ctx, AV_LOG_INFO, "Lt/Rt"); break;
99  case AV_DOWNMIX_TYPE_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
100  default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
101  }
103  av_log(ctx, AV_LOG_INFO, " Mix levels: center %f (%f ltrt) - "
104  "surround %f (%f ltrt) - lfe %f",
107  di->lfe_mix_level);
108 }
110 static void print_gain(AVFilterContext *ctx, const char *str, int32_t gain)
111 {
112  av_log(ctx, AV_LOG_INFO, "%s - ", str);
113  if (gain == INT32_MIN)
114  av_log(ctx, AV_LOG_INFO, "unknown");
115  else
116  av_log(ctx, AV_LOG_INFO, "%f", gain / 100000.0f);
117  av_log(ctx, AV_LOG_INFO, ", ");
118 }
120 static void print_peak(AVFilterContext *ctx, const char *str, uint32_t peak)
121 {
122  av_log(ctx, AV_LOG_INFO, "%s - ", str);
123  if (!peak)
124  av_log(ctx, AV_LOG_INFO, "unknown");
125  else
126  av_log(ctx, AV_LOG_INFO, "%f", (float)peak / UINT32_MAX);
127  av_log(ctx, AV_LOG_INFO, ", ");
128 }
131 {
132  AVReplayGain *rg;
134  av_log(ctx, AV_LOG_INFO, "replaygain: ");
135  if (sd->size < sizeof(*rg)) {
136  av_log(ctx, AV_LOG_INFO, "invalid data");
137  return;
138  }
139  rg = (AVReplayGain*)sd->data;
141  print_gain(ctx, "track gain", rg->track_gain);
142  print_peak(ctx, "track peak", rg->track_peak);
143  print_gain(ctx, "album gain", rg->album_gain);
144  print_peak(ctx, "album peak", rg->album_peak);
145 }
148 {
149  enum AVAudioServiceType *ast;
151  av_log(ctx, AV_LOG_INFO, "audio service type: ");
152  if (sd->size < sizeof(*ast)) {
153  av_log(ctx, AV_LOG_INFO, "invalid data");
154  return;
155  }
156  ast = (enum AVAudioServiceType*)sd->data;
157  switch (*ast) {
158  case AV_AUDIO_SERVICE_TYPE_MAIN: av_log(ctx, AV_LOG_INFO, "Main Audio Service"); break;
159  case AV_AUDIO_SERVICE_TYPE_EFFECTS: av_log(ctx, AV_LOG_INFO, "Effects"); break;
160  case AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED: av_log(ctx, AV_LOG_INFO, "Visually Impaired"); break;
161  case AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED: av_log(ctx, AV_LOG_INFO, "Hearing Impaired"); break;
162  case AV_AUDIO_SERVICE_TYPE_DIALOGUE: av_log(ctx, AV_LOG_INFO, "Dialogue"); break;
163  case AV_AUDIO_SERVICE_TYPE_COMMENTARY: av_log(ctx, AV_LOG_INFO, "Commentary"); break;
164  case AV_AUDIO_SERVICE_TYPE_EMERGENCY: av_log(ctx, AV_LOG_INFO, "Emergency"); break;
165  case AV_AUDIO_SERVICE_TYPE_VOICE_OVER: av_log(ctx, AV_LOG_INFO, "Voice Over"); break;
166  case AV_AUDIO_SERVICE_TYPE_KARAOKE: av_log(ctx, AV_LOG_INFO, "Karaoke"); break;
167  default: av_log(ctx, AV_LOG_INFO, "unknown"); break;
168  }
169 }
172 {
173  av_log(ctx, AV_LOG_INFO, "unknown side data type: %d, size %d bytes", sd->type, sd->size);
174 }
177 {
178  AVFilterContext *ctx = inlink->dst;
179  AShowInfoContext *s = ctx->priv;
180  char chlayout_str[128];
181  uint32_t checksum = 0;
182  int channels = inlink->channels;
184  int block_align = av_get_bytes_per_sample(buf->format) * (planar ? 1 : channels);
185  int data_size = buf->nb_samples * block_align;
186  int planes = planar ? channels : 1;
187  int i;
188  void *tmp_ptr = av_realloc_array(s->plane_checksums, channels, sizeof(*s->plane_checksums));
190  if (!tmp_ptr)
191  return AVERROR(ENOMEM);
192  s->plane_checksums = tmp_ptr;
194  for (i = 0; i < planes; i++) {
195  uint8_t *data = buf->extended_data[i];
197  s->plane_checksums[i] = av_adler32_update(0, data, data_size);
198  checksum = i ? av_adler32_update(checksum, data, data_size) :
199  s->plane_checksums[0];
200  }
202  av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), buf->channels,
203  buf->channel_layout);
205  av_log(ctx, AV_LOG_INFO,
206  "n:%"PRId64" pts:%s pts_time:%s pos:%"PRId64" "
207  "fmt:%s channels:%d chlayout:%s rate:%d nb_samples:%d "
208  "checksum:%08"PRIX32" ",
209  inlink->frame_count_out,
210  av_ts2str(buf->pts), av_ts2timestr(buf->pts, &inlink->time_base),
211  buf->pkt_pos,
212  av_get_sample_fmt_name(buf->format), buf->channels, chlayout_str,
213  buf->sample_rate, buf->nb_samples,
214  checksum);
216  av_log(ctx, AV_LOG_INFO, "plane_checksums: [ ");
217  for (i = 0; i < planes; i++)
218  av_log(ctx, AV_LOG_INFO, "%08"PRIX32" ", s->plane_checksums[i]);
219  av_log(ctx, AV_LOG_INFO, "]\n");
221  for (i = 0; i < buf->nb_side_data; i++) {
222  AVFrameSideData *sd = buf->side_data[i];
224  av_log(ctx, AV_LOG_INFO, " side data - ");
225  switch (sd->type) {
226  case AV_FRAME_DATA_MATRIXENCODING: dump_matrixenc (ctx, sd); break;
227  case AV_FRAME_DATA_DOWNMIX_INFO: dump_downmix (ctx, sd); break;
228  case AV_FRAME_DATA_REPLAYGAIN: dump_replaygain(ctx, sd); break;
230  default: dump_unknown (ctx, sd); break;
231  }
233  av_log(ctx, AV_LOG_INFO, "\n");
234  }
236  return ff_filter_frame(inlink->dst->outputs[0], buf);
237 }
239 static const AVFilterPad inputs[] = {
240  {
241  .name = "default",
242  .type = AVMEDIA_TYPE_AUDIO,
243  .filter_frame = filter_frame,
244  },
245  { NULL }
246 };
248 static const AVFilterPad outputs[] = {
249  {
250  .name = "default",
251  .type = AVMEDIA_TYPE_AUDIO,
252  },
253  { NULL }
254 };
257  .name = "ashowinfo",
258  .description = NULL_IF_CONFIG_SMALL("Show textual information for each audio frame."),
259  .priv_size = sizeof(AShowInfoContext),
260  .uninit = uninit,
261  .inputs = inputs,
262  .outputs = outputs,
263 };
audio downmix medatata
#define NULL
Definition: coverity.c:32
This structure describes decoded (raw) audio or video data.
Definition: frame.h:314
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
This side data must be associated with an audio frame and corresponds to enum AVAudioServiceType defi...
Definition: frame.h:113
int64_t pkt_pos
reordered pos from the last AVPacket that has been input into the decoder
Definition: frame.h:585
Something somehow does not look correct.
Definition: log.h:200
static const AVFilterPad outputs[]
Definition: af_ashowinfo.c:248
Main libavfilter public API header.
Memory handling functions.
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_S32, int32_t, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1<< 16)) 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/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(UINT64_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 *(UINT64_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 *(UINT64_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,(const uint8_t **) in->ch+ch, off *(out-> planar
Definition: audioconvert.c:56
static void dump_replaygain(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:130
double center_mix_level_ltrt
Absolute scale factor representing the nominal level of the center channel during an Lt/Rt compatible...
Definition: downmix_info.h:74
static const struct @322 planes[]
uint32_t track_peak
Peak track amplitude, with 100000 representing full scale (but values may overflow).
Definition: replaygain.h:39
Macro definitions for various function/variable attributes.
const char * name
Pad name.
Definition: internal.h:60
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1093
#define av_cold
Definition: attributes.h:88
timestamp utils, mostly useful for debugging/logging purposes
#define f(width, name)
Definition: cbs_vp9.c:255
unsigned long av_adler32_update(unsigned long adler, const uint8_t *buf, unsigned int len)
Calculate the Adler32 checksum of a buffer.
Definition: adler32.c:44
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:407
Structure to hold side data for an AVFrame.
Definition: frame.h:220
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
Definition: af_ashowinfo.c:176
double surround_mix_level_ltrt
Absolute scale factor representing the nominal level of the surround channels during an Lt/Rt compati...
Definition: downmix_info.h:86
Lt/Rt 2-channel downmix, Dolby Pro Logic II compatible.
Definition: downmix_info.h:48
Definition: aptx.h:33
Metadata relevant to a downmix procedure.
Definition: frame.h:72
int nb_side_data
Definition: frame.h:526
void * av_realloc_array(void *ptr, size_t nmemb, size_t size)
Definition: mem.c:198
int32_t album_gain
Same as track_gain, but for the whole album.
Definition: replaygain.h:43
AVFrameSideData ** side_data
Definition: frame.h:525
#define av_log(a,...)
double lfe_mix_level
Absolute scale factor representing the level at which the LFE data is mixed into L/R channels during ...
Definition: downmix_info.h:92
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
A filter pad used for either input or output.
Definition: internal.h:54
static void print_gain(AVFilterContext *ctx, const char *str, int32_t gain)
Definition: af_ashowinfo.c:110
This structure describes optional metadata relevant to a downmix procedure.
Definition: downmix_info.h:58
Definition: avcodec.h:239
#define av_ts2timestr(ts, tb)
Convenience macro, the return value should be used only directly in function arguments but never stan...
Definition: timestamp.h:76
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:115
void * priv
private data for use by the filter
Definition: avfilter.h:356
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:49
uint64_t channel_layout
Channel layout of the audio data.
Definition: frame.h:491
int channels
number of audio channels, only used for audio.
Definition: frame.h:620
audio channel layout utility functions
AVFilter ff_af_ashowinfo
Definition: af_ashowinfo.c:256
Lt/Rt 2-channel downmix, Dolby Surround compatible.
Definition: downmix_info.h:47
static void dump_downmix(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:83
static void print_peak(AVFilterContext *ctx, const char *str, uint32_t peak)
Definition: af_ashowinfo.c:120
AVFormatContext * ctx
Definition: movenc.c:48
static void dump_audio_service_type(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:147
Lo/Ro 2-channel downmix (Stereo).
Definition: downmix_info.h:46
#define s(width, name)
Definition: cbs_vp9.c:257
static volatile int checksum
Definition: adler32.c:30
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_ashowinfo.c:53
static void dump_matrixenc(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:59
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.
Public header for Adler-32 hash function implementation.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:387
uint32_t * plane_checksums
Scratch space for individual plane checksums for planar audio.
Definition: af_ashowinfo.c:50
#define AV_LOG_INFO
Standard information.
Definition: log.h:205
double surround_mix_level
Absolute scale factor representing the nominal level of the surround channels during a regular downmi...
Definition: downmix_info.h:80
Libavcodec external API header.
uint8_t * data
Definition: frame.h:222
int sample_rate
Sample rate of the audio data.
Definition: frame.h:486
Filter definition.
Definition: avfilter.h:145
static const AVFilterPad inputs[]
Definition: af_ashowinfo.c:239
const char * name
Filter name.
Definition: avfilter.h:149
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:353
enum AVFrameSideDataType type
Definition: frame.h:221
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
common internal and external API header
uint32_t album_peak
Same as track_peak, but for the whole album,.
Definition: replaygain.h:47
double center_mix_level
Absolute scale factor representing the nominal level of the center channel during a regular downmix...
Definition: downmix_info.h:68
#define av_ts2str(ts)
Convenience macro, the return value should be used only directly in function arguments but never stan...
Definition: timestamp.h:54
An instance of a filter.
Definition: avfilter.h:341
#define av_freep(p)
static void dump_unknown(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:171
int32_t track_gain
Track replay gain in microbels (divide by 100000 to get the value in dB).
Definition: replaygain.h:34
ReplayGain information in the form of the AVReplayGain struct.
Definition: frame.h:76
internal API functions
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
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:361
ReplayGain information (see http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1.0_specification).
Definition: replaygain.h:29
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:380
The data is the AVMatrixEncoding enum defined in libavutil/channel_layout.h.
Definition: frame.h:67
int i
Definition: input.c:407
enum AVDownmixType preferred_downmix_type
Type of downmix preferred by the mastering engineer.
Definition: downmix_info.h:62