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af_ashowinfo.c
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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
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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  */
20 
21 /**
22  * @file
23  * filter for showing textual audio frame information
24  */
25 
26 #include <inttypes.h>
27 #include <stddef.h>
28 
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"
39 
40 #include "libavcodec/avcodec.h"
41 
42 #include "audio.h"
43 #include "avfilter.h"
44 #include "internal.h"
45 
46 typedef struct AShowInfoContext {
47  /**
48  * Scratch space for individual plane checksums for planar audio
49  */
50  uint32_t *plane_checksums;
52 
54 {
55  AShowInfoContext *s = ctx->priv;
57 }
58 
60 {
61  enum AVMatrixEncoding enc;
62 
63  av_log(ctx, AV_LOG_INFO, "matrix encoding: ");
64 
65  if (sd->size < sizeof(enum AVMatrixEncoding)) {
66  av_log(ctx, AV_LOG_INFO, "invalid data");
67  return;
68  }
69 
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 }
82 
84 {
85  AVDownmixInfo *di;
86 
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  }
92 
93  di = (AVDownmixInfo *)sd->data;
94 
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  }
102 
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 }
109 
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 }
119 
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 }
129 
131 {
132  AVReplayGain *rg;
133 
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;
140 
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 }
146 
148 {
149  enum AVAudioServiceType *ast;
150 
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 }
170 
172 {
173  av_log(ctx, AV_LOG_INFO, "unknown side data type: %d, size %d bytes", sd->type, sd->size);
174 }
175 
176 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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));
189 
190  if (!tmp_ptr)
191  return AVERROR(ENOMEM);
192  s->plane_checksums = tmp_ptr;
193 
194  for (i = 0; i < planes; i++) {
195  uint8_t *data = buf->extended_data[i];
196 
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  }
201 
202  av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
203  buf->channel_layout);
204 
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),
212  av_get_sample_fmt_name(buf->format), av_frame_get_channels(buf), chlayout_str,
213  buf->sample_rate, buf->nb_samples,
214  checksum);
215 
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");
220 
221  for (i = 0; i < buf->nb_side_data; i++) {
222  AVFrameSideData *sd = buf->side_data[i];
223 
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  }
232 
233  av_log(ctx, AV_LOG_INFO, "\n");
234  }
235 
236  return ff_filter_frame(inlink->dst->outputs[0], buf);
237 }
238 
239 static const AVFilterPad inputs[] = {
240  {
241  .name = "default",
242  .type = AVMEDIA_TYPE_AUDIO,
243  .filter_frame = filter_frame,
244  },
245  { NULL }
246 };
247 
248 static const AVFilterPad outputs[] = {
249  {
250  .name = "default",
251  .type = AVMEDIA_TYPE_AUDIO,
252  },
253  { NULL }
254 };
255 
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
const char * s
Definition: avisynth_c.h:768
This structure describes decoded (raw) audio or video data.
Definition: frame.h:190
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
This side data must be associated with an audio frame and corresponds to enum AVAudioServiceType defi...
Definition: frame.h:112
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static const AVFilterPad outputs[]
Definition: af_ashowinfo.c:248
Main libavfilter public API header.
Memory handling functions.
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
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:59
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
timestamp utils, mostly useful for debugging/logging purposes
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:274
Structure to hold side data for an AVFrame.
Definition: frame.h:149
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
Metadata relevant to a downmix procedure.
Definition: frame.h:71
int nb_side_data
Definition: frame.h:394
void * av_realloc_array(void *ptr, size_t nmemb, size_t size)
Definition: mem.c:208
int32_t album_gain
Same as track_gain, but for the whole album.
Definition: replaygain.h:43
AVFrameSideData ** side_data
Definition: frame.h:393
#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:53
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
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch const uint8_t **in ch off *out planar
Definition: audioconvert.c:56
AVAudioServiceType
Definition: avcodec.h:790
#define AVERROR(e)
Definition: error.h:43
#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
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
void * priv
private data for use by the filter
Definition: avfilter.h:322
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:359
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
int32_t
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
static volatile int checksum
Definition: adler32.c:28
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:254
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:187
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:151
void * buf
Definition: avisynth_c.h:690
int sample_rate
Sample rate of the audio data.
Definition: frame.h:354
int av_frame_get_channels(const AVFrame *frame)
Filter definition.
Definition: avfilter.h:144
static const AVFilterPad inputs[]
Definition: af_ashowinfo.c:239
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:319
enum AVFrameSideDataType type
Definition: frame.h:150
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
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
int64_t av_frame_get_pkt_pos(const AVFrame *frame)
An instance of a filter.
Definition: avfilter.h:307
#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
AVMatrixEncoding
ReplayGain information in the form of the AVReplayGain struct.
Definition: frame.h:75
internal API functions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:237
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:247
The data is the AVMatrixEncoding enum defined in libavutil/channel_layout.h.
Definition: frame.h:66
enum AVDownmixType preferred_downmix_type
Type of downmix preferred by the mastering engineer.
Definition: downmix_info.h:62