FFmpeg
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 
28 #include "libavutil/adler32.h"
29 #include "libavutil/attributes.h"
31 #include "libavutil/downmix_info.h"
32 #include "libavutil/mem.h"
33 #include "libavutil/replaygain.h"
34 #include "libavutil/timestamp.h"
35 #include "libavutil/samplefmt.h"
36 
37 #include "libavcodec/defs.h"
38 
39 #include "audio.h"
40 #include "avfilter.h"
41 #include "filters.h"
42 
43 typedef struct AShowInfoContext {
44  /**
45  * Scratch space for individual plane checksums for planar audio
46  */
47  uint32_t *plane_checksums;
49 
51 {
52  AShowInfoContext *s = ctx->priv;
53  av_freep(&s->plane_checksums);
54 }
55 
57 {
58  enum AVMatrixEncoding enc;
59 
60  av_log(ctx, AV_LOG_INFO, "matrix encoding: ");
61 
62  if (sd->size < sizeof(enum AVMatrixEncoding)) {
63  av_log(ctx, AV_LOG_INFO, "invalid data");
64  return;
65  }
66 
67  enc = *(enum AVMatrixEncoding *)sd->data;
68  switch (enc) {
69  case AV_MATRIX_ENCODING_NONE: av_log(ctx, AV_LOG_INFO, "none"); break;
70  case AV_MATRIX_ENCODING_DOLBY: av_log(ctx, AV_LOG_INFO, "Dolby Surround"); break;
71  case AV_MATRIX_ENCODING_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
72  case AV_MATRIX_ENCODING_DPLIIX: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIx"); break;
73  case AV_MATRIX_ENCODING_DPLIIZ: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIz"); break;
74  case AV_MATRIX_ENCODING_DOLBYEX: av_log(ctx, AV_LOG_INFO, "Dolby EX"); break;
75  case AV_MATRIX_ENCODING_DOLBYHEADPHONE: av_log(ctx, AV_LOG_INFO, "Dolby Headphone"); break;
76  default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
77  }
78 }
79 
81 {
82  AVDownmixInfo *di;
83 
84  av_log(ctx, AV_LOG_INFO, "downmix: ");
85  if (sd->size < sizeof(*di)) {
86  av_log(ctx, AV_LOG_INFO, "invalid data");
87  return;
88  }
89 
90  di = (AVDownmixInfo *)sd->data;
91 
92  av_log(ctx, AV_LOG_INFO, "preferred downmix type - ");
93  switch (di->preferred_downmix_type) {
94  case AV_DOWNMIX_TYPE_LORO: av_log(ctx, AV_LOG_INFO, "Lo/Ro"); break;
95  case AV_DOWNMIX_TYPE_LTRT: av_log(ctx, AV_LOG_INFO, "Lt/Rt"); break;
96  case AV_DOWNMIX_TYPE_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
97  default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
98  }
99 
100  av_log(ctx, AV_LOG_INFO, " Mix levels: center %f (%f ltrt) - "
101  "surround %f (%f ltrt) - lfe %f",
104  di->lfe_mix_level);
105 }
106 
107 static void print_gain(AVFilterContext *ctx, const char *str, int32_t gain)
108 {
109  av_log(ctx, AV_LOG_INFO, "%s - ", str);
110  if (gain == INT32_MIN)
111  av_log(ctx, AV_LOG_INFO, "unknown");
112  else
113  av_log(ctx, AV_LOG_INFO, "%f", gain / 100000.0f);
114  av_log(ctx, AV_LOG_INFO, ", ");
115 }
116 
117 static void print_peak(AVFilterContext *ctx, const char *str, uint32_t peak)
118 {
119  av_log(ctx, AV_LOG_INFO, "%s - ", str);
120  if (!peak)
121  av_log(ctx, AV_LOG_INFO, "unknown");
122  else
123  av_log(ctx, AV_LOG_INFO, "%f", peak / 100000.0f);
124  av_log(ctx, AV_LOG_INFO, ", ");
125 }
126 
128 {
129  AVReplayGain *rg;
130 
131  av_log(ctx, AV_LOG_INFO, "replaygain: ");
132  if (sd->size < sizeof(*rg)) {
133  av_log(ctx, AV_LOG_INFO, "invalid data");
134  return;
135  }
136  rg = (AVReplayGain*)sd->data;
137 
138  print_gain(ctx, "track gain", rg->track_gain);
139  print_peak(ctx, "track peak", rg->track_peak);
140  print_gain(ctx, "album gain", rg->album_gain);
141  print_peak(ctx, "album peak", rg->album_peak);
142 }
143 
145 {
146  enum AVAudioServiceType *ast;
147 
148  av_log(ctx, AV_LOG_INFO, "audio service type: ");
149  if (sd->size < sizeof(*ast)) {
150  av_log(ctx, AV_LOG_INFO, "invalid data");
151  return;
152  }
153  ast = (enum AVAudioServiceType*)sd->data;
154  switch (*ast) {
155  case AV_AUDIO_SERVICE_TYPE_MAIN: av_log(ctx, AV_LOG_INFO, "Main Audio Service"); break;
156  case AV_AUDIO_SERVICE_TYPE_EFFECTS: av_log(ctx, AV_LOG_INFO, "Effects"); break;
157  case AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED: av_log(ctx, AV_LOG_INFO, "Visually Impaired"); break;
158  case AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED: av_log(ctx, AV_LOG_INFO, "Hearing Impaired"); break;
159  case AV_AUDIO_SERVICE_TYPE_DIALOGUE: av_log(ctx, AV_LOG_INFO, "Dialogue"); break;
160  case AV_AUDIO_SERVICE_TYPE_COMMENTARY: av_log(ctx, AV_LOG_INFO, "Commentary"); break;
161  case AV_AUDIO_SERVICE_TYPE_EMERGENCY: av_log(ctx, AV_LOG_INFO, "Emergency"); break;
162  case AV_AUDIO_SERVICE_TYPE_VOICE_OVER: av_log(ctx, AV_LOG_INFO, "Voice Over"); break;
163  case AV_AUDIO_SERVICE_TYPE_KARAOKE: av_log(ctx, AV_LOG_INFO, "Karaoke"); break;
164  default: av_log(ctx, AV_LOG_INFO, "unknown"); break;
165  }
166 }
167 
169 {
170  av_log(ctx, AV_LOG_INFO, "unknown side data type: %d, size "
171  "%"SIZE_SPECIFIER" bytes", sd->type, sd->size);
172 }
173 
175 {
177  AVFilterContext *ctx = inlink->dst;
178  AShowInfoContext *s = ctx->priv;
179  char chlayout_str[128];
180  uint32_t checksum = 0;
181  int channels = inlink->ch_layout.nb_channels;
183  int block_align = av_get_bytes_per_sample(buf->format) * (planar ? 1 : channels);
184  int data_size = buf->nb_samples * block_align;
185  int planes = planar ? channels : 1;
186  int i;
187  void *tmp_ptr = av_realloc_array(s->plane_checksums, channels, sizeof(*s->plane_checksums));
188 
189  if (!tmp_ptr)
190  return AVERROR(ENOMEM);
191  s->plane_checksums = tmp_ptr;
192 
193  for (i = 0; i < planes; i++) {
194  uint8_t *data = buf->extended_data[i];
195 
196  s->plane_checksums[i] = av_adler32_update(0, data, data_size);
197  checksum = i ? av_adler32_update(checksum, data, data_size) :
198  s->plane_checksums[0];
199  }
200 
201  av_channel_layout_describe(&buf->ch_layout, chlayout_str, sizeof(chlayout_str));
202 
204  "n:%"PRId64" pts:%s pts_time:%s "
205  "fmt:%s channels:%d chlayout:%s rate:%d nb_samples:%d "
206  "checksum:%08"PRIX32" ",
207  inl->frame_count_out,
208  av_ts2str(buf->pts), av_ts2timestr(buf->pts, &inlink->time_base),
209  av_get_sample_fmt_name(buf->format), buf->ch_layout.nb_channels, chlayout_str,
210  buf->sample_rate, buf->nb_samples,
211  checksum);
212 
213  av_log(ctx, AV_LOG_INFO, "plane_checksums: [ ");
214  for (i = 0; i < planes; i++)
215  av_log(ctx, AV_LOG_INFO, "%08"PRIX32" ", s->plane_checksums[i]);
216  av_log(ctx, AV_LOG_INFO, "]\n");
217 
218  for (i = 0; i < buf->nb_side_data; i++) {
219  AVFrameSideData *sd = buf->side_data[i];
220 
221  av_log(ctx, AV_LOG_INFO, " side data - ");
222  switch (sd->type) {
224  case AV_FRAME_DATA_DOWNMIX_INFO: dump_downmix (ctx, sd); break;
225  case AV_FRAME_DATA_REPLAYGAIN: dump_replaygain(ctx, sd); break;
227  default: dump_unknown (ctx, sd); break;
228  }
229 
230  av_log(ctx, AV_LOG_INFO, "\n");
231  }
232 
233  return ff_filter_frame(inlink->dst->outputs[0], buf);
234 }
235 
236 static const AVFilterPad inputs[] = {
237  {
238  .name = "default",
239  .type = AVMEDIA_TYPE_AUDIO,
240  .filter_frame = filter_frame,
241  },
242 };
243 
245  .name = "ashowinfo",
246  .description = NULL_IF_CONFIG_SMALL("Show textual information for each audio frame."),
247  .priv_size = sizeof(AShowInfoContext),
248  .uninit = uninit,
252 };
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:215
AVERROR
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
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1061
dump_unknown
static void dump_unknown(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:168
AVFrame::nb_side_data
int nb_side_data
Definition: frame.h:623
inlink
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
Definition: filter_design.txt:212
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: filters.h:262
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:389
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:501
AVReplayGain::album_gain
int32_t album_gain
Same as track_gain, but for the whole album.
Definition: replaygain.h:43
data
const char data[16]
Definition: mxf.c:149
planes
static const struct @467 planes[]
AV_AUDIO_SERVICE_TYPE_VOICE_OVER
@ AV_AUDIO_SERVICE_TYPE_VOICE_OVER
Definition: defs.h:232
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:205
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:327
replaygain.h
AVDownmixInfo::surround_mix_level_ltrt
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
ff_af_ashowinfo
const AVFilter ff_af_ashowinfo
Definition: af_ashowinfo.c:244
AV_FRAME_DATA_MATRIXENCODING
@ AV_FRAME_DATA_MATRIXENCODING
The data is the AVMatrixEncoding enum defined in libavutil/channel_layout.h.
Definition: frame.h:68
samplefmt.h
AVFrame::ch_layout
AVChannelLayout ch_layout
Channel layout of the audio data.
Definition: frame.h:790
AVDownmixInfo
This structure describes optional metadata relevant to a downmix procedure.
Definition: downmix_info.h:58
planar
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;} } 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
AVFilterPad
A filter pad used for either input or output.
Definition: filters.h:38
AVFrameSideData::size
size_t size
Definition: frame.h:268
av_cold
#define av_cold
Definition: attributes.h:90
AV_MATRIX_ENCODING_DOLBY
@ AV_MATRIX_ENCODING_DOLBY
Definition: channel_layout.h:260
av_channel_layout_describe
int av_channel_layout_describe(const AVChannelLayout *channel_layout, char *buf, size_t buf_size)
Get a human-readable string describing the channel layout properties.
Definition: channel_layout.c:651
AShowInfoContext::plane_checksums
uint32_t * plane_checksums
Scratch space for individual plane checksums for planar audio.
Definition: af_ashowinfo.c:47
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
Definition: af_ashowinfo.c:174
s
#define s(width, name)
Definition: cbs_vp9.c:198
av_realloc_array
void * av_realloc_array(void *ptr, size_t nmemb, size_t size)
Definition: mem.c:217
dump_downmix
static void dump_downmix(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:80
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AV_FRAME_DATA_AUDIO_SERVICE_TYPE
@ AV_FRAME_DATA_AUDIO_SERVICE_TYPE
This side data must be associated with an audio frame and corresponds to enum AVAudioServiceType defi...
Definition: frame.h:114
av_sample_fmt_is_planar
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:114
filters.h
ctx
AVFormatContext * ctx
Definition: movenc.c:49
channels
channels
Definition: aptx.h:31
AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED
@ AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED
Definition: defs.h:228
AV_MATRIX_ENCODING_DPLIIX
@ AV_MATRIX_ENCODING_DPLIIX
Definition: channel_layout.h:262
av_get_sample_fmt_name
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:51
AVReplayGain::track_peak
uint32_t track_peak
Peak track amplitude, with 100000 representing full scale (but values may overflow).
Definition: replaygain.h:39
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: filters.h:263
AVMatrixEncoding
AVMatrixEncoding
Definition: channel_layout.h:258
AV_MATRIX_ENCODING_DOLBYHEADPHONE
@ AV_MATRIX_ENCODING_DOLBYHEADPHONE
Definition: channel_layout.h:265
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_ashowinfo.c:50
dump_matrixenc
static void dump_matrixenc(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:56
AVDownmixInfo::surround_mix_level
double surround_mix_level
Absolute scale factor representing the nominal level of the surround channels during a regular downmi...
Definition: downmix_info.h:80
adler32.h
ff_audio_default_filterpad
const AVFilterPad ff_audio_default_filterpad[1]
An AVFilterPad array whose only entry has name "default" and is of type AVMEDIA_TYPE_AUDIO.
Definition: audio.c:34
AV_AUDIO_SERVICE_TYPE_EMERGENCY
@ AV_AUDIO_SERVICE_TYPE_EMERGENCY
Definition: defs.h:231
print_gain
static void print_gain(AVFilterContext *ctx, const char *str, int32_t gain)
Definition: af_ashowinfo.c:107
AVAudioServiceType
AVAudioServiceType
Definition: defs.h:224
ff_filter_link
static FilterLink * ff_filter_link(AVFilterLink *link)
Definition: filters.h:197
av_adler32_update
AVAdler av_adler32_update(AVAdler adler, const uint8_t *buf, size_t len)
Calculate the Adler32 checksum of a buffer.
Definition: adler32.c:44
f
f
Definition: af_crystalizer.c:122
av_ts2timestr
#define av_ts2timestr(ts, tb)
Convenience macro, the return value should be used only directly in function arguments but never stan...
Definition: timestamp.h:83
AV_FRAME_DATA_REPLAYGAIN
@ AV_FRAME_DATA_REPLAYGAIN
ReplayGain information in the form of the AVReplayGain struct.
Definition: frame.h:77
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:94
AVFrame::sample_rate
int sample_rate
Sample rate of the audio data.
Definition: frame.h:588
AV_MATRIX_ENCODING_NONE
@ AV_MATRIX_ENCODING_NONE
Definition: channel_layout.h:259
dump_audio_service_type
static void dump_audio_service_type(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:144
print_peak
static void print_peak(AVFilterContext *ctx, const char *str, uint32_t peak)
Definition: af_ashowinfo.c:117
AVFrameSideData::data
uint8_t * data
Definition: frame.h:267
AVDownmixInfo::center_mix_level_ltrt
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
AVReplayGain::track_gain
int32_t track_gain
Track replay gain in microbels (divide by 100000 to get the value in dB).
Definition: replaygain.h:34
AVFrame::format
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
Definition: frame.h:476
attributes.h
AVDownmixInfo::lfe_mix_level
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
AV_LOG_INFO
#define AV_LOG_INFO
Standard information.
Definition: log.h:220
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:469
AVDownmixInfo::center_mix_level
double center_mix_level
Absolute scale factor representing the nominal level of the center channel during a regular downmix.
Definition: downmix_info.h:68
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
downmix_info.h
av_get_bytes_per_sample
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:108
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:450
AVDownmixInfo::preferred_downmix_type
enum AVDownmixType preferred_downmix_type
Type of downmix preferred by the mastering engineer.
Definition: downmix_info.h:62
AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED
@ AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED
Definition: defs.h:227
AV_DOWNMIX_TYPE_LORO
@ AV_DOWNMIX_TYPE_LORO
Lo/Ro 2-channel downmix (Stereo).
Definition: downmix_info.h:46
AVFrame::side_data
AVFrameSideData ** side_data
Definition: frame.h:622
AVFilterPad::name
const char * name
Pad name.
Definition: filters.h:44
inputs
static const AVFilterPad inputs[]
Definition: af_ashowinfo.c:236
AShowInfoContext
Definition: af_ashowinfo.c:43
AVFilter
Filter definition.
Definition: avfilter.h:201
AV_AUDIO_SERVICE_TYPE_KARAOKE
@ AV_AUDIO_SERVICE_TYPE_KARAOKE
Definition: defs.h:233
SIZE_SPECIFIER
#define SIZE_SPECIFIER
Definition: internal.h:129
channel_layout.h
AV_MATRIX_ENCODING_DOLBYEX
@ AV_MATRIX_ENCODING_DOLBYEX
Definition: channel_layout.h:264
AV_AUDIO_SERVICE_TYPE_COMMENTARY
@ AV_AUDIO_SERVICE_TYPE_COMMENTARY
Definition: defs.h:230
AV_DOWNMIX_TYPE_DPLII
@ AV_DOWNMIX_TYPE_DPLII
Lt/Rt 2-channel downmix, Dolby Pro Logic II compatible.
Definition: downmix_info.h:48
avfilter.h
defs.h
AVReplayGain::album_peak
uint32_t album_peak
Same as track_peak, but for the whole album,.
Definition: replaygain.h:47
AVFILTER_FLAG_METADATA_ONLY
#define AVFILTER_FLAG_METADATA_ONLY
The filter is a "metadata" filter - it does not modify the frame data in any way.
Definition: avfilter.h:168
AVFrameSideData::type
enum AVFrameSideDataType type
Definition: frame.h:266
AVFilterContext
An instance of a filter.
Definition: avfilter.h:457
AVReplayGain
ReplayGain information (see http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1....
Definition: replaygain.h:29
mem.h
audio.h
AVFrameSideData
Structure to hold side data for an AVFrame.
Definition: frame.h:265
AV_AUDIO_SERVICE_TYPE_EFFECTS
@ AV_AUDIO_SERVICE_TYPE_EFFECTS
Definition: defs.h:226
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
AV_MATRIX_ENCODING_DPLIIZ
@ AV_MATRIX_ENCODING_DPLIIZ
Definition: channel_layout.h:263
int32_t
int32_t
Definition: audioconvert.c:56
timestamp.h
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:482
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
av_ts2str
#define av_ts2str(ts)
Convenience macro, the return value should be used only directly in function arguments but never stan...
Definition: timestamp.h:54
AV_AUDIO_SERVICE_TYPE_DIALOGUE
@ AV_AUDIO_SERVICE_TYPE_DIALOGUE
Definition: defs.h:229
dump_replaygain
static void dump_replaygain(AVFilterContext *ctx, AVFrameSideData *sd)
Definition: af_ashowinfo.c:127
AV_DOWNMIX_TYPE_LTRT
@ AV_DOWNMIX_TYPE_LTRT
Lt/Rt 2-channel downmix, Dolby Surround compatible.
Definition: downmix_info.h:47
AV_FRAME_DATA_DOWNMIX_INFO
@ AV_FRAME_DATA_DOWNMIX_INFO
Metadata relevant to a downmix procedure.
Definition: frame.h:73
AV_AUDIO_SERVICE_TYPE_MAIN
@ AV_AUDIO_SERVICE_TYPE_MAIN
Definition: defs.h:225
AV_MATRIX_ENCODING_DPLII
@ AV_MATRIX_ENCODING_DPLII
Definition: channel_layout.h:261