FFmpeg  4.3
af_volumedetect.c
Go to the documentation of this file.
1 /*
2  * Copyright (c) 2012 Nicolas George
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 License
8  * 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 License
17  * along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
18  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
22 #include "libavutil/avassert.h"
23 #include "audio.h"
24 #include "avfilter.h"
25 #include "internal.h"
26 
27 typedef struct VolDetectContext {
28  /**
29  * Number of samples at each PCM value.
30  * histogram[0x8000 + i] is the number of samples at value i.
31  * The extra element is there for symmetry.
32  */
33  uint64_t histogram[0x10001];
35 
37 {
38  static const enum AVSampleFormat sample_fmts[] = {
42  };
45  int ret;
46 
48  return AVERROR(ENOMEM);
49 
51  if (!layouts)
52  return AVERROR(ENOMEM);
54  if (ret < 0)
55  return ret;
56 
58 }
59 
61 {
62  AVFilterContext *ctx = inlink->dst;
63  VolDetectContext *vd = ctx->priv;
64  int nb_samples = samples->nb_samples;
65  int nb_channels = samples->channels;
66  int nb_planes = nb_channels;
67  int plane, i;
68  int16_t *pcm;
69 
70  if (!av_sample_fmt_is_planar(samples->format)) {
71  nb_samples *= nb_channels;
72  nb_planes = 1;
73  }
74  for (plane = 0; plane < nb_planes; plane++) {
75  pcm = (int16_t *)samples->extended_data[plane];
76  for (i = 0; i < nb_samples; i++)
77  vd->histogram[pcm[i] + 0x8000]++;
78  }
79 
80  return ff_filter_frame(inlink->dst->outputs[0], samples);
81 }
82 
83 #define MAX_DB 91
84 
85 static inline double logdb(uint64_t v)
86 {
87  double d = v / (double)(0x8000 * 0x8000);
88  if (!v)
89  return MAX_DB;
90  return -log10(d) * 10;
91 }
92 
94 {
95  VolDetectContext *vd = ctx->priv;
96  int i, max_volume, shift;
97  uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
98  uint64_t histdb[MAX_DB + 1] = { 0 };
99 
100  for (i = 0; i < 0x10000; i++)
101  nb_samples += vd->histogram[i];
102  av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);
103  if (!nb_samples)
104  return;
105 
106  /* If nb_samples > 1<<34, there is a risk of overflow in the
107  multiplication or the sum: shift all histogram values to avoid that.
108  The total number of samples must be recomputed to avoid rounding
109  errors. */
110  shift = av_log2(nb_samples >> 33);
111  for (i = 0; i < 0x10000; i++) {
112  nb_samples_shift += vd->histogram[i] >> shift;
113  power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift);
114  }
115  if (!nb_samples_shift)
116  return;
117  power = (power + nb_samples_shift / 2) / nb_samples_shift;
118  av_assert0(power <= 0x8000 * 0x8000);
119  av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power));
120 
121  max_volume = 0x8000;
122  while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] &&
123  !vd->histogram[0x8000 - max_volume])
124  max_volume--;
125  av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume));
126 
127  for (i = 0; i < 0x10000; i++)
128  histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i];
129  for (i = 0; i <= MAX_DB && !histdb[i]; i++);
130  for (; i <= MAX_DB && sum < nb_samples / 1000; i++) {
131  av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]);
132  sum += histdb[i];
133  }
134 }
135 
137 {
138  print_stats(ctx);
139 }
140 
142  {
143  .name = "default",
144  .type = AVMEDIA_TYPE_AUDIO,
145  .filter_frame = filter_frame,
146  },
147  { NULL }
148 };
149 
151  {
152  .name = "default",
153  .type = AVMEDIA_TYPE_AUDIO,
154  },
155  { NULL }
156 };
157 
159  .name = "volumedetect",
160  .description = NULL_IF_CONFIG_SMALL("Detect audio volume."),
161  .priv_size = sizeof(VolDetectContext),
163  .uninit = uninit,
166 };
formats
formats
Definition: signature.h:48
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:85
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
volumedetect_outputs
static const AVFilterPad volumedetect_outputs[]
Definition: af_volumedetect.c:150
ff_make_format_list
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_volumedetect.c:136
ff_set_common_channel_layouts
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates.
Definition: formats.c:581
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1075
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:716
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
ff_af_volumedetect
AVFilter ff_af_volumedetect
Definition: af_volumedetect.c:158
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
ff_all_channel_counts
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:440
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
MAX_DB
#define MAX_DB
Definition: af_volumedetect.c:83
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
VolDetectContext::histogram
uint64_t histogram[0x10001]
Number of samples at each PCM value.
Definition: af_volumedetect.c:33
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
avassert.h
av_cold
#define av_cold
Definition: attributes.h:90
ff_set_common_formats
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:600
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
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:112
outputs
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
ctx
AVFormatContext * ctx
Definition: movenc.c:48
NULL
#define NULL
Definition: coverity.c:32
inputs
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
Definition: filter_design.txt:243
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
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:186
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_volumedetect.c:36
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
av_log2
#define av_log2
Definition: intmath.h:83
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
AV_LOG_INFO
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
internal.h
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
Definition: af_volumedetect.c:60
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
AVFilter
Filter definition.
Definition: avfilter.h:144
ret
ret
Definition: filter_design.txt:187
channel_layout.h
avfilter.h
logdb
static double logdb(uint64_t v)
Definition: af_volumedetect.c:85
AVFilterContext
An instance of a filter.
Definition: avfilter.h:338
shift
static int shift(int a, int b)
Definition: sonic.c:82
VolDetectContext
Definition: af_volumedetect.c:27
audio.h
print_stats
static void print_stats(AVFilterContext *ctx)
Definition: af_volumedetect.c:93
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
volumedetect_inputs
static const AVFilterPad volumedetect_inputs[]
Definition: af_volumedetect.c:141
nb_channels
int nb_channels
Definition: channel_layout.c:76