af_earwax.c
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1 /*
2  * Copyright (c) 2011 Mina Nagy Zaki
3  * Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
4  * This source code is freely redistributable and may be used for any purpose.
5  * This copyright notice must be maintained. Edward Beingessner And Sundry
6  * Contributors are not responsible for the consequences of using this
7  * software.
8  *
9  * This file is part of FFmpeg.
10  *
11  * FFmpeg is free software; you can redistribute it and/or
12  * modify it under the terms of the GNU Lesser General Public
13  * License as published by the Free Software Foundation; either
14  * version 2.1 of the License, or (at your option) any later version.
15  *
16  * FFmpeg is distributed in the hope that it will be useful,
17  * but WITHOUT ANY WARRANTY; without even the implied warranty of
18  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19  * Lesser General Public License for more details.
20  *
21  * You should have received a copy of the GNU Lesser General Public
22  * License along with FFmpeg; if not, write to the Free Software
23  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24  */
25 
26 /**
27  * @file
28  * Stereo Widening Effect. Adds audio cues to move stereo image in
29  * front of the listener. Adapted from the libsox earwax effect.
30  */
31 
33 #include "avfilter.h"
34 #include "audio.h"
35 #include "formats.h"
36 
37 #define NUMTAPS 64
38 
39 static const int8_t filt[NUMTAPS] = {
40 /* 30° 330° */
41  4, -6, /* 32 tap stereo FIR filter. */
42  4, -11, /* One side filters as if the */
43  -1, -5, /* signal was from 30 degrees */
44  3, 3, /* from the ear, the other as */
45  -2, 5, /* if 330 degrees. */
46  -5, 0,
47  9, 1,
48  6, 3, /* Input */
49  -4, -1, /* Left Right */
50  -5, -3, /* __________ __________ */
51  -2, -5, /* | | | | */
52  -7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
53  6, -7, /* / |__________| |__________| \ */
54  30, -29, /* / \ / \ */
55  12, -3, /* / X \ */
56  -11, 4, /* / / \ \ */
57  -3, 7, /* ____V_____ __________V V__________ _____V____ */
58  -20, 23, /* | | | | | | | | */
59  2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
60  1, -6, /* |__________| |__________| |__________| |__________| */
61  -14, -5, /* \ ___ / \ ___ / */
62  15, -18, /* \ / \ / _____ \ / \ / */
63  6, 7, /* `->| + |<--' / \ `-->| + |<-' */
64  15, -10, /* \___/ _/ \_ \___/ */
65  -14, 22, /* \ / \ / \ / */
66  -7, -2, /* `--->| | | |<---' */
67  -4, 9, /* \_/ \_/ */
68  6, -12, /* */
69  6, -6, /* Headphones */
70  0, -11,
71  0, -5,
72  4, 0};
73 
74 typedef struct {
75  int16_t taps[NUMTAPS * 2];
77 
79 {
80  static const int sample_rates[] = { 44100, -1 };
81 
84 
86  ff_set_common_formats(ctx, formats);
88  ff_set_common_channel_layouts(ctx, layout);
90 
91  return 0;
92 }
93 
94 //FIXME: replace with DSPContext.scalarproduct_int16
95 static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, int16_t *out)
96 {
98  int16_t j;
99 
100  while (in < endin) {
101  sample = 0;
102  for (j = 0; j < NUMTAPS; j++)
103  sample += in[j] * filt[j];
104  *out = av_clip_int16(sample >> 6);
105  out++;
106  in++;
107  }
108 
109  return out;
110 }
111 
112 static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
113 {
114  AVFilterLink *outlink = inlink->dst->outputs[0];
115  int16_t *taps, *endin, *in, *out;
116  AVFrame *outsamples = ff_get_audio_buffer(inlink, insamples->nb_samples);
117 
118  if (!outsamples) {
119  av_frame_free(&insamples);
120  return AVERROR(ENOMEM);
121  }
122  av_frame_copy_props(outsamples, insamples);
123 
124  taps = ((EarwaxContext *)inlink->dst->priv)->taps;
125  out = (int16_t *)outsamples->data[0];
126  in = (int16_t *)insamples ->data[0];
127 
128  // copy part of new input and process with saved input
129  memcpy(taps+NUMTAPS, in, NUMTAPS * sizeof(*taps));
130  out = scalarproduct(taps, taps + NUMTAPS, out);
131 
132  // process current input
133  endin = in + insamples->nb_samples * 2 - NUMTAPS;
134  scalarproduct(in, endin, out);
135 
136  // save part of input for next round
137  memcpy(taps, endin, NUMTAPS * sizeof(*taps));
138 
139  av_frame_free(&insamples);
140  return ff_filter_frame(outlink, outsamples);
141 }
142 
143 static const AVFilterPad earwax_inputs[] = {
144  {
145  .name = "default",
146  .type = AVMEDIA_TYPE_AUDIO,
147  .filter_frame = filter_frame,
148  },
149  { NULL }
150 };
151 
152 static const AVFilterPad earwax_outputs[] = {
153  {
154  .name = "default",
155  .type = AVMEDIA_TYPE_AUDIO,
156  },
157  { NULL }
158 };
159 
161  .name = "earwax",
162  .description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
163  .query_formats = query_formats,
164  .priv_size = sizeof(EarwaxContext),
165  .inputs = earwax_inputs,
166  .outputs = earwax_outputs,
167 };
static const AVFilterPad earwax_outputs[]
Definition: af_earwax.c:152
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:424
This structure describes decoded (raw) audio or video data.
Definition: frame.h:76
static const AVFilterPad outputs[]
Definition: af_ashowinfo.c:117
external API header
About Git write you should know how to use GIT properly Luckily Git comes with excellent documentation git help man git shows you the available git< command > help man git< command > shows information about the subcommand< command > The most comprehensive manual is the website Git Reference visit they are quite exhaustive You do not need a special username or password All you need is to provide a ssh public key to the Git server admin What follows now is a basic introduction to Git and some FFmpeg specific guidelines Read it at least if you are granted commit privileges to the FFmpeg project you are expected to be familiar with these rules I if not You can get git from etc no matter how small Every one of them has been saved from looking like a fool by this many times It s very easy for stray debug output or cosmetic modifications to slip in
Definition: git-howto.txt:5
#define AV_CH_LAYOUT_STEREO
signed 16 bits
Definition: samplefmt.h:52
#define sample
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:308
const char * name
Pad name.
it can be given away to ff_start_frame *A reference passed to ff_filter_frame(or the deprecated ff_start_frame) is given away and must no longer be used.*A reference created with avfilter_ref_buffer belongs to the code that created it.*A reference obtained with ff_get_video_buffer or ff_get_audio_buffer belongs to the code that requested it.*A reference given as return value by the get_video_buffer or get_audio_buffer method is given away and must no longer be used.Link reference fields---------------------The AVFilterLink structure has a few AVFilterBufferRef fields.The cur_buf and out_buf were used with the deprecated start_frame/draw_slice/end_frame API and should no longer be used.src_buf
static const AVFilterPad earwax_inputs[]
Definition: af_earwax.c:143
void 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:545
static int query_formats(AVFilterContext *ctx)
Definition: af_earwax.c:78
A filter pad used for either input or output.
static int16_t * scalarproduct(const int16_t *in, const int16_t *endin, int16_t *out)
Definition: af_earwax.c:95
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:350
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:84
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void * priv
private data for use by the filter
Definition: avfilter.h:545
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:344
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
Definition: af_earwax.c:112
audio channel layout utility functions
int32_t
#define NUMTAPS
Definition: af_earwax.c:37
A list of supported channel layouts.
Definition: formats.h:85
NULL
Definition: eval.c:55
Filter definition.
Definition: avfilter.h:436
const char * name
filter name
Definition: avfilter.h:437
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer 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.Buffer references ownership and permissions
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:539
static const int8_t filt[NUMTAPS]
Definition: af_earwax.c:39
AVFilter avfilter_af_earwax
Definition: af_earwax.c:160
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:87
void ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:533
The official guide to swscale for confused that consecutive non overlapping rectangles of slice_bottom special converter These generally are unscaled converters of common formats
Definition: swscale.txt:33
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:108
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel layout
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:524
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;}return ac;}in_planar=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
void 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:526
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
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:127