FFmpeg  4.0
af_haas.c
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1 /*
2  * Copyright (c) 2001-2010 Vladimir Sadovnikov
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 
22 #include "libavutil/opt.h"
23 #include "avfilter.h"
24 #include "audio.h"
25 #include "formats.h"
26 
27 #define MAX_HAAS_DELAY 40
28 
29 typedef struct HaasContext {
30  const AVClass *class;
31 
33  double par_delay0;
34  double par_delay1;
38  double par_side_gain;
39  double par_gain0;
40  double par_gain1;
41  double par_balance0;
42  double par_balance1;
43  double level_in;
44  double level_out;
45 
46  double *buffer;
47  size_t buffer_size;
48  uint32_t write_ptr;
49  uint32_t delay[2];
50  double balance_l[2];
51  double balance_r[2];
52  double phase0;
53  double phase1;
54 } HaasContext;
55 
56 #define OFFSET(x) offsetof(HaasContext, x)
57 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
58 
59 static const AVOption haas_options[] = {
60  { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
61  { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
62  { "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
63  { "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, "source" },
64  { "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "source" },
65  { "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "source" },
66  { "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "source" },
67  { "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "source" },
68  { "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
69  { "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, MAX_HAAS_DELAY, A },
70  { "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A },
71  { "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
72  { "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
73  { "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, MAX_HAAS_DELAY, A },
74  { "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A },
75  { "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
76  { "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A },
77  { NULL }
78 };
79 
81 
83 {
86  int ret;
87 
88  if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
89  (ret = ff_set_common_formats (ctx , formats )) < 0 ||
90  (ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
91  (ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
92  return ret;
93 
94  formats = ff_all_samplerates();
95  return ff_set_common_samplerates(ctx, formats);
96 }
97 
98 static int config_input(AVFilterLink *inlink)
99 {
100  AVFilterContext *ctx = inlink->dst;
101  HaasContext *s = ctx->priv;
102  size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001);
103  size_t new_buf_size = 1;
104 
105  while (new_buf_size < min_buf_size)
106  new_buf_size <<= 1;
107 
108  av_freep(&s->buffer);
109  s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer));
110  if (!s->buffer)
111  return AVERROR(ENOMEM);
112 
113  s->buffer_size = new_buf_size;
114  s->write_ptr = 0;
115 
116  s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate);
117  s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate);
118 
119  s->phase0 = s->par_phase0 ? 1.0 : -1.0;
120  s->phase1 = s->par_phase1 ? 1.0 : -1.0;
121 
122  s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0;
123  s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0;
124  s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1;
125  s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1;
126 
127  return 0;
128 }
129 
130 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
131 {
132  AVFilterContext *ctx = inlink->dst;
133  AVFilterLink *outlink = ctx->outputs[0];
134  HaasContext *s = ctx->priv;
135  const double *src = (const double *)in->data[0];
136  const double level_in = s->level_in;
137  const double level_out = s->level_out;
138  const uint32_t mask = s->buffer_size - 1;
139  double *buffer = s->buffer;
140  AVFrame *out;
141  double *dst;
142  int n;
143 
144  if (av_frame_is_writable(in)) {
145  out = in;
146  } else {
147  out = ff_get_audio_buffer(outlink, in->nb_samples);
148  if (!out) {
149  av_frame_free(&in);
150  return AVERROR(ENOMEM);
151  }
153  }
154  dst = (double *)out->data[0];
155 
156  for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
157  double mid, side[2], side_l, side_r;
158  uint32_t s0_ptr, s1_ptr;
159 
160  switch (s->par_m_source) {
161  case 0: mid = src[0]; break;
162  case 1: mid = src[1]; break;
163  case 2: mid = (src[0] + src[1]) * 0.5; break;
164  case 3: mid = (src[0] - src[1]) * 0.5; break;
165  }
166 
167  mid *= level_in;
168 
169  buffer[s->write_ptr] = mid;
170 
171  s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask;
172  s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask;
173 
174  if (s->par_middle_phase)
175  mid = -mid;
176 
177  side[0] = buffer[s0_ptr] * s->par_side_gain;
178  side[1] = buffer[s1_ptr] * s->par_side_gain;
179  side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1];
180  side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0];
181 
182  dst[0] = (mid + side_l) * level_out;
183  dst[1] = (mid + side_r) * level_out;
184 
185  s->write_ptr = (s->write_ptr + 1) & mask;
186  }
187 
188  if (out != in)
189  av_frame_free(&in);
190  return ff_filter_frame(outlink, out);
191 }
192 
194 {
195  HaasContext *s = ctx->priv;
196 
197  av_freep(&s->buffer);
198  s->buffer_size = 0;
199 }
200 
201 static const AVFilterPad inputs[] = {
202  {
203  .name = "default",
204  .type = AVMEDIA_TYPE_AUDIO,
205  .filter_frame = filter_frame,
206  .config_props = config_input,
207  },
208  { NULL }
209 };
210 
211 static const AVFilterPad outputs[] = {
212  {
213  .name = "default",
214  .type = AVMEDIA_TYPE_AUDIO,
215  },
216  { NULL }
217 };
218 
220  .name = "haas",
221  .description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
222  .query_formats = query_formats,
223  .priv_size = sizeof(HaasContext),
224  .priv_class = &haas_class,
225  .uninit = uninit,
226  .inputs = inputs,
227  .outputs = outputs,
228 };
#define NULL
Definition: coverity.c:32
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:549
const char * s
Definition: avisynth_c.h:768
size_t buffer_size
Definition: af_haas.c:47
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
AVFILTER_DEFINE_CLASS(haas)
double par_gain1
Definition: af_haas.c:40
AVOption.
Definition: opt.h:246
Main libavfilter public API header.
double * buffer
Definition: af_haas.c:46
double balance_r[2]
Definition: af_haas.c:51
#define AV_CH_LAYOUT_STEREO
#define src
Definition: vp8dsp.c:254
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
int par_phase0
Definition: af_haas.c:35
const char * name
Pad name.
Definition: internal.h:60
double balance_l[2]
Definition: af_haas.c:50
double phase0
Definition: af_haas.c:52
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
#define av_cold
Definition: attributes.h:82
AVOptions.
static int query_formats(AVFilterContext *ctx)
Definition: af_haas.c:82
double par_gain0
Definition: af_haas.c:39
uint32_t write_ptr
Definition: af_haas.c:48
double phase1
Definition: af_haas.c:53
A filter pad used for either input or output.
Definition: internal.h:54
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:568
#define OFFSET(x)
Definition: af_haas.c:56
double level_in
Definition: af_haas.c:43
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:343
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
static const uint16_t mask[17]
Definition: lzw.c:38
int par_phase1
Definition: af_haas.c:36
int par_middle_phase
Definition: af_haas.c:37
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:337
AVFilter ff_af_haas
Definition: af_haas.c:219
static const AVOption haas_options[]
Definition: af_haas.c:59
static const AVFilterPad inputs[]
Definition: af_haas.c:201
audio channel layout utility functions
static const AVFilterPad outputs[]
Definition: af_haas.c:211
AVFormatContext * ctx
Definition: movenc.c:48
#define A
Definition: af_haas.c:57
int n
Definition: avisynth_c.h:684
if(ret< 0)
Definition: vf_mcdeint.c:279
A list of supported channel layouts.
Definition: formats.h:85
static int config_input(AVFilterLink *inlink)
Definition: af_haas.c:98
double par_balance1
Definition: af_haas.c:42
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:592
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_haas.c:130
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_haas.c:193
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=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);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_AARCH64) ff_audio_convert_init_aarch64(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_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
double par_delay1
Definition: af_haas.c:34
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
int par_m_source
Definition: af_haas.c:32
double par_balance0
Definition: af_haas.c:41
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:232
double par_side_gain
Definition: af_haas.c:38
#define MAX_HAAS_DELAY
Definition: af_haas.c:27
A list of supported formats for one end of a filter link.
Definition: formats.h:64
uint64_t layout
An instance of a filter.
Definition: avfilter.h:338
FILE * out
Definition: movenc.c:54
#define av_freep(p)
double par_delay0
Definition: af_haas.c:33
formats
Definition: signature.h:48
double level_out
Definition: af_haas.c:44
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:284
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:652
uint32_t delay[2]
Definition: af_haas.c:49