FFmpeg
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 
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 
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:550
size_t buffer_size
Definition: af_haas.c:47
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
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:1093
#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:569
#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:342
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
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:336
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 s(width, name)
Definition: cbs_vp9.c:257
#define A
Definition: af_haas.c:57
int n
Definition: avisynth_c.h:760
A list of supported channel layouts.
Definition: formats.h:85
if(ret)
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:594
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)\n", 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:394
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
double par_side_gain
Definition: af_haas.c:38
#define MAX_HAAS_DELAY
Definition: af_haas.c:27
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
A list of supported formats for one end of a filter link.
Definition: formats.h:64
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
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
double level_out
Definition: af_haas.c:44
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
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 layout
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:557
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654
GLuint buffer
Definition: opengl_enc.c:101
uint32_t delay[2]
Definition: af_haas.c:49