libavfilter/af_haas.c - third_party/ffmpeg - Git at Google

 /*
  * Copyright (c) 2001-2010 Vladimir Sadovnikov
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 #include "libavutil/channel_layout.h"
 #include "libavutil/opt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "formats.h"

 #define MAX_HAAS_DELAY 40

 typedef struct HaasContext {
     const AVClass *class;

     int par_m_source;
     double par_delay0;
     double par_delay1;
     int par_phase0;
     int par_phase1;
     int par_middle_phase;
     double par_side_gain;
     double par_gain0;
     double par_gain1;
     double par_balance0;
     double par_balance1;
     double level_in;
     double level_out;

     double *buffer;
     size_t buffer_size;
     uint32_t write_ptr;
     uint32_t delay[2];
     double balance_l[2];
     double balance_r[2];
     double phase0;
     double phase1;
 } HaasContext;

 #define OFFSET(x) offsetof(HaasContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption haas_options[] = {
     { "level_in",      "set level in",      OFFSET(level_in),         AV_OPT_TYPE_DOUBLE,  {.dbl=1}, 0.015625,  64, A },
     { "level_out",     "set level out",     OFFSET(level_out),        AV_OPT_TYPE_DOUBLE,  {.dbl=1}, 0.015625,  64, A },
     { "side_gain",     "set side gain",     OFFSET(par_side_gain),    AV_OPT_TYPE_DOUBLE,  {.dbl=1}, 0.015625,  64, A },
     { "middle_source", "set middle source", OFFSET(par_m_source),     AV_OPT_TYPE_INT,     {.i64=2},        0,   3, A, "source" },
     {   "left",        0,                   0,                        AV_OPT_TYPE_CONST,   {.i64=0},        0,   0, A, "source" },
     {   "right",       0,                   0,                        AV_OPT_TYPE_CONST,   {.i64=1},        0,   0, A, "source" },
     {   "mid",         "L+R",               0,                        AV_OPT_TYPE_CONST,   {.i64=2},        0,   0, A, "source" },
     {   "side",        "L-R",               0,                        AV_OPT_TYPE_CONST,   {.i64=3},        0,   0, A, "source" },
     { "middle_phase",  "set middle phase",  OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL,    {.i64=0},        0,   1, A },
     { "left_delay",    "set left delay",    OFFSET(par_delay0),       AV_OPT_TYPE_DOUBLE,  {.dbl=2.05},     0,  MAX_HAAS_DELAY, A },
     { "left_balance",  "set left balance",  OFFSET(par_balance0),     AV_OPT_TYPE_DOUBLE,  {.dbl=-1.0},    -1,   1, A },
     { "left_gain",     "set left gain",     OFFSET(par_gain0),        AV_OPT_TYPE_DOUBLE,  {.dbl=1}, 0.015625,  64, A },
     { "left_phase",    "set left phase",    OFFSET(par_phase0),       AV_OPT_TYPE_BOOL,    {.i64=0},        0,   1, A },
     { "right_delay",   "set right delay",   OFFSET(par_delay1),       AV_OPT_TYPE_DOUBLE,  {.dbl=2.12},     0,  MAX_HAAS_DELAY, A },
     { "right_balance", "set right balance", OFFSET(par_balance1),     AV_OPT_TYPE_DOUBLE,  {.dbl=1},       -1,   1, A },
     { "right_gain",    "set right gain",    OFFSET(par_gain1),        AV_OPT_TYPE_DOUBLE,  {.dbl=1}, 0.015625,  64, A },
     { "right_phase",   "set right phase",   OFFSET(par_phase1),       AV_OPT_TYPE_BOOL,    {.i64=1},        0,   1, A },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(haas);

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats = NULL;
     AVFilterChannelLayouts *layout = NULL;
     int ret;

     if ((ret = ff_add_format                 (&formats, AV_SAMPLE_FMT_DBL  )) < 0 ||
         (ret = ff_set_common_formats         (ctx     , formats            )) < 0 ||
         (ret = ff_add_channel_layout         (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
         (ret = ff_set_common_channel_layouts (ctx     , layout             )) < 0)
         return ret;

     formats = ff_all_samplerates();
     return ff_set_common_samplerates(ctx, formats);
 }

 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     HaasContext *s = ctx->priv;
     size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001);
     size_t new_buf_size = 1;

     while (new_buf_size < min_buf_size)
         new_buf_size <<= 1;

     av_freep(&s->buffer);
     s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer));
     if (!s->buffer)
         return AVERROR(ENOMEM);

     s->buffer_size = new_buf_size;
     s->write_ptr = 0;

     s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate);
     s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate);

     s->phase0 = s->par_phase0 ? 1.0 : -1.0;
     s->phase1 = s->par_phase1 ? 1.0 : -1.0;

     s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0;
     s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0;
     s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1;
     s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1;

     return 0;
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
     HaasContext *s = ctx->priv;
     const double *src = (const double *)in->data[0];
     const double level_in = s->level_in;
     const double level_out = s->level_out;
     const uint32_t mask = s->buffer_size - 1;
     double *buffer = s->buffer;
     AVFrame *out;
     double *dst;
     int n;

     if (av_frame_is_writable(in)) {
         out = in;
     } else {
         out = ff_get_audio_buffer(inlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);
         }
         av_frame_copy_props(out, in);
     }
     dst = (double *)out->data[0];

     for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
         double mid, side[2], side_l, side_r;
         uint32_t s0_ptr, s1_ptr;

         switch (s->par_m_source) {
         case 0: mid = src[0]; break;
         case 1: mid = src[1]; break;
         case 2: mid = (src[0] + src[1]) * 0.5; break;
         case 3: mid = (src[0] - src[1]) * 0.5; break;
         }

         mid *= level_in;

         buffer[s->write_ptr] = mid;

         s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask;
         s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask;

         if (s->par_middle_phase)
             mid = -mid;

         side[0] = buffer[s0_ptr] * s->par_side_gain;
         side[1] = buffer[s1_ptr] * s->par_side_gain;
         side_l  = side[0] * s->balance_l[0] - side[1] * s->balance_l[1];
         side_r  = side[1] * s->balance_r[1] - side[0] * s->balance_r[0];

         dst[0] = (mid + side_l) * level_out;
         dst[1] = (mid + side_r) * level_out;

         s->write_ptr = (s->write_ptr + 1) & mask;
     }

     if (out != in)
         av_frame_free(&in);
     return ff_filter_frame(outlink, out);
 }

 static av_cold void uninit(AVFilterContext *ctx)
 {
     HaasContext *s = ctx->priv;

     av_freep(&s->buffer);
     s->buffer_size = 0;
 }

 static const AVFilterPad inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
         .config_props = config_input,
     },
     { NULL }
 };

 static const AVFilterPad outputs[] = {
     {
         .name = "default",
         .type = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };

 AVFilter ff_af_haas = {
     .name           = "haas",
     .description    = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
     .query_formats  = query_formats,
     .priv_size      = sizeof(HaasContext),
     .priv_class     = &haas_class,
     .uninit         = uninit,
     .inputs         = inputs,
     .outputs        = outputs,
 };
	/*
	* Copyright (c) 2001-2010 Vladimir Sadovnikov
	*
	* This file is part of FFmpeg.
	*
	* FFmpeg is free software; you can redistribute it and/or
	* modify it under the terms of the GNU Lesser General Public
	* License as published by the Free Software Foundation; either
	* version 2.1 of the License, or (at your option) any later version.
	*
	* FFmpeg is distributed in the hope that it will be useful,
	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	* Lesser General Public License for more details.
	*
	* You should have received a copy of the GNU Lesser General Public
	* License along with FFmpeg; if not, write to the Free Software
	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	*/

	#include "libavutil/channel_layout.h"
	#include "libavutil/opt.h"
	#include "avfilter.h"
	#include "audio.h"
	#include "formats.h"

	#define MAX_HAAS_DELAY 40

	typedef struct HaasContext {
	const AVClass *class;

	int par_m_source;
	double par_delay0;
	double par_delay1;
	int par_phase0;
	int par_phase1;
	int par_middle_phase;
	double par_side_gain;
	double par_gain0;
	double par_gain1;
	double par_balance0;
	double par_balance1;
	double level_in;
	double level_out;

	double *buffer;
	size_t buffer_size;
	uint32_t write_ptr;
	uint32_t delay[2];
	double balance_l[2];
	double balance_r[2];
	double phase0;
	double phase1;
	} HaasContext;

	#define OFFSET(x) offsetof(HaasContext, x)
	#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM

	static const AVOption haas_options[] = {
	{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
	{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
	{ "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
	{ "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, "source" },
	{ "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "source" },
	{ "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "source" },
	{ "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "source" },
	{ "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "source" },
	{ "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
	{ "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, MAX_HAAS_DELAY, A },
	{ "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A },
	{ "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
	{ "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
	{ "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, MAX_HAAS_DELAY, A },
	{ "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A },
	{ "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
	{ "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(haas);

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterFormats *formats = NULL;
	AVFilterChannelLayouts *layout = NULL;
	int ret;

	if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 \|\|
	(ret = ff_set_common_formats (ctx , formats )) < 0 \|\|
	(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 \|\|
	(ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
	return ret;

	formats = ff_all_samplerates();
	return ff_set_common_samplerates(ctx, formats);
	}

	static int config_input(AVFilterLink *inlink)
	{
	AVFilterContext *ctx = inlink->dst;
	HaasContext *s = ctx->priv;
	size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001);
	size_t new_buf_size = 1;

	while (new_buf_size < min_buf_size)
	new_buf_size <<= 1;

	av_freep(&s->buffer);
	s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer));
	if (!s->buffer)
	return AVERROR(ENOMEM);

	s->buffer_size = new_buf_size;
	s->write_ptr = 0;

	s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate);
	s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate);

	s->phase0 = s->par_phase0 ? 1.0 : -1.0;
	s->phase1 = s->par_phase1 ? 1.0 : -1.0;

	s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0;
	s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0;
	s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1;
	s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1;

	return 0;
	}

	static int filter_frame(AVFilterLink inlink, AVFrame in)
	{
	AVFilterContext *ctx = inlink->dst;
	AVFilterLink *outlink = ctx->outputs[0];
	HaasContext *s = ctx->priv;
	const double src = (const double )in->data[0];
	const double level_in = s->level_in;
	const double level_out = s->level_out;
	const uint32_t mask = s->buffer_size - 1;
	double *buffer = s->buffer;
	AVFrame *out;
	double *dst;
	int n;

	if (av_frame_is_writable(in)) {
	out = in;
	} else {
	out = ff_get_audio_buffer(inlink, in->nb_samples);
	if (!out) {
	av_frame_free(&in);
	return AVERROR(ENOMEM);
	}
	av_frame_copy_props(out, in);
	}
	dst = (double *)out->data[0];

	for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
	double mid, side[2], side_l, side_r;
	uint32_t s0_ptr, s1_ptr;

	switch (s->par_m_source) {
	case 0: mid = src[0]; break;
	case 1: mid = src[1]; break;
	case 2: mid = (src[0] + src[1]) * 0.5; break;
	case 3: mid = (src[0] - src[1]) * 0.5; break;
	}

	mid *= level_in;

	buffer[s->write_ptr] = mid;

	s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask;
	s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask;

	if (s->par_middle_phase)
	mid = -mid;

	side[0] = buffer[s0_ptr] * s->par_side_gain;
	side[1] = buffer[s1_ptr] * s->par_side_gain;
	side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1];
	side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0];

	dst[0] = (mid + side_l) * level_out;
	dst[1] = (mid + side_r) * level_out;

	s->write_ptr = (s->write_ptr + 1) & mask;
	}

	if (out != in)
	av_frame_free(&in);
	return ff_filter_frame(outlink, out);
	}

	static av_cold void uninit(AVFilterContext *ctx)
	{
	HaasContext *s = ctx->priv;

	av_freep(&s->buffer);
	s->buffer_size = 0;
	}

	static const AVFilterPad inputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.filter_frame = filter_frame,
	.config_props = config_input,
	},
	{ NULL }
	};

	static const AVFilterPad outputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	},
	{ NULL }
	};

	AVFilter ff_af_haas = {
	.name = "haas",
	.description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
	.query_formats = query_formats,
	.priv_size = sizeof(HaasContext),
	.priv_class = &haas_class,
	.uninit = uninit,
	.inputs = inputs,
	.outputs = outputs,
	};