@@ -15,6 +15,7 @@ version <next>:

 - new ffmpeg options -filter_script and -filter_complex_script, which allow a

   filtergraph description to be read from a file

 - OpenCL support

+- audio phaser filter

 version 1.2:

@@ -6266,6 +6266,43 @@ following one, the permission might not be received as expected in that

 following filter. Inserting a @ref{format} or @ref{aformat} filter before the

 perms/aperms filter can avoid this problem.

+@section aphaser

+Add a phasing effect to the input audio.

+

+A phaser filter creates series of peaks and troughs in the frequency spectrum.

+The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.

+

+The filter accepts parameters as a list of @var{key}=@var{value}

+pairs, separated by ":".

+

+A description of the accepted parameters follows.

+

+@table @option

+@item in_gain

+Set input gain. Default is 0.4.

+

+@item out_gain

+Set output gain. Default is 0.74

+

+@item delay

+Set delay in milliseconds. Default is 3.0.

+

+@item decay

+Set decay. Default is 0.4.

+

+@item speed

+Set modulation speed in Hz. Default is 0.5.

+

+@item type

+Set modulation type. Default is triangular.

+

+It accepts the following values:

+@table @samp

+@item triangular, t

+@item sinusoidal, s

+@end table

+@end table

+

 @section aselect, select

 Select frames to pass in output.

@@ -58,6 +58,7 @@ OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o

 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o

 OBJS-$(CONFIG_APAD_FILTER)                   += af_apad.o

 OBJS-$(CONFIG_APERMS_FILTER)                 += f_perms.o

+OBJS-$(CONFIG_APHASER_FILTER)                += af_aphaser.o

 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o

 OBJS-$(CONFIG_ASELECT_FILTER)                += f_select.o

 OBJS-$(CONFIG_ASENDCMD_FILTER)               += f_sendcmd.o

new file mode 100644

@@ -0,0 +1,360 @@

+/*

+ * Copyright (c) 2013 Paul B Mahol

+ *

+ * 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

+ */

+

+/**

+ * @file

+ * phaser audio filter

+ */

+

+#include "libavutil/avassert.h"

+#include "libavutil/opt.h"

+#include "audio.h"

+#include "avfilter.h"

+#include "internal.h"

+

+enum WaveType {

+    WAVE_SIN,

+    WAVE_TRI,

+    WAVE_NB,

+};

+

+typedef struct AudioPhaserContext {

+    const AVClass *class;

+    double in_gain, out_gain;

+    double delay;

+    double decay;

+    double speed;

+

+    enum WaveType type;

+

+    int delay_buffer_length;

+    double *delay_buffer;

+

+    int modulation_buffer_length;

+    int32_t *modulation_buffer;

+

+    int delay_pos, modulation_pos;

+

+    void (*phaser)(struct AudioPhaserContext *p,

+                   uint8_t * const *src, uint8_t **dst,

+                   int nb_samples, int channels);

+} AudioPhaserContext;

+

+#define OFFSET(x) offsetof(AudioPhaserContext, x)

+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

+

+static const AVOption aphaser_options[] = {

+    { "in_gain",  "set input gain",            OFFSET(in_gain),  AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0,  1,   FLAGS },

+    { "out_gain", "set output gain",           OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0,  1e9, FLAGS },

+    { "delay",    "set delay in milliseconds", OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=3.},  0,  5,   FLAGS },

+    { "decay",    "set decay",                 OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0, .99,  FLAGS },

+    { "speed",    "set modulation speed",      OFFSET(speed),    AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1,  2,   FLAGS },

+    { "type",     "set modulation type",       OFFSET(type),     AV_OPT_TYPE_INT,    {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },

+    { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },

+    { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },

+    { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },

+    { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },

+    { NULL },

+};

+

+AVFILTER_DEFINE_CLASS(aphaser);

+

+static av_cold int init(AVFilterContext *ctx, const char *args)

+{

+    AudioPhaserContext *p = ctx->priv;

+

+    if (p->in_gain > (1 - p->decay * p->decay))

+        av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");

+    if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)

+        av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");

+

+    return 0;

+}

+

+static int query_formats(AVFilterContext *ctx)

+{

+    AVFilterFormats *formats;

+    AVFilterChannelLayouts *layouts;

+    static const enum AVSampleFormat sample_fmts[] = {

+        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,

+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,

+        AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,

+        AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,

+        AV_SAMPLE_FMT_NONE

+    };

+

+    layouts = ff_all_channel_layouts();

+    if (!layouts)

+        return AVERROR(ENOMEM);

+    ff_set_common_channel_layouts(ctx, layouts);

+

+    formats = ff_make_format_list(sample_fmts);

+    if (!formats)

+        return AVERROR(ENOMEM);

+    ff_set_common_formats(ctx, formats);

+

+    formats = ff_all_samplerates();

+    if (!formats)

+        return AVERROR(ENOMEM);

+    ff_set_common_samplerates(ctx, formats);

+

+    return 0;

+}

+

+static void generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt,

+                                void *table, int table_size,

+                                double min, double max, double phase)

+{

+    uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;

+

+    for (i = 0; i < table_size; i++) {

+        uint32_t point = (i + phase_offset) % table_size;

+        double d;

+

+        switch (wave_type) {

+        case WAVE_SIN:

+            d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;

+            break;

+        case WAVE_TRI:

+            d = (double)point * 2 / table_size;

+            switch (4 * point / table_size) {

+            case 0: d = d + 0.5; break;

+            case 1:

+            case 2: d = 1.5 - d; break;

+            case 3: d = d - 1.5; break;

+            }

+            break;

+        default:

+            av_assert0(0);

+        }

+

+        d  = d * (max - min) + min;

+        switch (sample_fmt) {

+        case AV_SAMPLE_FMT_FLT: {

+            float *fp = (float *)table;

+            *fp++ = (float)d;

+            table = fp;

+            continue; }

+        case AV_SAMPLE_FMT_DBL: {

+            double *dp = (double *)table;

+            *dp++ = d;

+            table = dp;

+            continue; }

+        }

+

+        d += d < 0 ? -0.5 : 0.5;

+        switch (sample_fmt) {

+        case AV_SAMPLE_FMT_S16: {

+            int16_t *sp = table;

+            *sp++ = (int16_t)d;

+            table = sp;

+            continue; }

+        case AV_SAMPLE_FMT_S32: {

+            int32_t *ip = table;

+            *ip++ = (int32_t)d;

+            table = ip;

+            continue; }

+        default:

+            av_assert0(0);

+        }

+    }

+}

+

+#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))

+

+#define PHASER_PLANAR(name, type)                                      \

+static void phaser_## name ##p(AudioPhaserContext *p,                  \

+                               uint8_t * const *src, uint8_t **dst,    \

+                               int nb_samples, int channels)           \

+{                                                                      \

+    int i, c, delay_pos, modulation_pos;                               \

+                                                                       \

+    for (c = 0; c < channels; c++) {                                   \

+        type *s = (type *)src[c];                                      \

+        type *d = (type *)dst[c];                                      \

+        double *buffer = p->delay_buffer +                             \

+                         c * p->delay_buffer_length;                   \

+                                                                       \

+        delay_pos      = p->delay_pos;                                 \

+        modulation_pos = p->modulation_pos;                            \

+                                                                       \

+        for (i = 0; i < nb_samples; i++, s++, d++) {                   \

+            double v = *s * p->in_gain + buffer[                       \

+                       MOD(delay_pos + p->modulation_buffer[           \

+                       modulation_pos],                                \

+                       p->delay_buffer_length)] * p->decay;            \

+                                                                       \

+            modulation_pos = MOD(modulation_pos + 1,                   \

+                             p->modulation_buffer_length);             \

+            delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);    \

+            buffer[delay_pos] = v;                                     \

+                                                                       \

+            *d = v * p->out_gain;                                      \

+        }                                                              \

+    }                                                                  \

+                                                                       \

+    p->delay_pos      = delay_pos;                                     \

+    p->modulation_pos = modulation_pos;                                \

+}

+

+#define PHASER(name, type)                                              \

+static void phaser_## name (AudioPhaserContext *p,                      \

+                            uint8_t * const *src, uint8_t **dst,        \

+                            int nb_samples, int channels)               \

+{                                                                       \

+    int i, c, delay_pos, modulation_pos;                                \

+    type *s = (type *)src[0];                                           \

+    type *d = (type *)dst[0];                                           \

+    double *buffer = p->delay_buffer;                                   \

+                                                                        \

+    delay_pos      = p->delay_pos;                                      \

+    modulation_pos = p->modulation_pos;                                 \

+                                                                        \

+    for (i = 0; i < nb_samples; i++) {                                  \

+        int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \

+                   p->delay_buffer_length) * channels;                  \

+        int npos;                                                       \

+                                                                        \

+        delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);         \

+        npos = delay_pos * channels;                                    \

+        for (c = 0; c < channels; c++, s++, d++) {                      \

+            double v = *s * p->in_gain + buffer[pos + c] * p->decay;    \

+                                                                        \

+            buffer[npos + c] = v;                                       \

+                                                                        \

+            *d = v * p->out_gain;                                       \

+        }                                                               \

+                                                                        \

+        modulation_pos = MOD(modulation_pos + 1,                        \

+                         p->modulation_buffer_length);                  \

+    }                                                                   \

+                                                                        \

+    p->delay_pos      = delay_pos;                                      \

+    p->modulation_pos = modulation_pos;                                 \

+}

+

+PHASER_PLANAR(dbl, double)

+PHASER_PLANAR(flt, float)

+PHASER_PLANAR(s16, int16_t)

+PHASER_PLANAR(s32, int32_t)

+

+PHASER(dbl, double)

+PHASER(flt, float)

+PHASER(s16, int16_t)

+PHASER(s32, int32_t)

+

+static int config_output(AVFilterLink *outlink)

+{

+    AudioPhaserContext *p = outlink->src->priv;

+    AVFilterLink *inlink = outlink->src->inputs[0];

+

+    p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;

+    p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);

+    p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;

+    p->modulation_buffer = av_malloc(p->modulation_buffer_length * sizeof(*p->modulation_buffer));

+

+    if (!p->modulation_buffer || !p->delay_buffer)

+        return AVERROR(ENOMEM);

+

+    generate_wave_table(p->type, AV_SAMPLE_FMT_S32,

+                        p->modulation_buffer, p->modulation_buffer_length,

+                        1., p->delay_buffer_length, M_PI / 2.0);

+

+    p->delay_pos = p->modulation_pos = 0;

+

+    switch (inlink->format) {

+    case AV_SAMPLE_FMT_DBL:  p->phaser = phaser_dbl;  break;

+    case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;

+    case AV_SAMPLE_FMT_FLT:  p->phaser = phaser_flt;  break;

+    case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;

+    case AV_SAMPLE_FMT_S16:  p->phaser = phaser_s16;  break;

+    case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;

+    case AV_SAMPLE_FMT_S32:  p->phaser = phaser_s32;  break;

+    case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;

+    default: av_assert0(0);

+    }

+

+    return 0;

+}

+

+static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)

+{

+    AudioPhaserContext *p = inlink->dst->priv;

+    AVFilterLink *outlink = inlink->dst->outputs[0];

+    AVFrame *outbuf;

+

+    if (av_frame_is_writable(inbuf)) {

+        outbuf = inbuf;

+    } else {

+        outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);

+        if (!outbuf)

+            return AVERROR(ENOMEM);

+        av_frame_copy_props(outbuf, inbuf);

+    }

+

+    p->phaser(p, inbuf->extended_data, outbuf->extended_data,

+              outbuf->nb_samples, av_frame_get_channels(outbuf));

+

+    if (inbuf != outbuf)

+        av_frame_free(&inbuf);

+

+    return ff_filter_frame(outlink, outbuf);

+}

+

+static av_cold void uninit(AVFilterContext *ctx)

+{

+    AudioPhaserContext *p = ctx->priv;

+

+    av_freep(&p->delay_buffer);

+    av_freep(&p->modulation_buffer);

+}

+

+static const AVFilterPad aphaser_inputs[] = {

+    {

+        .name         = "default",

+        .type         = AVMEDIA_TYPE_AUDIO,

+        .filter_frame = filter_frame,

+    },

+    { NULL }

+};

+

+static const AVFilterPad aphaser_outputs[] = {

+    {

+        .name         = "default",

+        .type         = AVMEDIA_TYPE_AUDIO,

+        .config_props = config_output,

+    },

+    { NULL }

+};

+

+static const char *const shorthand[] = { "in_gain", "out_gain", "delay", "decay", "speed", "type", NULL };

+

+AVFilter avfilter_af_aphaser = {

+    .name          = "aphaser",

+    .description   = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),

+    .query_formats = query_formats,

+    .priv_size     = sizeof(AudioPhaserContext),

+    .init          = init,

+    .uninit        = uninit,

+    .inputs        = aphaser_inputs,

+    .outputs       = aphaser_outputs,

+    .priv_class    = &aphaser_class,

+    .shorthand     = shorthand,

+};

@@ -54,6 +54,7 @@ void avfilter_register_all(void)

     REGISTER_FILTER(ANULL,          anull,          af);

     REGISTER_FILTER(APAD,           apad,           af);

     REGISTER_FILTER(APERMS,         aperms,         af);

+    REGISTER_FILTER(APHASER,        aphaser,        af);

     REGISTER_FILTER(ARESAMPLE,      aresample,      af);

     REGISTER_FILTER(ASELECT,        aselect,        af);

     REGISTER_FILTER(ASENDCMD,       asendcmd,       af);

@@ -29,8 +29,8 @@

 #include "libavutil/avutil.h"

 #define LIBAVFILTER_VERSION_MAJOR  3

-#define LIBAVFILTER_VERSION_MINOR  48

-#define LIBAVFILTER_VERSION_MICRO 105

+#define LIBAVFILTER_VERSION_MINOR  49

+#define LIBAVFILTER_VERSION_MICRO 100

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

                                                LIBAVFILTER_VERSION_MINOR, \