lavfi: add aresample filter

Based on a patch by Stefano.

Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>

doc/filters.texi

@@ -124,6 +124,19 @@ aformat=s16:mono\\,stereo:all
 
 Pass the audio source unchanged to the output.
 
+@section aresample
+
+Resample the input audio to the specified sample rate.
+
+The filter accepts exactly one parameter, the output sample rate. If not
+specified then the filter will automatically convert between its input
+and output sample rates.
+
+For example, to resample the input audio to 44100Hz:
+@example
+aresample=44100
+@end example
+
 @c man end AUDIO FILTERS
 
 @chapter Audio Sources

libavfilter/Makefile

@@ -2,6 +2,7 @@ include $(SUBDIR)../config.mak
 
 NAME = avfilter
 FFLIBS = avutil
+FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += avcodec
 FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
 FFLIBS-$(CONFIG_SCALE_FILTER) += swscale
 FFLIBS-$(CONFIG_MP_FILTER) += avcodec
@@ -20,6 +21,7 @@ OBJS-$(CONFIG_AVCODEC)                       += avcodec.o
 
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
+OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
 
 OBJS-$(CONFIG_ANULLSRC_FILTER)               += asrc_anullsrc.o
 

libavfilter/af_aresample.c

@@ -0,0 +1,351 @@
+/*
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2011 Mina Nagy Zaki
+ *
+ * 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
+ * resampling audio filter
+ */
+
+#include "libavutil/eval.h"
+#include "libavcodec/avcodec.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct {
+    struct AVResampleContext *resample;
+    int out_rate;
+    double ratio;
+    AVFilterBufferRef *outsamplesref;
+    int unconsumed_nb_samples,
+        max_cached_nb_samples;
+    int16_t *cached_data[8],
+            *resampled_data[8];
+} AResampleContext;
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+    AResampleContext *aresample = ctx->priv;
+    int ret;
+
+    if (args) {
+        if ((ret = ff_parse_sample_rate(&aresample->out_rate, args, ctx)) < 0)
+            return ret;
+    } else {
+        aresample->out_rate = -1;
+    }
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AResampleContext *aresample = ctx->priv;
+    if (aresample->outsamplesref) {
+        int nb_channels =
+            av_get_channel_layout_nb_channels(
+                aresample->outsamplesref->audio->channel_layout);
+        avfilter_unref_buffer(aresample->outsamplesref);
+        while (nb_channels--) {
+            av_freep(&(aresample->cached_data[nb_channels]));
+            av_freep(&(aresample->resampled_data[nb_channels]));
+        }
+    }
+
+    if (aresample->resample)
+        av_resample_close(aresample->resample);
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AVFilterLink *inlink = ctx->inputs[0];
+    AResampleContext *aresample = ctx->priv;
+
+    if (aresample->out_rate == -1)
+        aresample->out_rate = outlink->sample_rate;
+    else
+        outlink->sample_rate = aresample->out_rate;
+
+    //TODO: make the resampling parameters configurable
+    aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
+                                           16, 10, 0, 0.8);
+
+    aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
+
+    av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
+           inlink->sample_rate, outlink->sample_rate);
+    return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+
+    avfilter_add_format(&formats, AV_SAMPLE_FMT_S16);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    avfilter_set_common_sample_formats(ctx, formats);
+
+    formats = avfilter_all_channel_layouts();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    avfilter_set_common_channel_layouts(ctx, formats);
+
+    formats = avfilter_all_packing_formats();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    avfilter_set_common_packing_formats(ctx, formats);
+
+    return 0;
+}
+
+static void deinterleave(int16_t **outp, int16_t *in,
+                         int nb_channels, int nb_samples)
+{
+    int16_t *out[8];
+    memcpy(out, outp, nb_channels * sizeof(int16_t*));
+
+    switch (nb_channels) {
+    case 2:
+        while (nb_samples--) {
+            *out[0]++ = *in++;
+            *out[1]++ = *in++;
+        }
+        break;
+    case 3:
+        while (nb_samples--) {
+            *out[0]++ = *in++;
+            *out[1]++ = *in++;
+            *out[2]++ = *in++;
+        }
+        break;
+    case 4:
+        while (nb_samples--) {
+            *out[0]++ = *in++;
+            *out[1]++ = *in++;
+            *out[2]++ = *in++;
+            *out[3]++ = *in++;
+        }
+        break;
+    case 5:
+        while (nb_samples--) {
+            *out[0]++ = *in++;
+            *out[1]++ = *in++;
+            *out[2]++ = *in++;
+            *out[3]++ = *in++;
+            *out[4]++ = *in++;
+        }
+        break;
+    case 6:
+        while (nb_samples--) {
+            *out[0]++ = *in++;
+            *out[1]++ = *in++;
+            *out[2]++ = *in++;
+            *out[3]++ = *in++;
+            *out[4]++ = *in++;
+            *out[5]++ = *in++;
+        }
+        break;
+    case 8:
+        while (nb_samples--) {
+            *out[0]++ = *in++;
+            *out[1]++ = *in++;
+            *out[2]++ = *in++;
+            *out[3]++ = *in++;
+            *out[4]++ = *in++;
+            *out[5]++ = *in++;
+            *out[6]++ = *in++;
+            *out[7]++ = *in++;
+        }
+        break;
+    }
+}
+
+static void interleave(int16_t *out, int16_t **inp,
+        int nb_channels, int nb_samples)
+{
+    int16_t *in[8];
+    memcpy(in, inp, nb_channels * sizeof(int16_t*));
+
+    switch (nb_channels) {
+    case 2:
+        while (nb_samples--) {
+            *out++ = *in[0]++;
+            *out++ = *in[1]++;
+        }
+        break;
+    case 3:
+        while (nb_samples--) {
+            *out++ = *in[0]++;
+            *out++ = *in[1]++;
+            *out++ = *in[2]++;
+        }
+        break;
+    case 4:
+        while (nb_samples--) {
+            *out++ = *in[0]++;
+            *out++ = *in[1]++;
+            *out++ = *in[2]++;
+            *out++ = *in[3]++;
+        }
+        break;
+    case 5:
+        while (nb_samples--) {
+            *out++ = *in[0]++;
+            *out++ = *in[1]++;
+            *out++ = *in[2]++;
+            *out++ = *in[3]++;
+            *out++ = *in[4]++;
+        }
+        break;
+    case 6:
+        while (nb_samples--) {
+            *out++ = *in[0]++;
+            *out++ = *in[1]++;
+            *out++ = *in[2]++;
+            *out++ = *in[3]++;
+            *out++ = *in[4]++;
+            *out++ = *in[5]++;
+        }
+        break;
+    case 8:
+        while (nb_samples--) {
+            *out++ = *in[0]++;
+            *out++ = *in[1]++;
+            *out++ = *in[2]++;
+            *out++ = *in[3]++;
+            *out++ = *in[4]++;
+            *out++ = *in[5]++;
+            *out++ = *in[6]++;
+            *out++ = *in[7]++;
+        }
+        break;
+    }
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+{
+    AResampleContext *aresample  = inlink->dst->priv;
+    AVFilterLink * const outlink = inlink->dst->outputs[0];
+    int i,
+        in_nb_samples            = insamplesref->audio->nb_samples,
+        cached_nb_samples        = in_nb_samples + aresample->unconsumed_nb_samples,
+        requested_out_nb_samples = aresample->ratio * cached_nb_samples,
+        nb_channels              =
+            av_get_channel_layout_nb_channels(inlink->channel_layout);
+
+    if (cached_nb_samples > aresample->max_cached_nb_samples) {
+        for (i = 0; i < nb_channels; i++) {
+            aresample->cached_data[i]    =
+                av_realloc(aresample->cached_data[i], cached_nb_samples * sizeof(int16_t));
+            aresample->resampled_data[i] =
+                av_realloc(aresample->resampled_data[i],
+                           FFALIGN(sizeof(int16_t) * requested_out_nb_samples, 16));
+
+            if (aresample->cached_data[i] == NULL || aresample->resampled_data[i] == NULL)
+                return;
+        }
+        aresample->max_cached_nb_samples = cached_nb_samples;
+
+        if (aresample->outsamplesref)
+            avfilter_unref_buffer(aresample->outsamplesref);
+
+        aresample->outsamplesref = avfilter_get_audio_buffer(outlink,
+                                                            AV_PERM_WRITE | AV_PERM_REUSE2,
+                                                            inlink->format,
+                                                            requested_out_nb_samples,
+                                                            insamplesref->audio->channel_layout,
+                                                            insamplesref->audio->planar);
+
+        avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
+        aresample->outsamplesref->pts =
+            insamplesref->pts / inlink->sample_rate * outlink->sample_rate;
+        aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
+        outlink->out_buf = aresample->outsamplesref;
+    }
+
+    /* av_resample() works with planar audio buffers */
+    if (!inlink->planar && nb_channels > 1) {
+        int16_t *out[8];
+        for (i = 0; i < nb_channels; i++)
+            out[i] = aresample->cached_data[i] + aresample->unconsumed_nb_samples;
+
+        deinterleave(out, (int16_t *)insamplesref->data[0],
+                     nb_channels, in_nb_samples);
+    } else {
+        for (i = 0; i < nb_channels; i++)
+            memcpy(aresample->cached_data[i] + aresample->unconsumed_nb_samples,
+                   insamplesref->data[i],
+                   in_nb_samples * sizeof(int16_t));
+    }
+
+    for (i = 0; i < nb_channels; i++) {
+        int consumed_nb_samples;
+        const int is_last = i+1 == nb_channels;
+
+        aresample->outsamplesref->audio->nb_samples =
+            av_resample(aresample->resample,
+                        aresample->resampled_data[i], aresample->cached_data[i],
+                        &consumed_nb_samples,
+                        cached_nb_samples,
+                        requested_out_nb_samples, is_last);
+
+        /* move unconsumed data back to the beginning of the cache */
+        aresample->unconsumed_nb_samples = cached_nb_samples - consumed_nb_samples;
+        memmove(aresample->cached_data[i],
+                aresample->cached_data[i] + consumed_nb_samples,
+                aresample->unconsumed_nb_samples * sizeof(int16_t));
+    }
+
+
+    /* copy resampled data to the output samplesref */
+    if (!inlink->planar && nb_channels > 1) {
+        interleave((int16_t *)aresample->outsamplesref->data[0],
+                   aresample->resampled_data,
+                   nb_channels, aresample->outsamplesref->audio->nb_samples);
+    } else {
+        for (i = 0; i < nb_channels; i++)
+            memcpy(aresample->outsamplesref->data[i], aresample->resampled_data[i],
+                   aresample->outsamplesref->audio->nb_samples * sizeof(int16_t));
+    }
+
+    avfilter_filter_samples(outlink, avfilter_ref_buffer(aresample->outsamplesref, ~0));
+    avfilter_unref_buffer(insamplesref);
+}
+
+AVFilter avfilter_af_aresample = {
+    .name          = "aresample",
+    .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
+    .init          = init,
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .priv_size     = sizeof(AResampleContext),
+
+    .inputs    = (AVFilterPad[]) {{ .name            = "default",
+                                    .type            = AVMEDIA_TYPE_AUDIO,
+                                    .filter_samples  = filter_samples,
+                                    .min_perms       = AV_PERM_READ, },
+                                  { .name = NULL}},
+    .outputs   = (AVFilterPad[]) {{ .name            = "default",
+                                    .config_props    = config_output,
+                                    .type            = AVMEDIA_TYPE_AUDIO, },
+                                  { .name = NULL}},
+};

libavfilter/allfilters.c

@@ -36,6 +36,7 @@ void avfilter_register_all(void)
 
     REGISTER_FILTER (AFORMAT,     aformat,     af);
     REGISTER_FILTER (ANULL,       anull,       af);
+    REGISTER_FILTER (ARESAMPLE,   aresample,   af);
 
     REGISTER_FILTER (ANULLSRC,    anullsrc,    asrc);
 

libavfilter/avfilter.h

@@ -29,8 +29,8 @@
 #include "libavutil/rational.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  2
-#define LIBAVFILTER_VERSION_MINOR 31
-#define LIBAVFILTER_VERSION_MICRO  1
+#define LIBAVFILTER_VERSION_MINOR 32
+#define LIBAVFILTER_VERSION_MICRO  0
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
                                                LIBAVFILTER_VERSION_MINOR, \