[FFmpeg-devel] [PATCH 4/4] lavfi: add audio resample filter
Mina Nagy Zaki
mnzaki at gmail.com
Wed Jul 27 00:05:20 CEST 2011
From: Stefano Sabatini <stefano.sabatini-lala at poste.it>
---
libavfilter/Makefile | 1 +
libavfilter/af_aresample.c | 328 ++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
3 files changed, 330 insertions(+), 0 deletions(-)
create mode 100644 libavfilter/af_aresample.c
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 0e6051b..237b79b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -21,6 +21,7 @@ OBJS-$(CONFIG_AVCODEC) += avcodec.o
OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.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
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c
new file mode 100644
index 0000000..49450c4
--- /dev/null
+++ b/libavfilter/af_aresample.c
@@ -0,0 +1,328 @@
+/*
+ * 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"
+
+typedef struct {
+ struct AVResampleContext *resample;
+ int out_rate;
+ double ratio;
+ AVFilterBufferRef *outsamplesref;
+ int unconsumed_nb_samples,
+ cached_nb_samples;
+ int16_t *cached_data[8],
+ *resampled_data[8];
+} ResampleContext;
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+ ResampleContext *resample = ctx->priv;
+ char rate_str[128] = "", *tail;
+ resample->out_rate = -1;
+
+ if (args)
+ sscanf(args, "%127[a-z0-9]", rate_str);
+
+ if (*rate_str) {
+ double d = av_strtod(rate_str, &tail);
+ if (*tail || d < 0 || (int)d != d) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid value '%s' for rate",
+ rate_str);
+ return AVERROR(EINVAL);
+ }
+ resample->out_rate = d;
+ }
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ResampleContext *resample = ctx->priv;
+ int nb_channels;
+ if (resample->outsamplesref) {
+ nb_channels =
+ av_get_channel_layout_nb_channels(
+ resample->outsamplesref->audio->channel_layout);
+ avfilter_unref_buffer(resample->outsamplesref);
+
+ while (nb_channels--) {
+ av_freep(&resample->cached_data[nb_channels]);
+ av_freep(&resample->resampled_data[nb_channels]);
+ }
+ }
+
+ if (resample->resample)
+ av_resample_close(resample->resample);
+
+ resample->resample = NULL;
+}
+
+static int config_props(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AVFilterLink *inlink = ctx->inputs[0];
+ ResampleContext *resample = ctx->priv;
+
+ if (resample->out_rate == -1)
+ resample->out_rate = inlink->sample_rate;
+ outlink->sample_rate = resample->out_rate;
+
+ /* fixme: make the resampling parameters configurable */
+ resample->resample = av_resample_init(resample->out_rate, inlink->sample_rate,
+ 16, 10, 0, 0.8);
+
+ resample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
+
+ av_log(ctx, AV_LOG_INFO, "r:%"PRId64" -> r:%"PRId64"\n",
+ inlink->sample_rate, outlink->sample_rate);
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ };
+
+ avfilter_set_common_sample_formats(ctx, avfilter_make_format_list(sample_fmts));
+ avfilter_set_common_channel_layouts(ctx, avfilter_all_channel_layouts());
+ avfilter_set_common_packing_formats(ctx, avfilter_all_packing_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*));
+
+ if (nb_channels == 2) {
+ while (nb_samples--) {
+ *out[0]++ = *in++;
+ *out[1]++ = *in++;
+ }
+ } else if (nb_channels == 3) {
+ while (nb_samples--) {
+ *out[0]++ = *in++;
+ *out[1]++ = *in++;
+ *out[2]++ = *in++;
+ }
+ } else if (nb_channels == 4) {
+ while (nb_samples--) {
+ *out[0]++ = *in++;
+ *out[1]++ = *in++;
+ *out[2]++ = *in++;
+ *out[3]++ = *in++;
+ }
+ } else if (nb_channels == 5) {
+ while (nb_samples--) {
+ *out[0]++ = *in++;
+ *out[1]++ = *in++;
+ *out[2]++ = *in++;
+ *out[3]++ = *in++;
+ *out[4]++ = *in++;
+ }
+ } else if (nb_channels == 6) {
+ while (nb_samples--) {
+ *out[0]++ = *in++;
+ *out[1]++ = *in++;
+ *out[2]++ = *in++;
+ *out[3]++ = *in++;
+ *out[4]++ = *in++;
+ *out[5]++ = *in++;
+ }
+ } else if (nb_channels == 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++;
+ }
+ }
+}
+
+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*));
+
+ if (nb_channels == 2) {
+ while (nb_samples--) {
+ *out++ = *in[0]++;
+ *out++ = *in[1]++;
+ }
+ } else if (nb_channels == 3) {
+ while (nb_samples--) {
+ *out++ = *in[0]++;
+ *out++ = *in[1]++;
+ *out++ = *in[2]++;
+ }
+ } else if (nb_channels == 4) {
+ while (nb_samples--) {
+ *out++ = *in[0]++;
+ *out++ = *in[1]++;
+ *out++ = *in[2]++;
+ *out++ = *in[3]++;
+ }
+ } else if (nb_channels == 5) {
+ while (nb_samples--) {
+ *out++ = *in[0]++;
+ *out++ = *in[1]++;
+ *out++ = *in[2]++;
+ *out++ = *in[3]++;
+ *out++ = *in[4]++;
+ }
+ } else if (nb_channels == 6) {
+ while (nb_samples--) {
+ *out++ = *in[0]++;
+ *out++ = *in[1]++;
+ *out++ = *in[2]++;
+ *out++ = *in[3]++;
+ *out++ = *in[4]++;
+ *out++ = *in[5]++;
+ }
+ } else if (nb_channels == 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]++;
+ }
+ }
+
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+{
+ ResampleContext *resample = 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 + resample->unconsumed_nb_samples,
+ requested_out_nb_samples = resample->ratio * cached_nb_samples,
+ nb_channels =
+ av_get_channel_layout_nb_channels(inlink->channel_layout);
+ if (cached_nb_samples > resample->cached_nb_samples) {
+ for (i = 0; i < nb_channels; i++) {
+ resample->cached_data[i] =
+ av_realloc(resample->cached_data[i], cached_nb_samples * sizeof(int16_t));
+ resample->resampled_data[i] =
+ av_realloc(resample->resampled_data[i], 4 * requested_out_nb_samples + 16);
+
+ if (resample->cached_data[i] == NULL || resample->resampled_data[i] == NULL)
+ return;
+ }
+ if (resample->outsamplesref)
+ avfilter_unref_buffer(resample->outsamplesref);
+ resample->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);
+ resample->outsamplesref->audio->sample_rate = outlink->sample_rate;
+ resample->cached_nb_samples = cached_nb_samples;
+ outlink->out_buf = resample->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] = resample->cached_data[i] + resample->unconsumed_nb_samples;
+
+ deinterleave(out, (int16_t *)insamplesref->data[0],
+ nb_channels, in_nb_samples);
+ } else {
+ for (i = 0; i < nb_channels; i++)
+ memcpy(resample->cached_data[i] + resample->unconsumed_nb_samples,
+ insamplesref->data[i],
+ in_nb_samples * sizeof(int16_t));
+ }
+
+ for (i = 0; i < nb_channels; i++) {
+ int consumed;
+ const int is_last = i+1 == nb_channels;
+
+ resample->outsamplesref->audio->nb_samples =
+ av_resample(resample->resample,
+ resample->resampled_data[i], resample->cached_data[i],
+ &consumed,
+ cached_nb_samples,
+ requested_out_nb_samples, is_last);
+
+ /* move unconsumed data back to the beginning of the cache */
+ resample->unconsumed_nb_samples = cached_nb_samples - consumed;
+ memmove(resample->cached_data[i], resample->cached_data[i] + consumed,
+ resample->unconsumed_nb_samples * sizeof(int16_t));
+ }
+
+
+ /* copy resampled data to the output samplesref */
+ if (!inlink->planar && nb_channels > 1) {
+ interleave((int16_t *)resample->outsamplesref->data[0],
+ resample->resampled_data,
+ nb_channels, resample->outsamplesref->audio->nb_samples);
+ } else {
+ for (i = 0; i < nb_channels; i++)
+ memcpy(resample->outsamplesref->data[i], resample->resampled_data[i],
+ resample->outsamplesref->audio->nb_samples * sizeof(int16_t));
+ }
+
+ avfilter_filter_samples(outlink, avfilter_ref_buffer(resample->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(ResampleContext),
+
+ .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_props,
+ .type = AVMEDIA_TYPE_AUDIO, },
+ { .name = NULL}},
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 4a4642d..bfaca1e 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -37,6 +37,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ACONVERT, aconvert, af);
REGISTER_FILTER (AFORMAT, aformat, af);
REGISTER_FILTER (ANULL, anull, af);
+ REGISTER_FILTER (ARESAMPLE, aresample, af);
REGISTER_FILTER (ABUFFER, abuffer, asrc);
REGISTER_FILTER (ANULLSRC, anullsrc, asrc);
--
1.7.4.4
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