[FFmpeg-cvslog] lavfi: add amerge audio filter.
Nicolas George
git at videolan.org
Sat Dec 31 16:26:55 CET 2011
ffmpeg | branch: master | Nicolas George <nicolas.george at normalesup.org> | Sun Nov 6 21:28:05 2011 +0100| [4962edf88926ae2b06260eb50dbc96dcd4199784] | committer: Nicolas George
lavfi: add amerge audio filter.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=4962edf88926ae2b06260eb50dbc96dcd4199784
---
Changelog | 1 +
doc/filters.texi | 33 ++++++
libavfilter/Makefile | 1 +
libavfilter/af_amerge.c | 288 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/avfilter.h | 2 +-
6 files changed, 325 insertions(+), 1 deletions(-)
diff --git a/Changelog b/Changelog
index e24f39d..965b2b2 100644
--- a/Changelog
+++ b/Changelog
@@ -13,6 +13,7 @@ version next:
- asplit audio filter
- tinterlace video filter
- astreamsync audio filter
+- amerge audio filter
version 0.9:
diff --git a/doc/filters.texi b/doc/filters.texi
index 487ec21..3109d0d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -156,6 +156,39 @@ aformat=u8\\,s16:mono:packed
aformat=s16:mono\\,stereo:all
@end example
+ at section amerge
+
+Merge two audio streams into a single multi-channel stream.
+
+This filter does not need any argument.
+
+If the channel layouts of the inputs are disjoint, and therefore compatible,
+the channel layout of the output will be set accordingly and the channels
+will be reordered as necessary. If the channel layouts of the inputs are not
+disjoint, the output will have all the channels of the first input then all
+the channels of the second input, in that order, and the channel layout of
+the output will be the default value corresponding to the total number of
+channels.
+
+For example, if the first input is in 2.1 (FL+FR+LF) and the second input
+is FC+BL+BR, then the output will be in 5.1, with the channels in the
+following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the
+first input, b1 is the first channel of the second input).
+
+On the other hand, if both input are in stereo, the output channels will be
+in the default order: a1, a2, b1, b2, and the channel layout will be
+arbitrarily set to 4.0, which may or may not be the expected value.
+
+Both inputs must have the same sample rate, format and packing.
+
+If inputs do not have the same duration, the output will stop with the
+shortest.
+
+Example: merge two mono files into a stereo stream:
+ at example
+amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
+ at end example
+
@section anull
Pass the audio source unchanged to the output.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 318039e..0d8f120 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -26,6 +26,7 @@ OBJS-$(CONFIG_AVCODEC) += avcodec.o
OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
+OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
diff --git a/libavfilter/af_amerge.c b/libavfilter/af_amerge.c
new file mode 100644
index 0000000..736379c
--- /dev/null
+++ b/libavfilter/af_amerge.c
@@ -0,0 +1,288 @@
+/*
+ * Copyright (c) 2011 Nicolas George <nicolas.george at normalesup.org>
+ *
+ * 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 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
+ * Audio merging filter
+ */
+
+#include "libswresample/swresample.h" // only for SWR_CH_MAX
+#include "avfilter.h"
+#include "internal.h"
+
+#define QUEUE_SIZE 16
+
+typedef struct {
+ int nb_in_ch[2]; /**< number of channels for each input */
+ int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
+ int bps;
+ struct amerge_queue {
+ AVFilterBufferRef *buf[QUEUE_SIZE];
+ int nb_buf, nb_samples, pos;
+ } queue[2];
+} AMergeContext;
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AMergeContext *am = ctx->priv;
+ int i, j;
+
+ for (i = 0; i < 2; i++)
+ for (j = 0; j < am->queue[i].nb_buf; j++)
+ avfilter_unref_buffer(am->queue[i].buf[j]);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AMergeContext *am = ctx->priv;
+ int64_t inlayout[2], outlayout;
+ const int packing_fmts[] = { AVFILTER_PACKED, -1 };
+ AVFilterFormats *formats;
+ int i;
+
+ for (i = 0; i < 2; i++) {
+ if (!ctx->inputs[i]->in_chlayouts ||
+ !ctx->inputs[i]->in_chlayouts->format_count) {
+ av_log(ctx, AV_LOG_ERROR,
+ "No channel layout for input %d\n", i + 1);
+ return AVERROR(EINVAL);
+ }
+ inlayout[i] = ctx->inputs[i]->in_chlayouts->formats[0];
+ if (ctx->inputs[i]->in_chlayouts->format_count > 1) {
+ char buf[256];
+ av_get_channel_layout_string(buf, sizeof(buf), 0, inlayout[i]);
+ av_log(ctx, AV_LOG_INFO, "Using \"%s\" for input %d\n", buf, i + 1);
+ }
+ am->nb_in_ch[i] = av_get_channel_layout_nb_channels(inlayout[i]);
+ }
+ if (am->nb_in_ch[0] + am->nb_in_ch[1] > SWR_CH_MAX) {
+ av_log(ctx, AV_LOG_ERROR, "Too many channels (max %d)\n", SWR_CH_MAX);
+ return AVERROR(EINVAL);
+ }
+ if (inlayout[0] & inlayout[1]) {
+ av_log(ctx, AV_LOG_WARNING,
+ "Inputs overlap: output layout will be meaningless\n");
+ for (i = 0; i < am->nb_in_ch[0] + am->nb_in_ch[1]; i++)
+ am->route[i] = i;
+ outlayout = av_get_default_channel_layout(am->nb_in_ch[0] +
+ am->nb_in_ch[1]);
+ if (!outlayout)
+ outlayout = ((int64_t)1 << (am->nb_in_ch[0] + am->nb_in_ch[1])) - 1;
+ } else {
+ int *route[2] = { am->route, am->route + am->nb_in_ch[0] };
+ int c, out_ch_number = 0;
+
+ outlayout = inlayout[0] | inlayout[1];
+ for (c = 0; c < 64; c++)
+ for (i = 0; i < 2; i++)
+ if ((inlayout[i] >> c) & 1)
+ *(route[i]++) = out_ch_number++;
+ }
+ formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
+ avfilter_set_common_sample_formats(ctx, formats);
+ formats = avfilter_make_format_list(packing_fmts);
+ avfilter_set_common_packing_formats(ctx, formats);
+ for (i = 0; i < 2; i++) {
+ formats = NULL;
+ avfilter_add_format(&formats, inlayout[i]);
+ avfilter_formats_ref(formats, &ctx->inputs[i]->out_chlayouts);
+ }
+ formats = NULL;
+ avfilter_add_format(&formats, outlayout);
+ avfilter_formats_ref(formats, &ctx->outputs[0]->in_chlayouts);
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AMergeContext *am = ctx->priv;
+ int64_t layout;
+ char name[3][256];
+ int i;
+
+ if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Inputs must have the same sample rate "
+ "(%"PRIi64" vs %"PRIi64")\n",
+ ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
+ return AVERROR(EINVAL);
+ }
+ am->bps = av_get_bytes_per_sample(ctx->outputs[0]->format);
+ outlink->sample_rate = ctx->inputs[0]->sample_rate;
+ outlink->time_base = ctx->inputs[0]->time_base;
+ for (i = 0; i < 3; i++) {
+ layout = (i < 2 ? ctx->inputs[i] : ctx->outputs[0])->channel_layout;
+ av_get_channel_layout_string(name[i], sizeof(name[i]), -1, layout);
+ }
+ av_log(ctx, AV_LOG_INFO,
+ "in1:%s + in2:%s -> out:%s\n", name[0], name[1], name[2]);
+ return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AMergeContext *am = ctx->priv;
+ int i;
+
+ for (i = 0; i < 2; i++)
+ if (!am->queue[i].nb_samples)
+ avfilter_request_frame(ctx->inputs[i]);
+ return 0;
+}
+
+/**
+ * Copy samples from two input streams to one output stream.
+ * @param nb_in_ch number of channels in each input stream
+ * @param route routing values;
+ * input channel i goes to output channel route[i];
+ * i < nb_in_ch[0] are the channels from the first output;
+ * i >= nb_in_ch[0] are the channels from the second output
+ * @param ins pointer to the samples of each inputs, in packed format;
+ * will be left at the end of the copied samples
+ * @param outs pointer to the samples of the output, in packet format;
+ * must point to a buffer big enough;
+ * will be left at the end of the copied samples
+ * @param ns number of samples to copy
+ * @param bps bytes per sample
+ */
+static inline void copy_samples(int nb_in_ch[2], int *route, uint8_t *ins[2],
+ uint8_t **outs, int ns, int bps)
+{
+ int *route_cur;
+ int i, c;
+
+ while (ns--) {
+ route_cur = route;
+ for (i = 0; i < 2; i++) {
+ for (c = 0; c < nb_in_ch[i]; c++) {
+ memcpy((*outs) + bps * *(route_cur++), ins[i], bps);
+ ins[i] += bps;
+ }
+ }
+ *outs += (nb_in_ch[0] + nb_in_ch[1]) * bps;
+ }
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AMergeContext *am = ctx->priv;
+ int input_number = inlink == ctx->inputs[1];
+ struct amerge_queue *inq = &am->queue[input_number];
+ int nb_samples, ns, i;
+ AVFilterBufferRef *outbuf, **inbuf[2];
+ uint8_t *ins[2], *outs;
+
+ if (inq->nb_buf == QUEUE_SIZE) {
+ av_log(ctx, AV_LOG_ERROR, "Packet queue overflow; dropped\n");
+ avfilter_unref_buffer(insamples);
+ return;
+ }
+ inq->buf[inq->nb_buf++] = avfilter_ref_buffer(insamples, AV_PERM_READ |
+ AV_PERM_PRESERVE);
+ inq->nb_samples += insamples->audio->nb_samples;
+ avfilter_unref_buffer(insamples);
+ if (!am->queue[!input_number].nb_samples)
+ return;
+
+ nb_samples = FFMIN(am->queue[0].nb_samples,
+ am->queue[1].nb_samples);
+ outbuf = avfilter_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE,
+ nb_samples);
+ outs = outbuf->data[0];
+ for (i = 0; i < 2; i++) {
+ inbuf[i] = am->queue[i].buf;
+ ins[i] = (*inbuf[i])->data[0] +
+ am->queue[i].pos * am->nb_in_ch[i] * am->bps;
+ }
+ while (nb_samples) {
+ ns = nb_samples;
+ for (i = 0; i < 2; i++)
+ ns = FFMIN(ns, (*inbuf[i])->audio->nb_samples - am->queue[i].pos);
+ /* Unroll the most common sample formats: speed +~350% for the loop,
+ +~13% overall (including two common decoders) */
+ switch (am->bps) {
+ case 1:
+ copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 1);
+ break;
+ case 2:
+ copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 2);
+ break;
+ case 4:
+ copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 4);
+ break;
+ default:
+ copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, am->bps);
+ break;
+ }
+
+ nb_samples -= ns;
+ for (i = 0; i < 2; i++) {
+ am->queue[i].nb_samples -= ns;
+ am->queue[i].pos += ns;
+ if (am->queue[i].pos == (*inbuf[i])->audio->nb_samples) {
+ am->queue[i].pos = 0;
+ avfilter_unref_buffer(*inbuf[i]);
+ *inbuf[i] = NULL;
+ inbuf[i]++;
+ ins[i] = *inbuf[i] ? (*inbuf[i])->data[0] : NULL;
+ }
+ }
+ }
+ for (i = 0; i < 2; i++) {
+ int nbufused = inbuf[i] - am->queue[i].buf;
+ if (nbufused) {
+ am->queue[i].nb_buf -= nbufused;
+ memmove(am->queue[i].buf, inbuf[i],
+ am->queue[i].nb_buf * sizeof(**inbuf));
+ }
+ }
+ avfilter_filter_samples(ctx->outputs[0], outbuf);
+}
+
+AVFilter avfilter_af_amerge = {
+ .name = "amerge",
+ .description = NULL_IF_CONFIG_SMALL("Merge two audio streams into "
+ "a single multi-channel stream."),
+ .priv_size = sizeof(AMergeContext),
+ .uninit = uninit,
+ .query_formats = query_formats,
+
+ .inputs = (const AVFilterPad[]) {
+ { .name = "in1",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ, },
+ { .name = "in2",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ, },
+ { .name = NULL }
+ },
+ .outputs = (const AVFilterPad[]) {
+ { .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ .request_frame = request_frame, },
+ { .name = NULL }
+ },
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index cba7704..621568e 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -36,6 +36,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ACONVERT, aconvert, af);
REGISTER_FILTER (AFORMAT, aformat, af);
+ REGISTER_FILTER (AMERGE, amerge, af);
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (ARESAMPLE, aresample, af);
REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h
index 0daa84b..2ab2387 100644
--- a/libavfilter/avfilter.h
+++ b/libavfilter/avfilter.h
@@ -30,7 +30,7 @@
#include "libavcodec/avcodec.h"
#define LIBAVFILTER_VERSION_MAJOR 2
-#define LIBAVFILTER_VERSION_MINOR 56
+#define LIBAVFILTER_VERSION_MINOR 57
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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