[FFmpeg-devel] [PATCH] lavfi: reimplement MPlayer's af_pan filter for libavfilter.
Nicolas George
nicolas.george at normalesup.org
Wed Nov 16 14:32:43 CET 2011
From: Clément Bœsch <ubitux at gmail.com>
Signed-off-by: Nicolas George <nicolas.george at normalesup.org>
---
Changelog | 1 +
doc/filters.texi | 48 ++++++++
libavfilter/Makefile | 1 +
libavfilter/af_pan.c | 303 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
5 files changed, 354 insertions(+), 0 deletions(-)
create mode 100644 libavfilter/af_pan.c
diff --git a/Changelog b/Changelog
index f635e88..561cd0b 100644
--- a/Changelog
+++ b/Changelog
@@ -122,6 +122,7 @@ easier to use. The changes are:
- VBLE Decoder
- OS X Video Decoder Acceleration (VDA) support
- compact output in ffprobe
+- pan audio filter added
version 0.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index 9fa9c04..d5c05b2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -235,6 +235,54 @@ the listener (standard for speakers).
Ported from SoX.
+ at section pan
+
+Mix channels with specific gain levels. The filter accepts the output
+channel layout followed by a set of channels definitions.
+
+The filter accepts parameters of the form:
+"@var{l}:@var{outdef}:@var{outdef}:..."
+
+ at table @option
+ at item l
+output channel layout or number of channels
+
+ at item outdef
+output channel specification, of the form:
+"@var{out_name}=[@var{gain}*]@var{in_name}[+[@var{gain}*]@var{in_name}...]"
+
+ at item out_name
+output channel to define, either a channel name (FL, FR, etc.) or a channel
+number (c0, c1, etc.)
+
+ at item gain
+multiplicative coefficient for the channel, 1 leaving the volume unchanged
+
+ at item in_name
+input channel to use, see out_name for details; it is not possible to mix
+named and numbered input channels
+ at end table
+
+If the `=' in a channel specification is replaced by `<', then the gains for
+that specification will be renormalized so that the total is 1, thus
+avoiding clipping noise.
+
+For example, if you want to down-mix from stereo to mono, but with a bigger
+factor for the left channel:
+ at example
+af pan=1:c0=0.9*c0+0.1*c1
+ at end example
+
+A customized down-mix to stereo that works automatically for 3-, 4-, 5- and
+7-channels surround:
+ at example
+pan=stereo:FL<FL+0.5*FC+0.6*BL+0.6*SL:FR<FR+0.5*FC+0.6*BR+0.6*SR
+ at end example
+
+Note that @file{ffmpeg} integrates a default down-mix (and up-mix) system
+that should be preferred (see "-ac" option) unless you have very specific
+needs.
+
@section volume
Adjust the input audio volume.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8e43be8..cab6f2e 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -29,6 +29,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
+OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o
diff --git a/libavfilter/af_pan.c b/libavfilter/af_pan.c
new file mode 100644
index 0000000..03d4160
--- /dev/null
+++ b/libavfilter/af_pan.c
@@ -0,0 +1,303 @@
+/*
+ * Copyright (C) 2002 Anders Johansson <ajh at atri.curtin.edu.au>
+ * Copyright (C) 2011 Clément Bœsch <ubitux at gmail.com>
+ * 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 panning filter (channels mixing)
+ * Original code written by Anders Johansson for MPlayer,
+ * reimplemented for FFmpeg.
+ */
+
+#include <stdio.h>
+#include "libavutil/audioconvert.h"
+#include "libavutil/avstring.h"
+#include "avfilter.h"
+#include "internal.h"
+
+#define MAX_CHANNELS 63
+
+typedef struct {
+ int64_t out_channel_layout;
+ union {
+ double d[MAX_CHANNELS][MAX_CHANNELS];
+ int i[MAX_CHANNELS][MAX_CHANNELS]; // 1:7:8 fixed point
+ } gain;
+ int64_t need_renorm;
+ int need_renumber;
+ int nb_input_channels;
+ int nb_output_channels;
+} PanContext;
+
+static int parse_channel_name(char **arg, int *rchannel, int *rnamed)
+{
+ char buf[8];
+ int len, i, channel;
+ int64_t layout, layout0;
+
+ if (sscanf(*arg, " %7[A-Z] %n", buf, &len)) {
+ layout0 = layout = av_get_channel_layout(buf);
+ for (i = 32; i > 0; i >>= 1) {
+ if (layout >= (int64_t)1 << i) {
+ channel += i;
+ layout >>= i;
+ }
+ }
+ if (channel >= MAX_CHANNELS || layout0 != (int64_t)1 << channel)
+ return AVERROR(EINVAL);
+ *rchannel = channel;
+ *rnamed = 1;
+ *arg += len;
+ return 0;
+ }
+ if (sscanf(*arg, " c%d %n", &channel, &len) &&
+ channel >= 0 && channel < MAX_CHANNELS) {
+ *rchannel = channel;
+ *rnamed = 0;
+ *arg += len;
+ return 0;
+ }
+ return AVERROR(EINVAL);
+}
+
+static void skip_spaces(char **arg)
+{
+ int len = 0;
+
+ sscanf(*arg, " %n", &len);
+ *arg += len;
+}
+
+static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
+{
+ PanContext *const pan = ctx->priv;
+ char *arg, *arg0, *tokenizer, *args = av_strdup(args0);
+ int out_ch_id, in_ch_id, len, named;
+ int nb_named[2] = { 0, 0 }; // number of unnamed and named input channels
+ double gain;
+
+ if (!args)
+ return AVERROR(ENOMEM);
+ arg = av_strtok(args, ":", &tokenizer);
+ pan->out_channel_layout = av_get_channel_layout(arg);
+ if (!pan->out_channel_layout) {
+ av_log(ctx, AV_LOG_ERROR, "Unknown channel layout \"%s\"\n", arg);
+ return AVERROR(EINVAL);
+ }
+ pan->nb_output_channels = av_get_channel_layout_nb_channels(pan->out_channel_layout);
+
+ /* parse channel specifications */
+ while ((arg = arg0 = av_strtok(NULL, ":", &tokenizer))) {
+ /* channel name */
+ if (parse_channel_name(&arg, &out_ch_id, &named)) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Expected out channel name, got \"%.8s\"\n", arg);
+ return AVERROR(EINVAL);
+ }
+ if (named) {
+ if (!((pan->out_channel_layout >> out_ch_id) & 1)) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Channel \"%.8s\" does not exist in the chosen layout\n", arg0);
+ return AVERROR(EINVAL);
+ }
+ /* get the channel number in the output channel layout:
+ * out_channel_layout & ((1 << out_ch_id) - 1) are all the
+ * channels that come before out_ch_id,
+ * so their count is the index of out_ch_id */
+ out_ch_id = av_get_channel_layout_nb_channels(pan->out_channel_layout & (((int64_t)1 << out_ch_id) - 1));
+ }
+ if (out_ch_id < 0 || out_ch_id >= pan->nb_output_channels) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Invalid out channel name \"%.8s\"\n", arg0);
+ return AVERROR(EINVAL);
+ }
+ if (*arg == '=') {
+ arg++;
+ } else if (*arg == '<') {
+ pan->need_renorm |= (int64_t)1 << out_ch_id;
+ arg++;
+ } else {
+ av_log(ctx, AV_LOG_ERROR,
+ "Syntax error after channel name in \"%.8s\"\n", arg0);
+ return AVERROR(EINVAL);
+ }
+ /* gains */
+ while (1) {
+ gain = 1;
+ if (sscanf(arg, " %lf %n* %n", &gain, &len, &len))
+ arg += len;
+ if (parse_channel_name(&arg, &in_ch_id, &named)){
+ av_log(ctx, AV_LOG_ERROR,
+ "Expected in channel name, got \"%.8s\"\n", arg);
+ return AVERROR(EINVAL);
+ }
+ nb_named[named]++;
+ if (nb_named[!named]) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Can not mix named and numbered channels\n");
+ return AVERROR(EINVAL);
+ }
+ pan->gain.d[out_ch_id][in_ch_id] = gain;
+ if (!*arg)
+ break;
+ if (*arg != '+') {
+ av_log(ctx, AV_LOG_ERROR, "Syntax error near \"%.8s\"\n", arg);
+ return AVERROR(EINVAL);
+ }
+ arg++;
+ skip_spaces(&arg);
+ }
+ }
+ pan->need_renumber = !!nb_named[1];
+
+ av_free(args);
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ PanContext *pan = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFilterFormats *formats;
+
+ const enum AVSampleFormat sample_fmts[] = {AV_SAMPLE_FMT_S16, -1};
+ const int packing_fmts[] = {AVFILTER_PACKED, -1};
+
+ avfilter_set_common_sample_formats (ctx, avfilter_make_format_list(sample_fmts));
+ avfilter_set_common_packing_formats(ctx, avfilter_make_format_list(packing_fmts));
+
+ // inlink supports any channel layout
+ formats = avfilter_make_all_channel_layouts();
+ avfilter_formats_ref(formats, &inlink->out_chlayouts);
+
+ // outlink supports only requested output channel layout
+ formats = NULL;
+ avfilter_add_format(&formats, pan->out_channel_layout);
+ avfilter_formats_ref(formats, &outlink->in_chlayouts);
+ return 0;
+}
+
+static int config_props(AVFilterLink *link)
+{
+ AVFilterContext *ctx = link->dst;
+ PanContext *pan = ctx->priv;
+ char buf[1024], *cur;
+ int i, j, k, r;
+ double t;
+
+ pan->nb_input_channels = av_get_channel_layout_nb_channels(link->channel_layout);
+ if (pan->need_renumber) {
+ // input channels were given by their name: renumber them
+ for (i = j = 0; i < MAX_CHANNELS; i++) {
+ if ((link->channel_layout >> i) & 1) {
+ for (k = 0; k < pan->nb_output_channels; k++)
+ pan->gain.d[k][j] = pan->gain.d[k][i];
+ j++;
+ }
+ }
+ }
+ // renormalize
+ for (i = 0; i < pan->nb_output_channels; i++) {
+ if (!((pan->need_renorm >> i) & 1))
+ continue;
+ t = 0;
+ for (j = 0; j < pan->nb_input_channels; j++)
+ t += pan->gain.d[i][j];
+ if (t > -1E-5 && t < 1E-5) {
+ if (t)
+ av_log(ctx, AV_LOG_WARNING,
+ "Degenerate coefficients while renormalizing\n");
+ continue;
+ }
+ for (j = 0; j < pan->nb_input_channels; j++)
+ pan->gain.d[i][j] /= t;
+ }
+ // summary
+ for (i = 0; i < pan->nb_output_channels; i++) {
+ cur = buf;
+ for (j = 0; j < pan->nb_input_channels; j++) {
+ r = snprintf(cur, buf + sizeof(buf) - cur, "%s%.3g i%d",
+ j ? " + " : "", pan->gain.d[i][j], j);
+ cur += FFMIN(buf + sizeof(buf) - cur, r);
+ }
+ av_log(ctx, AV_LOG_INFO, "o%d = %s\n", i, buf);
+ }
+ // convert to integer
+ for (i = 0; i < pan->nb_output_channels; i++) {
+ for (j = 0; j < pan->nb_input_channels; j++) {
+ if (pan->gain.d[i][j] < -256 || pan->gain.d[i][j] > 256)
+ av_log(ctx, AV_LOG_WARNING, "Gain too large, clamped\n");
+ pan->gain.i[i][j] = FFMIN(256, FFMAX(-256, pan->gain.d[i][j])) * 256.0;
+ }
+ }
+ return 0;
+}
+
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+{
+ PanContext *const pan = inlink->dst->priv;
+ int i, o, n = insamples->audio->nb_samples;
+
+ /* input */
+ const int16_t *in = (int16_t *)insamples->data[0];
+ const int16_t *in_end = in + n * pan->nb_input_channels;
+
+ /* output */
+ AVFilterLink *const outlink = inlink->dst->outputs[0];
+ AVFilterBufferRef *outsamples = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n);
+ int16_t *out = (int16_t *)outsamples->data[0];
+
+ for (; in < in_end; in += pan->nb_input_channels) {
+ for (o = 0; o < pan->nb_output_channels; o++) {
+ int v = 0;
+ for (i = 0; i < pan->nb_input_channels; i++)
+ v += pan->gain.i[o][i] * in[i];
+ *(out++) = v >> 8;
+ }
+ }
+
+ avfilter_filter_samples(outlink, outsamples);
+ avfilter_unref_buffer(insamples);
+}
+
+AVFilter avfilter_af_pan = {
+ .name = "pan",
+ .description = NULL_IF_CONFIG_SMALL("Remix channels with coefficients (panning)"),
+ .priv_size = sizeof(PanContext),
+ .init = init,
+ .query_formats = query_formats,
+
+ .inputs = (const AVFilterPad[]) {
+ { .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_props,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ, },
+ { .name = NULL}
+ },
+ .outputs = (const AVFilterPad[]) {
+ { .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO, },
+ { .name = NULL}
+ },
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index e0e5c6f..c4b6972 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -40,6 +40,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ARESAMPLE, aresample, af);
REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
REGISTER_FILTER (EARWAX, earwax, af);
+ REGISTER_FILTER (PAN, pan, af);
REGISTER_FILTER (VOLUME, volume, af);
REGISTER_FILTER (ABUFFER, abuffer, asrc);
--
1.7.7.1
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