[FFmpeg-devel] [PATCH] lavfi: reimplement MPlayer's af_pan filter for libavfilter.
Stefano Sabatini
stefasab at gmail.com
Sun Nov 13 16:41:45 CET 2011
On date Saturday 2011-11-12 12:50:40 +0100, Nicolas George encoded:
> 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 | 304 ++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 5 files changed, 355 insertions(+), 0 deletions(-)
> create mode 100644 libavfilter/af_pan.c
>
> diff --git a/Changelog b/Changelog
> index 7dd82ef..02ca812 100644
> --- a/Changelog
> +++ b/Changelog
> @@ -120,6 +120,7 @@ easier to use. The changes are:
> - Encrypted OMA files support
> - Discworld II BMV decoding support
> - VBLE Decoder
> +- pan audio filter added
>
>
> version 0.8:
> diff --git a/doc/filters.texi b/doc/filters.texi
> index b45df57..7536d62 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..167661d
> --- /dev/null
> +++ b/libavfilter/af_pan.c
> @@ -0,0 +1,304 @@
> +/*
> + * 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_channels_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 int get_channel_number(int64_t layout, int num)
> +{
> + return av_get_channel_layout_nb_channels(layout & (((int64_t)1 << num) - 1));
> +}
this is used just once, please skip the obfuscating wrapper
> +
> +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 };
you may add a comment here explaining what contains nb_named[0] and
nb_named[1]
> + double gain;
> +
> + if (!args)
> + return AVERROR(ENOMEM);
> + arg = av_strtok(args, ":", &tokenizer);
> + pan->out_channels_layout = av_get_channel_layout(arg);
> + if (!pan->out_channels_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_channels_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_channels_layout >> out_ch_id) & 1)) {
> + av_log(ctx, AV_LOG_ERROR,
> + "Channel \"%.8s\" does not exist\n", arg0);
...does not exist in the chosen output layout
> + return AVERROR(EINVAL);
> + }
> + out_ch_id = get_channel_number(pan->out_channels_layout, out_ch_id);
uh? If my understanding is correct, this is a number of channels, so
maybe nb_channels or n may be better, the recursive definition and the
ambiguity on the out_ch_id variable is rather confusing.
> + }
> + 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))
Possibly dumb question: are the spaces really required? I mean why not
"%lf%n*%n"?
> + 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[1 - named]) {
Nit: !named should be more clear
> + 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_channels_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;
> + }
Could you comment on this? What does this condition represent?
> + 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];
nit+++: (int16_t *) is more customary
> + 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];
ditto
[...]
--
FFmpeg = Fantastic Frenzy Monstrous Portable Exuberant Genius
More information about the ffmpeg-devel
mailing list