[FFmpeg-devel] [PATCH 2/4] pan: add channel mapping capability.
Clément Bœsch
ubitux at gmail.com
Mon Jan 23 11:18:36 CET 2012
On Thu, Jan 19, 2012 at 10:19:50AM +0100, Nicolas George wrote:
> Le nonidi 29 nivôse, an CCXX, Clément Bœsch a écrit :
> > > pure_remap would be more accurate, no?
> > Well I think it's more correct to have a "pure gains" flag: the gains
> > matrix is pure, and this means there will be a remap (I'd say a remap is
> > always "pure")
>
> A remap is always a pure remap, but gains can be pure... what? But that does
> not matter much.
>
I meant "remap == gains are pure"; remap being opposed to panning by its
purity. This debate is slightly moving to a philosophical bikeshed, I'd
better avoid it; feel free to send a patch on top of mine if that disturbs
you too much :)
[...]
> > diff --git a/libavfilter/af_pan.c b/libavfilter/af_pan.c
> > index add14b0..8f4e35c 100644
> > --- a/libavfilter/af_pan.c
> > +++ b/libavfilter/af_pan.c
> > @@ -30,12 +30,14 @@
> > #include <stdio.h>
> > #include "libavutil/audioconvert.h"
> > #include "libavutil/avstring.h"
> > +#include "libavutil/opt.h"
> > +#include "libswresample/swresample.h"
> > #include "avfilter.h"
> > #include "internal.h"
> >
> > #define MAX_CHANNELS 63
> >
> > -typedef struct {
> > +typedef struct PanContext {
> > int64_t out_channel_layout;
> > union {
> > double d[MAX_CHANNELS][MAX_CHANNELS];
> > @@ -46,6 +48,16 @@ typedef struct {
> > int need_renumber;
> > int nb_input_channels;
> > int nb_output_channels;
> > +
> > + int pure_gains;
> > + void (*filter_samples)(struct PanContext*,
> > + AVFilterBufferRef*,
> > + AVFilterBufferRef*,
> > + int);
> > +
> > + /* channel mapping specific */
> > + int channel_map[SWR_CH_MAX];
>
> As far as I can see, channel_map is only used in config_props now: maybe it
> would be better local to that function.
>
libswresample just keeps a pointer to the channel mapping array internally, so
at the moment it is required. This could be changed eventually.
> > + struct SwrContext *swr;
> > } PanContext;
> >
> > static int parse_channel_name(char **arg, int *rchannel, int *rnamed)
> > @@ -179,6 +191,31 @@ static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
> > return 0;
> > }
> >
> > +static void filter_samples_channel_mapping(PanContext *pan, AVFilterBufferRef *outsamples, AVFilterBufferRef *insamples, int n);
> > +static void filter_samples_panning (PanContext *pan, AVFilterBufferRef *outsamples, AVFilterBufferRef *insamples, int n);
> > +
> > +static int are_gains_pure(const PanContext *pan)
> > +{
> > + int i, j;
> > +
> > + for (i = 0; i < MAX_CHANNELS; i++) {
> > + int nb_gain = 0;
> > +
> > + for (j = 0; j < MAX_CHANNELS; j++) {
> > + double gain = pan->gain.d[i][j];
> > +
> > + /* channel mapping is effective only if 0% or 100% of a channel is
> > + * selected... */
> > + if (gain != 0. && gain != 1.)
> > + return 0;
> > + /* ...and if the output channel is only composed of one input */
> > + if (gain && nb_gain++)
> > + return 0;
> > + }
> > + }
> > + return 1;
> > +}
> > +
> > static int query_formats(AVFilterContext *ctx)
> > {
> > PanContext *pan = ctx->priv;
> > @@ -186,11 +223,21 @@ static int query_formats(AVFilterContext *ctx)
> > AVFilterLink *outlink = ctx->outputs[0];
> > AVFilterFormats *formats;
> >
> > + if (pan->nb_output_channels <= SWR_CH_MAX)
> > + pan->pure_gains = are_gains_pure(pan);
>
> Don't you need to check also the number of input channels?
>
Mmh, I can't check it here, so I added a check in config_props() callback. But
I'm not sure the checks are really necessary actually, since AFAIK no existing
channel layouts have more channels than SWR_CH_MAX.
[...]
> > +static void filter_samples_channel_mapping(struct PanContext *pan,
> > + AVFilterBufferRef *outsamples,
> > + AVFilterBufferRef *insamples,
> > + int n)
> > +{
> > + swr_convert(pan->swr, outsamples->data, n, insamples->data, n);
> > +}
> >
> > -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
> > +static void filter_samples_panning(struct PanContext *pan,
> > + AVFilterBufferRef *outsamples,
> > + AVFilterBufferRef *insamples,
> > + int n)
> > {
> > - PanContext *const pan = inlink->dst->priv;
> > - int i, o, n = insamples->audio->nb_samples;
> > + int i, o;
> >
> > /* 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) {
> > @@ -284,16 +378,33 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
> > *(out++) = v >> 8;
> > }
> > }
> > +}
> > +
> > +static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
> > +{
> > + int n = insamples->audio->nb_samples;
> > + AVFilterLink *const outlink = inlink->dst->outputs[0];
> > + AVFilterBufferRef *outsamples = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n);
> > + PanContext *pan = inlink->dst->priv;
> > +
> > + pan->filter_samples(pan, outsamples, insamples, n);
>
> I wonder if the indirect function call does not make the generated code
> slightly bugger and slower, compared to simply a conditional branch. But if
> so, it would probably be negligible, so do not bother checking unless you
> really have nothing better to do.
>
Well, it looks like a standard function call to me:
__asm__("filter_samples_callback:");
pan->filter_samples(pan, outsamples, insamples, n);
__asm__("end_filter_samples_callback:");
==>
000007cf <filter_samples_callback>:
7cf: 89 7c 24 0c mov %edi,0xc(%esp)
7d3: 89 5c 24 08 mov %ebx,0x8(%esp)
7d7: 89 44 24 04 mov %eax,0x4(%esp)
7db: 89 14 24 mov %edx,(%esp)
7de: 89 44 24 1c mov %eax,0x1c(%esp)
7e2: ff 92 28 7c 00 00 call *0x7c28(%edx)
000007e8 <end_filter_samples_callback>:
What kind of "bugs" are you expecting? Possibly the branching is faster than
the callback, but do you think this is a bottleneck?
[...]
Since latest patch, I also:
- added the ctx context to the swr_alloc_set_opts() for logging
- added pan swresample dependency in configure
- bumped lavfi micro
Any other comment?
--
Clément B.
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From 95f57535fc5f418173502a52bdbdeb0def4fbc20 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Cl=C3=A9ment=20B=C5=93sch?= <clement.boesch at smartjog.com>
Date: Wed, 18 Jan 2012 12:00:16 +0100
Subject: [PATCH] pan: add channel mapping capability.
---
configure | 1 +
doc/filters.texi | 45 +++++++++++++++++
libavfilter/af_pan.c | 133 ++++++++++++++++++++++++++++++++++++++++++++++--
libavfilter/version.h | 2 +-
4 files changed, 174 insertions(+), 7 deletions(-)
diff --git a/configure b/configure
index 2a3eaa1..a06e63c 100755
--- a/configure
+++ b/configure
@@ -1659,6 +1659,7 @@ mp_filter_deps="gpl avcodec"
mptestsrc_filter_deps="gpl"
negate_filter_deps="lut_filter"
ocv_filter_deps="libopencv"
+pan_filter_deps="swresample"
scale_filter_deps="swscale"
tinterlace_filter_deps="gpl"
yadif_filter_deps="gpl"
diff --git a/doc/filters.texi b/doc/filters.texi
index 3c9f554..7d008bc 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -315,6 +315,9 @@ Ported from SoX.
Mix channels with specific gain levels. The filter accepts the output
channel layout followed by a set of channels definitions.
+This filter is also designed to remap efficiently the channels of an audio
+stream.
+
The filter accepts parameters of the form:
"@var{l}:@var{outdef}:@var{outdef}:..."
@@ -342,6 +345,8 @@ 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.
+ at subsection Mixing examples
+
For example, if you want to down-mix from stereo to mono, but with a bigger
factor for the left channel:
@example
@@ -358,6 +363,46 @@ Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
that should be preferred (see "-ac" option) unless you have very specific
needs.
+ at subsection Remapping examples
+
+The channel remapping will be effective if, and only if:
+
+ at itemize
+ at item gain coefficients are zeroes or ones,
+ at item only one input per channel output,
+ at item the number of output channels is supported by libswresample (16 at the
+ moment)
+ at c if SWR_CH_MAX changes, fix the line above.
+ at end itemize
+
+If all these conditions are satisfied, the filter will notify the user ("Pure
+channel mapping detected"), and use an optimized and lossless method to do the
+remapping.
+
+For example, if you have a 5.1 source and want a stereo audio stream by
+dropping the extra channels:
+ at example
+pan="stereo: c0=FL : c1=FR"
+ at end example
+
+Given the same source, you can also switch front left and front right channels
+and keep the input channel layout:
+ at example
+pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5"
+ at end example
+
+If the input is a stereo audio stream, you can mute the front left channel (and
+still keep the stereo channel layout) with:
+ at example
+pan="stereo:c1=c1"
+ at end example
+
+Still with a stereo audio stream input, you can copy the right channel in both
+front left and right:
+ at example
+pan="stereo: c0=FR : c1=FR"
+ at end example
+
@section silencedetect
Detect silence in an audio stream.
diff --git a/libavfilter/af_pan.c b/libavfilter/af_pan.c
index a62dea1..d6bb0dc 100644
--- a/libavfilter/af_pan.c
+++ b/libavfilter/af_pan.c
@@ -30,12 +30,14 @@
#include <stdio.h>
#include "libavutil/audioconvert.h"
#include "libavutil/avstring.h"
+#include "libavutil/opt.h"
+#include "libswresample/swresample.h"
#include "avfilter.h"
#include "internal.h"
#define MAX_CHANNELS 63
-typedef struct {
+typedef struct PanContext {
int64_t out_channel_layout;
union {
double d[MAX_CHANNELS][MAX_CHANNELS];
@@ -46,6 +48,16 @@ typedef struct {
int need_renumber;
int nb_input_channels;
int nb_output_channels;
+
+ int pure_gains;
+ void (*filter_samples)(struct PanContext*,
+ AVFilterBufferRef*,
+ AVFilterBufferRef*,
+ int);
+
+ /* channel mapping specific */
+ int channel_map[SWR_CH_MAX];
+ struct SwrContext *swr;
} PanContext;
static int parse_channel_name(char **arg, int *rchannel, int *rnamed)
@@ -179,6 +191,31 @@ static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
return 0;
}
+static void filter_samples_channel_mapping(PanContext *pan, AVFilterBufferRef *outsamples, AVFilterBufferRef *insamples, int n);
+static void filter_samples_panning (PanContext *pan, AVFilterBufferRef *outsamples, AVFilterBufferRef *insamples, int n);
+
+static int are_gains_pure(const PanContext *pan)
+{
+ int i, j;
+
+ for (i = 0; i < MAX_CHANNELS; i++) {
+ int nb_gain = 0;
+
+ for (j = 0; j < MAX_CHANNELS; j++) {
+ double gain = pan->gain.d[i][j];
+
+ /* channel mapping is effective only if 0% or 100% of a channel is
+ * selected... */
+ if (gain != 0. && gain != 1.)
+ return 0;
+ /* ...and if the output channel is only composed of one input */
+ if (gain && nb_gain++)
+ return 0;
+ }
+ }
+ return 1;
+}
+
static int query_formats(AVFilterContext *ctx)
{
PanContext *pan = ctx->priv;
@@ -186,11 +223,21 @@ static int query_formats(AVFilterContext *ctx)
AVFilterLink *outlink = ctx->outputs[0];
AVFilterFormats *formats;
+ if (pan->nb_output_channels <= SWR_CH_MAX)
+ pan->pure_gains = are_gains_pure(pan);
+ if (pan->pure_gains) {
+ /* libswr supports any sample and packing formats */
+ avfilter_set_common_sample_formats(ctx, avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO));
+ avfilter_set_common_packing_formats(ctx, avfilter_make_all_packing_formats());
+ pan->filter_samples = filter_samples_channel_mapping;
+ } else {
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));
+ pan->filter_samples = filter_samples_panning;
+ }
// inlink supports any channel layout
formats = avfilter_make_all_channel_layouts();
@@ -222,6 +269,44 @@ static int config_props(AVFilterLink *link)
}
}
}
+ // gains are pure, init the channel mapping
+ if (pan->pure_gains) {
+
+ // sanity check; can't be done in query_formats since the inlink
+ // channel layout is unknown at that time
+ if (pan->nb_input_channels >= SWR_CH_MAX) {
+ av_log(ctx, AV_LOG_ERROR,
+ "libswresample support a maximum of %d channels. "
+ "Feel free to ask for a higher limit.\n", SWR_CH_MAX);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // get channel map from the pure gains
+ for (i = 0; i < pan->nb_output_channels; i++) {
+ int ch_id = -1;
+ for (j = 0; j < pan->nb_input_channels; j++) {
+ if (pan->gain.d[i][j]) {
+ ch_id = j;
+ break;
+ }
+ }
+ pan->channel_map[i] = ch_id;
+ }
+
+ // init libswresample context
+ pan->swr = swr_alloc_set_opts(pan->swr,
+ pan->out_channel_layout, link->format, link->sample_rate,
+ link->channel_layout, link->format, link->sample_rate,
+ 0, ctx);
+ if (!pan->swr)
+ return AVERROR(ENOMEM);
+ av_opt_set_int(pan->swr, "icl", pan->out_channel_layout, 0);
+ av_opt_set_int(pan->swr, "uch", pan->nb_output_channels, 0);
+ swr_set_channel_mapping(pan->swr, pan->channel_map);
+ r = swr_init(pan->swr);
+ if (r < 0)
+ return r;
+ } else {
// renormalize
for (i = 0; i < pan->nb_output_channels; i++) {
if (!((pan->need_renorm >> i) & 1))
@@ -239,6 +324,7 @@ static int config_props(AVFilterLink *link)
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;
@@ -249,6 +335,17 @@ static int config_props(AVFilterLink *link)
}
av_log(ctx, AV_LOG_INFO, "o%d = %s\n", i, buf);
}
+ // add channel mapping summary if possible
+ if (pan->pure_gains) {
+ av_log(ctx, AV_LOG_INFO, "Pure channel mapping detected:");
+ for (i = 0; i < pan->nb_output_channels; i++)
+ if (pan->channel_map[i] < 0)
+ av_log(ctx, AV_LOG_INFO, " M");
+ else
+ av_log(ctx, AV_LOG_INFO, " %d", pan->channel_map[i]);
+ av_log(ctx, AV_LOG_INFO, "\n");
+ return 0;
+ }
// convert to integer
for (i = 0; i < pan->nb_output_channels; i++) {
for (j = 0; j < pan->nb_input_channels; j++) {
@@ -261,19 +358,26 @@ static int config_props(AVFilterLink *link)
return 0;
}
+static void filter_samples_channel_mapping(struct PanContext *pan,
+ AVFilterBufferRef *outsamples,
+ AVFilterBufferRef *insamples,
+ int n)
+{
+ swr_convert(pan->swr, outsamples->data, n, (void *)insamples->data, n);
+}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static void filter_samples_panning(struct PanContext *pan,
+ AVFilterBufferRef *outsamples,
+ AVFilterBufferRef *insamples,
+ int n)
{
- PanContext *const pan = inlink->dst->priv;
- int i, o, n = insamples->audio->nb_samples;
+ int i, o;
/* 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) {
@@ -284,16 +388,33 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
*(out++) = v >> 8;
}
}
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+{
+ int n = insamples->audio->nb_samples;
+ AVFilterLink *const outlink = inlink->dst->outputs[0];
+ AVFilterBufferRef *outsamples = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n);
+ PanContext *pan = inlink->dst->priv;
+
+ pan->filter_samples(pan, outsamples, insamples, n);
avfilter_filter_samples(outlink, outsamples);
avfilter_unref_buffer(insamples);
}
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ PanContext *pan = ctx->priv;
+ swr_free(&pan->swr);
+}
+
AVFilter avfilter_af_pan = {
.name = "pan",
.description = NULL_IF_CONFIG_SMALL("Remix channels with coefficients (panning)."),
.priv_size = sizeof(PanContext),
.init = init,
+ .uninit = uninit,
.query_formats = query_formats,
.inputs = (const AVFilterPad[]) {
diff --git a/libavfilter/version.h b/libavfilter/version.h
index d2af525..6e19dd7 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#define LIBAVFILTER_VERSION_MAJOR 2
#define LIBAVFILTER_VERSION_MINOR 59
-#define LIBAVFILTER_VERSION_MICRO 101
+#define LIBAVFILTER_VERSION_MICRO 102
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
--
1.7.8.3
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