pan: add channel mapping capability.

4 files changed, 174 insertions, 7 deletions

diff --git a/configure b/configure
index 2a3eaa1e94..a06e63cbe0 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 3c9f55434d..7d008bc736 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.
 
+@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.
 
+@subsection Remapping examples
+
+The channel remapping will be effective if, and only if:
+
+@itemize
+@item gain coefficients are zeroes or ones,
+@item only one input per channel output,
+@item the number of output channels is supported by libswresample (16 at the
+      moment)
+@c if SWR_CH_MAX changes, fix the line above.
+@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:
+@example
+pan="stereo: c0=FL : c1=FR"
+@end example
+
+Given the same source, you can also switch front left and front right channels
+and keep the input channel layout:
+@example
+pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5"
+@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:
+@example
+pan="stereo:c1=c1"
+@end example
+
+Still with a stereo audio stream input, you can copy the right channel in both
+front left and right:
+@example
+pan="stereo: c0=FR : c1=FR"
+@end example
+
 @section silencedetect
 
 Detect silence in an audio stream.
diff --git a/libavfilter/af_pan.c b/libavfilter/af_pan.c
index a62dea19e7..741e76aeb9 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(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(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 d2af525bb4..6e19dd70f2 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, \