*/
#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];
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)
int nb_in_channels[2] = { 0, 0 }; // number of unnamed and named input channels
double gain;
+ if (!args0) {
+ av_log(ctx, AV_LOG_ERROR,
+ "pan filter needs a channel layout and a set "
+ "of channels definitions as parameter\n");
+ return AVERROR(EINVAL);
+ }
if (!args)
return AVERROR(ENOMEM);
arg = av_strtok(args, ":", &tokenizer);
return 0;
}
-static int query_formats(AVFilterContext *ctx)
+static int are_gains_pure(const PanContext *pan)
{
- PanContext *pan = ctx->priv;
- AVFilterLink *inlink = ctx->inputs[0];
- AVFilterLink *outlink = ctx->outputs[0];
- AVFilterFormats *formats;
+ int i, j;
- const enum AVSampleFormat sample_fmts[] = {AV_SAMPLE_FMT_S16, -1};
- const int packing_fmts[] = {AVFILTER_PACKED, -1};
+ for (i = 0; i < MAX_CHANNELS; i++) {
+ int nb_gain = 0;
- avfilter_set_common_sample_formats (ctx, avfilter_make_format_list(sample_fmts));
- avfilter_set_common_packing_formats(ctx, avfilter_make_format_list(packing_fmts));
+ for (j = 0; j < MAX_CHANNELS; j++) {
+ double gain = pan->gain.d[i][j];
- // 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;
+ /* 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 config_props(AVFilterLink *link)
}
}
}
- // 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) {
- // t is almost 0 but not exactly, this is probably a mistake
- if (t)
- av_log(ctx, AV_LOG_WARNING,
- "Degenerate coefficients while renormalizing\n");
- continue;
+ // 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))
+ 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) {
+ // t is almost 0 but not exactly, this is probably a mistake
+ 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;
}
- 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++) {
}
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++) {
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) {
*(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 int query_formats(AVFilterContext *ctx)
+{
+ PanContext *pan = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ 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();
+ 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 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)"),
+ .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[]) {
projects / ffmpeg / blobdiff