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author | Clément Bœsch <clement.boesch@smartjog.com> | 2012-01-18 12:00:16 +0100 |
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committer | Clément Bœsch <ubitux@gmail.com> | 2012-01-24 10:43:41 +0100 |
commit | 6728dd37ac3967395fa9c7a7905ed0511fb164e0 (patch) | |
tree | 9bee3f733142782729acb2da806c0796760ba1fc /libavfilter/af_pan.c | |
parent | 66fdbcbbc9d24e18c9faaa1f56bce85ccbb25e07 (diff) | |
download | ffmpeg-6728dd37ac3967395fa9c7a7905ed0511fb164e0.tar.gz |
pan: add channel mapping capability.
Diffstat (limited to 'libavfilter/af_pan.c')
-rw-r--r-- | libavfilter/af_pan.c | 133 |
1 files changed, 127 insertions, 6 deletions
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[]) { |