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lavfi/aresample: use libswresample.
This commit is contained in:
Clément Bœsch
1
configure
vendored
1
configure
vendored
@@ -1649,6 +1649,7 @@ udp_protocol_deps="network"
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# filters
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amovie_filter_deps="avcodec avformat"
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aresample_filter_deps="swresample"
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ass_filter_deps="libass"
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blackframe_filter_deps="gpl"
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boxblur_filter_deps="gpl"
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@@ -5,7 +5,7 @@ FFLIBS = avutil
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FFLIBS-$(CONFIG_ACONVERT_FILTER) += avcodec
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FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
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FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += avcodec
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FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += swresample
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FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
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FFLIBS-$(CONFIG_PAN_FILTER) += swresample
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FFLIBS-$(CONFIG_SCALE_FILTER) += swscale
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@@ -24,20 +24,14 @@
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* resampling audio filter
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*/
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#include "libavutil/eval.h"
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#include "libavcodec/avcodec.h"
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#include "libswresample/swresample.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct {
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struct AVResampleContext *resample;
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int out_rate;
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double ratio;
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AVFilterBufferRef *outsamplesref;
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int unconsumed_nb_samples,
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max_cached_nb_samples;
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int16_t *cached_data[8],
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*resampled_data[8];
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struct SwrContext *swr;
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} AResampleContext;
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static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
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@@ -58,23 +52,12 @@ static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AResampleContext *aresample = ctx->priv;
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if (aresample->outsamplesref) {
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int nb_channels =
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av_get_channel_layout_nb_channels(
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aresample->outsamplesref->audio->channel_layout);
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avfilter_unref_buffer(aresample->outsamplesref);
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while (nb_channels--) {
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av_freep(&(aresample->cached_data[nb_channels]));
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av_freep(&(aresample->resampled_data[nb_channels]));
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}
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}
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if (aresample->resample)
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av_resample_close(aresample->resample);
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swr_free(&aresample->swr);
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}
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static int config_output(AVFilterLink *outlink)
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{
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int ret;
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AVFilterContext *ctx = outlink->src;
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AVFilterLink *inlink = ctx->inputs[0];
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AResampleContext *aresample = ctx->priv;
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@@ -85,9 +68,16 @@ static int config_output(AVFilterLink *outlink)
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outlink->sample_rate = aresample->out_rate;
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outlink->time_base = (AVRational) {1, aresample->out_rate};
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//TODO: make the resampling parameters configurable
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aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
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16, 10, 0, 0.8);
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//TODO: make the resampling parameters (filter size, phrase shift, linear, cutoff) configurable
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aresample->swr = swr_alloc_set_opts(aresample->swr,
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inlink->channel_layout, inlink->format, aresample->out_rate,
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inlink->channel_layout, inlink->format, inlink->sample_rate,
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0, ctx);
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if (!aresample->swr)
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return AVERROR(ENOMEM);
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ret = swr_init(aresample->swr);
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if (ret < 0)
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return ret;
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aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
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@@ -96,235 +86,24 @@ static int config_output(AVFilterLink *outlink)
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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avfilter_add_format(&formats, AV_SAMPLE_FMT_S16);
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if (!formats)
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return AVERROR(ENOMEM);
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avfilter_set_common_sample_formats(ctx, formats);
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formats = avfilter_make_all_channel_layouts();
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if (!formats)
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return AVERROR(ENOMEM);
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avfilter_set_common_channel_layouts(ctx, formats);
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formats = avfilter_make_all_packing_formats();
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if (!formats)
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return AVERROR(ENOMEM);
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avfilter_set_common_packing_formats(ctx, formats);
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return 0;
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}
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static void deinterleave(int16_t **outp, int16_t *in,
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int nb_channels, int nb_samples)
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{
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int16_t *out[8];
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memcpy(out, outp, nb_channels * sizeof(int16_t*));
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switch (nb_channels) {
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case 2:
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while (nb_samples--) {
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*out[0]++ = *in++;
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*out[1]++ = *in++;
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}
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break;
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case 3:
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while (nb_samples--) {
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*out[0]++ = *in++;
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*out[1]++ = *in++;
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*out[2]++ = *in++;
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}
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break;
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case 4:
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while (nb_samples--) {
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*out[0]++ = *in++;
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*out[1]++ = *in++;
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*out[2]++ = *in++;
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*out[3]++ = *in++;
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}
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break;
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case 5:
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while (nb_samples--) {
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*out[0]++ = *in++;
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*out[1]++ = *in++;
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*out[2]++ = *in++;
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*out[3]++ = *in++;
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*out[4]++ = *in++;
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}
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break;
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case 6:
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while (nb_samples--) {
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*out[0]++ = *in++;
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*out[1]++ = *in++;
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*out[2]++ = *in++;
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*out[3]++ = *in++;
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*out[4]++ = *in++;
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*out[5]++ = *in++;
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}
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break;
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case 8:
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while (nb_samples--) {
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*out[0]++ = *in++;
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*out[1]++ = *in++;
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*out[2]++ = *in++;
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*out[3]++ = *in++;
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*out[4]++ = *in++;
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*out[5]++ = *in++;
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*out[6]++ = *in++;
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*out[7]++ = *in++;
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}
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break;
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}
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}
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static void interleave(int16_t *out, int16_t **inp,
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int nb_channels, int nb_samples)
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{
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int16_t *in[8];
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memcpy(in, inp, nb_channels * sizeof(int16_t*));
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switch (nb_channels) {
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case 2:
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while (nb_samples--) {
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*out++ = *in[0]++;
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*out++ = *in[1]++;
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}
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break;
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case 3:
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while (nb_samples--) {
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*out++ = *in[0]++;
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*out++ = *in[1]++;
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*out++ = *in[2]++;
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}
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break;
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case 4:
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while (nb_samples--) {
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*out++ = *in[0]++;
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*out++ = *in[1]++;
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*out++ = *in[2]++;
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*out++ = *in[3]++;
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}
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break;
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case 5:
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while (nb_samples--) {
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*out++ = *in[0]++;
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*out++ = *in[1]++;
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*out++ = *in[2]++;
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*out++ = *in[3]++;
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*out++ = *in[4]++;
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}
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break;
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case 6:
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while (nb_samples--) {
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*out++ = *in[0]++;
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*out++ = *in[1]++;
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*out++ = *in[2]++;
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*out++ = *in[3]++;
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*out++ = *in[4]++;
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*out++ = *in[5]++;
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}
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break;
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case 8:
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while (nb_samples--) {
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*out++ = *in[0]++;
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*out++ = *in[1]++;
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*out++ = *in[2]++;
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*out++ = *in[3]++;
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*out++ = *in[4]++;
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*out++ = *in[5]++;
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*out++ = *in[6]++;
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*out++ = *in[7]++;
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}
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break;
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}
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}
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static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
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{
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AResampleContext *aresample = inlink->dst->priv;
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AVFilterLink * const outlink = inlink->dst->outputs[0];
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int i,
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in_nb_samples = insamplesref->audio->nb_samples,
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cached_nb_samples = in_nb_samples + aresample->unconsumed_nb_samples,
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requested_out_nb_samples = aresample->ratio * cached_nb_samples,
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nb_channels =
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av_get_channel_layout_nb_channels(inlink->channel_layout);
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const int n_in = insamplesref->audio->nb_samples;
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int n_out = n_in * aresample->ratio;
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AVFilterLink *const outlink = inlink->dst->outputs[0];
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AVFilterBufferRef *outsamplesref = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
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if (cached_nb_samples > aresample->max_cached_nb_samples) {
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for (i = 0; i < nb_channels; i++) {
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aresample->cached_data[i] =
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av_realloc(aresample->cached_data[i], cached_nb_samples * sizeof(int16_t));
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aresample->resampled_data[i] =
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av_realloc(aresample->resampled_data[i],
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FFALIGN(sizeof(int16_t) * requested_out_nb_samples, 16));
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n_out = swr_convert(aresample->swr, outsamplesref->data, n_out,
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(void *)insamplesref->data, n_in);
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if (aresample->cached_data[i] == NULL || aresample->resampled_data[i] == NULL)
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return;
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}
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aresample->max_cached_nb_samples = cached_nb_samples;
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avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
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outsamplesref->audio->sample_rate = outlink->sample_rate;
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outsamplesref->audio->nb_samples = n_out;
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outsamplesref->pts = av_rescale(outlink->sample_rate, insamplesref->pts,
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inlink ->sample_rate);
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if (aresample->outsamplesref)
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avfilter_unref_buffer(aresample->outsamplesref);
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aresample->outsamplesref =
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avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, requested_out_nb_samples);
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outlink->out_buf = aresample->outsamplesref;
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}
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avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
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aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
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aresample->outsamplesref->pts =
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av_rescale(outlink->sample_rate, insamplesref->pts, inlink->sample_rate);
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/* av_resample() works with planar audio buffers */
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if (!inlink->planar && nb_channels > 1) {
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int16_t *out[8];
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for (i = 0; i < nb_channels; i++)
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out[i] = aresample->cached_data[i] + aresample->unconsumed_nb_samples;
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deinterleave(out, (int16_t *)insamplesref->data[0],
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nb_channels, in_nb_samples);
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} else {
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for (i = 0; i < nb_channels; i++)
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memcpy(aresample->cached_data[i] + aresample->unconsumed_nb_samples,
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insamplesref->data[i],
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in_nb_samples * sizeof(int16_t));
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}
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for (i = 0; i < nb_channels; i++) {
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int consumed_nb_samples;
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const int is_last = i+1 == nb_channels;
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aresample->outsamplesref->audio->nb_samples =
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av_resample(aresample->resample,
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aresample->resampled_data[i], aresample->cached_data[i],
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&consumed_nb_samples,
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cached_nb_samples,
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requested_out_nb_samples, is_last);
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/* move unconsumed data back to the beginning of the cache */
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aresample->unconsumed_nb_samples = cached_nb_samples - consumed_nb_samples;
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memmove(aresample->cached_data[i],
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aresample->cached_data[i] + consumed_nb_samples,
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aresample->unconsumed_nb_samples * sizeof(int16_t));
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}
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/* copy resampled data to the output samplesref */
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if (!inlink->planar && nb_channels > 1) {
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interleave((int16_t *)aresample->outsamplesref->data[0],
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aresample->resampled_data,
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nb_channels, aresample->outsamplesref->audio->nb_samples);
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} else {
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for (i = 0; i < nb_channels; i++)
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memcpy(aresample->outsamplesref->data[i], aresample->resampled_data[i],
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aresample->outsamplesref->audio->nb_samples * sizeof(int16_t));
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}
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avfilter_filter_samples(outlink, avfilter_ref_buffer(aresample->outsamplesref, ~0));
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avfilter_filter_samples(outlink, outsamplesref);
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avfilter_unref_buffer(insamplesref);
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}
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@@ -333,7 +112,6 @@ AVFilter avfilter_af_aresample = {
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.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
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.init = init,
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.uninit = uninit,
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.query_formats = query_formats,
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.priv_size = sizeof(AResampleContext),
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.inputs = (const AVFilterPad[]) {{ .name = "default",
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