[FFmpeg-devel] [PATCH 2/4] lavfi/aresample: use libswresample.
Stefano Sabatini
stefasab at gmail.com
Tue Jan 31 15:49:35 CET 2012
On date Tuesday 2012-01-31 08:31:46 +0100, Clément Bœsch encoded:
> ---
> configure | 1 +
> libavfilter/Makefile | 2 +-
> libavfilter/af_aresample.c | 282 +++++---------------------------------------
> 3 files changed, 32 insertions(+), 253 deletions(-)
>
> diff --git a/configure b/configure
> index 348342d..b4e433c 100755
> --- a/configure
> +++ b/configure
> @@ -1649,6 +1649,7 @@ udp_protocol_deps="network"
>
> # filters
> amovie_filter_deps="avcodec avformat"
> +aresample_filter_deps="swresample"
> ass_filter_deps="libass"
> blackframe_filter_deps="gpl"
> boxblur_filter_deps="gpl"
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 01a1316..ff0ba75 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -5,7 +5,7 @@ FFLIBS = avutil
>
> FFLIBS-$(CONFIG_ACONVERT_FILTER) += avcodec
> FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
> -FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += avcodec
> +FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += swresample
> FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
> FFLIBS-$(CONFIG_PAN_FILTER) += swresample
> FFLIBS-$(CONFIG_SCALE_FILTER) += swscale
> diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c
> index cd9d80e..cde9adf 100644
> --- a/libavfilter/af_aresample.c
> +++ b/libavfilter/af_aresample.c
> @@ -24,20 +24,14 @@
> * resampling audio filter
> */
>
> -#include "libavutil/eval.h"
> -#include "libavcodec/avcodec.h"
> +#include "libswresample/swresample.h"
> #include "avfilter.h"
> #include "internal.h"
>
> typedef struct {
> - struct AVResampleContext *resample;
> int out_rate;
> double ratio;
> - AVFilterBufferRef *outsamplesref;
> - int unconsumed_nb_samples,
> - max_cached_nb_samples;
> - int16_t *cached_data[8],
> - *resampled_data[8];
> + struct SwrContext *swr;
> } AResampleContext;
>
> static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
> @@ -58,23 +52,12 @@ static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
> static av_cold void uninit(AVFilterContext *ctx)
> {
> AResampleContext *aresample = ctx->priv;
> - if (aresample->outsamplesref) {
> - int nb_channels =
> - av_get_channel_layout_nb_channels(
> - aresample->outsamplesref->audio->channel_layout);
> - avfilter_unref_buffer(aresample->outsamplesref);
> - while (nb_channels--) {
> - av_freep(&(aresample->cached_data[nb_channels]));
> - av_freep(&(aresample->resampled_data[nb_channels]));
> - }
> - }
> -
> - if (aresample->resample)
> - av_resample_close(aresample->resample);
> + swr_free(&aresample->swr);
> }
>
> static int config_output(AVFilterLink *outlink)
> {
> + int ret;
> AVFilterContext *ctx = outlink->src;
> AVFilterLink *inlink = ctx->inputs[0];
> AResampleContext *aresample = ctx->priv;
> @@ -85,9 +68,16 @@ static int config_output(AVFilterLink *outlink)
> outlink->sample_rate = aresample->out_rate;
> outlink->time_base = (AVRational) {1, aresample->out_rate};
>
> - //TODO: make the resampling parameters configurable
> - aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
> - 16, 10, 0, 0.8);
> + //TODO: make the resampling parameters (filter size, phrase shift, linear, cutoff) configurable
> + aresample->swr = swr_alloc_set_opts(aresample->swr,
> + inlink->channel_layout, inlink->format, aresample->out_rate,
> + inlink->channel_layout, inlink->format, inlink->sample_rate,
> + 0, ctx);
> + if (!aresample->swr)
> + return AVERROR(ENOMEM);
> + ret = swr_init(aresample->swr);
> + if (ret < 0)
> + return ret;
>
> aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
>
> @@ -96,235 +86,24 @@ static int config_output(AVFilterLink *outlink)
> return 0;
> }
>
> -static int query_formats(AVFilterContext *ctx)
> -{
> - AVFilterFormats *formats = NULL;
> -
> - avfilter_add_format(&formats, AV_SAMPLE_FMT_S16);
> - if (!formats)
> - return AVERROR(ENOMEM);
> - avfilter_set_common_sample_formats(ctx, formats);
> -
> - formats = avfilter_make_all_channel_layouts();
> - if (!formats)
> - return AVERROR(ENOMEM);
> - avfilter_set_common_channel_layouts(ctx, formats);
> -
> - formats = avfilter_make_all_packing_formats();
> - if (!formats)
> - return AVERROR(ENOMEM);
> - avfilter_set_common_packing_formats(ctx, formats);
> -
> - return 0;
> -}
> -
> -static void deinterleave(int16_t **outp, int16_t *in,
> - int nb_channels, int nb_samples)
> -{
> - int16_t *out[8];
> - memcpy(out, outp, nb_channels * sizeof(int16_t*));
> -
> - switch (nb_channels) {
> - case 2:
> - while (nb_samples--) {
> - *out[0]++ = *in++;
> - *out[1]++ = *in++;
> - }
> - break;
> - case 3:
> - while (nb_samples--) {
> - *out[0]++ = *in++;
> - *out[1]++ = *in++;
> - *out[2]++ = *in++;
> - }
> - break;
> - case 4:
> - while (nb_samples--) {
> - *out[0]++ = *in++;
> - *out[1]++ = *in++;
> - *out[2]++ = *in++;
> - *out[3]++ = *in++;
> - }
> - break;
> - case 5:
> - while (nb_samples--) {
> - *out[0]++ = *in++;
> - *out[1]++ = *in++;
> - *out[2]++ = *in++;
> - *out[3]++ = *in++;
> - *out[4]++ = *in++;
> - }
> - break;
> - case 6:
> - while (nb_samples--) {
> - *out[0]++ = *in++;
> - *out[1]++ = *in++;
> - *out[2]++ = *in++;
> - *out[3]++ = *in++;
> - *out[4]++ = *in++;
> - *out[5]++ = *in++;
> - }
> - break;
> - case 8:
> - while (nb_samples--) {
> - *out[0]++ = *in++;
> - *out[1]++ = *in++;
> - *out[2]++ = *in++;
> - *out[3]++ = *in++;
> - *out[4]++ = *in++;
> - *out[5]++ = *in++;
> - *out[6]++ = *in++;
> - *out[7]++ = *in++;
> - }
> - break;
> - }
> -}
> -
> -static void interleave(int16_t *out, int16_t **inp,
> - int nb_channels, int nb_samples)
> -{
> - int16_t *in[8];
> - memcpy(in, inp, nb_channels * sizeof(int16_t*));
> -
> - switch (nb_channels) {
> - case 2:
> - while (nb_samples--) {
> - *out++ = *in[0]++;
> - *out++ = *in[1]++;
> - }
> - break;
> - case 3:
> - while (nb_samples--) {
> - *out++ = *in[0]++;
> - *out++ = *in[1]++;
> - *out++ = *in[2]++;
> - }
> - break;
> - case 4:
> - while (nb_samples--) {
> - *out++ = *in[0]++;
> - *out++ = *in[1]++;
> - *out++ = *in[2]++;
> - *out++ = *in[3]++;
> - }
> - break;
> - case 5:
> - while (nb_samples--) {
> - *out++ = *in[0]++;
> - *out++ = *in[1]++;
> - *out++ = *in[2]++;
> - *out++ = *in[3]++;
> - *out++ = *in[4]++;
> - }
> - break;
> - case 6:
> - while (nb_samples--) {
> - *out++ = *in[0]++;
> - *out++ = *in[1]++;
> - *out++ = *in[2]++;
> - *out++ = *in[3]++;
> - *out++ = *in[4]++;
> - *out++ = *in[5]++;
> - }
> - break;
> - case 8:
> - while (nb_samples--) {
> - *out++ = *in[0]++;
> - *out++ = *in[1]++;
> - *out++ = *in[2]++;
> - *out++ = *in[3]++;
> - *out++ = *in[4]++;
> - *out++ = *in[5]++;
> - *out++ = *in[6]++;
> - *out++ = *in[7]++;
> - }
> - break;
> - }
> -}
> -
> static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
> {
> - AResampleContext *aresample = inlink->dst->priv;
> - AVFilterLink * const outlink = inlink->dst->outputs[0];
> - int i,
> - in_nb_samples = insamplesref->audio->nb_samples,
> - cached_nb_samples = in_nb_samples + aresample->unconsumed_nb_samples,
> - requested_out_nb_samples = aresample->ratio * cached_nb_samples,
> - nb_channels =
> - av_get_channel_layout_nb_channels(inlink->channel_layout);
> -
> - if (cached_nb_samples > aresample->max_cached_nb_samples) {
> - for (i = 0; i < nb_channels; i++) {
> - aresample->cached_data[i] =
> - av_realloc(aresample->cached_data[i], cached_nb_samples * sizeof(int16_t));
> - aresample->resampled_data[i] =
> - av_realloc(aresample->resampled_data[i],
> - FFALIGN(sizeof(int16_t) * requested_out_nb_samples, 16));
> -
> - if (aresample->cached_data[i] == NULL || aresample->resampled_data[i] == NULL)
> - return;
> - }
> - aresample->max_cached_nb_samples = cached_nb_samples;
> -
> - if (aresample->outsamplesref)
> - avfilter_unref_buffer(aresample->outsamplesref);
> -
> - aresample->outsamplesref =
> - avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, requested_out_nb_samples);
> - outlink->out_buf = aresample->outsamplesref;
> - }
> -
> - avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
> - aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
> - aresample->outsamplesref->pts =
> - av_rescale(outlink->sample_rate, insamplesref->pts, inlink->sample_rate);
> -
> - /* av_resample() works with planar audio buffers */
> - if (!inlink->planar && nb_channels > 1) {
> - int16_t *out[8];
> - for (i = 0; i < nb_channels; i++)
> - out[i] = aresample->cached_data[i] + aresample->unconsumed_nb_samples;
> -
> - deinterleave(out, (int16_t *)insamplesref->data[0],
> - nb_channels, in_nb_samples);
> - } else {
> - for (i = 0; i < nb_channels; i++)
> - memcpy(aresample->cached_data[i] + aresample->unconsumed_nb_samples,
> - insamplesref->data[i],
> - in_nb_samples * sizeof(int16_t));
> - }
> -
> - for (i = 0; i < nb_channels; i++) {
> - int consumed_nb_samples;
> - const int is_last = i+1 == nb_channels;
> -
> - aresample->outsamplesref->audio->nb_samples =
> - av_resample(aresample->resample,
> - aresample->resampled_data[i], aresample->cached_data[i],
> - &consumed_nb_samples,
> - cached_nb_samples,
> - requested_out_nb_samples, is_last);
> -
> - /* move unconsumed data back to the beginning of the cache */
> - aresample->unconsumed_nb_samples = cached_nb_samples - consumed_nb_samples;
> - memmove(aresample->cached_data[i],
> - aresample->cached_data[i] + consumed_nb_samples,
> - aresample->unconsumed_nb_samples * sizeof(int16_t));
> - }
> -
> -
> - /* copy resampled data to the output samplesref */
> - if (!inlink->planar && nb_channels > 1) {
> - interleave((int16_t *)aresample->outsamplesref->data[0],
> - aresample->resampled_data,
> - nb_channels, aresample->outsamplesref->audio->nb_samples);
> - } else {
> - for (i = 0; i < nb_channels; i++)
> - memcpy(aresample->outsamplesref->data[i], aresample->resampled_data[i],
> - aresample->outsamplesref->audio->nb_samples * sizeof(int16_t));
> - }
> -
> - avfilter_filter_samples(outlink, avfilter_ref_buffer(aresample->outsamplesref, ~0));
> + AResampleContext *aresample = inlink->dst->priv;
> + const int n_in = insamplesref->audio->nb_samples;
> + int n_out = n_in * aresample->ratio;
> + AVFilterLink *const outlink = inlink->dst->outputs[0];
> + AVFilterBufferRef *outsamplesref = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
> +
> + n_out = swr_convert(aresample->swr, outsamplesref->data, n_out,
> + (void *)insamplesref->data, n_in);
> +
> + avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
> + outsamplesref->audio->sample_rate = outlink->sample_rate;
> + outsamplesref->audio->nb_samples = n_out;
> + outsamplesref->pts = av_rescale(outlink->sample_rate, insamplesref->pts,
> + inlink ->sample_rate);
> +
> + avfilter_filter_samples(outlink, outsamplesref);
> avfilter_unref_buffer(insamplesref);
> }
>
> @@ -333,7 +112,6 @@ AVFilter avfilter_af_aresample = {
> .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
> .init = init,
> .uninit = uninit,
> - .query_formats = query_formats,
> .priv_size = sizeof(AResampleContext),
>
> .inputs = (const AVFilterPad[]) {{ .name = "default",
Looks good, and very nice simplification, a FATE test would be very
nice at this point for testing libswr and audio lavfi (but we still
need to implement -af in ffmpeg before).
--
FFmpeg = Free Faithful Minimal Picky Elfic Gadget
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