* resampling audio filter
*/
+#include "libavutil/audioconvert.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
double ratio;
struct SwrContext *swr;
int64_t next_pts;
+ int req_fullfilled;
} AResampleContext;
-static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+static av_cold int init(AVFilterContext *ctx, const char *args)
{
AResampleContext *aresample = ctx->priv;
int ret = 0;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
- AVFilterFormats *in_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
+ AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *out_formats;
AVFilterFormats *in_samplerates = ff_all_samplerates();
AVFilterFormats *out_samplerates;
AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
AVFilterChannelLayouts *out_layouts;
- avfilter_formats_ref (in_formats, &inlink->out_formats);
- avfilter_formats_ref (in_samplerates, &inlink->out_samplerates);
+ ff_formats_ref (in_formats, &inlink->out_formats);
+ ff_formats_ref (in_samplerates, &inlink->out_samplerates);
ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
if(out_rate > 0) {
- out_samplerates = avfilter_make_format_list((int[]){ out_rate, -1 });
+ out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
} else {
out_samplerates = ff_all_samplerates();
}
- avfilter_formats_ref(out_samplerates, &outlink->in_samplerates);
+ ff_formats_ref(out_samplerates, &outlink->in_samplerates);
if(out_format != AV_SAMPLE_FMT_NONE) {
- out_formats = avfilter_make_format_list((int[]){ out_format, -1 });
+ out_formats = ff_make_format_list((int[]){ out_format, -1 });
} else
- out_formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
- avfilter_formats_ref(out_formats, &outlink->in_formats);
+ out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
+ ff_formats_ref(out_formats, &outlink->in_formats);
if(out_layout) {
out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), -1, inlink ->channel_layout);
av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), -1, outlink->channel_layout);
- av_log(ctx, AV_LOG_INFO, "chl:%s fmt:%s r:%"PRId64"Hz -> chl:%s fmt:%s r:%"PRId64"Hz\n",
+
+ av_log(ctx, AV_LOG_VERBOSE, "chl:%s fmt:%s r:%dHz -> chl:%s fmt:%s r:%dHz\n",
inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
int n_out = n_in * aresample->ratio * 2 ;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
+ int ret;
- n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
- (void *)insamplesref->extended_data, n_in);
- if (n_out <= 0) {
- avfilter_unref_buffer(outsamplesref);
- avfilter_unref_buffer(insamplesref);
- return;
- }
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
- outsamplesref->audio->sample_rate = outlink->sample_rate;
- outsamplesref->audio->nb_samples = n_out;
-
-#if 0
- if(insamplesref->pts != AV_NOPTS_VALUE) {
- aresample->next_pts =
- outsamplesref->pts = av_rescale_q(insamplesref->pts, inlink->time_base, outlink->time_base)
- - swr_get_delay(aresample->swr, outlink->time_base.den);
- av_assert0(outlink->time_base.num == 1);
- } else{
- outsamplesref->pts = AV_NOPTS_VALUE; //aresample->next_pts;
- }
- if(aresample->next_pts != AV_NOPTS_VALUE)
- aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
-#else
if(insamplesref->pts != AV_NOPTS_VALUE) {
int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
int64_t outpts= swr_next_pts(aresample->swr, inpts);
} else {
outsamplesref->pts = AV_NOPTS_VALUE;
}
-#endif
- ff_filter_samples(outlink, outsamplesref);
+
+ n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
+ (void *)insamplesref->extended_data, n_in);
+ if (n_out <= 0) {
+ avfilter_unref_buffer(outsamplesref);
+ avfilter_unref_buffer(insamplesref);
+ return 0;
+ }
+
+ outsamplesref->audio->sample_rate = outlink->sample_rate;
+ outsamplesref->audio->nb_samples = n_out;
+
+ ret = ff_filter_samples(outlink, outsamplesref);
+ aresample->req_fullfilled= 1;
avfilter_unref_buffer(insamplesref);
+ return ret;
}
static int request_frame(AVFilterLink *outlink)
AVFilterContext *ctx = outlink->src;
AResampleContext *aresample = ctx->priv;
AVFilterLink *const inlink = outlink->src->inputs[0];
- int ret = avfilter_request_frame(ctx->inputs[0]);
+ int ret;
+
+ aresample->req_fullfilled = 0;
+ do{
+ ret = ff_request_frame(ctx->inputs[0]);
+ }while(!aresample->req_fullfilled && ret>=0);
if (ret == AVERROR_EOF) {
AVFilterBufferRef *outsamplesref;