X-Git-Url: https://git.piment-noir.org/?p=deb_ffmpeg.git;a=blobdiff_plain;f=ffmpeg%2Flibavfilter%2Faf_aresample.c;h=57ac3978be2a9b6ae60e553eaa00562eb43b6eb9;hp=5f34321fe8534f972f1520432efc85a3ce7b6513;hb=f6fa7814ccfe3e76514b36cf04f5cd3cb657c8cf;hpb=2ba45a602cbfa7b771effba9b11bb4245c21bc00 diff --git a/ffmpeg/libavfilter/af_aresample.c b/ffmpeg/libavfilter/af_aresample.c index 5f34321..57ac397 100644 --- a/ffmpeg/libavfilter/af_aresample.c +++ b/ffmpeg/libavfilter/af_aresample.c @@ -41,6 +41,7 @@ typedef struct { struct SwrContext *swr; int64_t next_pts; int req_fullfilled; + int more_data; } AResampleContext; static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts) @@ -103,6 +104,11 @@ static int query_formats(AVFilterContext *ctx) } else { out_samplerates = ff_all_samplerates(); } + if (!out_samplerates) { + av_log(ctx, AV_LOG_ERROR, "Cannot allocate output samplerates.\n"); + return AVERROR(ENOMEM); + } + ff_formats_ref(out_samplerates, &outlink->in_samplerates); if(out_format != AV_SAMPLE_FMT_NONE) { @@ -181,7 +187,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref) delay = swr_get_delay(aresample->swr, outlink->sample_rate); if (delay > 0) - n_out += delay; + n_out += FFMIN(delay, FFMAX(4096, n_out)); outsamplesref = ff_get_audio_buffer(outlink, n_out); @@ -210,6 +216,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref) return 0; } + aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers + outsamplesref->nb_samples = n_out; ret = ff_filter_frame(outlink, outsamplesref); @@ -218,40 +226,65 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref) return ret; } -static int request_frame(AVFilterLink *outlink) +static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret) { AVFilterContext *ctx = outlink->src; AResampleContext *aresample = ctx->priv; AVFilterLink *const inlink = outlink->src->inputs[0]; + AVFrame *outsamplesref; + int n_out = 4096; + int64_t pts; + + outsamplesref = ff_get_audio_buffer(outlink, n_out); + *outsamplesref_ret = outsamplesref; + if (!outsamplesref) + return AVERROR(ENOMEM); + + pts = swr_next_pts(aresample->swr, INT64_MIN); + pts = ROUNDED_DIV(pts, inlink->sample_rate); + + n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0); + if (n_out <= 0) { + av_frame_free(&outsamplesref); + return (n_out == 0) ? AVERROR_EOF : n_out; + } + + outsamplesref->sample_rate = outlink->sample_rate; + outsamplesref->nb_samples = n_out; + + outsamplesref->pts = pts; + + return 0; +} + +static int request_frame(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AResampleContext *aresample = ctx->priv; int ret; + // First try to get data from the internal buffers + if (aresample->more_data) { + AVFrame *outsamplesref; + + if (flush_frame(outlink, 0, &outsamplesref) >= 0) { + return ff_filter_frame(outlink, outsamplesref); + } + } + aresample->more_data = 0; + + // Second request more data from the input aresample->req_fullfilled = 0; do{ ret = ff_request_frame(ctx->inputs[0]); }while(!aresample->req_fullfilled && ret>=0); + // Third if we hit the end flush if (ret == AVERROR_EOF) { AVFrame *outsamplesref; - int n_out = 4096; - int64_t pts; - - outsamplesref = ff_get_audio_buffer(outlink, n_out); - if (!outsamplesref) - return AVERROR(ENOMEM); - - pts = swr_next_pts(aresample->swr, INT64_MIN); - pts = ROUNDED_DIV(pts, inlink->sample_rate); - - n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0); - if (n_out <= 0) { - av_frame_free(&outsamplesref); - return (n_out == 0) ? AVERROR_EOF : n_out; - } - - outsamplesref->sample_rate = outlink->sample_rate; - outsamplesref->nb_samples = n_out; - outsamplesref->pts = pts; + if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0) + return ret; return ff_filter_frame(outlink, outsamplesref); }