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author | Michael Niedermayer <michaelni@gmx.at> | 2012-07-09 22:10:38 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-07-09 22:40:12 +0200 |
commit | f8911b987de4a84ff8ae92f41ff492ece4acadb9 (patch) | |
tree | 0ebda51a6ba23d790da30a7168870928954da395 /libavfilter/af_amix.c | |
parent | bf5386385dc504a076453ad58f61f808677be747 (diff) | |
parent | 5467742232c312b7d61dca7ac57447f728d8d6c9 (diff) | |
download | ffmpeg-streaming-f8911b987de4a84ff8ae92f41ff492ece4acadb9.zip ffmpeg-streaming-f8911b987de4a84ff8ae92f41ff492ece4acadb9.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
mss3: use standard zigzag table
mss3: split DSP functions that are used in MTS2(MSS4) into separate file
motion-test: do not use getopt()
tcp: add initial timeout limit for incoming connections
configure: Change the rdtsc check to a linker check
avconv: propagate fatal errors from lavfi.
lavfi: add error handling to filter_samples().
fate-run: make avconv() properly deal with multiple inputs.
asplit: don't leak the input buffer.
af_resample: fix request_frame() behavior.
af_asyncts: fix request_frame() behavior.
libx264: support aspect ratio switching
matroskadec: honor error_recognition when encountering unknown elements.
lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
lavr: resampling: add filter type and Kaiser window beta to AVOptions
lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
lavr: mix: validate internal sample format in ff_audio_mix_init()
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/libx264.c
libavfilter/audio.c
libavfilter/split.c
libavformat/tcp.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavfilter/af_amix.c')
-rw-r--r-- | libavfilter/af_amix.c | 22 |
1 files changed, 13 insertions, 9 deletions
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c index 6dad3db..7f83750 100644 --- a/libavfilter/af_amix.c +++ b/libavfilter/af_amix.c @@ -305,9 +305,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples) if (s->next_pts != AV_NOPTS_VALUE) s->next_pts += nb_samples; - ff_filter_samples(outlink, out_buf); - - return 0; + return ff_filter_samples(outlink, out_buf); } /** @@ -448,31 +446,37 @@ static int request_frame(AVFilterLink *outlink) return output_frame(outlink, available_samples); } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) { AVFilterContext *ctx = inlink->dst; MixContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; - int i; + int i, ret = 0; for (i = 0; i < ctx->nb_inputs; i++) if (ctx->inputs[i] == inlink) break; if (i >= ctx->nb_inputs) { av_log(ctx, AV_LOG_ERROR, "unknown input link\n"); - return; + ret = AVERROR(EINVAL); + goto fail; } if (i == 0) { int64_t pts = av_rescale_q(buf->pts, inlink->time_base, outlink->time_base); - frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts); + ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts); + if (ret < 0) + goto fail; } - av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data, - buf->audio->nb_samples); + ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data, + buf->audio->nb_samples); +fail: avfilter_unref_buffer(buf); + + return ret; } static int init(AVFilterContext *ctx, const char *args) |