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authorMichael Niedermayer <michaelni@gmx.at>2012-07-09 22:10:38 +0200
committerMichael Niedermayer <michaelni@gmx.at>2012-07-09 22:40:12 +0200
commitf8911b987de4a84ff8ae92f41ff492ece4acadb9 (patch)
tree0ebda51a6ba23d790da30a7168870928954da395 /libavfilter/af_astreamsync.c
parentbf5386385dc504a076453ad58f61f808677be747 (diff)
parent5467742232c312b7d61dca7ac57447f728d8d6c9 (diff)
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_astreamsync.c')
-rw-r--r--libavfilter/af_astreamsync.c9
1 files changed, 6 insertions, 3 deletions
diff --git a/libavfilter/af_astreamsync.c b/libavfilter/af_astreamsync.c
index 8cf3f39b52..587d9a7662 100644
--- a/libavfilter/af_astreamsync.c
+++ b/libavfilter/af_astreamsync.c
@@ -107,11 +107,12 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
-static void send_out(AVFilterContext *ctx, int out_id)
+static int send_out(AVFilterContext *ctx, int out_id)
{
AStreamSyncContext *as = ctx->priv;
struct buf_queue *queue = &as->queue[out_id];
AVFilterBufferRef *buf = queue->buf[queue->tail];
+ int ret;
queue->buf[queue->tail] = NULL;
as->var_values[VAR_B1 + out_id]++;
@@ -121,11 +122,12 @@ static void send_out(AVFilterContext *ctx, int out_id)
av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
(double)ctx->inputs[out_id]->sample_rate;
- ff_filter_samples(ctx->outputs[out_id], buf);
+ ret = ff_filter_samples(ctx->outputs[out_id], buf);
queue->nb--;
queue->tail = (queue->tail + 1) % QUEUE_SIZE;
if (as->req[out_id])
as->req[out_id]--;
+ return ret;
}
static void send_next(AVFilterContext *ctx)
@@ -165,7 +167,7 @@ static int request_frame(AVFilterLink *outlink)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AStreamSyncContext *as = ctx->priv;
@@ -175,6 +177,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
insamples;
as->eof &= ~(1 << id);
send_next(ctx);
+ return 0;
}
AVFilter avfilter_af_astreamsync = {