[FFmpeg-cvslog] asyncts: fix the asyncts behavior when using the first_pts option
Justin Ruggles
git at videolan.org
Fri Dec 14 14:03:23 CET 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Tue Dec 11 17:36:09 2012 -0500| [c143de40c3bfacc0d6713b16c2305552494fe669] | committer: Justin Ruggles
asyncts: fix the asyncts behavior when using the first_pts option
Currently it will do padding, but it does not properly handle
start-of-stream trimming as documented.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c143de40c3bfacc0d6713b16c2305552494fe669
---
libavfilter/af_asyncts.c | 64 +++++++++++++++++++++++++++++++++++++++-------
1 file changed, 55 insertions(+), 9 deletions(-)
diff --git a/libavfilter/af_asyncts.c b/libavfilter/af_asyncts.c
index 087692e..02dce5b 100644
--- a/libavfilter/af_asyncts.c
+++ b/libavfilter/af_asyncts.c
@@ -33,6 +33,8 @@ typedef struct ASyncContext {
AVAudioResampleContext *avr;
int64_t pts; ///< timestamp in samples of the first sample in fifo
int min_delta; ///< pad/trim min threshold in samples
+ int first_frame; ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE
+ int64_t first_pts; ///< user-specified first expected pts, in samples
/* options */
int resample;
@@ -50,7 +52,7 @@ static const AVOption options[] = {
{ "min_delta", "Minimum difference between timestamps and audio data "
"(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { .dbl = 0.1 }, 0, INT_MAX, A },
{ "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A },
- { "first_pts", "Assume the first pts should be this value.", OFFSET(pts), AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_MAX, A },
+ { "first_pts", "Assume the first pts should be this value.", OFFSET(first_pts), AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_MAX, A },
{ NULL },
};
@@ -75,6 +77,9 @@ static int init(AVFilterContext *ctx, const char *args)
}
av_opt_free(s);
+ s->pts = AV_NOPTS_VALUE;
+ s->first_frame = 1;
+
return 0;
}
@@ -122,6 +127,20 @@ static int64_t get_delay(ASyncContext *s)
return avresample_available(s->avr) + avresample_get_delay(s->avr);
}
+static void handle_trimming(AVFilterContext *ctx)
+{
+ ASyncContext *s = ctx->priv;
+
+ if (s->pts < s->first_pts) {
+ int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr));
+ av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n",
+ delta);
+ avresample_read(s->avr, NULL, delta);
+ s->pts += delta;
+ } else if (s->first_frame)
+ s->pts = s->first_pts;
+}
+
static int request_frame(AVFilterLink *link)
{
AVFilterContext *ctx = link->src;
@@ -134,7 +153,11 @@ static int request_frame(AVFilterLink *link)
ret = ff_request_frame(ctx->inputs[0]);
/* flush the fifo */
- if (ret == AVERROR_EOF && (nb_samples = get_delay(s))) {
+ if (ret == AVERROR_EOF) {
+ if (s->first_pts != AV_NOPTS_VALUE)
+ handle_trimming(ctx);
+
+ if (nb_samples = get_delay(s)) {
AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
nb_samples);
if (!buf)
@@ -148,6 +171,7 @@ static int request_frame(AVFilterLink *link)
buf->pts = s->pts;
return ff_filter_frame(link, buf);
+ }
}
return ret;
@@ -185,12 +209,18 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
return write_to_fifo(s, buf);
}
+ if (s->first_pts != AV_NOPTS_VALUE) {
+ handle_trimming(ctx);
+ if (!avresample_available(s->avr))
+ return write_to_fifo(s, buf);
+ }
+
/* when we have two timestamps, compute how many samples would we have
* to add/remove to get proper sync between data and timestamps */
delta = pts - s->pts - get_delay(s);
out_size = avresample_available(s->avr);
- if (labs(delta) > s->min_delta) {
+ if (labs(delta) > s->min_delta || (s->first_frame && delta)) {
av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
out_size = av_clipl_int32((int64_t)out_size + delta);
} else {
@@ -210,18 +240,33 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
goto fail;
}
- avresample_read(s->avr, buf_out->extended_data, out_size);
- buf_out->pts = s->pts;
+ if (s->first_frame && delta > 0) {
+ int ch;
+
+ av_samples_set_silence(buf_out->extended_data, 0, delta,
+ nb_channels, buf->format);
+
+ for (ch = 0; ch < nb_channels; ch++)
+ buf_out->extended_data[ch] += delta;
- if (delta > 0) {
- av_samples_set_silence(buf_out->extended_data, out_size - delta,
- delta, nb_channels, buf->format);
+ avresample_read(s->avr, buf_out->extended_data, out_size);
+
+ for (ch = 0; ch < nb_channels; ch++)
+ buf_out->extended_data[ch] -= delta;
+ } else {
+ avresample_read(s->avr, buf_out->extended_data, out_size);
+
+ if (delta > 0) {
+ av_samples_set_silence(buf_out->extended_data, out_size - delta,
+ delta, nb_channels, buf->format);
+ }
}
+ buf_out->pts = s->pts;
ret = ff_filter_frame(outlink, buf_out);
if (ret < 0)
goto fail;
s->got_output = 1;
- } else {
+ } else if (avresample_available(s->avr)) {
av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
"whole buffer.\n");
}
@@ -233,6 +278,7 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
buf->linesize[0], buf->audio->nb_samples);
+ s->first_frame = 0;
fail:
avfilter_unref_buffer(buf);
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