I don't use pulseaudio, so I can't test the exact conditions that led to
bug 753, but basic conversion from Vorbis to Opus fails for me without
the attached patch (producing an initial page with a granule position 1
too small to be valid), so this is at least a step in the right direction.
I also notice that the flag AVFMT_TS_NEGATIVE is set for all muxer
classes in libavformat/oggenc.c, but I don't believe this is actually
true for any of them (the granule position value -1 is reserved as
invalid, but granule positions are otherwise treated as unsigned,
leaving no way to represent a negative timestamp). I didn't change
anything there, however, as I don't really understand all the
implications, and I don't want to paper over other problems like this one.
>From c54239a28993847fc52094f654be46bc64186422 Mon Sep 17 00:00:00 2001
From: "Timothy B. Terriberry" <[email protected]>
Date: Mon, 13 Oct 2014 17:46:00 -0700
Subject: [PATCH] resample: Avoid off-by-1 errors in PTS calcs.
The rounding used in the PTS calculations in filter_frame() does
not actually match the number of samples output by the resampler.
This leads to off-by-1 errors in the timestamps indicating gaps and
underruns, even when the input timestamps are all contiguous.
Bug-Id: 753
---
libavfilter/af_resample.c | 10 +++++++++-
1 file changed, 9 insertions(+), 1 deletion(-)
diff --git a/libavfilter/af_resample.c b/libavfilter/af_resample.c
index bc8fd8a..a59e6f8 100644
--- a/libavfilter/af_resample.c
+++ b/libavfilter/af_resample.c
@@ -37,16 +37,17 @@
#include "internal.h"
typedef struct ResampleContext {
const AVClass *class;
AVAudioResampleContext *avr;
AVDictionary *options;
int64_t next_pts;
+ int64_t next_in_pts;
/* set by filter_frame() to signal an output frame to request_frame() */
int got_output;
} ResampleContext;
static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
{
ResampleContext *s = ctx->priv;
@@ -149,16 +150,17 @@ static int config_output(AVFilterLink *outlink)
av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
if ((ret = avresample_open(s->avr)) < 0)
return ret;
outlink->time_base = (AVRational){ 1, outlink->sample_rate };
s->next_pts = AV_NOPTS_VALUE;
+ s->next_in_pts = AV_NOPTS_VALUE;
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(ctx, AV_LOG_VERBOSE,
"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
@@ -250,25 +252,31 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
ret = av_frame_copy_props(out, in);
if (ret < 0) {
av_frame_free(&out);
goto fail;
}
out->sample_rate = outlink->sample_rate;
- if (in->pts != AV_NOPTS_VALUE) {
+ /* Only convert in->pts if there is a discontinuous jump.
+ This ensures that out->pts tracks the number of samples actually
+ output by the resampler in the absence of such a jump.
+ Otherwise, the rounding in av_rescale_q() and av_rescale()
+ causes off-by-1 errors. */
+ if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
out->pts = av_rescale_q(in->pts, inlink->time_base,
outlink->time_base) -
av_rescale(delay, outlink->sample_rate,
inlink->sample_rate);
} else
out->pts = s->next_pts;
s->next_pts = out->pts + out->nb_samples;
+ s->next_in_pts = in->pts + in->nb_samples;
ret = ff_filter_frame(outlink, out);
s->got_output = 1;
}
fail:
av_frame_free(&in);
} else {
--
1.8.3.2
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