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;
{ "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|F },
{ "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A|F },
- { "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|F },
+ { "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|F },
{ NULL },
};
return ret;
av_opt_free(s);
+ s->pts = AV_NOPTS_VALUE;
+ s->first_frame = 1;
+
return 0;
}
return 0;
}
+/* get amount of data currently buffered, in samples */
+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;
ret = ff_request_frame(ctx->inputs[0]);
/* flush the fifo */
- if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
- AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
- nb_samples);
- if (!buf)
- return AVERROR(ENOMEM);
- ret = avresample_convert(s->avr, buf->extended_data,
- buf->linesize[0], nb_samples, NULL, 0, 0);
- if (ret <= 0) {
- avfilter_unref_bufferp(&buf);
- return (ret < 0) ? ret : AVERROR_EOF;
+ 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)
+ return AVERROR(ENOMEM);
+ ret = avresample_convert(s->avr, buf->extended_data,
+ buf->linesize[0], nb_samples, NULL, 0, 0);
+ if (ret <= 0) {
+ avfilter_unref_bufferp(&buf);
+ return (ret < 0) ? ret : AVERROR_EOF;
+ }
+
+ buf->pts = s->pts;
+ return ff_filter_frame(link, buf);
}
-
- buf->pts = s->pts;
- return ff_filter_frame(link, buf);
}
return ret;
return ret;
}
-/* get amount of data currently buffered, in samples */
-static int64_t get_delay(ASyncContext *s)
-{
- return avresample_available(s->avr) + avresample_get_delay(s->avr);
-}
-
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
int out_size, ret;
int64_t delta;
- /* buffer data until we get the first timestamp */
- if (s->pts == AV_NOPTS_VALUE) {
+ /* buffer data until we get the next timestamp */
+ if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) {
if (pts != AV_NOPTS_VALUE) {
s->pts = pts - get_delay(s);
}
return write_to_fifo(s, buf);
}
- /* now wait for the next timestamp */
- if (pts == AV_NOPTS_VALUE) {
- 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
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 && s->first_pts != AV_NOPTS_VALUE)) {
av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
out_size = av_clipl_int32((int64_t)out_size + delta);
} else {
if (s->resample) {
int comp = av_clip(delta, -s->max_comp, s->max_comp);
av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
- avresample_set_compensation(s->avr, delta, inlink->sample_rate);
+ avresample_set_compensation(s->avr, comp, inlink->sample_rate);
}
delta = 0;
}
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;
- if (delta > 0) {
- av_samples_set_silence(buf_out->extended_data, out_size - delta,
- delta, nb_channels, buf->format);
+ 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;
+
+ 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");
}
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);
projects / ffmpeg / blobdiff