static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ResampleContext *s = ctx->priv;
int ret = 0;
s->got_output = 0;
while (ret >= 0 && !s->got_output)
ret = ff_request_frame(ctx->inputs[0]);
/* flush the lavr delay buffer */
if (ret == AVERROR_EOF && s->avr) {
AVFrame *frame;
int nb_samples = avresample_get_out_samples(s->avr, 0);
if (!nb_samples)
return ret;
frame = ff_get_audio_buffer(outlink, nb_samples);
if (!frame)
return AVERROR(ENOMEM);
ret = avresample_convert(s->avr, frame->extended_data,
frame->linesize[0], nb_samples,
NULL, 0, 0);
if (ret <= 0) {
av_frame_free(&frame);
return (ret == 0) ? AVERROR_EOF : ret;
}
frame->pts = s->next_pts;
return ff_filter_frame(outlink, frame);
}
return ret;
}
static int get_out_samples(struct af_resample *s, int in_samples)
{
return avresample_get_out_samples(s->avrctx, in_samples);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret;
if (s->avr) {
AVFrame *out;
int delay, nb_samples;
/* maximum possible samples lavr can output */
delay = avresample_get_delay(s->avr);
nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out) {
ret = AVERROR(ENOMEM);
goto fail;
}
ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
nb_samples, in->extended_data, in->linesize[0],
in->nb_samples);
if (ret <= 0) {
av_frame_free(&out);
if (ret < 0)
goto fail;
}
av_assert0(!avresample_available(s->avr));
if (s->next_pts == AV_NOPTS_VALUE) {
if (in->pts == AV_NOPTS_VALUE) {
av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
"assuming 0.\n");
s->next_pts = 0;
} else
s->next_pts = av_rescale_q(in->pts, inlink->time_base,
outlink->time_base);
}
if (ret > 0) {
out->nb_samples = ret;
ret = av_frame_copy_props(out, in);
if (ret < 0) {
av_frame_free(&out);
goto fail;
}
out->sample_rate = outlink->sample_rate;
/* 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 {
in->format = outlink->format;
ret = ff_filter_frame(outlink, in);
s->got_output = 1;
}
return ret;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int process_audio_stream(AVFormatContext *oc, OutputStream *ost)
{
AVFrame *frame;
int got_output = 0;
int ret;
frame = get_audio_frame(ost);
got_output |= !!frame;
/* feed the data to lavr */
if (frame) {
ret = avresample_convert(ost->avr, NULL, 0, 0,
frame->extended_data, frame->linesize[0],
frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error feeding audio data to the resampler\n");
exit(1);
}
}
while ((frame && avresample_available(ost->avr) >= ost->frame->nb_samples) ||
(!frame && avresample_get_out_samples(ost->avr, 0))) {
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* the difference between the two avresample calls here is that the
* first one just reads the already converted data that is buffered in
* the lavr output buffer, while the second one also flushes the
* resampler */
if (frame) {
ret = avresample_read(ost->avr, ost->frame->extended_data,
ost->frame->nb_samples);
} else {
ret = avresample_convert(ost->avr, ost->frame->extended_data,
ost->frame->linesize[0], ost->frame->nb_samples,
NULL, 0, 0);
}
if (ret < 0) {
fprintf(stderr, "Error while resampling\n");
exit(1);
} else if (frame && ret != ost->frame->nb_samples) {
fprintf(stderr, "Too few samples returned from lavr\n");
exit(1);
}
ost->frame->nb_samples = ret;
ost->frame->pts = ost->next_pts;
ost->next_pts += ost->frame->nb_samples;
got_output |= encode_audio_frame(oc, ost, ret ? ost->frame : NULL);
}
return !got_output;
}
