static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
int ret;
int wanted_samples, available_samples;
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
if (s->input_state[0] == INPUT_OFF) {
ret = request_samples(ctx, 1);
if (ret < 0)
return ret;
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
available_samples = get_available_samples(s);
if (!available_samples)
return AVERROR(EAGAIN);
return output_frame(outlink, available_samples);
}
if (s->frame_list->nb_frames == 0) {
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF) {
s->input_state[0] = INPUT_OFF;
if (s->nb_inputs == 1)
return AVERROR_EOF;
else
return AVERROR(EAGAIN);
} else if (ret < 0)
return ret;
}
av_assert0(s->frame_list->nb_frames > 0);
wanted_samples = frame_list_next_frame_size(s->frame_list);
if (s->active_inputs > 1) {
ret = request_samples(ctx, wanted_samples);
if (ret < 0)
return ret;
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
}
if (s->active_inputs > 1) {
available_samples = get_available_samples(s);
if (!available_samples)
return AVERROR(EAGAIN);
available_samples = FFMIN(available_samples, wanted_samples);
} else {
available_samples = wanted_samples;
}
s->next_pts = frame_list_next_pts(s->frame_list);
frame_list_remove_samples(s->frame_list, available_samples);
return output_frame(outlink, available_samples);
}
/**
* Read samples from the input FIFOs, mix, and write to the output link.
*/
static int output_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
AVFrame *out_buf, *in_buf;
int nb_samples, ns, i;
if (s->input_state[0] & INPUT_ON) {
/* first input live: use the corresponding frame size */
nb_samples = frame_list_next_frame_size(s->frame_list);
for (i = 1; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
ns = av_audio_fifo_size(s->fifos[i]);
if (ns < nb_samples) {
if (!(s->input_state[i] & INPUT_EOF))
/* unclosed input with not enough samples */
return 0;
/* closed input to drain */
nb_samples = ns;
}
}
}
} else {
/* first input closed: use the available samples */
nb_samples = INT_MAX;
for (i = 1; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
ns = av_audio_fifo_size(s->fifos[i]);
nb_samples = FFMIN(nb_samples, ns);
}
}
if (nb_samples == INT_MAX) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
return 0;
}
}
s->next_pts = frame_list_next_pts(s->frame_list);
frame_list_remove_samples(s->frame_list, nb_samples);
calculate_scales(s, nb_samples);
if (nb_samples == 0)
return 0;
out_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!out_buf)
return AVERROR(ENOMEM);
in_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!in_buf) {
av_frame_free(&out_buf);
return AVERROR(ENOMEM);
}
for (i = 0; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
int planes, plane_size, p;
av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
nb_samples);
planes = s->planar ? s->nb_channels : 1;
plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
plane_size = FFALIGN(plane_size, 16);
if (out_buf->format == AV_SAMPLE_FMT_FLT ||
out_buf->format == AV_SAMPLE_FMT_FLTP) {
for (p = 0; p < planes; p++) {
s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
(float *) in_buf->extended_data[p],
s->input_scale[i], plane_size);
}
} else {
for (p = 0; p < planes; p++) {
s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
(double *) in_buf->extended_data[p],
s->input_scale[i], plane_size);
}
}
}
}
av_frame_free(&in_buf);
out_buf->pts = s->next_pts;
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += nb_samples;
return ff_filter_frame(outlink, out_buf);
}
