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audio-converter: Fix resampling when there's nothing to output

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Sometimes we can't output anything because we don't have enough
incoming frames. In that case, the resampler was trying to call
do_quantize() and do_resample() in a loop forever because there would
never be samples to output (so chain->samples would always be NULL).

Fix this by not calling chain->make_func() in a loop -- seems
completely unnecessary since calling it over and over won't change
anything if the make_func() can't output samples.

Also add some checks for the input and / or output being NULL when
doing conversion or quantization. This will happen when we have
nothing to output.

We can't bail early, because we need resampler->samples_avail to be
updated in gst_audio_resampler_resample(), so we must call that and
no-op everything along the way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1461>
This commit is contained in:
Nirbheek Chauhan 2021-12-20 21:37:18 +05:30 committed by GStreamer Marge Bot
parent 51e93408a9
commit 554a2a5145
2 changed files with 5 additions and 4 deletions
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5
subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c
View File

@ -253,7 +253,7 @@ audio_chain_get_samples (AudioChain * chain, gsize * avail)
{ {
gpointer *res; gpointer *res;
while (!chain->samples) if (!chain->samples)
chain->make_func (chain, chain->make_func_data); chain->make_func (chain, chain->make_func_data);
res = chain->samples; res = chain->samples;
@ -582,6 +582,7 @@ do_quantize (AudioChain * chain, gpointer user_data)
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples)); out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("quantize %p, %p %" G_GSIZE_FORMAT, in, out, num_samples); GST_LOG ("quantize %p, %p %" G_GSIZE_FORMAT, in, out, num_samples);
if (in && out)
gst_audio_quantize_samples (convert->quant, in, out, num_samples); gst_audio_quantize_samples (convert->quant, in, out, num_samples);
audio_chain_set_samples (chain, out, num_samples); audio_chain_set_samples (chain, out, num_samples);
@ -1274,7 +1275,7 @@ converter_generic (GstAudioConverter * convert,
/* get frames to pack */ /* get frames to pack */
tmp = audio_chain_get_samples (chain, &produced); tmp = audio_chain_get_samples (chain, &produced);
if (!convert->out_default) { if (!convert->out_default && tmp && out) {
GST_LOG ("pack %p, %p %" G_GSIZE_FORMAT, tmp, out, produced); GST_LOG ("pack %p, %p %" G_GSIZE_FORMAT, tmp, out, produced);
/* and pack if needed */ /* and pack if needed */
for (i = 0; i < chain->blocks; i++) for (i = 0; i < chain->blocks; i++)

2
subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c
View File

@ -1781,7 +1781,7 @@ gst_audio_resampler_resample (GstAudioResampler * resampler,
need = resampler->n_taps + resampler->samp_index; need = resampler->n_taps + resampler->samp_index;
if (G_UNLIKELY (samples_avail < need || out_frames == 0)) { if (G_UNLIKELY (samples_avail < need || out_frames == 0)) {
GST_LOG ("not enough samples to start: need %" G_GSIZE_FORMAT ", avail %" GST_LOG ("not enough samples to start: need %" G_GSIZE_FORMAT ", avail %"
G_GSIZE_FORMAT ", out %" G_GSIZE_FORMAT, samples_avail, need, G_GSIZE_FORMAT ", out %" G_GSIZE_FORMAT, need, samples_avail,
out_frames); out_frames);
/* not enough samples to start */ /* not enough samples to start */
return; return;