diff options
author | Nirbheek Chauhan <nirbheek@centricular.com> | 2021-12-20 21:37:18 +0530 |
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committer | Tim-Philipp Müller <tim@centricular.com> | 2022-01-17 23:50:09 +0000 |
commit | 471173244f77fe1b32f612da81867c2419dcca2a (patch) | |
tree | 91b3e0f7893f1403366faeebc3283768b034a0cb | |
parent | a8c40645808f494e0230e78bbf113ac5c20ae76b (diff) | |
download | gstreamer-plugins-base-471173244f77fe1b32f612da81867c2419dcca2a.tar.gz |
audio-converter: Fix resampling when there's nothing to output
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/gst-plugins-base/-/merge_requests/1289>
-rw-r--r-- | gst-libs/gst/audio/audio-converter.c | 7 | ||||
-rw-r--r-- | gst-libs/gst/audio/audio-resampler.c | 2 |
2 files changed, 5 insertions, 4 deletions
diff --git a/gst-libs/gst/audio/audio-converter.c b/gst-libs/gst/audio/audio-converter.c index 88747f996..06a2841dd 100644 --- a/gst-libs/gst/audio/audio-converter.c +++ b/gst-libs/gst/audio/audio-converter.c @@ -253,7 +253,7 @@ audio_chain_get_samples (AudioChain * chain, gsize * avail) { gpointer *res; - while (!chain->samples) + if (!chain->samples) chain->make_func (chain, chain->make_func_data); res = chain->samples; @@ -582,7 +582,8 @@ do_quantize (AudioChain * chain, gpointer user_data) 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_audio_quantize_samples (convert->quant, in, out, num_samples); + if (in && out) + gst_audio_quantize_samples (convert->quant, in, out, num_samples); audio_chain_set_samples (chain, out, num_samples); @@ -1274,7 +1275,7 @@ converter_generic (GstAudioConverter * convert, /* get frames to pack */ 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); /* and pack if needed */ for (i = 0; i < chain->blocks; i++) diff --git a/gst-libs/gst/audio/audio-resampler.c b/gst-libs/gst/audio/audio-resampler.c index 5e65da5b6..c67f86052 100644 --- a/gst-libs/gst/audio/audio-resampler.c +++ b/gst-libs/gst/audio/audio-resampler.c @@ -1781,7 +1781,7 @@ gst_audio_resampler_resample (GstAudioResampler * resampler, need = resampler->n_taps + resampler->samp_index; if (G_UNLIKELY (samples_avail < need || out_frames == 0)) { 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); /* not enough samples to start */ return; |