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obs-ffmpeg: Ensure sample rate is supported in audio encoder

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Ensures that the sample rate is supported in the audio encoders, and if
not, then make it automatically resample to the closest sample rate.
This commit is contained in:
jp9000
2017-07-31 14:51:21 -07:00
parent 0089d379b0
commit d2ffd0fdd2
1 changed files with 19 additions and 0 deletions
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19
plugins/obs-ffmpeg/obs-ffmpeg-audio-encoders.c
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@@ -163,6 +163,24 @@ static void *enc_create(obs_data_t *settings, obs_encoder_t *encoder,
enc->context->sample_fmt = enc->codec->sample_fmts ?
enc->codec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
/* check to make sure sample rate is supported */
if (enc->codec->supported_samplerates) {
const int *rate = enc->codec->supported_samplerates;
int cur_rate = enc->context->sample_rate;
int closest = 0;
while (rate) {
int dist = abs(cur_rate - *rate);
int closest_dist = abs(cur_rate - closest);
if (dist < closest_dist)
closest = *rate;
}
if (closest)
enc->context->sample_rate = closest;
}
/* if using FFmpeg's AAC encoder, at least set a cutoff value
* (recommended by konverter) */
if (strcmp(enc->codec->name, "aac") == 0) {
@@ -287,6 +305,7 @@ static void enc_audio_info(void *data, struct audio_convert_info *info)
{
struct enc_encoder *enc = data;
info->format = convert_ffmpeg_sample_format(enc->context->sample_fmt);
info->samples_per_sec = (uint32_t)enc->context->sample_rate;
}
static size_t enc_frame_size(void *data)