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

Ensures that the sample rate is supported in the audio encoders, and if
not, then make it automatically resample to the closest sample rate.

plugins/obs-ffmpeg/obs-ffmpeg-audio-encoders.c

@@ -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)