obs-ffmpeg: Use actual audio encoder sample rate

Uses the sample rate the audio encoder is running at for ffmpeg muxing
(in case the audio sample rate had to be changed by the encoder)
This commit is contained in:
jp9000
2015-07-09 10:44:24 -07:00
parent 50b47a6861
commit ad3d448f19

View File

@@ -95,7 +95,7 @@ static bool build_flv_meta_data(obs_output_t *context,
enc_str_val(&enc, end, "audiocodecid", "mp4a");
enc_num_val(&enc, end, "audiodatarate", encoder_bitrate(aencoder));
enc_num_val(&enc, end, "audiosamplerate",
(double)audio_output_get_sample_rate(audio));
(double)obs_encoder_get_sample_rate(aencoder));
enc_num_val(&enc, end, "audiosamplesize", 16.0);
enc_num_val(&enc, end, "audiochannels",
(double)audio_output_get_channels(audio));