OPUS_APPLICATION_VOIP should be used

See: http://opus-codec.org/docs/html_api-1.0.1/group__opus__encoder.html#gaa89264fd93c9da70362a0c9b96b9ca88

"VOIP" rather than "AUDIO":
> gives best quality at a given bitrate for voice signals. It enhances the input signal by high-pass filtering and emphasizing formants and harmonics. Optionally it includes in-band forward error correction to protect against packet loss. Use this mode for typical VoIP applications.

1 files changed, 1 insertions, 1 deletions

diff --git a/toxav/codec.c b/toxav/codec.c
index ae146ccf..71d24e87 100644
--- a/toxav/codec.c
+++ b/toxav/codec.c
@@ -294,7 +294,7 @@ static int init_audio_encoder(CSSession *cs)
 {
     int rc = OPUS_OK;
     cs->audio_encoder = opus_encoder_create(cs->audio_encoder_sample_rate,
-                                            cs->audio_encoder_channels, OPUS_APPLICATION_AUDIO, &rc);
+                                            cs->audio_encoder_channels, OPUS_APPLICATION_VOIP, &rc);
 
     if ( rc != OPUS_OK ) {
         LOGGER_ERROR("Error while starting audio encoder: %s", opus_strerror(rc));