diff --git a/README.md b/README.md index fd6ee32..05ec82f 100644 --- a/README.md +++ b/README.md @@ -74,7 +74,7 @@ measures to avoid hitting the timeout. Namely, we use the following strategies: approaching (this is done using the is_final field from the returned speech results as they are an estimate of when a pause has happened). - Close sessions that have been silent for a very long time prior to hitting the -timeout. If someone were to start talking, it is better that that be towards the +timeout. If someone were to start talking, it is better that be towards the beginning of the session. - Store a buffer of audio in between sessions and send it once the new session starts. The device may jump from one network to another, or from network to @@ -132,7 +132,7 @@ stream. For every block of audio that is pushed into the Ogg/Opus stream, we flush the Ogg stream rather than let the ogg library decide on its own when to push out the next block of data. This causes a slight increase in bitrate, but a significant reduction in latency. For the curious, deep in our encoder is a -"low_latency_mode" flag. As a user of this library nothing need be done to +"low_latency_mode" flag. As a user of this library nothing needs to be done to enable that. Just request the following settings for your CloudSpeechSessionParams: