| Index: webrtc/call/audio_send_stream.h
|
| diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h
|
| index 20635890649b077e0742654d126c922116af7b00..24792ff0d4ff8a32cb715eb5c1c62bb742290d2a 100644
|
| --- a/webrtc/call/audio_send_stream.h
|
| +++ b/webrtc/call/audio_send_stream.h
|
| @@ -100,6 +100,9 @@ class AudioSendStream {
|
| // string.
|
| rtc::Optional<std::string> audio_network_adaptor_config;
|
|
|
| + // Interval in which adapt codec is called. Default is an interval of 200ms.
|
| + uint32_t adapt_codec_interval_ms = 200;
|
| +
|
| struct SendCodecSpec {
|
| SendCodecSpec();
|
| std::string ToString() const;
|
|
|