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; |