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Unified Diff: webrtc/call/audio_send_stream.h

Issue 2546493002: Update smoothed bitrate. (Closed)
Patch Set: Response to comments. Created 4 years ago
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« webrtc/audio/audio_send_stream.cc ('K') | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/test/mock_voe_channel_proxy.h » ('j') | no next file with comments »
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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.
minyue-webrtc 2016/12/14 11:36:37 Time interval on which AdaptCodec is called. (no n
+ uint32_t adapt_codec_interval_ms = 200;
+
struct SendCodecSpec {
SendCodecSpec();
std::string ToString() const;
« webrtc/audio/audio_send_stream.cc ('K') | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/test/mock_voe_channel_proxy.h » ('j') | no next file with comments »

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