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Unified Diff: webrtc/call.h

Issue 1813763005: Updated structures and functions for setting the max bitrate limit to take rtc::Optional<int> Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
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Index: webrtc/call.h
diff --git a/webrtc/call.h b/webrtc/call.h
index 313c5e58c16f49446c8d6fbcdaa92950179628a8..a195aee5d3a3219579dd89d724e6c10363170b4c 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -17,6 +17,7 @@
#include "webrtc/audio_receive_stream.h"
#include "webrtc/audio_send_stream.h"
#include "webrtc/audio_state.h"
+#include "webrtc/base/optional.h"
#include "webrtc/base/socket.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@@ -77,7 +78,7 @@ class Call {
struct BitrateConfig {
int min_bitrate_bps = 0;
int start_bitrate_bps = kDefaultStartBitrateBps;
- int max_bitrate_bps = -1;
+ rtc::Optional<int> max_bitrate_bps;
} bitrate_config;
pthatcher1 2016/03/17 21:51:28 Should we just use kMaxBandwidth here and not have
skvlad 2016/03/18 00:49:17 It appears that the kMaxBandwidth value = 2Mbps is
// AudioState which is possibly shared between multiple calls.

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