webrtc/call.h - Issue 1813763005: Updated structures and functions for setting the max bitrate limit to take rtc::Optional<int>

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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: Code review feedback 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 3ba473fec07d09e5293a836218bba926ba8dcc1b..f9cec0c11e94dbd4b7b6c8678c9805579e732e62 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;

// AudioState which is possibly shared between multiple calls.

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