| Index: webrtc/media/engine/webrtcvideoengine2.h
|
| diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h
|
| index 8c05c60f6823606a951783a5485adebef2e3dc1b..511a8234b84ea741dea1b398a5307199690263f0 100644
|
| --- a/webrtc/media/engine/webrtcvideoengine2.h
|
| +++ b/webrtc/media/engine/webrtcvideoengine2.h
|
| @@ -192,7 +192,9 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| // These optionals are unset if not changed.
|
| rtc::Optional<VideoCodecSettings> codec;
|
| rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
|
| - rtc::Optional<int> max_bandwidth_bps;
|
| + // The outer optional is unset if the bandwidth limit has not changed;
|
| + // the inner optional indicates if a bandwidth limit is in effect.
|
| + rtc::Optional<rtc::Optional<int>> max_bandwidth_bps;
|
| rtc::Optional<bool> conference_mode;
|
| rtc::Optional<webrtc::RtcpMode> rtcp_mode;
|
| };
|
| @@ -239,7 +241,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| const VideoOptions& options,
|
| WebRtcVideoEncoderFactory* external_encoder_factory,
|
| bool enable_cpu_overuse_detection,
|
| - int max_bitrate_bps,
|
| + rtc::Optional<int> max_bitrate_bps,
|
| const rtc::Optional<VideoCodecSettings>& codec_settings,
|
| const std::vector<webrtc::RtpExtension>& rtp_extensions,
|
| const VideoSendParameters& send_params);
|
| @@ -274,11 +276,11 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| VideoSendStreamParameters(
|
| const webrtc::VideoSendStream::Config& config,
|
| const VideoOptions& options,
|
| - int max_bitrate_bps,
|
| + rtc::Optional<int> max_bitrate_bps,
|
| const rtc::Optional<VideoCodecSettings>& codec_settings);
|
| webrtc::VideoSendStream::Config config;
|
| VideoOptions options;
|
| - int max_bitrate_bps;
|
| + rtc::Optional<int> max_bitrate_bps;
|
| bool conference_mode;
|
| rtc::Optional<VideoCodecSettings> codec_settings;
|
| // Sent resolutions + bitrates etc. by the underlying VideoSendStream,
|
| @@ -319,12 +321,12 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| static std::vector<webrtc::VideoStream> CreateVideoStreams(
|
| const VideoCodec& codec,
|
| const VideoOptions& options,
|
| - int max_bitrate_bps,
|
| + rtc::Optional<int> max_bitrate_bps,
|
| size_t num_streams);
|
| static std::vector<webrtc::VideoStream> CreateSimulcastVideoStreams(
|
| const VideoCodec& codec,
|
| const VideoOptions& options,
|
| - int max_bitrate_bps,
|
| + rtc::Optional<int> max_bitrate_bps,
|
| size_t num_streams);
|
|
|
| void* ConfigureVideoEncoderSettings(const VideoCodec& codec)
|
|
|