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Unified Diff: webrtc/media/base/mediaengine.h

Issue 1788583004: Enable setting the maximum bitrate limit in RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed support for bitrate limits for audio streams; corrected code review issues Created 4 years, 9 months ago
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Index: webrtc/media/base/mediaengine.h
diff --git a/webrtc/media/base/mediaengine.h b/webrtc/media/base/mediaengine.h
index 479d1acf4e820af67f8bbf925eac50700c9dfb2a..883c34c834fb3d9181b21637c4d92c07948d7c91 100644
--- a/webrtc/media/base/mediaengine.h
+++ b/webrtc/media/base/mediaengine.h
@@ -19,6 +19,7 @@
#include <vector>
#include "webrtc/audio_state.h"
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/sigslotrepeater.h"
#include "webrtc/media/base/codec.h"
@@ -205,6 +206,8 @@ class DataEngineInterface {
virtual const std::vector<DataCodec>& data_codecs() = 0;
};
+webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
+
} // namespace cricket
#endif // WEBRTC_MEDIA_BASE_MEDIAENGINE_H_

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