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+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/. */
+
+#ifndef VideoStreamFactory_h
+#define VideoStreamFactory_h
+
+#include "CodecConfig.h"
+#include "mozilla/Atomics.h"
+#include "mozilla/DataMutex.h"
+#include "mozilla/gfx/Point.h"
+#include "mozilla/UniquePtr.h"
+#include "api/video/video_source_interface.h"
+#include "common_video/framerate_controller.h"
+#include "rtc_base/time_utils.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+class VideoFrame;
+}
+
+namespace mozilla {
+
+// Factory class for VideoStreams... vie_encoder.cc will call this to
+// reconfigure.
+class VideoStreamFactory
+ : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
+ public:
+ struct ResolutionAndBitrateLimits {
+ int resolution_in_mb;
+ int min_bitrate_bps;
+ int start_bitrate_bps;
+ int max_bitrate_bps;
+ };
+
+ static ResolutionAndBitrateLimits GetLimitsFor(unsigned int aWidth,
+ unsigned int aHeight,
+ int aCapBps = 0);
+
+ VideoStreamFactory(VideoCodecConfig aConfig,
+ webrtc::VideoCodecMode aCodecMode, int aMinBitrate,
+ int aStartBitrate, int aPrefMaxBitrate,
+ int aNegotiatedMaxBitrate,
+ const rtc::VideoSinkWants& aWants, bool aLockScaling)
+ : mCodecMode(aCodecMode),
+ mMaxFramerateForAllStreams(std::numeric_limits<unsigned int>::max()),
+ mCodecConfig(std::forward<VideoCodecConfig>(aConfig)),
+ mMinBitrate(aMinBitrate),
+ mStartBitrate(aStartBitrate),
+ mPrefMaxBitrate(aPrefMaxBitrate),
+ mNegotiatedMaxBitrate(aNegotiatedMaxBitrate),
+ mFramerateController("VideoStreamFactory::mFramerateController"),
+ mWants(aWants),
+ mLockScaling(aLockScaling) {}
+
+ // This gets called off-main thread and may hold internal webrtc.org
+ // locks. May *NOT* lock the conduit's mutex, to avoid deadlocks.
+ std::vector<webrtc::VideoStream> CreateEncoderStreams(
+ int aWidth, int aHeight,
+ const webrtc::VideoEncoderConfig& aConfig) override;
+ /**
+ * Function to select and change the encoding resolution based on incoming
+ * frame size and current available bandwidth.
+ * @param width, height: dimensions of the frame
+ */
+ void SelectMaxFramerateForAllStreams(unsigned short aWidth,
+ unsigned short aHeight);
+
+ /**
+ * Function to determine if the frame should be dropped based on the given
+ * frame's resolution (combined with the factory's scaleResolutionDownBy) or
+ * timestamp.
+ * @param aFrame frame to be evaluated.
+ * @return true if frame should be dropped, false otehrwise.
+ */
+ bool ShouldDropFrame(const webrtc::VideoFrame& aFrame);
+
+ private:
+ /**
+ * Function to calculate a scaled down width and height based on
+ * scaleDownByResolution, maxFS, and max pixel count settings.
+ * @param aWidth current frame width
+ * @param aHeight current frame height
+ * @param aScaleDownByResolution value to scale width and height down by.
+ * @param aMaxPixelCount maximum number of pixels wanted in a frame.
+ * @return a gfx:IntSize containing width and height to use. These may match
+ * the aWidth and aHeight passed in if no scaling was needed.
+ */
+ gfx::IntSize CalculateScaledResolution(int aWidth, int aHeight,
+ double aScaleDownByResolution,
+ unsigned int aMaxPixelCount);
+
+ /**
+ * Function to select and change the encoding frame rate based on incoming
+ * frame rate, current frame size and max-mbps setting.
+ * @param aOldFramerate current framerate
+ * @param aSendingWidth width of frames being sent
+ * @param aSendingHeight height of frames being sent
+ * @return new framerate meeting max-mbps requriements based on frame size
+ */
+ unsigned int SelectFrameRate(unsigned int aOldFramerate,
+ unsigned short aSendingWidth,
+ unsigned short aSendingHeight);
+
+ // Used to limit number of streams for screensharing.
+ Atomic<webrtc::VideoCodecMode> mCodecMode;
+
+ // The framerate we're currently sending at.
+ Atomic<unsigned int> mMaxFramerateForAllStreams;
+
+ // The current send codec config, containing simulcast layer configs.
+ const VideoCodecConfig mCodecConfig;
+
+ // Bitrate limits in bps.
+ const int mMinBitrate = 0;
+ const int mStartBitrate = 0;
+ const int mPrefMaxBitrate = 0;
+ const int mNegotiatedMaxBitrate = 0;
+
+ // DatamMutex used as object is mutated from a libwebrtc thread and
+ // a seperate thread used to pass video frames to libwebrtc.
+ DataMutex<webrtc::FramerateController> mFramerateController;
+
+ const rtc::VideoSinkWants mWants;
+ const bool mLockScaling;
+};
+
+} // namespace mozilla
+
+#endif