From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- .../webrtc/libwebrtcglue/VideoStreamFactory.h | 132 +++++++++++++++++++++ 1 file changed, 132 insertions(+) create mode 100644 dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h (limited to 'dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h') diff --git a/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h b/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h new file mode 100644 index 0000000000..832d1e9399 --- /dev/null +++ b/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h @@ -0,0 +1,132 @@ +/* -*- 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::max()), + mCodecConfig(std::forward(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 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 mCodecMode; + + // The framerate we're currently sending at. + Atomic 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 mFramerateController; + + const rtc::VideoSinkWants mWants; + const bool mLockScaling; +}; + +} // namespace mozilla + +#endif -- cgit v1.2.3