Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 132 insertions, 0 deletions

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<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