Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

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

1 files changed, 93 insertions, 0 deletions

diff --git a/third_party/libwebrtc/media/base/media_config.h b/third_party/libwebrtc/media/base/media_config.h
new file mode 100644
index 0000000000..b383c9aa3d
--- /dev/null
+++ b/third_party/libwebrtc/media/base/media_config.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MEDIA_BASE_MEDIA_CONFIG_H_
+#define MEDIA_BASE_MEDIA_CONFIG_H_
+
+namespace cricket {
+
+// Construction-time settings, passed on when creating
+// MediaChannels.
+struct MediaConfig {
+  // Set DSCP value on packets. This flag comes from the
+  // PeerConnection constraint 'googDscp'.
+  // TODO(https://crbug.com/1315574): Remove the ability to set it in Chromium
+  // and delete this flag.
+  bool enable_dscp = true;
+
+  // Video-specific config.
+  struct Video {
+    // Enable WebRTC CPU Overuse Detection. This flag comes from the
+    // PeerConnection constraint 'googCpuOveruseDetection'.
+    // TODO(https://crbug.com/1315569): Remove the ability to set it in Chromium
+    // and delete this flag.
+    bool enable_cpu_adaptation = true;
+
+    // Enable WebRTC suspension of video. No video frames will be sent
+    // when the bitrate is below the configured minimum bitrate. This
+    // flag comes from the PeerConnection constraint
+    // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
+    // to VideoSendStream::Config::suspend_below_min_bitrate.
+    // TODO(https://crbug.com/1315564): Remove the ability to set it in Chromium
+    // and delete this flag.
+    bool suspend_below_min_bitrate = false;
+
+    // Enable buffering and playout timing smoothing of decoded frames.
+    // If set to true, then WebRTC will buffer and potentially drop decoded
+    // frames in order to keep a smooth rendering.
+    // If set to false, then WebRTC will hand over the frame from the decoder
+    // to the renderer as soon as possible, meaning that the renderer is
+    // responsible for smooth rendering.
+    // Note that even if this flag is set to false, dropping of frames can
+    // still happen pre-decode, e.g., dropping of higher temporal layers.
+    // This flag comes from the PeerConnection RtcConfiguration.
+    bool enable_prerenderer_smoothing = true;
+
+    // Enables periodic bandwidth probing in application-limited region.
+    bool periodic_alr_bandwidth_probing = false;
+
+    // Enables the new method to estimate the cpu load from encoding, used for
+    // cpu adaptation. This flag is intended to be controlled primarily by a
+    // Chrome origin-trial.
+    // TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed
+    // together with the old method of estimation.
+    bool experiment_cpu_load_estimator = false;
+
+    // Time interval between RTCP report for video
+    int rtcp_report_interval_ms = 1000;
+  } video;
+
+  // Audio-specific config.
+  struct Audio {
+    // Time interval between RTCP report for audio
+    int rtcp_report_interval_ms = 5000;
+  } audio;
+
+  bool operator==(const MediaConfig& o) const {
+    return enable_dscp == o.enable_dscp &&
+           video.enable_cpu_adaptation == o.video.enable_cpu_adaptation &&
+           video.suspend_below_min_bitrate ==
+               o.video.suspend_below_min_bitrate &&
+           video.enable_prerenderer_smoothing ==
+               o.video.enable_prerenderer_smoothing &&
+           video.periodic_alr_bandwidth_probing ==
+               o.video.periodic_alr_bandwidth_probing &&
+           video.experiment_cpu_load_estimator ==
+               o.video.experiment_cpu_load_estimator &&
+           video.rtcp_report_interval_ms == o.video.rtcp_report_interval_ms &&
+           audio.rtcp_report_interval_ms == o.audio.rtcp_report_interval_ms;
+  }
+
+  bool operator!=(const MediaConfig& o) const { return !(*this == o); }
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_MEDIA_CONFIG_H_