Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 06:15:48 +02:00
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
--- a/modules/bitrate_controller/include/bitrate_controller.h
+++ b/modules/bitrate_controller/include/bitrate_controller.h
@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2012 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.
+ *
+ *  Usage: this class will register multiple RtcpBitrateObserver's one at each
+ *  RTCP module. It will aggregate the results and run one bandwidth estimation
+ *  and push the result to the encoders via BitrateObserver(s).
+ */
+
+#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
+#define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
+
+#include <map>
+
+#include "webrtc/modules/congestion_controller/delay_based_bwe.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+// Deprecated
+// TODO(perkj): Remove BitrateObserver when no implementations use it.
+class BitrateObserver {
+  // Observer class for bitrate changes announced due to change in bandwidth
+  // estimate or due to bitrate allocation changes. Fraction loss and rtt is
+  // also part of this callback to allow the obsevrer to optimize its settings
+  // for different types of network environments. The bitrate does not include
+  // packet headers and is measured in bits per second.
+ public:
+  virtual void OnNetworkChanged(uint32_t bitrate_bps,
+                                uint8_t fraction_loss,  // 0 - 255.
+                                int64_t rtt_ms) = 0;
+  // TODO(gnish): Merge these two into one function.
+  virtual void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
+                                uint32_t bitrate_for_pacer_bps,
+                                bool in_probe_rtt,
+                                int64_t target_set_time,
+                                uint64_t congestion_window) {}
+  virtual void OnBytesAcked(size_t bytes) {}
+  virtual size_t pacer_queue_size_in_bytes() { return 0; }
+  virtual ~BitrateObserver() {}
+};
+
+class BitrateController : public Module, public RtcpBandwidthObserver {
+  // This class collects feedback from all streams sent to a peer (via
+  // RTCPBandwidthObservers). It does one  aggregated send side bandwidth
+  // estimation and divide the available bitrate between all its registered
+  // BitrateObservers.
+ public:
+  static const int kDefaultStartBitratebps = 300000;
+
+  // Deprecated:
+  // TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
+  // Remove this method once other other projects does not use it.
+  static BitrateController* CreateBitrateController(const Clock* clock,
+                                                    BitrateObserver* observer,
+                                                    RtcEventLog* event_log);
+
+  static BitrateController* CreateBitrateController(const Clock* clock,
+                                                    RtcEventLog* event_log);
+
+  virtual ~BitrateController() {}
+
+  // Creates RtcpBandwidthObserver caller responsible to delete.
+  virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 0;
+
+  // Deprecated
+  virtual void SetStartBitrate(int start_bitrate_bps) = 0;
+  // Deprecated
+  virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
+  virtual void SetBitrates(int start_bitrate_bps,
+                           int min_bitrate_bps,
+                           int max_bitrate_bps) = 0;
+
+  virtual void ResetBitrates(int bitrate_bps,
+                             int min_bitrate_bps,
+                             int max_bitrate_bps) = 0;
+
+  virtual void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) = 0;
+
+  // Gets the available payload bandwidth in bits per second. Note that
+  // this bandwidth excludes packet headers.
+  virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
+
+  virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0;
+
+  virtual bool GetNetworkParameters(uint32_t* bitrate,
+                                    uint8_t* fraction_loss,
+                                    int64_t* rtt) = 0;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_