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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /webrtc/modules/bitrate_controller/include/bitrate_controller.h
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
Diffstat (limited to 'webrtc/modules/bitrate_controller/include/bitrate_controller.h')
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+/*
+ * 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/interface/module.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+struct PacketInfo;
+
+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;
+
+ virtual ~BitrateObserver() {}
+};
+
+class BitrateController : public Module {
+ // 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 kDefaultStartBitrateKbps = 300;
+
+ static BitrateController* CreateBitrateController(Clock* clock,
+ BitrateObserver* observer);
+ virtual ~BitrateController() {}
+
+ virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 0;
+
+ virtual void SetStartBitrate(int start_bitrate_bps) = 0;
+ virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 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;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_