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803e3ff298
This removes the reserved bitrate feature as it is not currently used. This saves debugging time as there is less code to investigate. This also makes it easier to compare the code with the task queue based version which lacks this feature. Bug: webrtc:9586 Change-Id: I207624ceb8d203c88c3d01bfc753d60523f99fe4 Reviewed-on: https://webrtc-review.googlesource.com/92622 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24357}
100 lines
4.3 KiB
C++
100 lines
4.3 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*
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* Usage: this class will register multiple RtcpBitrateObserver's one at each
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* RTCP module. It will aggregate the results and run one bandwidth estimation
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* and push the result to the encoders via BitrateObserver(s).
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*/
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#ifndef MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
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#define MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
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#include <map>
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#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
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#include "modules/include/module.h"
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#include "modules/pacing/paced_sender.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class RtcEventLog;
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// Deprecated
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// TODO(perkj): Remove BitrateObserver when no implementations use it.
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class BitrateObserver {
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// Observer class for bitrate changes announced due to change in bandwidth
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// estimate or due to bitrate allocation changes. Fraction loss and rtt is
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// also part of this callback to allow the obsevrer to optimize its settings
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// for different types of network environments. The bitrate does not include
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// packet headers and is measured in bits per second.
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public:
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virtual void OnNetworkChanged(uint32_t bitrate_bps,
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uint8_t fraction_loss, // 0 - 255.
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int64_t rtt_ms) = 0;
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// TODO(gnish): Merge these two into one function.
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virtual void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
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uint32_t bitrate_for_pacer_bps,
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bool in_probe_rtt,
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int64_t target_set_time,
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uint64_t congestion_window) {}
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virtual void OnBytesAcked(size_t bytes) {}
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virtual size_t pacer_queue_size_in_bytes();
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virtual ~BitrateObserver() {}
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};
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class BitrateController : public Module, public RtcpBandwidthObserver {
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// This class collects feedback from all streams sent to a peer (via
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// RTCPBandwidthObservers). It does one aggregated send side bandwidth
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// estimation and divide the available bitrate between all its registered
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// BitrateObservers.
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public:
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static const int kDefaultStartBitratebps = 300000;
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// Deprecated:
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// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
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// Remove this method once other other projects does not use it.
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static BitrateController* CreateBitrateController(const Clock* clock,
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BitrateObserver* observer,
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RtcEventLog* event_log);
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static BitrateController* CreateBitrateController(const Clock* clock,
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RtcEventLog* event_log);
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~BitrateController() override {}
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// Deprecated, use raw pointer to BitrateController instance instead.
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// Creates RtcpBandwidthObserver caller responsible to delete.
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RTC_DEPRECATED virtual RtcpBandwidthObserver*
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CreateRtcpBandwidthObserver() = 0;
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// Deprecated
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virtual void SetStartBitrate(int start_bitrate_bps) = 0;
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// Deprecated
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virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
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virtual void SetBitrates(int start_bitrate_bps,
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int min_bitrate_bps,
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int max_bitrate_bps) = 0;
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virtual void ResetBitrates(int bitrate_bps,
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int min_bitrate_bps,
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int max_bitrate_bps) = 0;
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virtual void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) = 0;
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// Gets the available payload bandwidth in bits per second. Note that
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// this bandwidth excludes packet headers.
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virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
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virtual bool GetNetworkParameters(uint32_t* bitrate,
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uint8_t* fraction_loss,
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int64_t* rtt) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
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