blob: f479e52d763f5e24cdbbf899387b356295838b3f [file] [log] [blame]
* 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).
#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.
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.
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