perkj 825eb58d59 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )
Reason for revert:
Fails in the waterfall here:

https://build.chromium.org/p/client.webrtc/builders/Win32%20Debug/builds/7832/steps/rtc_media_unittests/logs/stdio

Original issue's description:
> Remove SendPacer from ViEEncoder
>
> This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to  Call (and BitrateAllocator)
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/857c5ccdb56e4c94196f7c6227abd5993c95abe2
> Cr-Commit-Position: refs/heads/master@{#12620}

TBR=stefan@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1947853002
Cr-Commit-Position: refs/heads/master@{#12621}
2016-05-04 08:08:15 +00:00

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2.9 KiB
C++

/*
* 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/include/module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class CriticalSectionWrapper;
class RtcEventLog;
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;
// 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 UpdateDelayBasedEstimate(uint32_t bitrate_bps) = 0;
virtual void SetEventLog(RtcEventLog* event_log) = 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_