ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
67 lines
2.6 KiB
C++
67 lines
2.6 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;
|
|
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_
|