kjellander@webrtc.org 14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00

165 lines
5.7 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.
*/
#ifndef WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
#define WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
#include <list>
#include <set>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class BitrateProber;
class Clock;
class CriticalSectionWrapper;
namespace paced_sender {
class IntervalBudget;
struct Packet;
class PacketQueue;
} // namespace paced_sender
class PacedSender : public Module {
public:
enum Priority {
kHighPriority = 0, // Pass through; will be sent immediately.
kNormalPriority = 2, // Put in back of the line.
kLowPriority = 3, // Put in back of the low priority line.
};
// Low priority packets are mixed with the normal priority packets
// while we are paused.
class Callback {
public:
// Note: packets sent as a result of a callback should not pass by this
// module again.
// Called when it's time to send a queued packet.
// Returns false if packet cannot be sent.
virtual bool TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission) = 0;
// Called when it's a good time to send a padding data.
// Returns the number of bytes sent.
virtual size_t TimeToSendPadding(size_t bytes) = 0;
protected:
virtual ~Callback() {}
};
static const int64_t kDefaultMaxQueueLengthMs = 2000;
// Pace in kbits/s until we receive first estimate.
static const int kDefaultInitialPaceKbps = 2000;
// Pacing-rate relative to our target send rate.
// Multiplicative factor that is applied to the target bitrate to calculate
// the number of bytes that can be transmitted per interval.
// Increasing this factor will result in lower delays in cases of bitrate
// overshoots from the encoder.
static const float kDefaultPaceMultiplier;
PacedSender(Clock* clock,
Callback* callback,
int bitrate_kbps,
int max_bitrate_kbps,
int min_bitrate_kbps);
virtual ~PacedSender();
// Enable/disable pacing.
void SetStatus(bool enable);
bool Enabled() const;
// Temporarily pause all sending.
void Pause();
// Resume sending packets.
void Resume();
// Enable bitrate probing. Enabled by default, mostly here to simplify
// testing. Must be called before any packets are being sent to have an
// effect.
void SetProbingEnabled(bool enabled);
// Set target bitrates for the pacer.
// We will pace out bursts of packets at a bitrate of |max_bitrate_kbps|.
// |bitrate_kbps| is our estimate of what we are allowed to send on average.
// Padding packets will be utilized to reach |min_bitrate| unless enough media
// packets are available.
void UpdateBitrate(int bitrate_kbps,
int max_bitrate_kbps,
int min_bitrate_kbps);
// Returns true if we send the packet now, else it will add the packet
// information to the queue and call TimeToSendPacket when it's time to send.
virtual bool SendPacket(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission);
// Returns the time since the oldest queued packet was enqueued.
virtual int64_t QueueInMs() const;
virtual size_t QueueSizePackets() const;
// Returns the number of milliseconds it will take to send the current
// packets in the queue, given the current size and bitrate, ignoring prio.
virtual int64_t ExpectedQueueTimeMs() const;
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
int64_t TimeUntilNextProcess() override;
// Process any pending packets in the queue(s).
int32_t Process() override;
private:
// Updates the number of bytes that can be sent for the next time interval.
void UpdateBytesPerInterval(int64_t delta_time_in_ms)
EXCLUSIVE_LOCKS_REQUIRED(critsect_);
bool SendPacket(const paced_sender::Packet& packet)
EXCLUSIVE_LOCKS_REQUIRED(critsect_);
void SendPadding(size_t padding_needed) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
Clock* const clock_;
Callback* const callback_;
rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
bool enabled_ GUARDED_BY(critsect_);
bool paused_ GUARDED_BY(critsect_);
bool probing_enabled_;
// This is the media budget, keeping track of how many bits of media
// we can pace out during the current interval.
rtc::scoped_ptr<paced_sender::IntervalBudget> media_budget_
GUARDED_BY(critsect_);
// This is the padding budget, keeping track of how many bits of padding we're
// allowed to send out during the current interval. This budget will be
// utilized when there's no media to send.
rtc::scoped_ptr<paced_sender::IntervalBudget> padding_budget_
GUARDED_BY(critsect_);
rtc::scoped_ptr<BitrateProber> prober_ GUARDED_BY(critsect_);
int bitrate_bps_ GUARDED_BY(critsect_);
int64_t time_last_update_us_ GUARDED_BY(critsect_);
rtc::scoped_ptr<paced_sender::PacketQueue> packets_ GUARDED_BY(critsect_);
uint64_t packet_counter_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_