TEST=try and vie_auto_test with tsan. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2163004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4673 4adac7df-926f-26a2-2b94-8c16560cd09d
68 lines
1.6 KiB
C++
68 lines
1.6 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.
|
|
*/
|
|
|
|
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
|
|
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
|
|
|
|
#include <stdio.h>
|
|
|
|
#include <list>
|
|
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class Clock;
|
|
class CriticalSectionWrapper;
|
|
|
|
class Bitrate {
|
|
public:
|
|
explicit Bitrate(Clock* clock);
|
|
|
|
// Calculates rates.
|
|
void Process();
|
|
|
|
// Update with a packet.
|
|
void Update(const int32_t bytes);
|
|
|
|
// Packet rate last second, updated roughly every 100 ms.
|
|
uint32_t PacketRate() const;
|
|
|
|
// Bitrate last second, updated roughly every 100 ms.
|
|
uint32_t BitrateLast() const;
|
|
|
|
// Bitrate last second, updated now.
|
|
uint32_t BitrateNow() const;
|
|
|
|
int64_t time_last_rate_update() const;
|
|
|
|
protected:
|
|
Clock* clock_;
|
|
|
|
private:
|
|
scoped_ptr<CriticalSectionWrapper> crit_;
|
|
uint32_t packet_rate_;
|
|
uint32_t bitrate_;
|
|
uint8_t bitrate_next_idx_;
|
|
int64_t packet_rate_array_[10];
|
|
int64_t bitrate_array_[10];
|
|
int64_t bitrate_diff_ms_[10];
|
|
int64_t time_last_rate_update_;
|
|
uint32_t bytes_count_;
|
|
uint32_t packet_count_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
|