Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions

View File

@ -0,0 +1,75 @@
/*
* 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_SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_
#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_
#include <list>
#include "webrtc/system_wrappers/include/ntp_time.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Class for converting an RTP timestamp to the NTP domain in milliseconds.
// The class needs to be trained with (at least 2) RTP/NTP timestamp pairs from
// RTCP sender reports before the convertion can be done.
class RtpToNtpEstimator {
public:
RtpToNtpEstimator();
~RtpToNtpEstimator();
// RTP and NTP timestamp pair from a RTCP SR report.
struct RtcpMeasurement {
RtcpMeasurement(uint32_t ntp_secs, uint32_t ntp_frac, uint32_t timestamp);
bool IsEqual(const RtcpMeasurement& other) const;
NtpTime ntp_time;
uint32_t rtp_timestamp;
};
// Estimated parameters from RTP and NTP timestamp pairs in |measurements_|.
struct Parameters {
double frequency_khz = 0.0;
double offset_ms = 0.0;
bool calculated = false;
};
// Updates measurements with RTP/NTP timestamp pair from a RTCP sender report.
// |new_rtcp_sr| is set to true if a new report is added.
bool UpdateMeasurements(uint32_t ntp_secs,
uint32_t ntp_frac,
uint32_t rtp_timestamp,
bool* new_rtcp_sr);
// Converts an RTP timestamp to the NTP domain in milliseconds.
// Returns true on success, false otherwise.
bool Estimate(int64_t rtp_timestamp, int64_t* rtp_timestamp_ms) const;
const Parameters& params() const { return params_; }
static const int kMaxInvalidSamples = 3;
private:
void UpdateParameters();
int consecutive_invalid_samples_;
std::list<RtcpMeasurement> measurements_;
Parameters params_;
};
// Returns:
// 1: forward wrap around.
// 0: no wrap around.
// -1: backwards wrap around (i.e. reordering).
int CheckForWrapArounds(uint32_t new_timestamp, uint32_t old_timestamp);
} // namespace webrtc
#endif // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_