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/*
* Copyright (c) 2016 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 VIDEO_SEND_DELAY_STATS_H_
#define VIDEO_SEND_DELAY_STATS_H_
#include <stddef.h>
#include <stdint.h>
#include <map>
#include "api/units/timestamp.h"
#include "call/video_send_stream.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
#include "video/stats_counter.h"
namespace webrtc {
// Used to collect delay stats for video streams. The class gets callbacks
// from more than one threads and internally uses a mutex for data access
// synchronization.
// TODO(bugs.webrtc.org/11993): OnSendPacket and OnSentPacket will eventually
// be called consistently on the same thread. Once we're there, we should be
// able to avoid locking (at least for the fast path).
class SendDelayStats {
public:
explicit SendDelayStats(Clock* clock);
~SendDelayStats();
// Adds the configured ssrcs for the rtp streams.
// Stats will be calculated for these streams.
void AddSsrcs(const VideoSendStream::Config& config);
// Called when a packet is sent (leaving socket).
bool OnSentPacket(int packet_id, Timestamp time);
// Called when a packet is sent to the transport.
void OnSendPacket(uint16_t packet_id, Timestamp capture_time, uint32_t ssrc);
private:
// Map holding sent packets (mapped by sequence number).
struct SequenceNumberOlderThan {
bool operator()(uint16_t seq1, uint16_t seq2) const {
return IsNewerSequenceNumber(seq2, seq1);
}
};
struct Packet {
AvgCounter* send_delay;
Timestamp capture_time;
Timestamp send_time;
};
void UpdateHistograms();
void RemoveOld(Timestamp now) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
Clock* const clock_;
Mutex mutex_;
std::map<uint16_t, Packet, SequenceNumberOlderThan> packets_
RTC_GUARDED_BY(mutex_);
size_t num_old_packets_ RTC_GUARDED_BY(mutex_);
size_t num_skipped_packets_ RTC_GUARDED_BY(mutex_);
// Mapped by SSRC.
std::map<uint32_t, AvgCounter> send_delay_counters_ RTC_GUARDED_BY(mutex_);
};
} // namespace webrtc
#endif // VIDEO_SEND_DELAY_STATS_H_
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