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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 <memory>
#include <set>
#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 SendPacketObserver {
public:
explicit SendDelayStats(Clock* clock);
~SendDelayStats() override;
// 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, int64_t time_ms);
protected:
// From SendPacketObserver.
// Called when a packet is sent to the transport.
void OnSendPacket(uint16_t packet_id,
int64_t capture_time_ms,
uint32_t ssrc) override;
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 {
Packet(uint32_t ssrc, int64_t capture_time_ms, int64_t send_time_ms)
: ssrc(ssrc),
capture_time_ms(capture_time_ms),
send_time_ms(send_time_ms) {}
uint32_t ssrc;
int64_t capture_time_ms;
int64_t send_time_ms;
};
typedef std::map<uint16_t, Packet, SequenceNumberOlderThan> PacketMap;
void UpdateHistograms();
void RemoveOld(int64_t now, PacketMap* packets)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
AvgCounter* GetSendDelayCounter(uint32_t ssrc)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
Clock* const clock_;
Mutex mutex_;
PacketMap packets_ RTC_GUARDED_BY(mutex_);
size_t num_old_packets_ RTC_GUARDED_BY(mutex_);
size_t num_skipped_packets_ RTC_GUARDED_BY(mutex_);
std::set<uint32_t> ssrcs_ RTC_GUARDED_BY(mutex_);
// Mapped by SSRC.
std::map<uint32_t, std::unique_ptr<AvgCounter>> send_delay_counters_
RTC_GUARDED_BY(mutex_);
};
} // namespace webrtc
#endif // VIDEO_SEND_DELAY_STATS_H_
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