BUG=1788,1667 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/44509004 Cr-Commit-Position: refs/heads/master@{#8575} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8575 4adac7df-926f-26a2-2b94-8c16560cd09d
129 lines
4.3 KiB
C++
129 lines
4.3 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video/receive_statistics_proxy.h"
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#include "webrtc/system_wrappers/interface/clock.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock)
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: clock_(clock),
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crit_(CriticalSectionWrapper::CreateCriticalSection()),
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// 1000ms window, scale 1000 for ms to s.
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decode_fps_estimator_(1000, 1000),
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renders_fps_estimator_(1000, 1000) {
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stats_.ssrc = ssrc;
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}
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ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {}
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VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
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CriticalSectionScoped lock(crit_.get());
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return stats_;
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}
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void ReceiveStatisticsProxy::IncomingRate(const int video_channel,
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const unsigned int framerate,
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const unsigned int bitrate_bps) {
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CriticalSectionScoped lock(crit_.get());
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stats_.network_frame_rate = framerate;
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stats_.total_bitrate_bps = bitrate_bps;
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}
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void ReceiveStatisticsProxy::DecoderTiming(int decode_ms,
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int max_decode_ms,
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int current_delay_ms,
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int target_delay_ms,
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int jitter_buffer_ms,
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int min_playout_delay_ms,
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int render_delay_ms) {
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CriticalSectionScoped lock(crit_.get());
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stats_.decode_ms = decode_ms;
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stats_.max_decode_ms = max_decode_ms;
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stats_.current_delay_ms = current_delay_ms;
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stats_.target_delay_ms = target_delay_ms;
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stats_.jitter_buffer_ms = jitter_buffer_ms;
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stats_.min_playout_delay_ms = min_playout_delay_ms;
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stats_.render_delay_ms = render_delay_ms;
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}
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void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
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uint32_t ssrc,
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const RtcpPacketTypeCounter& packet_counter) {
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CriticalSectionScoped lock(crit_.get());
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if (stats_.ssrc != ssrc)
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return;
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stats_.rtcp_packet_type_counts = packet_counter;
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}
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void ReceiveStatisticsProxy::StatisticsUpdated(
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const webrtc::RtcpStatistics& statistics,
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uint32_t ssrc) {
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CriticalSectionScoped lock(crit_.get());
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// TODO(pbos): Handle both local and remote ssrcs here and DCHECK that we
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// receive stats from one of them.
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if (stats_.ssrc != ssrc)
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return;
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stats_.rtcp_stats = statistics;
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}
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void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
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CriticalSectionScoped lock(crit_.get());
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// TODO(pbos): Handle both local and remote ssrcs here and DCHECK that we
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// receive stats from one of them.
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if (stats_.ssrc != ssrc)
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return;
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stats_.c_name = cname;
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}
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void ReceiveStatisticsProxy::DataCountersUpdated(
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const webrtc::StreamDataCounters& counters,
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uint32_t ssrc) {
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CriticalSectionScoped lock(crit_.get());
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if (stats_.ssrc != ssrc)
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return;
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stats_.rtp_stats = counters;
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}
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void ReceiveStatisticsProxy::OnDecodedFrame() {
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uint64_t now = clock_->TimeInMilliseconds();
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CriticalSectionScoped lock(crit_.get());
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decode_fps_estimator_.Update(1, now);
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stats_.decode_frame_rate = decode_fps_estimator_.Rate(now);
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}
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void ReceiveStatisticsProxy::OnRenderedFrame() {
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uint64_t now = clock_->TimeInMilliseconds();
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CriticalSectionScoped lock(crit_.get());
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renders_fps_estimator_.Update(1, now);
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stats_.render_frame_rate = renders_fps_estimator_.Rate(now);
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}
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void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
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uint32_t frameRate) {
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}
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void ReceiveStatisticsProxy::OnFrameCountsUpdated(
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const FrameCounts& frame_counts) {
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CriticalSectionScoped lock(crit_.get());
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stats_.frame_counts = frame_counts;
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}
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void ReceiveStatisticsProxy::OnDiscardedPacketsUpdated(int discarded_packets) {
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CriticalSectionScoped lock(crit_.get());
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stats_.discarded_packets = discarded_packets;
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}
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} // namespace webrtc
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