зеркало из https://github.com/mozilla/gecko-dev.git
Bug 1616875: Use timestamps for SSRC/CSRC statistics that are consistent with JS timestamps. r=ng
This means we won't have to re-implement all of our JSImpl workarounds in c++. Also, rename a variable in a way that made it easier to read this code. Differential Revision: https://phabricator.services.mozilla.com/D50395 --HG-- extra : moz-landing-system : lando
This commit is contained in:
Byron Campen [:bwc]
Родитель
f995b16e1b
Коммит
2c26ca9900
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@ -1,4 +1,5 @@
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#include <RtpSourceObserver.h>
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#include "RTCStatsReport.h"
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#include "webrtc/modules/include/module_common_types.h"
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#define GTEST_HAS_RTTI 0
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#include "gtest/gtest.h"
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@ -19,7 +20,7 @@ class RtpSourcesTest : public ::testing::Test {
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EXPECT_EQ(history.mDetailedHistory.size(), static_cast<size_t>(0));
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const auto& e = history.mLatestEviction;
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EXPECT_FALSE(history.mHasEvictedEntry);
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EXPECT_EQ(e.jitterAdjustedTimestamp, 0);
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EXPECT_EQ(e.predictedPlayoutTime, 0);
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EXPECT_FALSE(e.hasAudioLevel);
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EXPECT_EQ(e.audioLevel, 0);
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}
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@ -40,7 +41,7 @@ class RtpSourcesTest : public ::testing::Test {
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auto entry = history.FindClosestNotAfter(i + jitter);
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ASSERT_NE(entry, nullptr);
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if (entry) {
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EXPECT_EQ(entry->jitterAdjustedTimestamp, i + jitter);
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EXPECT_EQ(entry->predictedPlayoutTime, i + jitter);
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EXPECT_EQ(entry->hasAudioLevel, hasAudioLevel);
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EXPECT_EQ(entry->audioLevel, audioLevel);
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}
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@ -74,25 +75,25 @@ class RtpSourcesTest : public ::testing::Test {
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history.Insert(timeNow, time1, 1, true, 0);
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// Check that the jitter window buffer hasn't been used
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EXPECT_TRUE(history.Empty());
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ASSERT_EQ(history.mLatestEviction.jitterAdjustedTimestamp, time1);
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ASSERT_EQ(history.mLatestEviction.predictedPlayoutTime, time1);
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EXPECT_TRUE(history.mHasEvictedEntry);
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// time2
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history.Insert(timeNow, time2, 2, true, 0);
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EXPECT_TRUE(history.Empty());
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ASSERT_EQ(history.mLatestEviction.jitterAdjustedTimestamp, time2);
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ASSERT_EQ(history.mLatestEviction.predictedPlayoutTime, time2);
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EXPECT_TRUE(history.mHasEvictedEntry);
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// time3
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history.Insert(timeNow, time3, 3, true, 0);
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EXPECT_TRUE(history.Empty());
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ASSERT_EQ(history.mLatestEviction.jitterAdjustedTimestamp, time2);
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ASSERT_EQ(history.mLatestEviction.predictedPlayoutTime, time2);
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EXPECT_TRUE(history.mHasEvictedEntry);
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// pruneTime0
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history.Prune(pruneTime0);
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EXPECT_TRUE(history.Empty());
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ASSERT_EQ(history.mLatestEviction.jitterAdjustedTimestamp, time2);
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ASSERT_EQ(history.mLatestEviction.predictedPlayoutTime, time2);
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EXPECT_TRUE(history.mHasEvictedEntry);
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// pruneTime1
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@ -134,7 +135,7 @@ class RtpSourcesTest : public ::testing::Test {
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ASSERT_EQ(history.mMaxJitterWindow, jitterWindow);
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EXPECT_EQ(history.mDetailedHistory.size(), static_cast<size_t>(1));
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EXPECT_TRUE(history.mHasEvictedEntry);
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ASSERT_EQ(history.mLatestEviction.jitterAdjustedTimestamp, time0);
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ASSERT_EQ(history.mLatestEviction.predictedPlayoutTime, time0);
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ASSERT_EQ(history.mLatestEviction.hasAudioLevel, false);
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ASSERT_EQ(history.mLatestEviction.audioLevel, 1);
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@ -143,7 +144,7 @@ class RtpSourcesTest : public ::testing::Test {
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EXPECT_EQ(history.mMaxJitterWindow, jitterWindow);
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EXPECT_EQ(history.mDetailedHistory.size(), static_cast<size_t>(0));
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EXPECT_TRUE(history.mHasEvictedEntry);
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EXPECT_EQ(history.mLatestEviction.jitterAdjustedTimestamp, time1);
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EXPECT_EQ(history.mLatestEviction.predictedPlayoutTime, time1);
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EXPECT_EQ(history.mLatestEviction.hasAudioLevel, true);
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EXPECT_EQ(history.mLatestEviction.audioLevel, 2);
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}
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@ -217,7 +218,7 @@ class RtpSourcesTest : public ::testing::Test {
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history.Prune(timeNow + (jitter * 3) + 1);
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EXPECT_EQ(history.mDetailedHistory.size(), static_cast<size_t>(0));
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EXPECT_TRUE(history.mHasEvictedEntry);
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EXPECT_EQ(jitterAdjusted, history.mLatestEviction.jitterAdjustedTimestamp);
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EXPECT_EQ(jitterAdjusted, history.mLatestEviction.predictedPlayoutTime);
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}
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// Observer tests that keys are properly handled
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@ -242,7 +243,8 @@ class RtpSourcesTest : public ::testing::Test {
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// Observer a header with a single Csrc
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void TestObserveOneCsrc() {
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RtpSourceObserver observer;
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RtpSourceObserver observer =
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RtpSourceObserver(dom::RTCStatsTimestampMaker());
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webrtc::RTPHeader header;
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constexpr unsigned int ssrc = 857265;
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constexpr unsigned int csrc = 3268365;
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@ -257,7 +259,7 @@ class RtpSourcesTest : public ::testing::Test {
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// One for the SSRC, one for the CSRC
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EXPECT_EQ(observer.mRtpSources.size(), static_cast<size_t>(2));
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nsTArray<dom::RTCRtpSourceEntry> outLevels;
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observer.GetRtpSources(timestamp, outLevels);
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observer.GetRtpSources(outLevels);
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EXPECT_EQ(outLevels.Length(), static_cast<size_t>(2));
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bool ssrcFound = false;
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bool csrcFound = true;
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@ -278,7 +280,8 @@ class RtpSourcesTest : public ::testing::Test {
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// Observer a header with two CSRCs
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void TestObserveTwoCsrcs() {
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RtpSourceObserver observer;
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RtpSourceObserver observer =
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RtpSourceObserver(dom::RTCStatsTimestampMaker());
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webrtc::RTPHeader header;
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constexpr unsigned int ssrc = 239485;
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constexpr unsigned int csrc0 = 3425;
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@ -295,7 +298,7 @@ class RtpSourcesTest : public ::testing::Test {
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// One for the SSRC, two for the CSRCs
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EXPECT_EQ(observer.mRtpSources.size(), static_cast<size_t>(3));
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nsTArray<dom::RTCRtpSourceEntry> outLevels;
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observer.GetRtpSources(timestamp, outLevels);
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observer.GetRtpSources(outLevels);
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EXPECT_EQ(outLevels.Length(), static_cast<size_t>(3));
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bool ssrcFound = false;
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bool csrc0Found = true;
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@ -322,7 +325,8 @@ class RtpSourcesTest : public ::testing::Test {
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// Observer a header with a CSRC with audio level extension
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void TestObserveCsrcWithAudioLevel() {
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RtpSourceObserver observer;
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RtpSourceObserver observer =
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RtpSourceObserver(dom::RTCStatsTimestampMaker());
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webrtc::RTPHeader header;
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}
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};
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@ -311,9 +311,9 @@ void WebrtcAudioConduit::SetRtcpEventObserver(
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}
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void WebrtcAudioConduit::GetRtpSources(
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const int64_t aTimeNow, nsTArray<dom::RTCRtpSourceEntry>& outSources) {
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nsTArray<dom::RTCRtpSourceEntry>& outSources) {
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MOZ_ASSERT(NS_IsMainThread());
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return mRtpSourceObserver.GetRtpSources(aTimeNow, outSources);
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return mRtpSourceObserver.GetRtpSources(outSources);
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}
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// test-only: inserts a CSRC entry in a RtpSourceObserver's history for
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@ -197,6 +197,7 @@ class WebrtcAudioConduit : public AudioSessionConduit,
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mSendChannel(-1),
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mDtmfEnabled(false),
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mMutex("WebrtcAudioConduit::mMutex"),
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mRtpSourceObserver(mCall->GetTimestampMaker()),
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mStsThread(aStsThread) {}
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virtual ~WebrtcAudioConduit();
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@ -241,8 +242,7 @@ class WebrtcAudioConduit : public AudioSessionConduit,
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bool InsertDTMFTone(int channel, int eventCode, bool outOfBand, int lengthMs,
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int attenuationDb) override;
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void GetRtpSources(const int64_t aTimeNow,
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nsTArray<dom::RTCRtpSourceEntry>& outSources) override;
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void GetRtpSources(nsTArray<dom::RTCRtpSourceEntry>& outSources) override;
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void OnRtpPacket(const webrtc::RTPHeader& aRtpHeader,
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const int64_t aTimestamp, const uint32_t aJitter) override;
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@ -312,6 +312,10 @@ class WebRtcCallWrapper : public RefCounted<WebRtcCallWrapper> {
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DOMHighResTimeStamp GetNow() const { return mTimestampMaker.GetNow(); }
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const dom::RTCStatsTimestampMaker& GetTimestampMaker() const {
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return mTimestampMaker;
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}
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MOZ_DECLARE_REFCOUNTED_TYPENAME(WebRtcCallWrapper)
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> mDecoderFactory;
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@ -597,8 +601,7 @@ class AudioSessionConduit : public MediaSessionConduit {
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virtual bool InsertDTMFTone(int channel, int eventCode, bool outOfBand,
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int lengthMs, int attenuationDb) = 0;
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virtual void GetRtpSources(const int64_t aTimeNow,
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nsTArray<dom::RTCRtpSourceEntry>& outSources) = 0;
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virtual void GetRtpSources(nsTArray<dom::RTCRtpSourceEntry>& outSources) = 0;
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};
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} // namespace mozilla
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#endif
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@ -22,21 +22,35 @@ double RtpSourceObserver::RtpSourceEntry::ToLinearAudioLevel() const {
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return std::pow(10, -static_cast<double>(audioLevel) / 20);
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}
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RtpSourceObserver::RtpSourceObserver()
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: mMaxJitterWindow(0), mLevelGuard("RtpSourceObserver::mLevelGuard") {}
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RtpSourceObserver::RtpSourceObserver(
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const dom::RTCStatsTimestampMaker& aTimestampMaker)
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: mMaxJitterWindow(0),
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mLevelGuard("RtpSourceObserver::mLevelGuard"),
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mTimestampMaker(aTimestampMaker) {}
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void RtpSourceObserver::OnRtpPacket(const webrtc::RTPHeader& aHeader,
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const int64_t aTimestamp,
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const uint32_t aJitter) {
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MutexAutoLock lock(mLevelGuard);
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{
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// We assume that aTimestamp is not significantly in the past, and just get
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// a JS timestamp for the current time instead of converting aTimestamp.
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DOMHighResTimeStamp jsNow = mTimestampMaker.GetNow();
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mMaxJitterWindow =
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std::max(mMaxJitterWindow, static_cast<int64_t>(aJitter) * 2);
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const auto jitterAdjusted = aTimestamp + aJitter;
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// We are supposed to report the time at which this packet was played out,
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// but we have only just received the packet. We try to guess when it will
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// be played out.
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// TODO: We need to move where we update these stats to MediaPipeline, where
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// we send frames to the media track graph. In order to do that, we will
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// need to have the ssrc (and csrc) for decoded frames, but we don't have
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// that right now. Once we move this to the correct place, we will no longer
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// need to keep anything but the most recent data.
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const auto predictedPlayoutTime = jsNow + aJitter;
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auto& hist = mRtpSources[GetKey(aHeader.ssrc, EntryType::Synchronization)];
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hist.Prune(aTimestamp);
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hist.Prune(jsNow);
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// ssrc-audio-level handling
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hist.Insert(aTimestamp, jitterAdjusted, aHeader.timestamp,
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hist.Insert(jsNow, predictedPlayoutTime, aHeader.timestamp,
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aHeader.extension.hasAudioLevel, aHeader.extension.audioLevel);
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// csrc-audio-level handling
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@ -44,27 +58,27 @@ void RtpSourceObserver::OnRtpPacket(const webrtc::RTPHeader& aHeader,
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for (uint8_t i = 0; i < aHeader.numCSRCs; i++) {
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const uint32_t& csrc = aHeader.arrOfCSRCs[i];
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auto& hist = mRtpSources[GetKey(csrc, EntryType::Contributing)];
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hist.Prune(aTimestamp);
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hist.Prune(jsNow);
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bool hasLevel = i < list.numAudioLevels;
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uint8_t level = hasLevel ? list.arrOfAudioLevels[i] : 0;
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hist.Insert(aTimestamp, jitterAdjusted, aHeader.timestamp, hasLevel,
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hist.Insert(jsNow, predictedPlayoutTime, aHeader.timestamp, hasLevel,
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level);
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}
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}
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}
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void RtpSourceObserver::GetRtpSources(
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const int64_t aTimeNow,
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nsTArray<dom::RTCRtpSourceEntry>& outSources) const {
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MutexAutoLock lock(mLevelGuard);
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outSources.Clear();
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for (const auto& it : mRtpSources) {
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const RtpSourceEntry* entry = it.second.FindClosestNotAfter(aTimeNow);
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const RtpSourceEntry* entry =
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it.second.FindClosestNotAfter(mTimestampMaker.GetNow());
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if (entry) {
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dom::RTCRtpSourceEntry domEntry;
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domEntry.mSource = GetSourceFromKey(it.first);
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domEntry.mSourceType = GetTypeFromKey(it.first);
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domEntry.mTimestamp = entry->jitterAdjustedTimestamp;
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domEntry.mTimestamp = entry->predictedPlayoutTime;
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domEntry.mRtpTimestamp = entry->rtpTimestamp;
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if (entry->hasAudioLevel) {
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domEntry.mAudioLevel.Construct(entry->ToLinearAudioLevel());
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@ -74,10 +88,6 @@ void RtpSourceObserver::GetRtpSources(
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}
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}
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int64_t RtpSourceObserver::NowInReportClockTime() {
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return webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds();
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}
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const RtpSourceObserver::RtpSourceEntry*
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RtpSourceObserver::RtpSourceHistory::FindClosestNotAfter(int64_t aTime) const {
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// This method scans the history for the entry whose timestamp is closest to a
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@ -88,7 +98,7 @@ RtpSourceObserver::RtpSourceHistory::FindClosestNotAfter(int64_t aTime) const {
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auto lastFound = mDetailedHistory.cbegin();
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bool found = false;
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for (const auto& it : mDetailedHistory) {
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if (it.second.jitterAdjustedTimestamp > aTime) {
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if (it.second.predictedPlayoutTime > aTime) {
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break;
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}
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// lastFound can't start before begin, so the first inc must be skipped
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@ -100,7 +110,7 @@ RtpSourceObserver::RtpSourceHistory::FindClosestNotAfter(int64_t aTime) const {
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if (found) {
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return &lastFound->second;
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}
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if (HasEvicted() && aTime >= mLatestEviction.jitterAdjustedTimestamp) {
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if (HasEvicted() && aTime >= mLatestEviction.predictedPlayoutTime) {
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return &mLatestEviction;
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}
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return nullptr;
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@ -113,7 +123,7 @@ void RtpSourceObserver::RtpSourceHistory::Prune(const int64_t aTimeNow) {
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// New lower bound of the map
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auto lower = mDetailedHistory.begin();
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for (auto& it : mDetailedHistory) {
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if (it.second.jitterAdjustedTimestamp > aTimeT) {
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if (it.second.predictedPlayoutTime > aTimeT) {
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found = true;
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break;
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}
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@ -123,7 +133,7 @@ void RtpSourceObserver::RtpSourceHistory::Prune(const int64_t aTimeNow) {
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found = true;
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}
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if (found) {
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if (lower->second.jitterAdjustedTimestamp > aTimePrehistory) {
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if (lower->second.predictedPlayoutTime > aTimePrehistory) {
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mLatestEviction = lower->second;
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mHasEvictedEntry = true;
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}
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@ -131,7 +141,7 @@ void RtpSourceObserver::RtpSourceHistory::Prune(const int64_t aTimeNow) {
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mDetailedHistory.erase(mDetailedHistory.begin(), lower);
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}
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if (HasEvicted() &&
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(mLatestEviction.jitterAdjustedTimestamp + kHistoryWindow) < aTimeNow) {
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(mLatestEviction.predictedPlayoutTime + kHistoryWindow) < aTimeNow) {
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mHasEvictedEntry = false;
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}
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}
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@ -156,7 +166,7 @@ RtpSourceObserver::RtpSourceEntry& RtpSourceObserver::RtpSourceHistory::Insert(
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// x or x T J
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// |------Z-----|ABC| -> |------Z-----|ABC|
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if ((aTimestamp + kHistoryWindow) < aTimeNow ||
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aTimestamp < mLatestEviction.jitterAdjustedTimestamp) {
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aTimestamp < mLatestEviction.predictedPlayoutTime) {
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return mPrehistory; // A.K.A. /dev/null
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}
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mMaxJitterWindow = std::max(mMaxJitterWindow, (aTimestamp - aTimeNow) * 2);
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@ -14,6 +14,7 @@
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#include "nsISupportsImpl.h"
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#include "mozilla/dom/RTCRtpSourcesBinding.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_packet_observer.h"
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#include "RTCStatsReport.h"
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// Unit Test class
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namespace test {
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@ -29,29 +30,21 @@ namespace mozilla {
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*/
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class RtpSourceObserver : public webrtc::RtpPacketObserver {
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public:
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RtpSourceObserver();
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explicit RtpSourceObserver(
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const dom::RTCStatsTimestampMaker& aTimestampMaker);
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virtual ~RtpSourceObserver(){};
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void OnRtpPacket(const webrtc::RTPHeader& aRtpHeader,
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const int64_t aTimestamp, const uint32_t aJitter) override;
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/* Get the local time in MS from the same clock source that is used
|
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* to generate the capture timestamps. Use for computing the age of
|
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* an entry relative to another clock, e.g. the JS
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* @return time of now in MS
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*/
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static int64_t NowInReportClockTime();
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/*
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* Get the most recent 10 second window of CSRC and SSRC sources.
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* @param aTimeNow the current report clock time, @see NowInReportClockTime.
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* @param outLevels will be popluted with source entries
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||||
* Note: this takes jitter into account when calculating the window so
|
||||
* the window is actually [time - jitter - 10 sec .. time - jitter]
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*/
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void GetRtpSources(const int64_t aTimeNow,
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nsTArray<dom::RTCRtpSourceEntry>& outSources) const;
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void GetRtpSources(nsTArray<dom::RTCRtpSourceEntry>& outSources) const;
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private:
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||||
// Note: these are pool allocated
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||||
|
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@ -59,7 +52,7 @@ class RtpSourceObserver : public webrtc::RtpPacketObserver {
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|||
RtpSourceEntry() = default;
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||||
void Update(const int64_t aTimestamp, const uint32_t aRtpTimestamp,
|
||||
const bool aHasAudioLevel, const uint8_t aAudioLevel) {
|
||||
jitterAdjustedTimestamp = aTimestamp;
|
||||
predictedPlayoutTime = aTimestamp;
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||||
rtpTimestamp = aRtpTimestamp;
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||||
// Audio level range is 0 - 127 inclusive
|
||||
hasAudioLevel = aHasAudioLevel && !(aAudioLevel & 0x80);
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||||
|
|
@ -69,12 +62,19 @@ class RtpSourceObserver : public webrtc::RtpPacketObserver {
|
|||
// outlined in the webrtc-pc spec.
|
||||
double ToLinearAudioLevel() const;
|
||||
// Time this information was received + jitter
|
||||
int64_t jitterAdjustedTimestamp = 0;
|
||||
int64_t predictedPlayoutTime = 0;
|
||||
// The original RTP timestamp in the received packet
|
||||
uint32_t rtpTimestamp = 0;
|
||||
bool hasAudioLevel = false;
|
||||
uint8_t audioLevel = 0;
|
||||
};
|
||||
|
||||
/* Why this is needed:
|
||||
* We are supposed to only report source stats for packets that have already
|
||||
* been rendered. Unfortunately, we only know when these packets are
|
||||
* _received_ right now. So, we need to make a guess at when each packet will
|
||||
* be rendered, and hide its statistics until the clock reaches that estimate.
|
||||
*/
|
||||
/* Maintains a history of packets for reporting with getContributingSources
|
||||
* and getSynchronizationSources. It is expected that entries will not always
|
||||
* be observed in chronological order, and that the correct entry for a query
|
||||
|
|
@ -166,6 +166,7 @@ class RtpSourceObserver : public webrtc::RtpPacketObserver {
|
|||
int64_t mMaxJitterWindow;
|
||||
// Guards statistics
|
||||
mutable Mutex mLevelGuard;
|
||||
dom::RTCStatsTimestampMaker mTimestampMaker;
|
||||
|
||||
// Unit test
|
||||
friend test::RtpSourcesTest;
|
||||
|
|
|
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