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Add callbacks for receive channel RTCP statistics.

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This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.

TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.

BUG=2235
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
sprang@webrtc.org
2013-12-19 13:26:02 +00:00
parent e682aa5077
commit 54ae4ffb9e
12 changed files with 260 additions and 60 deletions
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83
webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
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@@ -131,4 +131,87 @@ TEST_F(ReceiveStatisticsTest, ActiveStatisticians) {
EXPECT_EQ(200u, bytes_received);
EXPECT_EQ(2u, packets_received);
}
TEST_F(ReceiveStatisticsTest, Callbacks) {
class TestCallback : public RtcpStatisticsCallback {
public:
TestCallback()
: RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {}
virtual ~TestCallback() {}
virtual void StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) {
ssrc_ = ssrc;
stats_ = statistics;
++num_calls_;
}
uint32_t num_calls_;
uint32_t ssrc_;
RtcpStatistics stats_;
} callback;
receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
// Add some arbitrary data, with loss and jitter.
header1_.sequenceNumber = 1;
clock_.AdvanceTimeMilliseconds(7);
header1_.timestamp += 3;
receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
header1_.sequenceNumber += 2;
clock_.AdvanceTimeMilliseconds(9);
header1_.timestamp += 9;
receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
--header1_.sequenceNumber;
clock_.AdvanceTimeMilliseconds(13);
header1_.timestamp += 47;
receive_statistics_->IncomingPacket(header1_, kPacketSize1, true);
header1_.sequenceNumber += 3;
clock_.AdvanceTimeMilliseconds(11);
header1_.timestamp += 17;
receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
++header1_.sequenceNumber;
EXPECT_EQ(0u, callback.num_calls_);
// Call GetStatistics, simulating a timed rtcp sender thread.
RtcpStatistics statistics;
receive_statistics_->GetStatistician(kSsrc1)
->GetStatistics(&statistics, true);
EXPECT_EQ(1u, callback.num_calls_);
EXPECT_EQ(callback.ssrc_, kSsrc1);
EXPECT_EQ(statistics.cumulative_lost, callback.stats_.cumulative_lost);
EXPECT_EQ(statistics.extended_max_sequence_number,
callback.stats_.extended_max_sequence_number);
EXPECT_EQ(statistics.fraction_lost, callback.stats_.fraction_lost);
EXPECT_EQ(statistics.jitter, callback.stats_.jitter);
receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
// Add some more data.
header1_.sequenceNumber = 1;
clock_.AdvanceTimeMilliseconds(7);
header1_.timestamp += 3;
receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
header1_.sequenceNumber += 2;
clock_.AdvanceTimeMilliseconds(9);
header1_.timestamp += 9;
receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
--header1_.sequenceNumber;
clock_.AdvanceTimeMilliseconds(13);
header1_.timestamp += 47;
receive_statistics_->IncomingPacket(header1_, kPacketSize1, true);
header1_.sequenceNumber += 3;
clock_.AdvanceTimeMilliseconds(11);
header1_.timestamp += 17;
receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
++header1_.sequenceNumber;
receive_statistics_->GetStatistician(kSsrc1)
->GetStatistics(&statistics, true);
// Should not have been called after deregister.
EXPECT_EQ(1u, callback.num_calls_);
}
} // namespace webrtc