Add support for transport wide sequence numbers
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc
index fd159f5..cf65ea0 100644
--- a/webrtc/video/call_perf_tests.cc
+++ b/webrtc/video/call_perf_tests.cc
@@ -528,6 +528,7 @@
static const int kMinAcceptableTransmitBitrate = 130;
static const int kMaxAcceptableTransmitBitrate = 170;
static const int kNumBitrateObservationsInRange = 100;
+ static const int kAcceptableBitrateErrorMargin = 15; // +- 7
class BitrateObserver : public test::EndToEndTest, public PacketReceiver {
public:
explicit BitrateObserver(bool using_min_transmit_bitrate)
@@ -567,8 +568,10 @@
}
} else {
// Expect bitrate stats to roughly match the max encode bitrate.
- if (bitrate_kbps > kMaxEncodeBitrateKbps - 5 &&
- bitrate_kbps < kMaxEncodeBitrateKbps + 5) {
+ if (bitrate_kbps > (kMaxEncodeBitrateKbps -
+ kAcceptableBitrateErrorMargin / 2) &&
+ bitrate_kbps < (kMaxEncodeBitrateKbps +
+ kAcceptableBitrateErrorMargin / 2)) {
++num_bitrate_observations_in_range_;
}
}
@@ -629,7 +632,9 @@
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
time_to_reconfigure_(webrtc::EventWrapper::Create()),
- encoder_inits_(0) {}
+ encoder_inits_(0),
+ last_set_bitrate_(0),
+ send_stream_(nullptr) {}
int32_t InitEncode(const VideoCodec* config,
int32_t number_of_cores,
