Wire up bandwidth stats to the new API and webrtcvideoengine2.
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index b863957..3ab11274 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -962,8 +962,8 @@
config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc));
// Check for data populated by various sources. RTCP excluded as this
// data is received from remote side. Tested in call tests instead.
- const StreamStats& entry = stats.substreams[ssrc];
- if (entry.key_frames > 0u && entry.bitrate_bps > 0 &&
+ const SsrcStats& entry = stats.substreams[ssrc];
+ if (entry.key_frames > 0u && entry.total_bitrate_bps > 0 &&
entry.rtp_stats.packets > 0u && entry.avg_delay_ms > 0 &&
entry.max_delay_ms > 0) {
return true;
@@ -1045,20 +1045,20 @@
VideoSendStream::Stats stats = stream_->GetStats();
if (!stats.substreams.empty()) {
EXPECT_EQ(1u, stats.substreams.size());
- int bitrate_bps = stats.substreams.begin()->second.bitrate_bps;
- test::PrintResult(
- "bitrate_stats_",
- "min_transmit_bitrate_low_remb",
- "bitrate_bps",
- static_cast<size_t>(bitrate_bps),
- "bps",
- false);
- if (bitrate_bps > kHighBitrateBps) {
+ int total_bitrate_bps =
+ stats.substreams.begin()->second.total_bitrate_bps;
+ test::PrintResult("bitrate_stats_",
+ "min_transmit_bitrate_low_remb",
+ "bitrate_bps",
+ static_cast<size_t>(total_bitrate_bps),
+ "bps",
+ false);
+ if (total_bitrate_bps > kHighBitrateBps) {
rtp_rtcp_->SetREMBData(kRembBitrateBps, 1, &header.ssrc);
rtp_rtcp_->Process();
bitrate_capped_ = true;
} else if (bitrate_capped_ &&
- bitrate_bps < kRembRespectedBitrateBps) {
+ total_bitrate_bps < kRembRespectedBitrateBps) {
observation_complete_->Set();
}
}