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Revert "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."

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This reverts commit 870bca1f418a1abf445169a638a61f9a649d557f.

Reason for revert: it breaks internal tests and builds

Original change's description:
> Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
>
> We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> to report the metrics in pc/ and p2p/ that are currently been reported
> using MetricsObserverInterface.
>
> TBR=tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> Reviewed-on: https://webrtc-review.googlesource.com/83782
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23914}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hta@webrtc.org,tommi@webrtc.org

Change-Id: I1afd92d44f3b8cf3ae9aa6e6daa9a3a272e8097f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23916}
This commit is contained in:
Qingsi Wang
2018-07-10 20:53:10 +00:00
committed by Commit Bot
parent 79abc3d61a
commit 13f4c896d5
25 changed files with 510 additions and 307 deletions
Download Patch File Download Diff File Expand all files Collapse all files

41
pc/srtpsession_unittest.cc
View File

@ -13,21 +13,20 @@
#include <string>
#include "absl/memory/memory.h"
#include "api/fakemetricsobserver.h"
#include "media/base/fakertp.h"
#include "pc/srtptestutil.h"
#include "rtc_base/gunit.h"
#include "rtc_base/sslstreamadapter.h" // For rtc::SRTP_*
#include "system_wrappers/include/metrics_default.h"
#include "third_party/libsrtp/include/srtp.h"
namespace rtc {
using webrtc::FakeMetricsObserver;
std::vector<int> kEncryptedHeaderExtensionIds;
class SrtpSessionTest : public testing::Test {
public:
SrtpSessionTest() { webrtc::metrics::Reset(); }
protected:
virtual void SetUp() {
rtp_len_ = sizeof(kPcmuFrame);
@ -137,6 +136,9 @@ TEST_F(SrtpSessionTest, TestGetSendStreamPacketIndex) {
// Test that we fail to unprotect if someone tampers with the RTP/RTCP paylaods.
TEST_F(SrtpSessionTest, TestTamperReject) {
rtc::scoped_refptr<FakeMetricsObserver> metrics_observer(
new rtc::RefCountedObject<FakeMetricsObserver>());
s2_.SetMetricsObserver(metrics_observer);
int out_len;
EXPECT_TRUE(s1_.SetSend(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
kEncryptedHeaderExtensionIds));
@ -147,38 +149,29 @@ TEST_F(SrtpSessionTest, TestTamperReject) {
rtp_packet_[0] = 0x12;
rtcp_packet_[1] = 0x34;
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
EXPECT_EQ(1, webrtc::metrics::NumSamples(
"WebRTC.PeerConnection.SrtpUnprotectError"));
EXPECT_EQ(
1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
srtp_err_status_bad_param));
EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
webrtc::kEnumCounterSrtpUnprotectError, srtp_err_status_bad_param));
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
EXPECT_EQ(1, webrtc::metrics::NumSamples(
"WebRTC.PeerConnection.SrtcpUnprotectError"));
EXPECT_EQ(
1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
srtp_err_status_auth_fail));
EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
webrtc::kEnumCounterSrtcpUnprotectError, srtp_err_status_auth_fail));
}
// Test that we fail to unprotect if the payloads are not authenticated.
TEST_F(SrtpSessionTest, TestUnencryptReject) {
rtc::scoped_refptr<FakeMetricsObserver> metrics_observer(
new rtc::RefCountedObject<FakeMetricsObserver>());
s2_.SetMetricsObserver(metrics_observer);
int out_len;
EXPECT_TRUE(s1_.SetSend(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
kEncryptedHeaderExtensionIds));
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
EXPECT_EQ(1, webrtc::metrics::NumSamples(
"WebRTC.PeerConnection.SrtpUnprotectError"));
EXPECT_EQ(
1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
srtp_err_status_auth_fail));
EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
webrtc::kEnumCounterSrtpUnprotectError, srtp_err_status_auth_fail));
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
EXPECT_EQ(1, webrtc::metrics::NumSamples(
"WebRTC.PeerConnection.SrtcpUnprotectError"));
EXPECT_EQ(
1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
srtp_err_status_cant_check));
EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
webrtc::kEnumCounterSrtcpUnprotectError, srtp_err_status_cant_check));
}
// Test that we fail when using buffers that are too small.