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Remove expired histograms WebRTC.PeerConnection.SrtpCryptoSuite

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Fixed: chromium:1448119
Change-Id: Ibf903626f78860e2fb33e5f58b37276c106fdcbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308380
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40254}
This commit is contained in:
Johannes Kron 2023-06-09 10:49:09 +00:00 committed by WebRTC LUCI CQ
parent c7695a5b3a
commit 4133797557
3 changed files with 0 additions and 37 deletions
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25
pc/peer_connection.cc
View File

@ -2852,31 +2852,6 @@ void PeerConnection::ReportNegotiatedCiphers(
return;
}
if (srtp_crypto_suite != rtc::kSrtpInvalidCryptoSuite) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
rtc::kSrtpCryptoSuiteMaxValue);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
rtc::kSrtpCryptoSuiteMaxValue);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
rtc::kSrtpCryptoSuiteMaxValue);
break;
default:
RTC_DCHECK_NOTREACHED();
continue;
}
}
}
if (ssl_cipher_suite != rtc::kTlsNullWithNullNull) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {

8
pc/peer_connection_integrationtest.cc
View File

@ -1503,10 +1503,6 @@ TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
}
// Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
@ -1525,10 +1521,6 @@ TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the

4
pc/test/integration_test_helpers.h
View File

@ -1874,10 +1874,6 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
expected_cipher_suite));
}
void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,