Revert "Piping audio interruption metrics to API layer"
This reverts commit 299c4e68461f1c4428b2a919913b161115025dff.
Reason for revert: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/2753
../../chrome/browser/media/webrtc/webrtc_browsertest_base.cc(539): error: Expected equality of these values:
"ok-got-stats"
ExecuteJavascript("verifyLegacyStatsGenerated()", tab)
Which is: "Test failed: Error: \"googInterruptionCount\" is not a whitelisted stat. Exposing new metrics in the legacy getStats() API is not allowed. Please follow the standardization process: https://docs.google.com/document/d/1q1CJVUqJ6YW9NNRc0tENkLNny8AHrKZfqjy3SL89zjc/edit?usp=sharing\n at failTest (http://127.0.0.1:50650/webrtc/test_functions.js:46:15)\n at http://127.0.0.1:50650/webrtc/peerconnection.js:481:19"
With diff:
@@ -1,1 +1,3 @@
-ok-got-stats
+Test failed: Error: \"googInterruptionCount\" is not a whitelisted stat. Exposing new metrics in the legacy getStats() API is not allowed. Please follow the standardization process: https://docs.google.com/document/d/1q1CJVUqJ6YW9NNRc0tENkLNny8AHrKZfqjy3SL89zjc/edit?usp=sharing
+ at failTest (http://127.0.0.1:50650/webrtc/test_functions.js:46:15)
+ at http://127.0.0.1:50650/webrtc/peerconnection.js:481:19
Original change's description:
> Piping audio interruption metrics to API layer
>
> Bug: webrtc:10549
> Change-Id: Ie6abe5819c5b73dc5f5f89bdc375bad77f44ce97
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134303
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27788}
TBR=henrik.lundin@webrtc.org,kwiberg@webrtc.org,ivoc@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10549
Change-Id: I345306ba9758c0a3b1597724fa860d3e3fdb8995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134464
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27802}
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index c10a71c..da7d621 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -259,9 +259,6 @@
neteq_lifetime_stat.delayed_packet_outage_samples;
acm_stat->relativePacketArrivalDelayMs =
neteq_lifetime_stat.relative_packet_arrival_delay_ms;
- acm_stat->interruptionCount = neteq_lifetime_stat.interruption_count;
- acm_stat->totalInterruptionDurationMs =
- neteq_lifetime_stat.total_interruption_duration_ms;
NetEqOperationsAndState neteq_operations_and_state =
neteq_->GetOperationsAndState();
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index 621c478..8063a29 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -130,10 +130,6 @@
uint64_t delayedPacketOutageSamples;
// arrival delay of incoming packets
uint64_t relativePacketArrivalDelayMs;
- // number of audio interruptions
- int32_t interruptionCount;
- // total duration of audio interruptions
- int32_t totalInterruptionDurationMs;
};
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index d91850f..c0c836f 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -90,8 +90,8 @@
// An interruption is a loss-concealment event lasting at least 150 ms. The
// two stats below count the number os such events and the total duration of
// these events.
- int32_t interruption_count = 0;
- int32_t total_interruption_duration_ms = 0;
+ uint64_t interruption_count = 0;
+ uint64_t total_interruption_duration_ms = 0;
};
// Metrics that describe the operations performed in NetEq, and the internal
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 8269526..025ed69 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -745,7 +745,7 @@
}
auto lifetime_stats = neteq_->GetLifetimeStatistics();
- EXPECT_EQ(0, lifetime_stats.interruption_count);
+ EXPECT_EQ(0u, lifetime_stats.interruption_count);
}
// This test verifies that NetEq can handle comfort noise and enters/quits codec
diff --git a/modules/audio_coding/neteq/statistics_calculator_unittest.cc b/modules/audio_coding/neteq/statistics_calculator_unittest.cc
index abfa3c5..a851074 100644
--- a/modules/audio_coding/neteq/statistics_calculator_unittest.cc
+++ b/modules/audio_coding/neteq/statistics_calculator_unittest.cc
@@ -135,31 +135,31 @@
stats.DecodedOutputPlayed();
stats.EndExpandEvent(fs_hz);
auto lts = stats.GetLifetimeStatistics();
- EXPECT_EQ(0, lts.interruption_count);
- EXPECT_EQ(0, lts.total_interruption_duration_ms);
+ EXPECT_EQ(0u, lts.interruption_count);
+ EXPECT_EQ(0u, lts.total_interruption_duration_ms);
// Add an event that is shorter than 150 ms. Should not be logged.
stats.ExpandedVoiceSamples(10 * fs_khz, false); // 10 ms.
stats.ExpandedNoiseSamples(139 * fs_khz, false); // 139 ms.
stats.EndExpandEvent(fs_hz);
lts = stats.GetLifetimeStatistics();
- EXPECT_EQ(0, lts.interruption_count);
+ EXPECT_EQ(0u, lts.interruption_count);
// Add an event that is longer than 150 ms. Should be logged.
stats.ExpandedVoiceSamples(140 * fs_khz, false); // 140 ms.
stats.ExpandedNoiseSamples(11 * fs_khz, false); // 11 ms.
stats.EndExpandEvent(fs_hz);
lts = stats.GetLifetimeStatistics();
- EXPECT_EQ(1, lts.interruption_count);
- EXPECT_EQ(151, lts.total_interruption_duration_ms);
+ EXPECT_EQ(1u, lts.interruption_count);
+ EXPECT_EQ(151u, lts.total_interruption_duration_ms);
// Add one more long event.
stats.ExpandedVoiceSamples(100 * fs_khz, false); // 100 ms.
stats.ExpandedNoiseSamples(5000 * fs_khz, false); // 5000 ms.
stats.EndExpandEvent(fs_hz);
lts = stats.GetLifetimeStatistics();
- EXPECT_EQ(2, lts.interruption_count);
- EXPECT_EQ(5100 + 151, lts.total_interruption_duration_ms);
+ EXPECT_EQ(2u, lts.interruption_count);
+ EXPECT_EQ(5100u + 151u, lts.total_interruption_duration_ms);
}
TEST(StatisticsCalculator, InterruptionCounterDoNotLogBeforeDecoding) {
@@ -172,7 +172,7 @@
stats.ExpandedVoiceSamples(151 * fs_khz, false); // 151 ms.
stats.EndExpandEvent(fs_hz);
auto lts = stats.GetLifetimeStatistics();
- EXPECT_EQ(0, lts.interruption_count);
+ EXPECT_EQ(0u, lts.interruption_count);
// Call DecodedOutputPlayed(). Logging should happen after this.
stats.DecodedOutputPlayed();
@@ -181,7 +181,7 @@
stats.ExpandedVoiceSamples(151 * fs_khz, false); // 151 ms.
stats.EndExpandEvent(fs_hz);
lts = stats.GetLifetimeStatistics();
- EXPECT_EQ(1, lts.interruption_count);
+ EXPECT_EQ(1u, lts.interruption_count);
}
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc b/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
index 7c3aed8..c7bec9c 100644
--- a/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
+++ b/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
@@ -79,8 +79,9 @@
const auto lifetime_stats_vector = stats_getter_->lifetime_stats();
if (!lifetime_stats_vector->empty()) {
auto lifetime_stats = lifetime_stats_vector->back().second;
- printf(" num_interruptions: %d\n", lifetime_stats.interruption_count);
- printf(" sum_interruption_length_ms: %d ms\n",
+ printf(" num_interruptions: %" PRId64 "\n",
+ lifetime_stats.interruption_count);
+ printf(" sum_interruption_length_ms: %" PRId64 " ms\n",
lifetime_stats.total_interruption_duration_ms);
printf(" interruption ratio: %f%%\n",
100.0 * lifetime_stats.total_interruption_duration_ms /