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Revert "Piping audio interruption metrics to API layer"

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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}
This commit is contained in:
Henrik Andreassson 2019-04-29 10:52:23 +00:00 committed by Commit Bot
parent 98499d5a20
commit fc02a793c2
14 changed files with 15 additions and 48 deletions
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4
api/stats_types.cc
View File

@ -565,8 +565,6 @@ const char* StatsReport::Value::display_name() const {
return "googInitiator";
case kStatsValueNameInterframeDelayMaxMs:
return "googInterframeDelayMax";
case kStatsValueNameInterruptionCount:
return "googInterruptionCount";
case kStatsValueNameIssuerId:
return "googIssuerId";
case kStatsValueNameJitterReceived:
@ -649,8 +647,6 @@ const char* StatsReport::Value::display_name() const {
return "googTrackId";
case kStatsValueNameTimingFrameInfo:
return "googTimingFrameInfo";
case kStatsValueNameTotalInterruptionDurationMs:
return "googTotalInterruptionDurationMs";
case kStatsValueNameTypingNoiseState:
return "googTypingNoiseState";
case kStatsValueNameWritable:

2
api/stats_types.h
View File

@ -192,7 +192,6 @@ class StatsReport {
kStatsValueNameHugeFramesSent,
kStatsValueNameInitiator,
kStatsValueNameInterframeDelayMaxMs, // Max over last 10 seconds.
kStatsValueNameInterruptionCount,
kStatsValueNameIssuerId,
kStatsValueNameJitterBufferMs,
kStatsValueNameJitterReceived,
@ -233,7 +232,6 @@ class StatsReport {
kStatsValueNameTargetDelayMs,
kStatsValueNameTargetEncBitrate,
kStatsValueNameTimingFrameInfo, // Result of |TimingFrameInfo::ToString|
kStatsValueNameTotalInterruptionDurationMs,
kStatsValueNameTrackId,
kStatsValueNameTransmitBitrate,
kStatsValueNameTransportType,

2
audio/audio_receive_stream.cc
View File

@ -226,8 +226,6 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
stats.relative_packet_arrival_delay_seconds =
static_cast<double>(ns.relativePacketArrivalDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec);
stats.interruption_count = ns.interruptionCount;
stats.total_interruption_duration_ms = ns.totalInterruptionDurationMs;
auto ds = channel_receive_->GetDecodingCallStatistics();
stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;

2
call/audio_receive_stream.h
View File

@ -79,8 +79,6 @@ class AudioReceiveStream {
absl::optional<int64_t> last_packet_received_timestamp_ms;
uint64_t jitter_buffer_flushes = 0;
double relative_packet_arrival_delay_seconds = 0.0;
int32_t interruption_count = 0;
int32_t total_interruption_duration_ms = 0;
};
struct Config {

4
media/base/media_channel.h
View File

@ -514,10 +514,6 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
uint64_t delayed_packet_outage_samples = 0;
// Arrival delay of received audio packets.
double relative_packet_arrival_delay_seconds = 0.0;
// Count and total duration of audio interruptions (loss-concealement periods
// longer than 150 ms).
int32_t interruption_count = 0;
int32_t total_interruption_duration_ms = 0;
};
struct VideoSenderInfo : public MediaSenderInfo {

2
media/engine/webrtc_voice_engine.cc
View File

@ -2264,8 +2264,6 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
rinfo.jitter_buffer_flushes = stats.jitter_buffer_flushes;
rinfo.relative_packet_arrival_delay_seconds =
stats.relative_packet_arrival_delay_seconds;
rinfo.interruption_count = stats.interruption_count;
rinfo.total_interruption_duration_ms = stats.total_interruption_duration_ms;
info->receivers.push_back(rinfo);
}

3
modules/audio_coding/acm2/acm_receiver.cc
View File

@ -259,9 +259,6 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
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();

4
modules/audio_coding/include/audio_coding_module_typedefs.h
View File

@ -130,10 +130,6 @@ struct NetworkStatistics {
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

4
modules/audio_coding/neteq/include/neteq.h
View File

@ -90,8 +90,8 @@ struct NetEqLifetimeStatistics {
// 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

2
modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -745,7 +745,7 @@ TEST_F(NetEqImplTest, NoAudioInterruptionLoggedBeforeFirstDecode) {
}
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

18
modules/audio_coding/neteq/statistics_calculator_unittest.cc
View File

@ -135,31 +135,31 @@ TEST(StatisticsCalculator, InterruptionCounter) {
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 @@ TEST(StatisticsCalculator, InterruptionCounterDoNotLogBeforeDecoding) {
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 @@ TEST(StatisticsCalculator, InterruptionCounterDoNotLogBeforeDecoding) {
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

5
modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
View File

@ -79,8 +79,9 @@ void NetEqStatsPlotter::SimulationEnded(int64_t simulation_time_ms) {
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 /

3
pc/stats_collector.cc
View File

@ -165,9 +165,6 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
{StatsReport::kStatsValueNamePacketsReceived, info.packets_rcvd},
{StatsReport::kStatsValueNamePreferredJitterBufferMs,
info.jitter_buffer_preferred_ms},
{StatsReport::kStatsValueNameInterruptionCount, info.interruption_count},
{StatsReport::kStatsValueNameTotalInterruptionDurationMs,
info.total_interruption_duration_ms},
};
for (const auto& f : floats)

8
pc/stats_collector_unittest.cc
View File

@ -383,14 +383,6 @@ void VerifyVoiceReceiverInfoReport(const StatsReport* report,
EXPECT_EQ(rtc::ToString(info.decoding_muted_output), value_in_report);
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameCodecName,
&value_in_report));
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameInterruptionCount,
&value_in_report));
EXPECT_EQ(rtc::ToString(info.interruption_count), value_in_report);
EXPECT_TRUE(GetValue(report,
StatsReport::kStatsValueNameTotalInterruptionDurationMs,
&value_in_report));
EXPECT_EQ(rtc::ToString(info.total_interruption_duration_ms),
value_in_report);
}
void VerifyVoiceSenderInfoReport(const StatsReport* report,