Replacing NetEq discard rate with secondary discarded rate.
NetEq network statistics contains discard rate but has not been used and even not been implemented until recently.
According to w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsdiscarded,
this statistics needs to be replaced with an accumulative stats. Such work will be carried out separately.
Meanwhile, we need to add a rate to reflect rate of discarded redundant packets. See webrtc:8025.
In this CL, we replace the existing discard rate with secondary discarded rate, so as to
1. fulfill the requests on webrtc:8025
2. get ready to implement an accumulative statistics for discarded packets.
BUG: webrtc:7903,webrtc:8025
Change-Id: Idbf143a105db76ca15f0af54848e1448f2a810ec
Reviewed-on: https://chromium-review.googlesource.com/582863
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#19495}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 0c3ca753c5e9bafeb33ef304e36272aa152f45b7diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 0fbfe51..0b1d47a 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -65,7 +65,7 @@
const CodecInst kCodecInst = {
123, "codec_name_recv", 96000, -187, 0, -103};
const NetworkStatistics kNetworkStats = {
- 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
+ 123, 456, false, 0, {}, 789, 12, 345, 678, 901, 0, -1, -1, -1, -1, -1, 0};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
diff --git a/common_types.h b/common_types.h
index ecfa15d..4034d54 100644
--- a/common_types.h
+++ b/common_types.h
@@ -370,7 +370,9 @@
// Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
uint16_t currentPacketLossRate;
// Late loss rate; fraction between 0 and 1, scaled to Q14.
- uint16_t currentDiscardRate;
+ union {
+ RTC_DEPRECATED uint16_t currentDiscardRate;
+ };
// fraction (of original stream) of synthesized audio inserted through
// expansion (in Q14)
uint16_t currentExpandRate;
@@ -384,6 +386,8 @@
uint16_t currentAccelerateRate;
// fraction of data coming from secondary decoding (in Q14)
uint16_t currentSecondaryDecodedRate;
+ // fraction of secondary data that is discarded (in Q14).
+ uint16_t currentSecondaryDiscardedRate;
// clock-drift in parts-per-million (negative or positive)
int32_t clockDriftPPM;
// average packet waiting time in the jitter buffer (ms)
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 2e0d8bc..b512f84 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -320,12 +320,12 @@
acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
- acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
acm_stat->currentExpandRate = neteq_stat.expand_rate;
acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
+ acm_stat->currentSecondaryDiscardedRate = neteq_stat.secondary_discarded_rate;
acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm;
acm_stat->addedSamples = neteq_stat.added_zero_samples;
acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index 61bdd1f..d05b76e 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -35,7 +35,6 @@
uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
// jitter; 0 otherwise.
uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
- uint16_t packet_discard_rate; // Late loss rate in Q14.
uint16_t expand_rate; // Fraction (of original stream) of synthesized
// audio inserted through expansion (in Q14).
uint16_t speech_expand_rate; // Fraction (of original stream) of synthesized
@@ -44,8 +43,10 @@
// expansion (in Q14).
uint16_t accelerate_rate; // Fraction of data removed through acceleration
// (in Q14).
- uint16_t secondary_decoded_rate; // Fraction of data coming from secondary
+ uint16_t secondary_decoded_rate; // Fraction of data coming from FEC/RED
// decoding (in Q14).
+ uint16_t secondary_discarded_rate; // Fraction of discarded FEC/RED data (in
+ // Q14).
int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
// (positive or negative).
size_t added_zero_samples; // Number of zero samples added in "off" mode.
diff --git a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
index da4e3ed..c854b24 100644
--- a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
+++ b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
@@ -19,8 +19,8 @@
class MockStatisticsCalculator : public StatisticsCalculator {
public:
- // For current unittest, we mock only one method.
MOCK_METHOD1(PacketsDiscarded, void(size_t num_packets));
+ MOCK_METHOD1(SecondaryPacketsDiscarded, void(size_t num_packets));
};
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index 4feb448..bfa9362 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -136,12 +136,12 @@
logic preferred_buffer_size_ms;
logic jitter_peaks_found;
logic packet_loss_rate;
- logic packet_discard_rate;
logic expand_rate;
logic speech_expand_rate;
logic preemptive_rate;
logic accelerate_rate;
logic secondary_decoded_rate;
+ logic secondary_discarded_rate;
logic clockdrift_ppm;
logic added_zero_samples;
NetEqNetworkStatistics stats_ref;
@@ -201,12 +201,12 @@
CHECK_NETEQ_NETWORK_STATS(preferred_buffer_size_ms);
CHECK_NETEQ_NETWORK_STATS(jitter_peaks_found);
CHECK_NETEQ_NETWORK_STATS(packet_loss_rate);
- CHECK_NETEQ_NETWORK_STATS(packet_discard_rate);
CHECK_NETEQ_NETWORK_STATS(expand_rate);
CHECK_NETEQ_NETWORK_STATS(speech_expand_rate);
CHECK_NETEQ_NETWORK_STATS(preemptive_rate);
CHECK_NETEQ_NETWORK_STATS(accelerate_rate);
CHECK_NETEQ_NETWORK_STATS(secondary_decoded_rate);
+ CHECK_NETEQ_NETWORK_STATS(secondary_discarded_rate);
CHECK_NETEQ_NETWORK_STATS(clockdrift_ppm);
CHECK_NETEQ_NETWORK_STATS(added_zero_samples);
@@ -249,12 +249,12 @@
kIgnore, // preferred_buffer_size_ms
kIgnore, // jitter_peaks_found
kEqual, // packet_loss_rate
- kEqual, // packet_discard_rate
kEqual, // expand_rate
kEqual, // voice_expand_rate
kIgnore, // preemptive_rate
kEqual, // accelerate_rate
kEqual, // decoded_fec_rate
+ kEqual, // discarded_fec_rate
kIgnore, // clockdrift_ppm
kEqual, // added_zero_samples
{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}
@@ -273,7 +273,7 @@
expects.stats_ref.packet_loss_rate = 0;
expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 0;
expects.stats_ref.secondary_decoded_rate = 2006;
- expects.stats_ref.packet_discard_rate = 13374;
+ expects.stats_ref.secondary_discarded_rate = 14336;
RunTest(50, expects);
}
@@ -283,12 +283,12 @@
kIgnore, // preferred_buffer_size_ms
kIgnore, // jitter_peaks_found
kEqual, // packet_loss_rate
- kEqual, // packet_discard_rate
kEqual, // expand_rate
kEqual, // speech_expand_rate
kIgnore, // preemptive_rate
kEqual, // accelerate_rate
kEqual, // decoded_fec_rate
+ kEqual, // discard_fec_rate
kIgnore, // clockdrift_ppm
kEqual, // added_zero_samples
{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 31fb74e..ec7e2bb 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -75,12 +75,12 @@
stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
stats->set_packet_loss_rate(stats_raw.packet_loss_rate);
- stats->set_packet_discard_rate(stats_raw.packet_discard_rate);
stats->set_expand_rate(stats_raw.expand_rate);
stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
stats->set_preemptive_rate(stats_raw.preemptive_rate);
stats->set_accelerate_rate(stats_raw.accelerate_rate);
stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
+ stats->set_secondary_discarded_rate(stats_raw.secondary_discarded_rate);
stats->set_clockdrift_ppm(stats_raw.clockdrift_ppm);
stats->set_added_zero_samples(stats_raw.added_zero_samples);
stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
@@ -445,10 +445,10 @@
"759fef89a5de52bd17e733dc255c671ce86be909");
const std::string network_stats_checksum =
- PlatformChecksum("f7c2158761a531dd2804d13da0480033faa7be12",
- "8b5e3c8247dce48cb33923eaa1a502ca91429d5e",
- "f7c2158761a531dd2804d13da0480033faa7be12",
- "f7c2158761a531dd2804d13da0480033faa7be12");
+ PlatformChecksum("5b4262ca328e5f066af5d34f3380521583dd20de",
+ "80235b6d727281203acb63b98f9a9e85d95f7ec0",
+ "5b4262ca328e5f066af5d34f3380521583dd20de",
+ "5b4262ca328e5f066af5d34f3380521583dd20de");
const std::string rtcp_stats_checksum = PlatformChecksum(
"b8880bf9fed2487efbddcb8d94b9937a29ae521d",
@@ -481,10 +481,10 @@
"721e1e0c6effe4b2401536a4eef11512c9fb709c");
const std::string network_stats_checksum =
- PlatformChecksum("dda4cee006d9369c7114a03790c5761346cf5e23",
- "dda4cee006d9369c7114a03790c5761346cf5e23",
- "dda4cee006d9369c7114a03790c5761346cf5e23",
- "dda4cee006d9369c7114a03790c5761346cf5e23");
+ PlatformChecksum("4e749c46e2611877120ac7a20cbbe555cfbd70ea",
+ "4e749c46e2611877120ac7a20cbbe555cfbd70ea",
+ "4e749c46e2611877120ac7a20cbbe555cfbd70ea",
+ "4e749c46e2611877120ac7a20cbbe555cfbd70ea");
const std::string rtcp_stats_checksum = PlatformChecksum(
"e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
diff --git a/modules/audio_coding/neteq/neteq_unittest.proto b/modules/audio_coding/neteq/neteq_unittest.proto
index 07f21ff..344dcf6 100644
--- a/modules/audio_coding/neteq/neteq_unittest.proto
+++ b/modules/audio_coding/neteq/neteq_unittest.proto
@@ -3,16 +3,18 @@
package webrtc.neteq_unittest;
message NetEqNetworkStatistics {
+ // Next field number 18.
optional uint32 current_buffer_size_ms = 1;
optional uint32 preferred_buffer_size_ms = 2;
optional uint32 jitter_peaks_found = 3;
optional uint32 packet_loss_rate = 4;
- optional uint32 packet_discard_rate = 5;
+ optional uint32 packet_discard_rate = 5 [deprecated = true];
optional uint32 expand_rate = 6;
optional uint32 speech_expand_rate = 7;
optional uint32 preemptive_rate = 8;
optional uint32 accelerate_rate = 9;
optional uint32 secondary_decoded_rate = 10;
+ optional uint32 secondary_discarded_rate = 17;
optional int32 clockdrift_ppm = 11;
optional uint64 added_zero_samples = 12;
optional int32 mean_waiting_time_ms = 13;
diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc
index 1e71525..e0e2e9a 100644
--- a/modules/audio_coding/neteq/packet_buffer.cc
+++ b/modules/audio_coding/neteq/packet_buffer.cc
@@ -48,6 +48,16 @@
auto* di2 = decoder_database.GetDecoderInfo(pt2);
return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
}
+
+void LogPacketDiscarded(int codec_level, StatisticsCalculator* stats) {
+ RTC_CHECK(stats);
+ if (codec_level > 0) {
+ stats->SecondaryPacketsDiscarded(1);
+ } else {
+ stats->PacketsDiscarded(1);
+ }
+}
+
} // namespace
PacketBuffer::PacketBuffer(size_t max_number_of_packets,
@@ -99,8 +109,7 @@
// timestamp as |rit|, which has a higher priority, do not insert the new
// packet to list.
if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) {
- RTC_CHECK(stats);
- stats->PacketsDiscarded(1);
+ LogPacketDiscarded(packet.priority.codec_level, stats);
return return_val;
}
@@ -109,9 +118,8 @@
// packet.
PacketList::iterator it = rit.base();
if (it != buffer_.end() && packet.timestamp == it->timestamp) {
+ LogPacketDiscarded(packet.priority.codec_level, stats);
it = buffer_.erase(it);
- RTC_CHECK(stats);
- stats->PacketsDiscarded(1);
}
buffer_.insert(it, std::move(packet)); // Insert the packet at that position.
@@ -218,25 +226,24 @@
return kBufferEmpty;
}
// Assert that the packet sanity checks in InsertPacket method works.
- RTC_DCHECK(!buffer_.front().empty());
+ const Packet& packet = buffer_.front();
+ RTC_DCHECK(!packet.empty());
+ LogPacketDiscarded(packet.priority.codec_level, stats);
buffer_.pop_front();
- RTC_CHECK(stats);
- stats->PacketsDiscarded(1);
return kOK;
}
void PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
uint32_t horizon_samples,
StatisticsCalculator* stats) {
- const size_t old_size = buffer_.size();
- buffer_.remove_if([timestamp_limit, horizon_samples](const Packet& p) {
- return timestamp_limit != p.timestamp &&
- IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples);
+ buffer_.remove_if([timestamp_limit, horizon_samples, stats](const Packet& p) {
+ if (timestamp_limit == p.timestamp ||
+ !IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples)) {
+ return false;
+ }
+ LogPacketDiscarded(p.priority.codec_level, stats);
+ return true;
});
- if (old_size > buffer_.size()) {
- RTC_CHECK(stats);
- stats->PacketsDiscarded(old_size - buffer_.size());
- }
}
void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit,
@@ -246,20 +253,13 @@
void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type,
StatisticsCalculator* stats) {
- int packets_discarded = 0;
- for (auto it = buffer_.begin(); it != buffer_.end(); /* */) {
- const Packet& packet = *it;
- if (packet.payload_type == payload_type) {
- it = buffer_.erase(it);
- ++packets_discarded;
- } else {
- ++it;
+ buffer_.remove_if([payload_type, stats](const Packet& p) {
+ if (p.payload_type != payload_type) {
+ return false;
}
- }
- if (packets_discarded > 0) {
- RTC_CHECK(stats);
- stats->PacketsDiscarded(packets_discarded);
- }
+ LogPacketDiscarded(p.priority.codec_level, stats);
+ return true;
+ });
}
size_t PacketBuffer::NumPacketsInBuffer() const {
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 2622c0c..7dcf6c4 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -292,9 +292,13 @@
packet_facts[i].payload_type,
kFrameSize);
Packet packet = gen.NextPacket(kPayloadLength);
- packet.priority.red_level = packet_facts[i].primary ? 0 : 1;
+ packet.priority.codec_level = packet_facts[i].primary ? 0 : 1;
if (packet_facts[i].extract_order < 0) {
- EXPECT_CALL(mock_stats, PacketsDiscarded(1));
+ if (packet.priority.codec_level > 0) {
+ EXPECT_CALL(mock_stats, SecondaryPacketsDiscarded(1));
+ } else {
+ EXPECT_CALL(mock_stats, PacketsDiscarded(1));
+ }
}
EXPECT_CALL(check, Call(i));
EXPECT_EQ(PacketBuffer::kOK,
@@ -358,7 +362,8 @@
// This will discard all remaining packets but one. The oldest packet is older
// than the indicated horizon_samples, and will thus be left in the buffer.
constexpr size_t kSkipPackets = 1;
- EXPECT_CALL(mock_stats, PacketsDiscarded(kRemainingPackets - kSkipPackets));
+ EXPECT_CALL(mock_stats, PacketsDiscarded(1))
+ .Times(kRemainingPackets - kSkipPackets);
EXPECT_CALL(check, Call(17)); // Arbitrary id number.
buffer.DiscardOldPackets(start_ts + kTotalPackets * ts_increment,
kRemainingPackets * ts_increment, &mock_stats);
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index 145d99a..3faed62 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -123,14 +123,14 @@
lost_timestamps_(0),
timestamps_since_last_report_(0),
secondary_decoded_samples_(0),
+ discarded_secondary_packets_(0),
delayed_packet_outage_counter_(
"WebRTC.Audio.DelayedPacketOutageEventsPerMinute",
60000, // 60 seconds report interval.
100),
excess_buffer_delay_("WebRTC.Audio.AverageExcessBufferDelayMs",
60000, // 60 seconds report interval.
- 1000) {
-}
+ 1000) {}
StatisticsCalculator::~StatisticsCalculator() = default;
@@ -141,6 +141,7 @@
expanded_speech_samples_ = 0;
expanded_noise_samples_ = 0;
secondary_decoded_samples_ = 0;
+ discarded_secondary_packets_ = 0;
waiting_times_.clear();
}
@@ -184,6 +185,10 @@
discarded_packets_ += num_packets;
}
+void StatisticsCalculator::SecondaryPacketsDiscarded(size_t num_packets) {
+ discarded_secondary_packets_ += num_packets;
+}
+
void StatisticsCalculator::LostSamples(size_t num_samples) {
lost_timestamps_ += num_samples;
}
@@ -249,10 +254,6 @@
stats->packet_loss_rate =
CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_);
- const size_t discarded_samples = discarded_packets_ * samples_per_packet;
- stats->packet_discard_rate =
- CalculateQ14Ratio(discarded_samples, timestamps_since_last_report_);
-
stats->accelerate_rate =
CalculateQ14Ratio(accelerate_samples_, timestamps_since_last_report_);
@@ -271,6 +272,12 @@
CalculateQ14Ratio(secondary_decoded_samples_,
timestamps_since_last_report_);
+ const size_t discarded_secondary_samples =
+ discarded_secondary_packets_ * samples_per_packet;
+ stats->secondary_discarded_rate = CalculateQ14Ratio(
+ discarded_secondary_samples,
+ discarded_secondary_samples + secondary_decoded_samples_);
+
if (waiting_times_.size() == 0) {
stats->mean_waiting_time_ms = -1;
stats->median_waiting_time_ms = -1;
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index 0f8bdee..2877a16 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -66,6 +66,9 @@
// Reports that |num_packets| packets were discarded.
virtual void PacketsDiscarded(size_t num_packets);
+ // Reports that |num_packets| packets samples were discarded.
+ virtual void SecondaryPacketsDiscarded(size_t num_samples);
+
// Reports that |num_samples| were lost.
void LostSamples(size_t num_samples);
@@ -165,6 +168,7 @@
uint32_t timestamps_since_last_report_;
std::deque<int> waiting_times_;
uint32_t secondary_decoded_samples_;
+ size_t discarded_secondary_packets_;
PeriodicUmaCount delayed_packet_outage_counter_;
PeriodicUmaAverage excess_buffer_delay_;
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 3fd36bc..996d8d8 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -352,12 +352,12 @@
double preferred_buffer_size_ms = 0.0;
double jitter_peaks_found = 0.0;
double packet_loss_rate = 0.0;
- double packet_discard_rate = 0.0;
double expand_rate = 0.0;
double speech_expand_rate = 0.0;
double preemptive_rate = 0.0;
double accelerate_rate = 0.0;
double secondary_decoded_rate = 0.0;
+ double secondary_discarded_rate = 0.0;
double clockdrift_ppm = 0.0;
double added_zero_samples = 0.0;
double mean_waiting_time_ms = 0.0;
@@ -410,12 +410,12 @@
a.preferred_buffer_size_ms += b.preferred_buffer_size_ms;
a.jitter_peaks_found += b.jitter_peaks_found;
a.packet_loss_rate += b.packet_loss_rate / 16384.0;
- a.packet_discard_rate += b.packet_discard_rate / 16384.0;
a.expand_rate += b.expand_rate / 16384.0;
a.speech_expand_rate += b.speech_expand_rate / 16384.0;
a.preemptive_rate += b.preemptive_rate / 16384.0;
a.accelerate_rate += b.accelerate_rate / 16384.0;
a.secondary_decoded_rate += b.secondary_decoded_rate / 16384.0;
+ a.secondary_discarded_rate += b.secondary_discarded_rate / 16384.0;
a.clockdrift_ppm += b.clockdrift_ppm;
a.added_zero_samples += b.added_zero_samples;
a.mean_waiting_time_ms += b.mean_waiting_time_ms;
@@ -429,12 +429,12 @@
sum_stats.preferred_buffer_size_ms /= stats_.size();
sum_stats.jitter_peaks_found /= stats_.size();
sum_stats.packet_loss_rate /= stats_.size();
- sum_stats.packet_discard_rate /= stats_.size();
sum_stats.expand_rate /= stats_.size();
sum_stats.speech_expand_rate /= stats_.size();
sum_stats.preemptive_rate /= stats_.size();
sum_stats.accelerate_rate /= stats_.size();
sum_stats.secondary_decoded_rate /= stats_.size();
+ sum_stats.secondary_discarded_rate /= stats_.size();
sum_stats.clockdrift_ppm /= stats_.size();
sum_stats.added_zero_samples /= stats_.size();
sum_stats.mean_waiting_time_ms /= stats_.size();
@@ -644,13 +644,14 @@
printf(" output duration: %" PRId64 " ms\n", test_duration_ms);
auto stats = stats_getter.AverageStats();
printf(" packet_loss_rate: %f %%\n", 100.0 * stats.packet_loss_rate);
- printf(" packet_discard_rate: %f %%\n", 100.0 * stats.packet_discard_rate);
printf(" expand_rate: %f %%\n", 100.0 * stats.expand_rate);
printf(" speech_expand_rate: %f %%\n", 100.0 * stats.speech_expand_rate);
printf(" preemptive_rate: %f %%\n", 100.0 * stats.preemptive_rate);
printf(" accelerate_rate: %f %%\n", 100.0 * stats.accelerate_rate);
printf(" secondary_decoded_rate: %f %%\n",
100.0 * stats.secondary_decoded_rate);
+ printf(" secondary_discarded_rate: %f %%\n",
+ 100.0 * stats.secondary_discarded_rate);
printf(" clockdrift_ppm: %f ppm\n", stats.clockdrift_ppm);
printf(" mean_waiting_time_ms: %f ms\n", stats.mean_waiting_time_ms);
printf(" median_waiting_time_ms: %f ms\n", stats.median_waiting_time_ms);
diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index d18cdef..7e67e3f 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -752,8 +752,6 @@
networkStat.jitterPeaksFound);
fprintf(stdout, "packet-size rate........ %d\n",
networkStat.currentPacketLossRate);
- fprintf(stdout, "discard rate............ %d\n",
- networkStat.currentDiscardRate);
fprintf(stdout, "expand rate............. %d\n",
networkStat.currentExpandRate);
fprintf(stdout, "speech expand rate...... %d\n",
@@ -764,6 +762,8 @@
networkStat.currentAccelerateRate);
fprintf(stdout, "Secondary decoded rate.. %d\n",
networkStat.currentSecondaryDecodedRate);
+ fprintf(stdout, "Secondary discarded rate.%d\n",
+ networkStat.currentSecondaryDiscardedRate);
fprintf(stdout, "Clock-drift............. %d\n", networkStat.clockDriftPPM);
fprintf(stdout, "Mean waiting time....... %d\n",
networkStat.meanWaitingTimeMs);
