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Adding speech_expand_rate to NetEQ Network Statistics.

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There have been requests for separating rate of expanded speech samples from noise samples.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37309004

Cr-Commit-Position: refs/heads/master@{#8404}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8404 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org 2015-02-18 10:01:53 +00:00
parent 3864363e2c
commit 7d721eea14
5 changed files with 53 additions and 10 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/audio_coding/neteq/interface/neteq.h
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@ -33,7 +33,9 @@ struct NetEqNetworkStatistics {
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
// speech inserted through expansion (in Q14).
// audio inserted through expansion (in Q14).
uint16_t speech_expand_rate; // Fraction (of original stream) of synthesized
// speech inserted through expansion (in Q14).
uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
// expansion (in Q14).
uint16_t accelerate_rate; // Fraction of data removed through acceleration

41
webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
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@ -90,6 +90,7 @@ struct NetEqNetworkStatsCheck {
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;
@ -153,6 +154,7 @@ struct NetEqNetworkStatsCheck {
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);
@ -198,30 +200,55 @@ struct NetEqNetworkStatsCheck {
EQUAL, // packet_loss_rate
EQUAL, // packet_discard_rate
EQUAL, // expand_rate
EQUAL, // voice_expand_rate
IGNORE, // preemptive_rate
EQUAL, // accelerate_rate
EQUAL, // decoded_fec_rate
IGNORE, // clockdrift_ppm
EQUAL, // added_zero_samples
{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}
{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}
};
RunTest(50, expects);
// Next we introduce packet losses.
SetPacketLossRate(0.1);
expects.stats_ref.packet_loss_rate = 1337;
expects.stats_ref.expand_rate = 1065;
expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 1065;
RunTest(50, expects);
// Next we enable Opus FEC.
external_decoder_->set_fec_enabled(true);
// If FEC fills in the lost packets, no packet loss will be counted.
expects.stats_ref.packet_loss_rate = 0;
expects.stats_ref.expand_rate = 0;
expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 0;
expects.stats_ref.secondary_decoded_rate = 2006;
RunTest(50, expects);
}
void NoiseExpansionTest() {
NetEqNetworkStatsCheck expects = {
IGNORE, // current_buffer_size_ms
IGNORE, // preferred_buffer_size_ms
IGNORE, // jitter_peaks_found
EQUAL, // packet_loss_rate
EQUAL, // packet_discard_rate
EQUAL, // expand_rate
EQUAL, // speech_expand_rate
IGNORE, // preemptive_rate
EQUAL, // accelerate_rate
EQUAL, // decoded_fec_rate
IGNORE, // clockdrift_ppm
EQUAL, // added_zero_samples
{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}
};
RunTest(50, expects);
SetPacketLossRate(1);
expects.stats_ref.expand_rate = 16384;
expects.stats_ref.speech_expand_rate = 5324;
RunTest(10, expects); // Lost 10 * 20ms in a row.
}
private:
MockAudioDecoderOpus* external_decoder_;
const int samples_per_ms_;
@ -250,6 +277,14 @@ TEST(NetEqNetworkStatsTest, StereoOpusDecodeFec) {
EXPECT_CALL(decoder, Die()).Times(1);
}
TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
MockAudioDecoderOpus decoder(1);
EXPECT_CALL(decoder, Init());
NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
test.NoiseExpansionTest();
EXPECT_CALL(decoder, Die()).Times(1);
}
} // namespace test
} // namespace webrtc

4
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -147,7 +147,7 @@ void RefFiles::ReadFromFileAndCompare(
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
// speech inserted through expansion (in Q14).
// audio inserted through expansion (in Q14).
uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
// expansion (in Q14).
uint16_t accelerate_rate; // Fraction of data removed through acceleration
@ -168,11 +168,13 @@ void RefFiles::ReadFromFileAndCompare(
ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found);
ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate);
ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate);
ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate);
ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate);
ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate);
ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm);
ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples);
ASSERT_EQ(stats.secondary_decoded_rate, 0);
ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate);
}
}

12
webrtc/modules/audio_coding/neteq/statistics_calculator.cc
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@ -22,7 +22,7 @@ StatisticsCalculator::StatisticsCalculator()
: preemptive_samples_(0),
accelerate_samples_(0),
added_zero_samples_(0),
expanded_voice_samples_(0),
expanded_speech_samples_(0),
expanded_noise_samples_(0),
discarded_packets_(0),
lost_timestamps_(0),
@ -37,7 +37,7 @@ void StatisticsCalculator::Reset() {
preemptive_samples_ = 0;
accelerate_samples_ = 0;
added_zero_samples_ = 0;
expanded_voice_samples_ = 0;
expanded_speech_samples_ = 0;
expanded_noise_samples_ = 0;
secondary_decoded_samples_ = 0;
}
@ -55,7 +55,7 @@ void StatisticsCalculator::ResetWaitingTimeStatistics() {
}
void StatisticsCalculator::ExpandedVoiceSamples(int num_samples) {
expanded_voice_samples_ += num_samples;
expanded_speech_samples_ += num_samples;
}
void StatisticsCalculator::ExpandedNoiseSamples(int num_samples) {
@ -143,9 +143,13 @@ void StatisticsCalculator::GetNetworkStatistics(
CalculateQ14Ratio(preemptive_samples_, timestamps_since_last_report_);
stats->expand_rate =
CalculateQ14Ratio(expanded_voice_samples_ + expanded_noise_samples_,
CalculateQ14Ratio(expanded_speech_samples_ + expanded_noise_samples_,
timestamps_since_last_report_);
stats->speech_expand_rate =
CalculateQ14Ratio(expanded_speech_samples_,
timestamps_since_last_report_);
stats->secondary_decoded_rate =
CalculateQ14Ratio(secondary_decoded_samples_,
timestamps_since_last_report_);

2
webrtc/modules/audio_coding/neteq/statistics_calculator.h
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@ -96,7 +96,7 @@ class StatisticsCalculator {
uint32_t preemptive_samples_;
uint32_t accelerate_samples_;
int added_zero_samples_;
uint32_t expanded_voice_samples_;
uint32_t expanded_speech_samples_;
uint32_t expanded_noise_samples_;
int discarded_packets_;
uint32_t lost_timestamps_;