NetEq: Simplify the dependencies of GetNetworkStatistics
Adds a new method PopulateDelayManagerStats which takes care of the
fields that needed information from the DelayManager.
Also adds a new test for StatisticsCalculator made practically
feasible by the refactoring.
Bug: webrtc:7554
Change-Id: Iff5cb5e209c276bd2784f2ccf73be8f619b1d955
Reviewed-on: https://webrtc-review.googlesource.com/3181
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19957}diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index cdf590b..2d50225 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -374,9 +374,11 @@
sync_buffer_->FutureLength();
assert(delay_manager_.get());
assert(decision_logic_.get());
+ const int ms_per_packet = rtc::dchecked_cast<int>(
+ decision_logic_->packet_length_samples() / (fs_hz_ / 1000));
+ stats_.PopulateDelayManagerStats(ms_per_packet, *delay_manager_.get(), stats);
stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
- decoder_frame_length_, *delay_manager_.get(),
- *decision_logic_.get(), stats);
+ decoder_frame_length_, stats);
return 0;
}
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index 09ced6a..4e034e6 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -14,7 +14,6 @@
#include <string.h> // memset
#include <algorithm>
-#include "modules/audio_coding/neteq/decision_logic.h"
#include "modules/audio_coding/neteq/delay_manager.h"
#include "rtc_base/checks.h"
#include "rtc_base/safe_conversions.h"
@@ -255,24 +254,13 @@
int fs_hz,
size_t num_samples_in_buffers,
size_t samples_per_packet,
- const DelayManager& delay_manager,
- const DecisionLogic& decision_logic,
NetEqNetworkStatistics *stats) {
- if (fs_hz <= 0 || !stats) {
- assert(false);
- return;
- }
+ RTC_DCHECK_GT(fs_hz, 0);
+ RTC_DCHECK(stats);
stats->added_zero_samples = added_zero_samples_;
stats->current_buffer_size_ms =
static_cast<uint16_t>(num_samples_in_buffers * 1000 / fs_hz);
- const int ms_per_packet = rtc::dchecked_cast<int>(
- decision_logic.packet_length_samples() / (fs_hz / 1000));
- stats->preferred_buffer_size_ms = (delay_manager.TargetLevel() >> 8) *
- ms_per_packet;
- stats->jitter_peaks_found = delay_manager.PeakFound();
- stats->clockdrift_ppm =
- rtc::saturated_cast<int32_t>(delay_manager.EstimatedClockDriftPpm());
stats->packet_loss_rate =
CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_);
@@ -331,6 +319,18 @@
Reset();
}
+void StatisticsCalculator::PopulateDelayManagerStats(
+ int ms_per_packet,
+ const DelayManager& delay_manager,
+ NetEqNetworkStatistics* stats) {
+ RTC_DCHECK(stats);
+ stats->preferred_buffer_size_ms =
+ (delay_manager.TargetLevel() >> 8) * ms_per_packet;
+ stats->jitter_peaks_found = delay_manager.PeakFound();
+ stats->clockdrift_ppm =
+ rtc::saturated_cast<int32_t>(delay_manager.EstimatedClockDriftPpm());
+}
+
NetEqLifetimeStatistics StatisticsCalculator::GetLifetimeStatistics() const {
return lifetime_stats_;
}
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index 232ca9a..5c2fbf3 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -20,8 +20,6 @@
namespace webrtc {
-// Forward declarations.
-class DecisionLogic;
class DelayManager;
// This class handles various network statistics in NetEq.
@@ -91,14 +89,22 @@
// Returns the current network statistics in |stats|. The current sample rate
// is |fs_hz|, the total number of samples in packet buffer and sync buffer
// yet to play out is |num_samples_in_buffers|, and the number of samples per
- // packet is |samples_per_packet|.
+ // packet is |samples_per_packet|. The method does not populate
+ // |preferred_buffer_size_ms|, |jitter_peaks_found| or |clockdrift_ppm|; use
+ // the PopulateDelayManagerStats method for those.
void GetNetworkStatistics(int fs_hz,
size_t num_samples_in_buffers,
size_t samples_per_packet,
- const DelayManager& delay_manager,
- const DecisionLogic& decision_logic,
NetEqNetworkStatistics *stats);
+ // Populates |preferred_buffer_size_ms|, |jitter_peaks_found| and
+ // |clockdrift_ppm| in |stats|. This is a convenience method, and does not
+ // strictly have to be in the StatisticsCalculator class, but it makes sense
+ // since all other stats fields are populated by that class.
+ static void PopulateDelayManagerStats(int ms_per_packet,
+ const DelayManager& delay_manager,
+ NetEqNetworkStatistics* stats);
+
// Returns a copy of this class's lifetime statistics. These statistics are
// never reset.
NetEqLifetimeStatistics GetLifetimeStatistics() const;
diff --git a/modules/audio_coding/neteq/statistics_calculator_unittest.cc b/modules/audio_coding/neteq/statistics_calculator_unittest.cc
index 0cc868a..0a4901d 100644
--- a/modules/audio_coding/neteq/statistics_calculator_unittest.cc
+++ b/modules/audio_coding/neteq/statistics_calculator_unittest.cc
@@ -63,4 +63,45 @@
EXPECT_EQ(200u, stats.GetLifetimeStatistics().concealed_samples);
}
+TEST(StatisticsCalculator, ExpandedSamplesCorrection) {
+ StatisticsCalculator stats;
+ NetEqNetworkStatistics stats_output;
+ constexpr int kSampleRateHz = 48000;
+ constexpr int k10MsSamples = kSampleRateHz / 100;
+ constexpr int kPacketSizeMs = 20;
+ constexpr size_t kSamplesPerPacket = kPacketSizeMs * kSampleRateHz / 1000;
+ // Assume 2 packets in the buffer.
+ constexpr size_t kNumSamplesInBuffer = 2 * kSamplesPerPacket;
+
+ // Advance time by 10 ms.
+ stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+
+ stats.GetNetworkStatistics(kSampleRateHz, kNumSamplesInBuffer,
+ kSamplesPerPacket, &stats_output);
+
+ EXPECT_EQ(0u, stats_output.expand_rate);
+ EXPECT_EQ(0u, stats_output.speech_expand_rate);
+
+ // Correct with a negative value.
+ stats.ExpandedVoiceSamplesCorrection(-100);
+ stats.ExpandedNoiseSamplesCorrection(-100);
+ stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+ stats.GetNetworkStatistics(kSampleRateHz, kNumSamplesInBuffer,
+ kSamplesPerPacket, &stats_output);
+ // Expect no change, since negative values are disallowed.
+ EXPECT_EQ(0u, stats_output.expand_rate);
+ EXPECT_EQ(0u, stats_output.speech_expand_rate);
+
+ // Correct with a positive value.
+ stats.ExpandedVoiceSamplesCorrection(50);
+ stats.ExpandedNoiseSamplesCorrection(200);
+ stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+ stats.GetNetworkStatistics(kSampleRateHz, kNumSamplesInBuffer,
+ kSamplesPerPacket, &stats_output);
+ // Calculate expected rates in Q14. Expand rate is noise + voice, while
+ // speech expand rate is only voice.
+ EXPECT_EQ(((50u + 200u) << 14) / k10MsSamples, stats_output.expand_rate);
+ EXPECT_EQ((50u << 14) / k10MsSamples, stats_output.speech_expand_rate);
+}
+
} // namespace webrtc
