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NetEq: Simplify the dependencies of GetNetworkStatistics

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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}
This commit is contained in:
Henrik Lundin
2017-09-25 12:30:58 +02:00
committed by Commit Bot
parent 13b668222a
commit dccfc405a6
4 changed files with 70 additions and 21 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -374,9 +374,11 @@ int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
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;
}

28
modules/audio_coding/neteq/statistics_calculator.cc
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@ -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 @@ void StatisticsCalculator::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) {
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 @@ void StatisticsCalculator::GetNetworkStatistics(
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_;
}

16
modules/audio_coding/neteq/statistics_calculator.h
View File

@ -20,8 +20,6 @@
namespace webrtc {
// Forward declarations.
class DecisionLogic;
class DelayManager;
// This class handles various network statistics in NetEq.
@ -91,14 +89,22 @@ class StatisticsCalculator {
// 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;

41
modules/audio_coding/neteq/statistics_calculator_unittest.cc
View File

@ -63,4 +63,45 @@ TEST(LifetimeStatistics, NoUpdateOnTimeStretch) {
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