Add new histograms WebRTC.Audio.(Speech)ExpandRatePercent
These two new histograms relate to the packet-loss concealment that
happens when audio packets are lost or late for decoding, and the
NetEq must resort to extrapolating audio from the previously
decoded data.
Bug: webrtc:9126
Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b
Reviewed-on: https://webrtc-review.googlesource.com/67861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22812}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index efa43ea..f6c6920 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1039,6 +1039,8 @@
"neteq/dtmf_tone_generator.h",
"neteq/expand.cc",
"neteq/expand.h",
+ "neteq/expand_uma_logger.cc",
+ "neteq/expand_uma_logger.h",
"neteq/include/neteq.h",
"neteq/merge.cc",
"neteq/merge.h",
diff --git a/modules/audio_coding/neteq/expand_uma_logger.cc b/modules/audio_coding/neteq/expand_uma_logger.cc
new file mode 100644
index 0000000..c656eed
--- /dev/null
+++ b/modules/audio_coding/neteq/expand_uma_logger.cc
@@ -0,0 +1,69 @@
+/* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/expand_uma_logger.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+std::unique_ptr<TickTimer::Countdown> GetNewCountdown(
+ const TickTimer& tick_timer,
+ int logging_period_s) {
+ return tick_timer.GetNewCountdown((logging_period_s * 1000) /
+ tick_timer.ms_per_tick());
+}
+} // namespace
+
+ExpandUmaLogger::ExpandUmaLogger(std::string uma_name,
+ int logging_period_s,
+ const TickTimer* tick_timer)
+ : uma_name_(uma_name),
+ logging_period_s_(logging_period_s),
+ tick_timer_(*tick_timer),
+ timer_(GetNewCountdown(tick_timer_, logging_period_s_)) {
+ RTC_DCHECK(tick_timer);
+ RTC_DCHECK_GT(logging_period_s_, 0);
+}
+
+ExpandUmaLogger::~ExpandUmaLogger() = default;
+
+void ExpandUmaLogger::UpdateSampleCounter(uint64_t samples,
+ int sample_rate_hz) {
+ if ((last_logged_value_ && *last_logged_value_ > samples) ||
+ sample_rate_hz_ != sample_rate_hz) {
+ // Sanity checks. The incremental counter moved backwards, or sample rate
+ // changed.
+ last_logged_value_.reset();
+ }
+ last_value_ = samples;
+ sample_rate_hz_ = sample_rate_hz;
+ if (!last_logged_value_) {
+ last_logged_value_ = rtc::Optional<uint64_t>(samples);
+ }
+
+ if (!timer_->Finished()) {
+ // Not yet time to log.
+ return;
+ }
+
+ RTC_DCHECK(last_logged_value_);
+ RTC_DCHECK_GE(last_value_, *last_logged_value_);
+ const uint64_t diff = last_value_ - *last_logged_value_;
+ last_logged_value_ = rtc::Optional<uint64_t>(last_value_);
+ // Calculate rate in percent.
+ RTC_DCHECK_GT(sample_rate_hz, 0);
+ const int rate = (100 * diff) / (sample_rate_hz * logging_period_s_);
+ RTC_DCHECK_GE(rate, 0);
+ RTC_DCHECK_LE(rate, 100);
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE(uma_name_, rate);
+ timer_ = GetNewCountdown(tick_timer_, logging_period_s_);
+}
+
+} // namespace webrtc
diff --git a/modules/audio_coding/neteq/expand_uma_logger.h b/modules/audio_coding/neteq/expand_uma_logger.h
new file mode 100644
index 0000000..70af39b
--- /dev/null
+++ b/modules/audio_coding/neteq/expand_uma_logger.h
@@ -0,0 +1,54 @@
+/* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
+
+#include <memory>
+#include <string>
+
+#include "api/optional.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "rtc_base/constructormagic.h"
+
+namespace webrtc {
+
+// This class is used to periodically log values to a UMA histogram. The caller
+// is expected to update this class with an incremental sample counter which
+// counts expand samples. At the end of each logging period, the class will
+// calculate the fraction of samples that were expand samples during that period
+// and report that in percent. The logging period must be strictly positive.
+// Does not take ownership of tick_timer and the pointer must refer to a valid
+// object that outlives the one constructed.
+class ExpandUmaLogger {
+ public:
+ ExpandUmaLogger(std::string uma_name,
+ int logging_period_s,
+ const TickTimer* tick_timer);
+
+ ~ExpandUmaLogger();
+
+ // In this call, value should be an incremental sample counter. The sample
+ // rate must be strictly positive.
+ void UpdateSampleCounter(uint64_t value, int sample_rate_hz);
+
+ private:
+ const std::string uma_name_;
+ const int logging_period_s_;
+ const TickTimer& tick_timer_;
+ std::unique_ptr<TickTimer::Countdown> timer_;
+ rtc::Optional<uint64_t> last_logged_value_;
+ uint64_t last_value_ = 0;
+ int sample_rate_hz_ = 0;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(ExpandUmaLogger);
+};
+
+} // namespace webrtc
+#endif // MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 4e780c3..6ce6a12 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -106,7 +106,13 @@
playout_mode_(config.playout_mode),
enable_fast_accelerate_(config.enable_fast_accelerate),
nack_enabled_(false),
- enable_muted_state_(config.enable_muted_state) {
+ enable_muted_state_(config.enable_muted_state),
+ expand_uma_logger_("WebRTC.Audio.ExpandRatePercent",
+ 10, // Report once every 10 s.
+ tick_timer_.get()),
+ speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
+ 10, // Report once every 10 s.
+ tick_timer_.get()) {
RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@@ -837,6 +843,11 @@
last_decoded_timestamps_.clear();
tick_timer_->Increment();
stats_.IncreaseCounter(output_size_samples_, fs_hz_);
+ const auto lifetime_stats = stats_.GetLifetimeStatistics();
+ expand_uma_logger_.UpdateSampleCounter(lifetime_stats.concealed_samples,
+ fs_hz_);
+ speech_expand_uma_logger_.UpdateSampleCounter(
+ lifetime_stats.voice_concealed_samples, fs_hz_);
// Check for muted state.
if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) {
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index bdeb020..3b7070f 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -17,6 +17,7 @@
#include "api/optional.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/defines.h"
+#include "modules/audio_coding/neteq/expand_uma_logger.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/packet.h" // Declare PacketList.
#include "modules/audio_coding/neteq/random_vector.h"
@@ -440,6 +441,8 @@
std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
RTC_GUARDED_BY(crit_sect_);
std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
+ ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
+ ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
