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Seperate NetEq stats getter to use in other tools.

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Bug: webrtc:9147
Change-Id: I251618bbb542d89b3d38c3ea424b1e55c0a5f2b2
Reviewed-on: https://webrtc-review.googlesource.com/69806
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22971}
This commit is contained in:
Minyue Li
2018-04-16 14:33:53 +02:00
committed by Commit Bot
parent 04d5f1d2e5
commit 2b415da8d0
4 changed files with 242 additions and 174 deletions
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181
modules/audio_coding/neteq/tools/neteq_rtpplay.cc
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@ -10,22 +10,18 @@
#include <errno.h>
#include <inttypes.h>
#include <iostream>
#include <limits.h> // For ULONG_MAX returned by strtoul.
#include <memory>
#include <stdio.h>
#include <stdlib.h> // For strtoul.
#include <string.h>
#include <algorithm>
#include <ios>
#include <iostream>
#include <memory>
#include <numeric>
#include <string>
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
@ -308,168 +304,6 @@ class SsrcSwitchDetector : public NetEqPostInsertPacket {
rtc::Optional<uint32_t> last_ssrc_;
};
class StatsGetter : public NetEqGetAudioCallback {
public:
// This struct is a replica of webrtc::NetEqNetworkStatistics, but with all
// values stored in double precision.
struct Stats {
double current_buffer_size_ms = 0.0;
double preferred_buffer_size_ms = 0.0;
double jitter_peaks_found = 0.0;
double packet_loss_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;
double median_waiting_time_ms = 0.0;
double min_waiting_time_ms = 0.0;
double max_waiting_time_ms = 0.0;
};
struct ConcealmentEvent {
uint64_t duration_ms;
size_t concealment_event_number;
int64_t time_from_previous_event_end_ms;
friend std::ostream& operator<<(std::ostream& stream,
const ConcealmentEvent& concealment_event) {
stream << "ConcealmentEvent duration_ms:" << concealment_event.duration_ms
<< " event_number:" << concealment_event.concealment_event_number
<< " time_from_previous_event_end_ms:"
<< concealment_event.time_from_previous_event_end_ms << "\n";
return stream;
}
};
// Takes a pointer to another callback object, which will be invoked after
// this object finishes. This does not transfer ownership, and null is a
// valid value.
explicit StatsGetter(NetEqGetAudioCallback* other_callback)
: other_callback_(other_callback) {}
void BeforeGetAudio(NetEq* neteq) override {
if (other_callback_) {
other_callback_->BeforeGetAudio(neteq);
}
}
void AfterGetAudio(int64_t time_now_ms,
const AudioFrame& audio_frame,
bool muted,
NetEq* neteq) override {
if (++counter_ >= 100) {
counter_ = 0;
NetEqNetworkStatistics stats;
RTC_CHECK_EQ(neteq->NetworkStatistics(&stats), 0);
stats_.push_back(stats);
}
const auto lifetime_stat = neteq->GetLifetimeStatistics();
if (current_concealment_event_ != lifetime_stat.concealment_events &&
voice_concealed_samples_until_last_event_ <
lifetime_stat.voice_concealed_samples) {
if (last_event_end_time_ms_ > 0) {
// Do not account for the first event to avoid start of the call
// skewing.
ConcealmentEvent concealment_event;
uint64_t last_event_voice_concealed_samples =
lifetime_stat.voice_concealed_samples -
voice_concealed_samples_until_last_event_;
RTC_CHECK_GT(last_event_voice_concealed_samples, 0);
concealment_event.duration_ms = last_event_voice_concealed_samples /
(audio_frame.sample_rate_hz_ / 1000);
concealment_event.concealment_event_number = current_concealment_event_;
concealment_event.time_from_previous_event_end_ms =
time_now_ms - last_event_end_time_ms_;
concealment_events_.emplace_back(concealment_event);
voice_concealed_samples_until_last_event_ =
lifetime_stat.voice_concealed_samples;
}
last_event_end_time_ms_ = time_now_ms;
voice_concealed_samples_until_last_event_ =
lifetime_stat.voice_concealed_samples;
current_concealment_event_ = lifetime_stat.concealment_events;
}
if (other_callback_) {
other_callback_->AfterGetAudio(time_now_ms, audio_frame, muted, neteq);
}
}
double AverageSpeechExpandRate() const {
double sum_speech_expand =
std::accumulate(stats_.begin(), stats_.end(), double{0.0},
[](double a, NetEqNetworkStatistics b) {
return a + static_cast<double>(b.speech_expand_rate);
});
return sum_speech_expand / 16384.0 / stats_.size();
}
const std::vector<ConcealmentEvent>& concealment_events() {
// Do not account for the last concealment event to avoid potential end
// call skewing.
return concealment_events_;
}
Stats AverageStats() const {
Stats sum_stats = std::accumulate(
stats_.begin(), stats_.end(), Stats(),
[](Stats a, NetEqNetworkStatistics b) {
a.current_buffer_size_ms += b.current_buffer_size_ms;
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.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;
a.median_waiting_time_ms += b.median_waiting_time_ms;
a.min_waiting_time_ms =
std::min(a.min_waiting_time_ms,
static_cast<double>(b.min_waiting_time_ms));
a.max_waiting_time_ms =
std::max(a.max_waiting_time_ms,
static_cast<double>(b.max_waiting_time_ms));
return a;
});
sum_stats.current_buffer_size_ms /= stats_.size();
sum_stats.preferred_buffer_size_ms /= stats_.size();
sum_stats.jitter_peaks_found /= stats_.size();
sum_stats.packet_loss_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();
sum_stats.median_waiting_time_ms /= stats_.size();
return sum_stats;
}
private:
NetEqGetAudioCallback* other_callback_;
size_t counter_ = 0;
std::vector<NetEqNetworkStatistics> stats_;
size_t current_concealment_event_ = 1;
uint64_t voice_concealed_samples_until_last_event_ = 0;
std::vector<ConcealmentEvent> concealment_events_;
int64_t last_event_end_time_ms_ = 0;
};
int RunTest(int argc, char* argv[]) {
std::string program_name = argv[0];
std::string usage = "Tool for decoding an RTP dump file using NetEq.\n"
@ -675,7 +509,7 @@ int RunTest(int argc, char* argv[]) {
SsrcSwitchDetector ssrc_switch_detector(delay_analyzer.get());
callbacks.post_insert_packet = &ssrc_switch_detector;
StatsGetter stats_getter(delay_analyzer.get());
NetEqStatsGetter stats_getter(std::move(delay_analyzer));
callbacks.get_audio_callback = &stats_getter;
NetEq::Config config;
config.sample_rate_hz = *sample_rate_hz;
@ -721,11 +555,10 @@ int RunTest(int argc, char* argv[]) {
printf(" current_buffer_size_ms: %f ms\n", stats.current_buffer_size_ms);
printf(" preferred_buffer_size_ms: %f ms\n", stats.preferred_buffer_size_ms);
if (FLAG_concealment_events) {
std::cout << " concealment_events_ms:"
<< "\n";
std::cout << " concealment_events_ms:" << std::endl;
for (auto concealment_event : stats_getter.concealment_events())
std::cout << concealment_event;
std::cout << " end of concealment_events_ms\n";
std::cout << concealment_event.ToString() << std::endl;
std::cout << " end of concealment_events_ms" << std::endl;
}
return 0;
}

140
modules/audio_coding/neteq/tools/neteq_stats_getter.cc Normal file
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@ -0,0 +1,140 @@
/*
* 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/tools/neteq_stats_getter.h"
#include <algorithm>
#include <numeric>
#include <utility>
#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
namespace test {
std::string NetEqStatsGetter::ConcealmentEvent::ToString() const {
char ss_buf[256];
rtc::SimpleStringBuilder ss(ss_buf);
ss << "ConcealmentEvent duration_ms:" << duration_ms
<< " event_number:" << concealment_event_number
<< " time_from_previous_event_end_ms:"
<< time_from_previous_event_end_ms;
return ss.str();
}
NetEqStatsGetter::NetEqStatsGetter(
std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer)
: delay_analyzer_(std::move(delay_analyzer)) {}
void NetEqStatsGetter::BeforeGetAudio(NetEq* neteq) {
if (delay_analyzer_) {
delay_analyzer_->BeforeGetAudio(neteq);
}
}
void NetEqStatsGetter::AfterGetAudio(int64_t time_now_ms,
const AudioFrame& audio_frame,
bool muted,
NetEq* neteq) {
if (++counter_ >= 100) {
counter_ = 0;
NetEqNetworkStatistics stats;
RTC_CHECK_EQ(neteq->NetworkStatistics(&stats), 0);
stats_.push_back(stats);
}
const auto lifetime_stat = neteq->GetLifetimeStatistics();
if (current_concealment_event_ != lifetime_stat.concealment_events &&
voice_concealed_samples_until_last_event_ <
lifetime_stat.voice_concealed_samples) {
if (last_event_end_time_ms_ > 0) {
// Do not account for the first event to avoid start of the call
// skewing.
ConcealmentEvent concealment_event;
uint64_t last_event_voice_concealed_samples =
lifetime_stat.voice_concealed_samples -
voice_concealed_samples_until_last_event_;
RTC_CHECK_GT(last_event_voice_concealed_samples, 0);
concealment_event.duration_ms = last_event_voice_concealed_samples /
(audio_frame.sample_rate_hz_ / 1000);
concealment_event.concealment_event_number = current_concealment_event_;
concealment_event.time_from_previous_event_end_ms =
time_now_ms - last_event_end_time_ms_;
concealment_events_.emplace_back(concealment_event);
voice_concealed_samples_until_last_event_ =
lifetime_stat.voice_concealed_samples;
}
last_event_end_time_ms_ = time_now_ms;
voice_concealed_samples_until_last_event_ =
lifetime_stat.voice_concealed_samples;
current_concealment_event_ = lifetime_stat.concealment_events;
}
if (delay_analyzer_) {
delay_analyzer_->AfterGetAudio(time_now_ms, audio_frame, muted, neteq);
}
}
double NetEqStatsGetter::AverageSpeechExpandRate() const {
double sum_speech_expand =
std::accumulate(stats_.begin(), stats_.end(), double{0.0},
[](double a, NetEqNetworkStatistics b) {
return a + static_cast<double>(b.speech_expand_rate);
});
return sum_speech_expand / 16384.0 / stats_.size();
}
NetEqStatsGetter::Stats NetEqStatsGetter::AverageStats() const {
Stats sum_stats = std::accumulate(
stats_.begin(), stats_.end(), Stats(),
[](Stats a, NetEqNetworkStatistics b) {
a.current_buffer_size_ms += b.current_buffer_size_ms;
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.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;
a.median_waiting_time_ms += b.median_waiting_time_ms;
a.min_waiting_time_ms =
std::min(a.min_waiting_time_ms,
static_cast<double>(b.min_waiting_time_ms));
a.max_waiting_time_ms =
std::max(a.max_waiting_time_ms,
static_cast<double>(b.max_waiting_time_ms));
return a;
});
sum_stats.current_buffer_size_ms /= stats_.size();
sum_stats.preferred_buffer_size_ms /= stats_.size();
sum_stats.jitter_peaks_found /= stats_.size();
sum_stats.packet_loss_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();
sum_stats.median_waiting_time_ms /= stats_.size();
return sum_stats;
}
} // namespace test
} // namespace webrtc

93
modules/audio_coding/neteq/tools/neteq_stats_getter.h Normal file
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@ -0,0 +1,93 @@
/*
* 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_TOOLS_NETEQ_STATS_GETTER_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
#include <memory>
#include <string>
#include <vector>
#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
namespace webrtc {
namespace test {
class NetEqStatsGetter : public NetEqGetAudioCallback {
public:
// This struct is a replica of webrtc::NetEqNetworkStatistics, but with all
// values stored in double precision.
struct Stats {
double current_buffer_size_ms = 0.0;
double preferred_buffer_size_ms = 0.0;
double jitter_peaks_found = 0.0;
double packet_loss_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;
double median_waiting_time_ms = 0.0;
double min_waiting_time_ms = 0.0;
double max_waiting_time_ms = 0.0;
};
struct ConcealmentEvent {
uint64_t duration_ms;
size_t concealment_event_number;
int64_t time_from_previous_event_end_ms;
std::string ToString() const;
};
// Takes a pointer to another callback object, which will be invoked after
// this object finishes. This does not transfer ownership, and null is a
// valid value.
explicit NetEqStatsGetter(std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer);
void BeforeGetAudio(NetEq* neteq) override;
void AfterGetAudio(int64_t time_now_ms,
const AudioFrame& audio_frame,
bool muted,
NetEq* neteq) override;
double AverageSpeechExpandRate() const;
NetEqDelayAnalyzer* delay_analyzer() const {
return delay_analyzer_.get();
}
const std::vector<ConcealmentEvent>& concealment_events() const {
// Do not account for the last concealment event to avoid potential end
// call skewing.
return concealment_events_;
}
Stats AverageStats() const;
private:
std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer_;
size_t counter_ = 0;
std::vector<NetEqNetworkStatistics> stats_;
size_t current_concealment_event_ = 1;
uint64_t voice_concealed_samples_until_last_event_ = 0;
std::vector<ConcealmentEvent> concealment_events_;
int64_t last_event_end_time_ms_ = 0;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_