mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00

Bug: webrtc:9693 Change-Id: I0135e934c638ec391546928ba1e623d137b27b75 Reviewed-on: https://webrtc-review.googlesource.com/98600 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24668}
308 lines
12 KiB
C++
308 lines
12 KiB
C++
/*
|
|
* Copyright (c) 2017 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_delay_analyzer.h"
|
|
|
|
#include <algorithm>
|
|
#include <fstream>
|
|
#include <ios>
|
|
#include <iterator>
|
|
#include <limits>
|
|
#include <utility>
|
|
|
|
#include "absl/strings/string_view.h"
|
|
#include "modules/include/module_common_types.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
namespace {
|
|
constexpr char kArrivalDelayX[] = "arrival_delay_x";
|
|
constexpr char kArrivalDelayY[] = "arrival_delay_y";
|
|
constexpr char kTargetDelayX[] = "target_delay_x";
|
|
constexpr char kTargetDelayY[] = "target_delay_y";
|
|
constexpr char kPlayoutDelayX[] = "playout_delay_x";
|
|
constexpr char kPlayoutDelayY[] = "playout_delay_y";
|
|
|
|
// Helper function for NetEqDelayAnalyzer::CreateGraphs. Returns the
|
|
// interpolated value of a function at the point x. Vector x_vec contains the
|
|
// sample points, and y_vec contains the function values at these points. The
|
|
// return value is a linear interpolation between y_vec values.
|
|
double LinearInterpolate(double x,
|
|
const std::vector<int64_t>& x_vec,
|
|
const std::vector<int64_t>& y_vec) {
|
|
// Find first element which is larger than x.
|
|
auto it = std::upper_bound(x_vec.begin(), x_vec.end(), x);
|
|
if (it == x_vec.end()) {
|
|
--it;
|
|
}
|
|
const size_t upper_ix = it - x_vec.begin();
|
|
|
|
size_t lower_ix;
|
|
if (upper_ix == 0 || x_vec[upper_ix] <= x) {
|
|
lower_ix = upper_ix;
|
|
} else {
|
|
lower_ix = upper_ix - 1;
|
|
}
|
|
double y;
|
|
if (lower_ix == upper_ix) {
|
|
y = y_vec[lower_ix];
|
|
} else {
|
|
RTC_DCHECK_NE(x_vec[lower_ix], x_vec[upper_ix]);
|
|
y = (x - x_vec[lower_ix]) * (y_vec[upper_ix] - y_vec[lower_ix]) /
|
|
(x_vec[upper_ix] - x_vec[lower_ix]) +
|
|
y_vec[lower_ix];
|
|
}
|
|
return y;
|
|
}
|
|
|
|
void PrintDelays(const NetEqDelayAnalyzer::Delays& delays,
|
|
int64_t ref_time_ms,
|
|
absl::string_view var_name_x,
|
|
absl::string_view var_name_y,
|
|
std::ofstream& output,
|
|
const std::string& terminator = "") {
|
|
output << var_name_x << " = [ ";
|
|
for (const std::pair<int64_t, float>& delay : delays) {
|
|
output << (delay.first - ref_time_ms) / 1000.f << ", ";
|
|
}
|
|
output << "]" << terminator << std::endl;
|
|
|
|
output << var_name_y << " = [ ";
|
|
for (const std::pair<int64_t, float>& delay : delays) {
|
|
output << delay.second << ", ";
|
|
}
|
|
output << "]" << terminator << std::endl;
|
|
}
|
|
|
|
} // namespace
|
|
|
|
void NetEqDelayAnalyzer::AfterInsertPacket(
|
|
const test::NetEqInput::PacketData& packet,
|
|
NetEq* neteq) {
|
|
data_.insert(
|
|
std::make_pair(packet.header.timestamp, TimingData(packet.time_ms)));
|
|
ssrcs_.insert(packet.header.ssrc);
|
|
payload_types_.insert(packet.header.payloadType);
|
|
}
|
|
|
|
void NetEqDelayAnalyzer::BeforeGetAudio(NetEq* neteq) {
|
|
last_sync_buffer_ms_ = neteq->SyncBufferSizeMs();
|
|
}
|
|
|
|
void NetEqDelayAnalyzer::AfterGetAudio(int64_t time_now_ms,
|
|
const AudioFrame& audio_frame,
|
|
bool /*muted*/,
|
|
NetEq* neteq) {
|
|
get_audio_time_ms_.push_back(time_now_ms);
|
|
// Check what timestamps were decoded in the last GetAudio call.
|
|
std::vector<uint32_t> dec_ts = neteq->LastDecodedTimestamps();
|
|
// Find those timestamps in data_, insert their decoding time and sync
|
|
// delay.
|
|
for (uint32_t ts : dec_ts) {
|
|
auto it = data_.find(ts);
|
|
if (it == data_.end()) {
|
|
// This is a packet that was split out from another packet. Skip it.
|
|
continue;
|
|
}
|
|
auto& it_timing = it->second;
|
|
RTC_CHECK(!it_timing.decode_get_audio_count)
|
|
<< "Decode time already written";
|
|
it_timing.decode_get_audio_count = get_audio_count_;
|
|
RTC_CHECK(!it_timing.sync_delay_ms) << "Decode time already written";
|
|
it_timing.sync_delay_ms = last_sync_buffer_ms_;
|
|
it_timing.target_delay_ms = neteq->TargetDelayMs();
|
|
it_timing.current_delay_ms = neteq->FilteredCurrentDelayMs();
|
|
}
|
|
last_sample_rate_hz_ = audio_frame.sample_rate_hz_;
|
|
++get_audio_count_;
|
|
}
|
|
|
|
void NetEqDelayAnalyzer::CreateGraphs(Delays* arrival_delay_ms,
|
|
Delays* corrected_arrival_delay_ms,
|
|
Delays* playout_delay_ms,
|
|
Delays* target_delay_ms) const {
|
|
if (get_audio_time_ms_.empty()) {
|
|
return;
|
|
}
|
|
// Create nominal_get_audio_time_ms, a vector starting at
|
|
// get_audio_time_ms_[0] and increasing by 10 for each element.
|
|
std::vector<int64_t> nominal_get_audio_time_ms(get_audio_time_ms_.size());
|
|
nominal_get_audio_time_ms[0] = get_audio_time_ms_[0];
|
|
std::transform(
|
|
nominal_get_audio_time_ms.begin(), nominal_get_audio_time_ms.end() - 1,
|
|
nominal_get_audio_time_ms.begin() + 1, [](int64_t& x) { return x + 10; });
|
|
RTC_DCHECK(
|
|
std::is_sorted(get_audio_time_ms_.begin(), get_audio_time_ms_.end()));
|
|
|
|
std::vector<double> rtp_timestamps_ms;
|
|
double offset = std::numeric_limits<double>::max();
|
|
TimestampUnwrapper unwrapper;
|
|
// This loop traverses data_ and populates rtp_timestamps_ms as well as
|
|
// calculates the base offset.
|
|
for (auto& d : data_) {
|
|
rtp_timestamps_ms.push_back(
|
|
static_cast<double>(unwrapper.Unwrap(d.first)) /
|
|
rtc::CheckedDivExact(last_sample_rate_hz_, 1000));
|
|
offset =
|
|
std::min(offset, d.second.arrival_time_ms - rtp_timestamps_ms.back());
|
|
}
|
|
|
|
// This loop traverses the data again and populates the graph vectors. The
|
|
// reason to have two loops and traverse twice is that the offset cannot be
|
|
// known until the first traversal is done. Meanwhile, the final offset must
|
|
// be known already at the start of this second loop.
|
|
size_t i = 0;
|
|
for (const auto& data : data_) {
|
|
const double offset_send_time_ms = rtp_timestamps_ms[i++] + offset;
|
|
const auto& timing = data.second;
|
|
corrected_arrival_delay_ms->push_back(std::make_pair(
|
|
timing.arrival_time_ms,
|
|
LinearInterpolate(timing.arrival_time_ms, get_audio_time_ms_,
|
|
nominal_get_audio_time_ms) -
|
|
offset_send_time_ms));
|
|
arrival_delay_ms->push_back(std::make_pair(
|
|
timing.arrival_time_ms, timing.arrival_time_ms - offset_send_time_ms));
|
|
|
|
if (timing.decode_get_audio_count) {
|
|
// This packet was decoded.
|
|
RTC_DCHECK(timing.sync_delay_ms);
|
|
const int64_t get_audio_time =
|
|
*timing.decode_get_audio_count * 10 + get_audio_time_ms_[0];
|
|
const float playout_ms =
|
|
get_audio_time + *timing.sync_delay_ms - offset_send_time_ms;
|
|
playout_delay_ms->push_back(std::make_pair(get_audio_time, playout_ms));
|
|
RTC_DCHECK(timing.target_delay_ms);
|
|
RTC_DCHECK(timing.current_delay_ms);
|
|
const float target =
|
|
playout_ms - *timing.current_delay_ms + *timing.target_delay_ms;
|
|
target_delay_ms->push_back(std::make_pair(get_audio_time, target));
|
|
}
|
|
}
|
|
}
|
|
|
|
void NetEqDelayAnalyzer::CreateMatlabScript(
|
|
const std::string& script_name) const {
|
|
Delays arrival_delay_ms;
|
|
Delays corrected_arrival_delay_ms;
|
|
Delays playout_delay_ms;
|
|
Delays target_delay_ms;
|
|
CreateGraphs(&arrival_delay_ms, &corrected_arrival_delay_ms,
|
|
&playout_delay_ms, &target_delay_ms);
|
|
|
|
// Maybe better to find the actually smallest timestamp, to surely avoid
|
|
// x-axis starting from negative.
|
|
const int64_t ref_time_ms = arrival_delay_ms.front().first;
|
|
|
|
// Create an output file stream to Matlab script file.
|
|
std::ofstream output(script_name);
|
|
|
|
PrintDelays(corrected_arrival_delay_ms, ref_time_ms, kArrivalDelayX,
|
|
kArrivalDelayY, output, ";");
|
|
|
|
// PrintDelays(corrected_arrival_delay_x, kCorrectedArrivalDelayX,
|
|
// kCorrectedArrivalDelayY, output);
|
|
|
|
PrintDelays(playout_delay_ms, ref_time_ms, kPlayoutDelayX, kPlayoutDelayY,
|
|
output, ";");
|
|
|
|
PrintDelays(target_delay_ms, ref_time_ms, kTargetDelayX, kTargetDelayY,
|
|
output, ";");
|
|
|
|
output << "h=plot(" << kArrivalDelayX << ", " << kArrivalDelayY << ", "
|
|
<< kTargetDelayX << ", " << kTargetDelayY << ", 'g.', "
|
|
<< kPlayoutDelayX << ", " << kPlayoutDelayY << ");" << std::endl;
|
|
output << "set(h(1),'color',0.75*[1 1 1]);" << std::endl;
|
|
output << "set(h(2),'markersize',6);" << std::endl;
|
|
output << "set(h(3),'linew',1.5);" << std::endl;
|
|
output << "ax1=axis;" << std::endl;
|
|
output << "axis tight" << std::endl;
|
|
output << "ax2=axis;" << std::endl;
|
|
output << "axis([ax2(1:3) ax1(4)])" << std::endl;
|
|
output << "xlabel('time [s]');" << std::endl;
|
|
output << "ylabel('relative delay [ms]');" << std::endl;
|
|
if (!ssrcs_.empty()) {
|
|
auto ssrc_it = ssrcs_.cbegin();
|
|
output << "title('SSRC: 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
|
|
while (ssrc_it != ssrcs_.end()) {
|
|
output << ", 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
|
|
}
|
|
output << std::dec;
|
|
auto pt_it = payload_types_.cbegin();
|
|
output << "; Payload Types: " << *pt_it++;
|
|
while (pt_it != payload_types_.end()) {
|
|
output << ", " << *pt_it++;
|
|
}
|
|
output << "');" << std::endl;
|
|
}
|
|
}
|
|
|
|
void NetEqDelayAnalyzer::CreatePythonScript(
|
|
const std::string& script_name) const {
|
|
Delays arrival_delay_ms;
|
|
Delays corrected_arrival_delay_ms;
|
|
Delays playout_delay_ms;
|
|
Delays target_delay_ms;
|
|
CreateGraphs(&arrival_delay_ms, &corrected_arrival_delay_ms,
|
|
&playout_delay_ms, &target_delay_ms);
|
|
|
|
// Maybe better to find the actually smallest timestamp, to surely avoid
|
|
// x-axis starting from negative.
|
|
const int64_t ref_time_ms = arrival_delay_ms.front().first;
|
|
|
|
// Create an output file stream to the python script file.
|
|
std::ofstream output(script_name);
|
|
|
|
// Necessary includes
|
|
output << "import numpy as np" << std::endl;
|
|
output << "import matplotlib.pyplot as plt" << std::endl;
|
|
|
|
PrintDelays(corrected_arrival_delay_ms, ref_time_ms, kArrivalDelayX,
|
|
kArrivalDelayY, output);
|
|
|
|
// PrintDelays(corrected_arrival_delay_x, kCorrectedArrivalDelayX,
|
|
// kCorrectedArrivalDelayY, output);
|
|
|
|
PrintDelays(playout_delay_ms, ref_time_ms, kPlayoutDelayX, kPlayoutDelayY,
|
|
output);
|
|
|
|
PrintDelays(target_delay_ms, ref_time_ms, kTargetDelayX, kTargetDelayY,
|
|
output);
|
|
|
|
output << "if __name__ == '__main__':" << std::endl;
|
|
output << " h=plt.plot(" << kArrivalDelayX << ", " << kArrivalDelayY << ", "
|
|
<< kTargetDelayX << ", " << kTargetDelayY << ", 'g.', "
|
|
<< kPlayoutDelayX << ", " << kPlayoutDelayY << ")" << std::endl;
|
|
output << " plt.setp(h[0],'color',[.75, .75, .75])" << std::endl;
|
|
output << " plt.setp(h[1],'markersize',6)" << std::endl;
|
|
output << " plt.setp(h[2],'linewidth',1.5)" << std::endl;
|
|
output << " plt.axis('tight')" << std::endl;
|
|
output << " plt.xlabel('time [s]')" << std::endl;
|
|
output << " plt.ylabel('relative delay [ms]')" << std::endl;
|
|
if (!ssrcs_.empty()) {
|
|
auto ssrc_it = ssrcs_.cbegin();
|
|
output << " plt.title('SSRC: 0x" << std::hex
|
|
<< static_cast<int64_t>(*ssrc_it++);
|
|
while (ssrc_it != ssrcs_.end()) {
|
|
output << ", 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
|
|
}
|
|
output << std::dec;
|
|
auto pt_it = payload_types_.cbegin();
|
|
output << "; Payload Types: " << *pt_it++;
|
|
while (pt_it != payload_types_.end()) {
|
|
output << ", " << *pt_it++;
|
|
}
|
|
output << "')" << std::endl;
|
|
}
|
|
output << " plt.show()" << std::endl;
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|