blob: becc0044abbcfe79e0ec15743dcdfb38a185ec75 [file] [log] [blame]
/*
* Copyright (c) 2020 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 "rtc_tools/rtc_event_log_visualizer/analyze_audio.h"
#include <memory>
#include <set>
#include <utility>
#include <vector>
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "rtc_base/ref_counted_object.h"
namespace webrtc {
void CreateAudioEncoderTargetBitrateGraph(const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
Plot* plot) {
TimeSeries time_series("Audio encoder target bitrate", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaBitrateBps = [](const LoggedAudioNetworkAdaptationEvent& ana_event)
-> absl::optional<float> {
if (ana_event.config.bitrate_bps)
return absl::optional<float>(
static_cast<float>(*ana_event.config.bitrate_bps));
return absl::nullopt;
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaBitrateBps,
parsed_log.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Bitrate (bps)", kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder target bitrate");
}
void CreateAudioEncoderFrameLengthGraph(const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
Plot* plot) {
TimeSeries time_series("Audio encoder frame length", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaFrameLengthMs =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.frame_length_ms)
return absl::optional<float>(
static_cast<float>(*ana_event.config.frame_length_ms));
return absl::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaFrameLengthMs,
parsed_log.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Frame length (ms)", kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder frame length");
}
void CreateAudioEncoderPacketLossGraph(const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
Plot* plot) {
TimeSeries time_series("Audio encoder uplink packet loss fraction",
LineStyle::kLine, PointStyle::kHighlight);
auto GetAnaPacketLoss =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.uplink_packet_loss_fraction)
return absl::optional<float>(static_cast<float>(
*ana_event.config.uplink_packet_loss_fraction));
return absl::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaPacketLoss,
parsed_log.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
kTopMargin);
plot->SetTitle("Reported audio encoder lost packets");
}
void CreateAudioEncoderEnableFecGraph(const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
Plot* plot) {
TimeSeries time_series("Audio encoder FEC", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaFecEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_fec)
return absl::optional<float>(
static_cast<float>(*ana_event.config.enable_fec));
return absl::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaFecEnabled,
parsed_log.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "FEC (false/true)", kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder FEC");
}
void CreateAudioEncoderEnableDtxGraph(const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
Plot* plot) {
TimeSeries time_series("Audio encoder DTX", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaDtxEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_dtx)
return absl::optional<float>(
static_cast<float>(*ana_event.config.enable_dtx));
return absl::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaDtxEnabled,
parsed_log.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "DTX (false/true)", kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder DTX");
}
void CreateAudioEncoderNumChannelsGraph(const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
Plot* plot) {
TimeSeries time_series("Audio encoder number of channels", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaNumChannels =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.num_channels)
return absl::optional<float>(
static_cast<float>(*ana_event.config.num_channels));
return absl::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaNumChannels,
parsed_log.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder number of channels");
}
class NetEqStreamInput : public test::NetEqInput {
public:
// Does not take any ownership, and all pointers must refer to valid objects
// that outlive the one constructed.
NetEqStreamInput(const std::vector<LoggedRtpPacketIncoming>* packet_stream,
const std::vector<LoggedAudioPlayoutEvent>* output_events,
absl::optional<int64_t> end_time_ms)
: packet_stream_(*packet_stream),
packet_stream_it_(packet_stream_.begin()),
output_events_it_(output_events->begin()),
output_events_end_(output_events->end()),
end_time_ms_(end_time_ms) {
RTC_DCHECK(packet_stream);
RTC_DCHECK(output_events);
}
absl::optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return absl::nullopt;
}
if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
return absl::nullopt;
}
return packet_stream_it_->rtp.log_time_ms();
}
absl::optional<int64_t> NextOutputEventTime() const override {
if (output_events_it_ == output_events_end_) {
return absl::nullopt;
}
if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
return absl::nullopt;
}
return output_events_it_->log_time_ms();
}
std::unique_ptr<PacketData> PopPacket() override {
if (packet_stream_it_ == packet_stream_.end()) {
return std::unique_ptr<PacketData>();
}
std::unique_ptr<PacketData> packet_data(new PacketData());
packet_data->header = packet_stream_it_->rtp.header;
packet_data->time_ms = packet_stream_it_->rtp.log_time_ms();
// This is a header-only "dummy" packet. Set the payload to all zeros, with
// length according to the virtual length.
packet_data->payload.SetSize(packet_stream_it_->rtp.total_length -
packet_stream_it_->rtp.header_length);
std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
++packet_stream_it_;
return packet_data;
}
void AdvanceOutputEvent() override {
if (output_events_it_ != output_events_end_) {
++output_events_it_;
}
}
bool ended() const override { return !NextEventTime(); }
absl::optional<RTPHeader> NextHeader() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return absl::nullopt;
}
return packet_stream_it_->rtp.header;
}
private:
const std::vector<LoggedRtpPacketIncoming>& packet_stream_;
std::vector<LoggedRtpPacketIncoming>::const_iterator packet_stream_it_;
std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_it_;
const std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_end_;
const absl::optional<int64_t> end_time_ms_;
};
namespace {
// Factory to create a "replacement decoder" that produces the decoded audio
// by reading from a file rather than from the encoded payloads.
class ReplacementAudioDecoderFactory : public AudioDecoderFactory {
public:
ReplacementAudioDecoderFactory(const absl::string_view replacement_file_name,
int file_sample_rate_hz)
: replacement_file_name_(replacement_file_name),
file_sample_rate_hz_(file_sample_rate_hz) {}
std::vector<AudioCodecSpec> GetSupportedDecoders() override {
RTC_NOTREACHED();
return {};
}
bool IsSupportedDecoder(const SdpAudioFormat& format) override {
return true;
}
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id) override {
auto replacement_file = std::make_unique<test::ResampleInputAudioFile>(
replacement_file_name_, file_sample_rate_hz_);
replacement_file->set_output_rate_hz(48000);
return std::make_unique<test::FakeDecodeFromFile>(
std::move(replacement_file), 48000, false);
}
private:
const std::string replacement_file_name_;
const int file_sample_rate_hz_;
};
// Creates a NetEq test object and all necessary input and output helpers. Runs
// the test and returns the NetEqDelayAnalyzer object that was used to
// instrument the test.
std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
const std::vector<LoggedRtpPacketIncoming>* packet_stream,
const std::vector<LoggedAudioPlayoutEvent>* output_events,
absl::optional<int64_t> end_time_ms,
const std::string& replacement_file_name,
int file_sample_rate_hz) {
std::unique_ptr<test::NetEqInput> input(
new NetEqStreamInput(packet_stream, output_events, end_time_ms));
constexpr int kReplacementPt = 127;
std::set<uint8_t> cn_types;
std::set<uint8_t> forbidden_types;
input.reset(new test::NetEqReplacementInput(std::move(input), kReplacementPt,
cn_types, forbidden_types));
NetEq::Config config;
config.max_packets_in_buffer = 200;
config.enable_fast_accelerate = true;
std::unique_ptr<test::VoidAudioSink> output(new test::VoidAudioSink());
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
new rtc::RefCountedObject<ReplacementAudioDecoderFactory>(
replacement_file_name, file_sample_rate_hz);
test::NetEqTest::DecoderMap codecs = {
{kReplacementPt, SdpAudioFormat("l16", 48000, 1)}};
std::unique_ptr<test::NetEqDelayAnalyzer> delay_cb(
new test::NetEqDelayAnalyzer);
std::unique_ptr<test::NetEqStatsGetter> neteq_stats_getter(
new test::NetEqStatsGetter(std::move(delay_cb)));
test::DefaultNetEqTestErrorCallback error_cb;
test::NetEqTest::Callbacks callbacks;
callbacks.error_callback = &error_cb;
callbacks.post_insert_packet = neteq_stats_getter->delay_analyzer();
callbacks.get_audio_callback = neteq_stats_getter.get();
test::NetEqTest test(config, decoder_factory, codecs, /*text_log=*/nullptr,
/*factory=*/nullptr, std::move(input), std::move(output),
callbacks);
test.Run();
return neteq_stats_getter;
}
} // namespace
NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
const std::string& replacement_file_name,
int file_sample_rate_hz) {
NetEqStatsGetterMap neteq_stats;
for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
const uint32_t ssrc = stream.ssrc;
if (!IsAudioSsrc(parsed_log, kIncomingPacket, ssrc))
continue;
const std::vector<LoggedRtpPacketIncoming>* audio_packets =
&stream.incoming_packets;
if (audio_packets == nullptr) {
// No incoming audio stream found.
continue;
}
RTC_DCHECK(neteq_stats.find(ssrc) == neteq_stats.end());
std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>::const_iterator
output_events_it = parsed_log.audio_playout_events().find(ssrc);
if (output_events_it == parsed_log.audio_playout_events().end()) {
// Could not find output events with SSRC matching the input audio stream.
// Using the first available stream of output events.
output_events_it = parsed_log.audio_playout_events().cbegin();
}
int64_t end_time_ms = parsed_log.first_log_segment().stop_time_ms();
neteq_stats[ssrc] = CreateNetEqTestAndRun(
audio_packets, &output_events_it->second, end_time_ms,
replacement_file_name, file_sample_rate_hz);
}
return neteq_stats;
}
// Given a NetEqStatsGetter and the SSRC that the NetEqStatsGetter was created
// for, this method generates a plot for the jitter buffer delay profile.
void CreateAudioJitterBufferGraph(const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
uint32_t ssrc,
const test::NetEqStatsGetter* stats_getter,
Plot* plot) {
test::NetEqDelayAnalyzer::Delays arrival_delay_ms;
test::NetEqDelayAnalyzer::Delays corrected_arrival_delay_ms;
test::NetEqDelayAnalyzer::Delays playout_delay_ms;
test::NetEqDelayAnalyzer::Delays target_delay_ms;
stats_getter->delay_analyzer()->CreateGraphs(
&arrival_delay_ms, &corrected_arrival_delay_ms, &playout_delay_ms,
&target_delay_ms);
TimeSeries time_series_packet_arrival("packet arrival delay",
LineStyle::kLine);
TimeSeries time_series_relative_packet_arrival(
"Relative packet arrival delay", LineStyle::kLine);
TimeSeries time_series_play_time("Playout delay", LineStyle::kLine);
TimeSeries time_series_target_time("Target delay", LineStyle::kLine,
PointStyle::kHighlight);
for (const auto& data : arrival_delay_ms) {
const float x = config.GetCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_packet_arrival.points.emplace_back(TimeSeriesPoint(x, y));
}
for (const auto& data : corrected_arrival_delay_ms) {
const float x = config.GetCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_relative_packet_arrival.points.emplace_back(
TimeSeriesPoint(x, y));
}
for (const auto& data : playout_delay_ms) {
const float x = config.GetCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_play_time.points.emplace_back(TimeSeriesPoint(x, y));
}
for (const auto& data : target_delay_ms) {
const float x = config.GetCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_target_time.points.emplace_back(TimeSeriesPoint(x, y));
}
plot->AppendTimeSeries(std::move(time_series_packet_arrival));
plot->AppendTimeSeries(std::move(time_series_relative_packet_arrival));
plot->AppendTimeSeries(std::move(time_series_play_time));
plot->AppendTimeSeries(std::move(time_series_target_time));
plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Relative delay (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("NetEq timing for " +
GetStreamName(parsed_log, kIncomingPacket, ssrc));
}
template <typename NetEqStatsType>
void CreateNetEqStatsGraphInternal(
const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
const NetEqStatsGetterMap& neteq_stats,
rtc::FunctionView<const std::vector<std::pair<int64_t, NetEqStatsType>>*(
const test::NetEqStatsGetter*)> data_extractor,
rtc::FunctionView<float(const NetEqStatsType&)> stats_extractor,
const std::string& plot_name,
Plot* plot) {
std::map<uint32_t, TimeSeries> time_series;
for (const auto& st : neteq_stats) {
const uint32_t ssrc = st.first;
const std::vector<std::pair<int64_t, NetEqStatsType>>* data_vector =
data_extractor(st.second.get());
for (const auto& data : *data_vector) {
const float time = config.GetCallTimeSec(data.first * 1000); // ms to us.
const float value = stats_extractor(data.second);
time_series[ssrc].points.emplace_back(TimeSeriesPoint(time, value));
}
}
for (auto& series : time_series) {
series.second.label =
GetStreamName(parsed_log, kIncomingPacket, series.first);
series.second.line_style = LineStyle::kLine;
plot->AppendTimeSeries(std::move(series.second));
}
plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, plot_name, kBottomMargin, kTopMargin);
plot->SetTitle(plot_name);
}
void CreateNetEqNetworkStatsGraph(
const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
const NetEqStatsGetterMap& neteq_stats,
rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
const std::string& plot_name,
Plot* plot) {
CreateNetEqStatsGraphInternal<NetEqNetworkStatistics>(
parsed_log, config, neteq_stats,
[](const test::NetEqStatsGetter* stats_getter) {
return stats_getter->stats();
},
stats_extractor, plot_name, plot);
}
void CreateNetEqLifetimeStatsGraph(
const ParsedRtcEventLog& parsed_log,
const AnalyzerConfig& config,
const NetEqStatsGetterMap& neteq_stats,
rtc::FunctionView<float(const NetEqLifetimeStatistics&)> stats_extractor,
const std::string& plot_name,
Plot* plot) {
CreateNetEqStatsGraphInternal<NetEqLifetimeStatistics>(
parsed_log, config, neteq_stats,
[](const test::NetEqStatsGetter* stats_getter) {
return stats_getter->lifetime_stats();
},
stats_extractor, plot_name, plot);
}
} // namespace webrtc