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webrtc_m130/modules/audio_coding/neteq/tools/neteq_test.cc
Jakob Ivarsson b5f4006589 Inject field trials in NetEqTest instead of setting global. 
We can avoid using the global now that field trials from Environment are
used in NetEq. This allows running multiple instances in parallel with
different settings.

Bug: webrtc:42220378
Change-Id: Icff8539e3ae9b61c86bb393d9a313e786e032b93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359720
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42894}
2024-08-30 09:11:50 +00:00

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/*
* Copyright (c) 2016 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_test.h"
#include <iomanip>
#include <iostream>
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/field_trials.h"
#include "api/units/timestamp.h"
#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace test {
namespace {
absl::optional<NetEq::Operation> ActionToOperations(
absl::optional<NetEqSimulator::Action> a) {
if (!a) {
return absl::nullopt;
}
switch (*a) {
case NetEqSimulator::Action::kAccelerate:
return absl::make_optional(NetEq::Operation::kAccelerate);
case NetEqSimulator::Action::kExpand:
return absl::make_optional(NetEq::Operation::kExpand);
case NetEqSimulator::Action::kNormal:
return absl::make_optional(NetEq::Operation::kNormal);
case NetEqSimulator::Action::kPreemptiveExpand:
return absl::make_optional(NetEq::Operation::kPreemptiveExpand);
}
}
std::unique_ptr<NetEq> CreateNetEq(
const Environment& env,
const NetEq::Config& config,
scoped_refptr<AudioDecoderFactory> decoder_factory) {
return DefaultNetEqFactory().Create(env, config, std::move(decoder_factory));
}
} // namespace
void DefaultNetEqTestErrorCallback::OnInsertPacketError(
const NetEqInput::PacketData& packet) {
std::cerr << "InsertPacket returned an error." << std::endl;
std::cerr << "Packet data: " << packet.ToString() << std::endl;
RTC_FATAL();
}
void DefaultNetEqTestErrorCallback::OnGetAudioError() {
std::cerr << "GetAudio returned an error." << std::endl;
RTC_FATAL();
}
NetEqTest::NetEqTest(const NetEq::Config& config,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
const DecoderMap& codecs,
std::unique_ptr<std::ofstream> text_log,
NetEqFactory* neteq_factory,
std::unique_ptr<NetEqInput> input,
std::unique_ptr<AudioSink> output,
Callbacks callbacks,
absl::string_view field_trials)
: input_(std::move(input)),
clock_(Timestamp::Millis(input_->NextEventTime().value_or(0))),
env_(CreateEnvironment(
&clock_,
FieldTrials::CreateNoGlobal(std::string(field_trials)))),
neteq_(
neteq_factory
? neteq_factory->Create(env_, config, std::move(decoder_factory))
: CreateNetEq(env_, config, std::move(decoder_factory))),
output_(std::move(output)),
callbacks_(callbacks),
sample_rate_hz_(config.sample_rate_hz),
text_log_(std::move(text_log)) {
RegisterDecoders(codecs);
}
NetEqTest::~NetEqTest() = default;
int64_t NetEqTest::Run() {
int64_t simulation_time = 0;
SimulationStepResult step_result;
do {
step_result = RunToNextGetAudio();
simulation_time += step_result.simulation_step_ms;
} while (!step_result.is_simulation_finished);
if (callbacks_.simulation_ended_callback) {
callbacks_.simulation_ended_callback->SimulationEnded(simulation_time);
}
return simulation_time;
}
NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() {
SimulationStepResult result;
const int64_t start_time_ms = *input_->NextEventTime();
int64_t time_now_ms = clock_.CurrentTime().ms();
current_state_.packet_iat_ms.clear();
while (!input_->ended()) {
// Advance time to next event.
RTC_DCHECK(input_->NextEventTime());
clock_.AdvanceTimeMilliseconds(*input_->NextEventTime() - time_now_ms);
time_now_ms = *input_->NextEventTime();
// Check if it is time to insert packet.
if (input_->NextPacketTime() && time_now_ms >= *input_->NextPacketTime()) {
std::unique_ptr<NetEqInput::PacketData> packet_data = input_->PopPacket();
RTC_CHECK(packet_data);
const size_t payload_data_length =
packet_data->payload.size() - packet_data->header.paddingLength;
if (payload_data_length != 0) {
int error = neteq_->InsertPacket(
packet_data->header,
rtc::ArrayView<const uint8_t>(packet_data->payload),
Timestamp::Millis(time_now_ms));
if (error != NetEq::kOK && callbacks_.error_callback) {
callbacks_.error_callback->OnInsertPacketError(*packet_data);
}
if (callbacks_.post_insert_packet) {
callbacks_.post_insert_packet->AfterInsertPacket(*packet_data,
neteq_.get());
}
} else {
neteq_->InsertEmptyPacket(packet_data->header);
}
if (last_packet_time_ms_) {
current_state_.packet_iat_ms.push_back(time_now_ms -
*last_packet_time_ms_);
}
if (text_log_) {
const auto ops_state = neteq_->GetOperationsAndState();
const auto delta_wallclock =
last_packet_time_ms_ ? (time_now_ms - *last_packet_time_ms_) : -1;
const auto delta_timestamp =
last_packet_timestamp_
? (static_cast<int64_t>(packet_data->header.timestamp) -
*last_packet_timestamp_) *
1000 / sample_rate_hz_
: -1;
const auto packet_size_bytes =
packet_data->payload.size() == 12
? ByteReader<uint32_t>::ReadLittleEndian(
&packet_data->payload[8])
: -1;
*text_log_ << "Packet - wallclock: " << std::setw(5) << time_now_ms
<< ", delta wc: " << std::setw(4) << delta_wallclock
<< ", seq_no: " << packet_data->header.sequenceNumber
<< ", timestamp: " << std::setw(10)
<< packet_data->header.timestamp
<< ", delta ts: " << std::setw(4) << delta_timestamp
<< ", size: " << std::setw(5) << packet_size_bytes
<< ", frame size: " << std::setw(3)
<< ops_state.current_frame_size_ms
<< ", buffer size: " << std::setw(4)
<< ops_state.current_buffer_size_ms << std::endl;
}
last_packet_time_ms_ = absl::make_optional<int>(time_now_ms);
last_packet_timestamp_ =
absl::make_optional<uint32_t>(packet_data->header.timestamp);
}
if (input_->NextSetMinimumDelayInfo().has_value() &&
time_now_ms >= input_->NextSetMinimumDelayInfo().value().timestamp_ms) {
neteq_->SetBaseMinimumDelayMs(
input_->NextSetMinimumDelayInfo().value().delay_ms);
input_->AdvanceSetMinimumDelay();
}
// Check if it is time to get output audio.
if (input_->NextOutputEventTime() &&
time_now_ms >= *input_->NextOutputEventTime()) {
if (callbacks_.get_audio_callback) {
callbacks_.get_audio_callback->BeforeGetAudio(neteq_.get());
}
AudioFrame out_frame;
int error = neteq_->GetAudio(&out_frame, nullptr, nullptr,
ActionToOperations(next_action_));
next_action_ = absl::nullopt;
if (error != NetEq::kOK) {
if (callbacks_.error_callback) {
callbacks_.error_callback->OnGetAudioError();
}
} else {
sample_rate_hz_ = out_frame.sample_rate_hz_;
}
if (callbacks_.get_audio_callback) {
callbacks_.get_audio_callback->AfterGetAudio(
time_now_ms, out_frame, out_frame.muted(), neteq_.get());
}
if (output_) {
RTC_CHECK(output_->WriteArray(
out_frame.data(),
out_frame.samples_per_channel_ * out_frame.num_channels_));
}
input_->AdvanceOutputEvent();
result.simulation_step_ms =
input_->NextEventTime().value_or(time_now_ms) - start_time_ms;
const auto operations_state = neteq_->GetOperationsAndState();
current_state_.current_delay_ms = operations_state.current_buffer_size_ms;
current_state_.packet_size_ms = operations_state.current_frame_size_ms;
current_state_.next_packet_available =
operations_state.next_packet_available;
current_state_.packet_buffer_flushed =
operations_state.packet_buffer_flushes >
prev_ops_state_.packet_buffer_flushes;
// TODO(ivoc): Add more accurate reporting by tracking the origin of
// samples in the sync buffer.
result.action_times_ms[Action::kExpand] = 0;
result.action_times_ms[Action::kAccelerate] = 0;
result.action_times_ms[Action::kPreemptiveExpand] = 0;
result.action_times_ms[Action::kNormal] = 0;
if (out_frame.speech_type_ == AudioFrame::SpeechType::kPLC ||
out_frame.speech_type_ == AudioFrame::SpeechType::kPLCCNG) {
// Consider the whole frame to be the result of expansion.
result.action_times_ms[Action::kExpand] = 10;
} else if (operations_state.accelerate_samples -
prev_ops_state_.accelerate_samples >
0) {
// Consider the whole frame to be the result of acceleration.
result.action_times_ms[Action::kAccelerate] = 10;
} else if (operations_state.preemptive_samples -
prev_ops_state_.preemptive_samples >
0) {
// Consider the whole frame to be the result of preemptive expansion.
result.action_times_ms[Action::kPreemptiveExpand] = 10;
} else {
// Consider the whole frame to be the result of normal playout.
result.action_times_ms[Action::kNormal] = 10;
}
auto lifetime_stats = LifetimeStats();
if (text_log_) {
const bool plc =
(out_frame.speech_type_ == AudioFrame::SpeechType::kPLC) ||
(out_frame.speech_type_ == AudioFrame::SpeechType::kPLCCNG);
const bool cng = out_frame.speech_type_ == AudioFrame::SpeechType::kCNG;
const bool voice_concealed =
(lifetime_stats.concealed_samples -
lifetime_stats.silent_concealed_samples) >
(prev_lifetime_stats_.concealed_samples -
prev_lifetime_stats_.silent_concealed_samples);
*text_log_ << "GetAudio - wallclock: " << std::setw(5) << time_now_ms
<< ", delta wc: " << std::setw(4)
<< (input_->NextEventTime().value_or(time_now_ms) -
start_time_ms)
<< ", CNG: " << cng << ", PLC: " << plc
<< ", voice concealed: " << voice_concealed
<< ", buffer size: " << std::setw(4)
<< current_state_.current_delay_ms << std::endl;
if (lifetime_stats.packets_discarded >
prev_lifetime_stats_.packets_discarded) {
*text_log_ << "Discarded "
<< (lifetime_stats.packets_discarded -
prev_lifetime_stats_.packets_discarded)
<< " primary packets." << std::endl;
}
if (operations_state.packet_buffer_flushes >
prev_ops_state_.packet_buffer_flushes) {
*text_log_ << "Flushed packet buffer "
<< (operations_state.packet_buffer_flushes -
prev_ops_state_.packet_buffer_flushes)
<< " times." << std::endl;
}
}
prev_lifetime_stats_ = lifetime_stats;
const bool no_more_packets_to_decode =
!input_->NextPacketTime() && !operations_state.next_packet_available;
result.is_simulation_finished =
no_more_packets_to_decode || input_->ended();
prev_ops_state_ = operations_state;
return result;
}
}
result.simulation_step_ms =
input_->NextEventTime().value_or(time_now_ms) - start_time_ms;
result.is_simulation_finished = true;
return result;
}
void NetEqTest::SetNextAction(NetEqTest::Action next_operation) {
next_action_ = absl::optional<Action>(next_operation);
}
NetEqTest::NetEqState NetEqTest::GetNetEqState() {
return current_state_;
}
NetEqNetworkStatistics NetEqTest::SimulationStats() {
NetEqNetworkStatistics stats;
RTC_CHECK_EQ(neteq_->NetworkStatistics(&stats), 0);
return stats;
}
NetEqLifetimeStatistics NetEqTest::LifetimeStats() const {
return neteq_->GetLifetimeStatistics();
}
NetEqTest::DecoderMap NetEqTest::StandardDecoderMap() {
DecoderMap codecs = {{0, SdpAudioFormat("pcmu", 8000, 1)},
{8, SdpAudioFormat("pcma", 8000, 1)},
#ifdef WEBRTC_CODEC_ILBC
{102, SdpAudioFormat("ilbc", 8000, 1)},
#endif
#ifdef WEBRTC_CODEC_OPUS
{111, SdpAudioFormat("opus", 48000, 2)},
{63, SdpAudioFormat("red", 48000, 2)},
#endif
{93, SdpAudioFormat("l16", 8000, 1)},
{94, SdpAudioFormat("l16", 16000, 1)},
{95, SdpAudioFormat("l16", 32000, 1)},
{96, SdpAudioFormat("l16", 48000, 1)},
{9, SdpAudioFormat("g722", 8000, 1)},
{106, SdpAudioFormat("telephone-event", 8000, 1)},
{114, SdpAudioFormat("telephone-event", 16000, 1)},
{115, SdpAudioFormat("telephone-event", 32000, 1)},
{116, SdpAudioFormat("telephone-event", 48000, 1)},
{117, SdpAudioFormat("red", 8000, 1)},
{13, SdpAudioFormat("cn", 8000, 1)},
{98, SdpAudioFormat("cn", 16000, 1)},
{99, SdpAudioFormat("cn", 32000, 1)},
{100, SdpAudioFormat("cn", 48000, 1)}};
return codecs;
}
void NetEqTest::RegisterDecoders(const DecoderMap& codecs) {
for (const auto& c : codecs) {
RTC_CHECK(neteq_->RegisterPayloadType(c.first, c.second))
<< "Cannot register " << c.second.name << " to payload type "
<< c.first;
}
}
} // namespace test
} // namespace webrtc
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