| /* |
| * 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 "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 NetEq::Config& config, |
| Clock* clock, |
| const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) { |
| return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock); |
| } |
| |
| } // 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) |
| : clock_(0), |
| neteq_(neteq_factory |
| ? neteq_factory->CreateNetEq(config, decoder_factory, &clock_) |
| : CreateNetEq(config, &clock_, decoder_factory)), |
| input_(std::move(input)), |
| output_(std::move(output)), |
| callbacks_(callbacks), |
| sample_rate_hz_(config.sample_rate_hz), |
| text_log_(std::move(text_log)) { |
| RTC_CHECK(!config.enable_muted_state) |
| << "The code does not handle enable_muted_state"; |
| 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, |
| neteq_.get()); |
| } |
| return simulation_time; |
| } |
| |
| NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() { |
| SimulationStepResult result; |
| const int64_t start_time_ms = *input_->NextEventTime(); |
| int64_t time_now_ms = start_time_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)); |
| 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); |
| } |
| |
| // 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; |
| bool muted; |
| int error = neteq_->GetAudio(&out_frame, &muted, |
| ActionToOperations(next_action_)); |
| next_action_ = absl::nullopt; |
| RTC_CHECK(!muted) << "The code does not handle enable_muted_state"; |
| 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, |
| 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 (operations_state.discarded_primary_packets > |
| prev_ops_state_.discarded_primary_packets) { |
| *text_log_ << "Discarded " |
| << (operations_state.discarded_primary_packets - |
| prev_ops_state_.discarded_primary_packets) |
| << " 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; |
| // End the simulation if the gap is too large. This indicates an issue |
| // with the event log file. |
| const bool simulation_step_too_large = result.simulation_step_ms > 1000; |
| if (simulation_step_too_large) { |
| // If we don't reset the step time, the large gap will be included in |
| // the simulation time, which can be a large distortion. |
| result.simulation_step_ms = 10; |
| } |
| result.is_simulation_finished = simulation_step_too_large || |
| 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 |
| {103, SdpAudioFormat("isac", 16000, 1)}, |
| #if !defined(WEBRTC_ANDROID) |
| {104, SdpAudioFormat("isac", 32000, 1)}, |
| #endif |
| #ifdef WEBRTC_CODEC_OPUS |
| {111, SdpAudioFormat("opus", 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 |
