blob: 981504b23908ab85bec59986e1039b37e4411b77 [file] [log] [blame]
/*
* 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_test_factory.h"
#include <errno.h>
#include <limits.h> // For ULONG_MAX returned by strtoul.
#include <stdio.h>
#include <stdlib.h> // For strtoul.
#include <fstream>
#include <iostream>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/neteq/neteq.h"
#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/initial_packet_inserter_neteq_input.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_event_log_input.h"
#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
#include "modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_plotter.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/checks.h"
#include "test/function_audio_decoder_factory.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace test {
namespace {
absl::optional<int> CodecSampleRate(
uint8_t payload_type,
webrtc::test::NetEqTestFactory::Config config) {
if (payload_type == config.pcmu || payload_type == config.pcma ||
payload_type == config.ilbc || payload_type == config.pcm16b ||
payload_type == config.cn_nb || payload_type == config.avt)
return 8000;
if (payload_type == config.isac || payload_type == config.pcm16b_wb ||
payload_type == config.g722 || payload_type == config.cn_wb ||
payload_type == config.avt_16)
return 16000;
if (payload_type == config.isac_swb || payload_type == config.pcm16b_swb32 ||
payload_type == config.cn_swb32 || payload_type == config.avt_32)
return 32000;
if (payload_type == config.opus || payload_type == config.pcm16b_swb48 ||
payload_type == config.cn_swb48 || payload_type == config.avt_48)
return 48000;
if (payload_type == config.red)
return 0;
return absl::nullopt;
}
} // namespace
// A callback class which prints whenver the inserted packet stream changes
// the SSRC.
class SsrcSwitchDetector : public NetEqPostInsertPacket {
public:
// 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 SsrcSwitchDetector(NetEqPostInsertPacket* other_callback)
: other_callback_(other_callback) {}
void AfterInsertPacket(const NetEqInput::PacketData& packet,
NetEq* neteq) override {
if (last_ssrc_ && packet.header.ssrc != *last_ssrc_) {
std::cout << "Changing streams from 0x" << std::hex << *last_ssrc_
<< " to 0x" << std::hex << packet.header.ssrc << std::dec
<< " (payload type "
<< static_cast<int>(packet.header.payloadType) << ")"
<< std::endl;
}
last_ssrc_ = packet.header.ssrc;
if (other_callback_) {
other_callback_->AfterInsertPacket(packet, neteq);
}
}
private:
NetEqPostInsertPacket* other_callback_;
absl::optional<uint32_t> last_ssrc_;
};
NetEqTestFactory::NetEqTestFactory() = default;
NetEqTestFactory::~NetEqTestFactory() = default;
NetEqTestFactory::Config::Config() = default;
NetEqTestFactory::Config::Config(const Config& other) = default;
NetEqTestFactory::Config::~Config() = default;
std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTestFromString(
absl::string_view input_string,
NetEqFactory* factory,
const Config& config) {
ParsedRtcEventLog parsed_log;
auto status = parsed_log.ParseString(input_string);
if (!status.ok()) {
std::cerr << "Failed to parse event log: " << status.message() << std::endl;
return nullptr;
}
std::unique_ptr<NetEqInput> input =
CreateNetEqEventLogInput(parsed_log, config.ssrc_filter);
if (!input) {
std::cerr << "Error: Cannot parse input string" << std::endl;
return nullptr;
}
return InitializeTest(std::move(input), factory, config);
}
std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTestFromFile(
absl::string_view input_file_name,
NetEqFactory* factory,
const Config& config) {
// Gather RTP header extensions in a map.
std::map<int, RTPExtensionType> rtp_ext_map = {
{config.audio_level, kRtpExtensionAudioLevel},
{config.abs_send_time, kRtpExtensionAbsoluteSendTime},
{config.transport_seq_no, kRtpExtensionTransportSequenceNumber},
{config.video_content_type, kRtpExtensionVideoContentType},
{config.video_timing, kRtpExtensionVideoTiming}};
std::unique_ptr<NetEqInput> input;
if (RtpFileSource::ValidRtpDump(input_file_name) ||
RtpFileSource::ValidPcap(input_file_name)) {
input = CreateNetEqRtpDumpInput(input_file_name, rtp_ext_map,
config.ssrc_filter);
} else {
ParsedRtcEventLog parsed_log;
auto status = parsed_log.ParseFile(input_file_name);
if (!status.ok()) {
std::cerr << "Failed to parse event log: " << status.message()
<< std::endl;
return nullptr;
}
input = CreateNetEqEventLogInput(parsed_log, config.ssrc_filter);
}
std::cout << "Input file: " << input_file_name << std::endl;
if (!input) {
std::cerr << "Error: Cannot open input file" << std::endl;
return nullptr;
}
return InitializeTest(std::move(input), factory, config);
}
std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTest(
std::unique_ptr<NetEqInput> input,
NetEqFactory* factory,
const Config& config) {
if (input->ended()) {
std::cerr << "Error: Input is empty" << std::endl;
return nullptr;
}
if (!config.field_trial_string.empty()) {
field_trials_ =
std::make_unique<ScopedFieldTrials>(config.field_trial_string);
}
// Skip some initial events/packets if requested.
if (config.skip_get_audio_events > 0) {
std::cout << "Skipping " << config.skip_get_audio_events
<< " get_audio events" << std::endl;
if (!input->NextPacketTime() || !input->NextOutputEventTime()) {
std::cerr << "No events found" << std::endl;
return nullptr;
}
for (int i = 0; i < config.skip_get_audio_events; i++) {
input->AdvanceOutputEvent();
if (!input->NextOutputEventTime()) {
std::cerr << "Not enough get_audio events found" << std::endl;
return nullptr;
}
}
while (*input->NextPacketTime() < *input->NextOutputEventTime()) {
input->PopPacket();
if (!input->NextPacketTime()) {
std::cerr << "Not enough incoming packets found" << std::endl;
return nullptr;
}
}
}
// Check the sample rate.
absl::optional<int> sample_rate_hz;
std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
while (absl::optional<RTPHeader> first_rtp_header = input->NextHeader()) {
RTC_DCHECK(first_rtp_header);
sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType, config);
if (sample_rate_hz) {
std::cout << "Found valid packet with payload type "
<< static_cast<int>(first_rtp_header->payloadType)
<< " and SSRC 0x" << std::hex << first_rtp_header->ssrc
<< std::dec << std::endl;
if (config.initial_dummy_packets > 0) {
std::cout << "Nr of initial dummy packets: "
<< config.initial_dummy_packets << std::endl;
input = std::make_unique<InitialPacketInserterNetEqInput>(
std::move(input), config.initial_dummy_packets, *sample_rate_hz);
}
break;
}
// Discard this packet and move to the next. Keep track of discarded payload
// types and SSRCs.
discarded_pt_and_ssrc.emplace(first_rtp_header->payloadType,
first_rtp_header->ssrc);
input->PopPacket();
}
if (!discarded_pt_and_ssrc.empty()) {
std::cout << "Discarded initial packets with the following payload types "
"and SSRCs:"
<< std::endl;
for (const auto& d : discarded_pt_and_ssrc) {
std::cout << "PT " << d.first << "; SSRC 0x" << std::hex
<< static_cast<int>(d.second) << std::dec << std::endl;
}
}
if (!sample_rate_hz) {
std::cerr << "Cannot find any packets with known payload types"
<< std::endl;
return nullptr;
}
// If an output file is requested, open it.
std::unique_ptr<AudioSink> output;
if (!config.output_audio_filename.has_value()) {
output = std::make_unique<VoidAudioSink>();
std::cout << "No output audio file" << std::endl;
} else if (config.output_audio_filename->size() >= 4 &&
config.output_audio_filename->substr(
config.output_audio_filename->size() - 4) == ".wav") {
// Open a wav file with the known sample rate.
output = std::make_unique<OutputWavFile>(*config.output_audio_filename,
*sample_rate_hz);
std::cout << "Output WAV file: " << *config.output_audio_filename
<< std::endl;
} else {
// Open a pcm file.
output = std::make_unique<OutputAudioFile>(*config.output_audio_filename);
std::cout << "Output PCM file: " << *config.output_audio_filename
<< std::endl;
}
NetEqTest::DecoderMap codecs = NetEqTest::StandardDecoderMap();
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
CreateBuiltinAudioDecoderFactory();
// Check if a replacement audio file was provided.
if (config.replacement_audio_file.size() > 0) {
// Find largest unused payload type.
int replacement_pt = 127;
while (codecs.find(replacement_pt) != codecs.end()) {
--replacement_pt;
if (replacement_pt <= 0) {
std::cerr << "Error: Unable to find available replacement payload type"
<< std::endl;
return nullptr;
}
}
auto std_set_int32_to_uint8 = [](const std::set<int32_t>& a) {
std::set<uint8_t> b;
for (auto& x : a) {
b.insert(static_cast<uint8_t>(x));
}
return b;
};
std::set<uint8_t> cn_types = std_set_int32_to_uint8(
{config.cn_nb, config.cn_wb, config.cn_swb32, config.cn_swb48});
std::set<uint8_t> forbidden_types =
std_set_int32_to_uint8({config.g722, config.red, config.avt,
config.avt_16, config.avt_32, config.avt_48});
input.reset(new NetEqReplacementInput(std::move(input), replacement_pt,
cn_types, forbidden_types));
// Note that capture-by-copy implies that the lambda captures the value of
// decoder_factory before it's reassigned on the left-hand side.
decoder_factory = rtc::make_ref_counted<FunctionAudioDecoderFactory>(
[decoder_factory, config](
const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id) {
std::unique_ptr<AudioDecoder> decoder =
decoder_factory->MakeAudioDecoder(format, codec_pair_id);
if (!decoder && format.name == "replacement") {
decoder = std::make_unique<FakeDecodeFromFile>(
std::make_unique<InputAudioFile>(config.replacement_audio_file),
format.clockrate_hz, format.num_channels > 1);
}
return decoder;
});
if (!codecs
.insert({replacement_pt, SdpAudioFormat("replacement", 48000, 1)})
.second) {
std::cerr << "Error: Unable to insert replacement audio codec"
<< std::endl;
return nullptr;
}
}
// Create a text log output stream if needed.
std::unique_ptr<std::ofstream> text_log;
if (config.textlog && config.textlog_filename.has_value()) {
// Write to file.
text_log = std::make_unique<std::ofstream>(*config.textlog_filename);
} else if (config.textlog) {
// Print to stdout.
text_log = std::make_unique<std::ofstream>();
text_log->basic_ios<char>::rdbuf(std::cout.rdbuf());
}
NetEqTest::Callbacks callbacks;
stats_plotter_ = std::make_unique<NetEqStatsPlotter>(
config.matlabplot, config.pythonplot, config.concealment_events,
config.plot_scripts_basename.value_or(""));
ssrc_switch_detector_.reset(
new SsrcSwitchDetector(stats_plotter_->stats_getter()->delay_analyzer()));
callbacks.post_insert_packet = ssrc_switch_detector_.get();
callbacks.get_audio_callback = stats_plotter_->stats_getter();
callbacks.simulation_ended_callback = stats_plotter_.get();
NetEq::Config neteq_config;
neteq_config.sample_rate_hz = *sample_rate_hz;
neteq_config.max_packets_in_buffer = config.max_nr_packets_in_buffer;
neteq_config.enable_fast_accelerate = config.enable_fast_accelerate;
return std::make_unique<NetEqTest>(
neteq_config, decoder_factory, codecs, std::move(text_log), factory,
std::move(input), std::move(output), callbacks);
}
} // namespace test
} // namespace webrtc