blob: 1d38bc442036f59f164f0d11622b2e054da150f6 [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/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/fake_decode_from_file.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_packet_source_input.h"
#include "modules/audio_coding/neteq/tools/neteq_replacement_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 "rtc_base/flags.h"
#include "rtc_base/ref_counted_object.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::InitializeTest(
std::string input_file_name,
std::string output_file_name,
const Config& config) {
// Gather RTP header extensions in a map.
NetEqPacketSourceInput::RtpHeaderExtensionMap 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.reset(new NetEqRtpDumpInput(input_file_name, rtp_ext_map,
config.ssrc_filter));
} else {
input.reset(new NetEqEventLogInput(input_file_name, config.ssrc_filter));
}
std::cout << "Input file: " << input_file_name << std::endl;
RTC_CHECK(input) << "Cannot open input file";
RTC_CHECK(!input->ended()) << "Input file is empty";
// 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;
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::cout << "Cannot find any packets with known payload types"
<< std::endl;
RTC_NOTREACHED();
}
// Open the output file now that we know the sample rate. (Rate is only needed
// for wav files.)
std::unique_ptr<AudioSink> output;
if (output_file_name.size() >= 4 &&
output_file_name.substr(output_file_name.size() - 4) == ".wav") {
// Open a wav file.
output.reset(new OutputWavFile(output_file_name, *sample_rate_hz));
} else {
// Open a pcm file.
output.reset(new OutputAudioFile(output_file_name));
}
std::cout << "Output file: " << output_file_name << 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;
RTC_CHECK_GE(replacement_pt, 0);
}
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 = new rtc::RefCountedObject<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 = absl::make_unique<FakeDecodeFromFile>(
absl::make_unique<InputAudioFile>(
config.replacement_audio_file),
format.clockrate_hz, format.num_channels > 1);
}
return decoder;
});
RTC_CHECK(
codecs.insert({replacement_pt, SdpAudioFormat("replacement", 48000, 1)})
.second);
}
// Create a text log file if needed.
std::unique_ptr<std::ofstream> text_log;
if (config.textlog) {
text_log =
absl::make_unique<std::ofstream>(output_file_name + ".text_log.txt");
}
NetEqTest::Callbacks callbacks;
stats_plotter_.reset(
new NetEqStatsPlotter(config.matlabplot, config.pythonplot,
config.concealment_events, output_file_name));
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 absl::make_unique<NetEqTest>(neteq_config, decoder_factory, codecs,
std::move(text_log), std::move(input),
std::move(output), callbacks);
}
} // namespace test
} // namespace webrtc