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Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219}
341 lines
13 KiB
C++
341 lines
13 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
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#include <errno.h>
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#include <limits.h> // For ULONG_MAX returned by strtoul.
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#include <stdio.h>
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#include <stdlib.h> // For strtoul.
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#include <fstream>
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#include <iostream>
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#include <memory>
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#include <set>
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#include <string>
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#include <utility>
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/neteq/neteq.h"
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
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#include "modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h"
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
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#include "modules/audio_coding/neteq/tools/neteq_event_log_input.h"
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#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
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#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
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#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
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#include "modules/audio_coding/neteq/tools/neteq_stats_plotter.h"
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#include "modules/audio_coding/neteq/tools/neteq_test.h"
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#include "modules/audio_coding/neteq/tools/output_audio_file.h"
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#include "modules/audio_coding/neteq/tools/output_wav_file.h"
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#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
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#include "rtc_base/checks.h"
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#include "test/function_audio_decoder_factory.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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namespace test {
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namespace {
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absl::optional<int> CodecSampleRate(
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uint8_t payload_type,
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webrtc::test::NetEqTestFactory::Config config) {
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if (payload_type == config.pcmu || payload_type == config.pcma ||
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payload_type == config.ilbc || payload_type == config.pcm16b ||
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payload_type == config.cn_nb || payload_type == config.avt)
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return 8000;
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if (payload_type == config.isac || payload_type == config.pcm16b_wb ||
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payload_type == config.g722 || payload_type == config.cn_wb ||
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payload_type == config.avt_16)
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return 16000;
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if (payload_type == config.isac_swb || payload_type == config.pcm16b_swb32 ||
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payload_type == config.cn_swb32 || payload_type == config.avt_32)
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return 32000;
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if (payload_type == config.opus || payload_type == config.pcm16b_swb48 ||
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payload_type == config.cn_swb48 || payload_type == config.avt_48)
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return 48000;
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if (payload_type == config.red)
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return 0;
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return absl::nullopt;
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}
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} // namespace
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// A callback class which prints whenver the inserted packet stream changes
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// the SSRC.
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class SsrcSwitchDetector : public NetEqPostInsertPacket {
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public:
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// Takes a pointer to another callback object, which will be invoked after
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// this object finishes. This does not transfer ownership, and null is a
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// valid value.
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explicit SsrcSwitchDetector(NetEqPostInsertPacket* other_callback)
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: other_callback_(other_callback) {}
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void AfterInsertPacket(const NetEqInput::PacketData& packet,
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NetEq* neteq) override {
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if (last_ssrc_ && packet.header.ssrc != *last_ssrc_) {
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std::cout << "Changing streams from 0x" << std::hex << *last_ssrc_
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<< " to 0x" << std::hex << packet.header.ssrc << std::dec
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<< " (payload type "
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<< static_cast<int>(packet.header.payloadType) << ")"
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<< std::endl;
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}
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last_ssrc_ = packet.header.ssrc;
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if (other_callback_) {
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other_callback_->AfterInsertPacket(packet, neteq);
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}
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}
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private:
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NetEqPostInsertPacket* other_callback_;
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absl::optional<uint32_t> last_ssrc_;
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};
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NetEqTestFactory::NetEqTestFactory() = default;
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NetEqTestFactory::~NetEqTestFactory() = default;
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NetEqTestFactory::Config::Config() = default;
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NetEqTestFactory::Config::Config(const Config& other) = default;
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NetEqTestFactory::Config::~Config() = default;
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std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTestFromString(
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const std::string& input_string,
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NetEqFactory* factory,
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const Config& config) {
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std::unique_ptr<NetEqInput> input(
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NetEqEventLogInput::CreateFromString(input_string, config.ssrc_filter));
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if (!input) {
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std::cerr << "Error: Cannot parse input string" << std::endl;
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return nullptr;
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}
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return InitializeTest(std::move(input), factory, config);
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}
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std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTestFromFile(
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const std::string& input_file_name,
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NetEqFactory* factory,
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const Config& config) {
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// Gather RTP header extensions in a map.
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NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
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{config.audio_level, kRtpExtensionAudioLevel},
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{config.abs_send_time, kRtpExtensionAbsoluteSendTime},
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{config.transport_seq_no, kRtpExtensionTransportSequenceNumber},
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{config.video_content_type, kRtpExtensionVideoContentType},
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{config.video_timing, kRtpExtensionVideoTiming}};
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std::unique_ptr<NetEqInput> input;
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if (RtpFileSource::ValidRtpDump(input_file_name) ||
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RtpFileSource::ValidPcap(input_file_name)) {
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input.reset(new NetEqRtpDumpInput(input_file_name, rtp_ext_map,
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config.ssrc_filter));
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} else {
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input.reset(NetEqEventLogInput::CreateFromFile(input_file_name,
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config.ssrc_filter));
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}
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std::cout << "Input file: " << input_file_name << std::endl;
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if (!input) {
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std::cerr << "Error: Cannot open input file" << std::endl;
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return nullptr;
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}
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return InitializeTest(std::move(input), factory, config);
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}
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std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTest(
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std::unique_ptr<NetEqInput> input,
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NetEqFactory* factory,
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const Config& config) {
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if (input->ended()) {
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std::cerr << "Error: Input is empty" << std::endl;
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return nullptr;
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}
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if (!config.field_trial_string.empty()) {
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field_trials_ =
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std::make_unique<ScopedFieldTrials>(config.field_trial_string);
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}
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// Skip some initial events/packets if requested.
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if (config.skip_get_audio_events > 0) {
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std::cout << "Skipping " << config.skip_get_audio_events
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<< " get_audio events" << std::endl;
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if (!input->NextPacketTime() || !input->NextOutputEventTime()) {
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std::cerr << "No events found" << std::endl;
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return nullptr;
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}
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for (int i = 0; i < config.skip_get_audio_events; i++) {
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input->AdvanceOutputEvent();
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if (!input->NextOutputEventTime()) {
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std::cerr << "Not enough get_audio events found" << std::endl;
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return nullptr;
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}
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}
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while (*input->NextPacketTime() < *input->NextOutputEventTime()) {
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input->PopPacket();
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if (!input->NextPacketTime()) {
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std::cerr << "Not enough incoming packets found" << std::endl;
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return nullptr;
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}
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}
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}
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// Check the sample rate.
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absl::optional<int> sample_rate_hz;
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std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
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while (absl::optional<RTPHeader> first_rtp_header = input->NextHeader()) {
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RTC_DCHECK(first_rtp_header);
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sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType, config);
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if (sample_rate_hz) {
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std::cout << "Found valid packet with payload type "
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<< static_cast<int>(first_rtp_header->payloadType)
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<< " and SSRC 0x" << std::hex << first_rtp_header->ssrc
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<< std::dec << std::endl;
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if (config.initial_dummy_packets > 0) {
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std::cout << "Nr of initial dummy packets: "
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<< config.initial_dummy_packets << std::endl;
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input = std::make_unique<InitialPacketInserterNetEqInput>(
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std::move(input), config.initial_dummy_packets, *sample_rate_hz);
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}
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break;
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}
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// Discard this packet and move to the next. Keep track of discarded payload
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// types and SSRCs.
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discarded_pt_and_ssrc.emplace(first_rtp_header->payloadType,
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first_rtp_header->ssrc);
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input->PopPacket();
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}
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if (!discarded_pt_and_ssrc.empty()) {
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std::cout << "Discarded initial packets with the following payload types "
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"and SSRCs:"
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<< std::endl;
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for (const auto& d : discarded_pt_and_ssrc) {
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std::cout << "PT " << d.first << "; SSRC 0x" << std::hex
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<< static_cast<int>(d.second) << std::dec << std::endl;
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}
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}
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if (!sample_rate_hz) {
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std::cerr << "Cannot find any packets with known payload types"
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<< std::endl;
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return nullptr;
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}
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// If an output file is requested, open it.
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std::unique_ptr<AudioSink> output;
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if (!config.output_audio_filename.has_value()) {
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output = std::make_unique<VoidAudioSink>();
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std::cout << "No output audio file" << std::endl;
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} else if (config.output_audio_filename->size() >= 4 &&
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config.output_audio_filename->substr(
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config.output_audio_filename->size() - 4) == ".wav") {
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// Open a wav file with the known sample rate.
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output = std::make_unique<OutputWavFile>(*config.output_audio_filename,
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*sample_rate_hz);
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std::cout << "Output WAV file: " << *config.output_audio_filename
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<< std::endl;
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} else {
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// Open a pcm file.
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output = std::make_unique<OutputAudioFile>(*config.output_audio_filename);
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std::cout << "Output PCM file: " << *config.output_audio_filename
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<< std::endl;
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}
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NetEqTest::DecoderMap codecs = NetEqTest::StandardDecoderMap();
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
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CreateBuiltinAudioDecoderFactory();
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// Check if a replacement audio file was provided.
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if (config.replacement_audio_file.size() > 0) {
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// Find largest unused payload type.
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int replacement_pt = 127;
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while (codecs.find(replacement_pt) != codecs.end()) {
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--replacement_pt;
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if (replacement_pt <= 0) {
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std::cerr << "Error: Unable to find available replacement payload type"
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<< std::endl;
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return nullptr;
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}
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}
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auto std_set_int32_to_uint8 = [](const std::set<int32_t>& a) {
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std::set<uint8_t> b;
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for (auto& x : a) {
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b.insert(static_cast<uint8_t>(x));
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}
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return b;
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};
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std::set<uint8_t> cn_types = std_set_int32_to_uint8(
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{config.cn_nb, config.cn_wb, config.cn_swb32, config.cn_swb48});
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std::set<uint8_t> forbidden_types =
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std_set_int32_to_uint8({config.g722, config.red, config.avt,
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config.avt_16, config.avt_32, config.avt_48});
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input.reset(new NetEqReplacementInput(std::move(input), replacement_pt,
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cn_types, forbidden_types));
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// Note that capture-by-copy implies that the lambda captures the value of
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// decoder_factory before it's reassigned on the left-hand side.
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decoder_factory = rtc::make_ref_counted<FunctionAudioDecoderFactory>(
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[decoder_factory, config](
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const SdpAudioFormat& format,
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absl::optional<AudioCodecPairId> codec_pair_id) {
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std::unique_ptr<AudioDecoder> decoder =
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decoder_factory->MakeAudioDecoder(format, codec_pair_id);
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if (!decoder && format.name == "replacement") {
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decoder = std::make_unique<FakeDecodeFromFile>(
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std::make_unique<InputAudioFile>(config.replacement_audio_file),
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format.clockrate_hz, format.num_channels > 1);
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}
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return decoder;
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});
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if (!codecs
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.insert({replacement_pt, SdpAudioFormat("replacement", 48000, 1)})
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.second) {
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std::cerr << "Error: Unable to insert replacement audio codec"
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<< std::endl;
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return nullptr;
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}
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}
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// Create a text log output stream if needed.
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std::unique_ptr<std::ofstream> text_log;
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if (config.textlog && config.textlog_filename.has_value()) {
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// Write to file.
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text_log = std::make_unique<std::ofstream>(*config.textlog_filename);
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} else if (config.textlog) {
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// Print to stdout.
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text_log = std::make_unique<std::ofstream>();
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text_log->basic_ios<char>::rdbuf(std::cout.rdbuf());
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}
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NetEqTest::Callbacks callbacks;
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stats_plotter_ = std::make_unique<NetEqStatsPlotter>(
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config.matlabplot, config.pythonplot, config.concealment_events,
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config.plot_scripts_basename.value_or(""));
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ssrc_switch_detector_.reset(
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new SsrcSwitchDetector(stats_plotter_->stats_getter()->delay_analyzer()));
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callbacks.post_insert_packet = ssrc_switch_detector_.get();
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callbacks.get_audio_callback = stats_plotter_->stats_getter();
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callbacks.simulation_ended_callback = stats_plotter_.get();
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NetEq::Config neteq_config;
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neteq_config.sample_rate_hz = *sample_rate_hz;
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neteq_config.max_packets_in_buffer = config.max_nr_packets_in_buffer;
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neteq_config.enable_fast_accelerate = config.enable_fast_accelerate;
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return std::make_unique<NetEqTest>(
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neteq_config, decoder_factory, codecs, std::move(text_log), factory,
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std::move(input), std::move(output), callbacks);
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}
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} // namespace test
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} // namespace webrtc
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