Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 354 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
new file mode 100644
index 0000000000..981504b239
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
@@ -0,0 +1,354 @@
+/*
+ *  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