Adding upstream version 110.0.1.upstream/110.0.1upstream

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

1 files changed, 167 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.cc b/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.cc
new file mode 100644
index 0000000000..799e9c5b1f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.cc
@@ -0,0 +1,167 @@
+/*
+ *  Copyright (c) 2014 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/test/PacketLossTest.h"
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+ReceiverWithPacketLoss::ReceiverWithPacketLoss()
+    : loss_rate_(0),
+      burst_length_(1),
+      packet_counter_(0),
+      lost_packet_counter_(0),
+      burst_lost_counter_(burst_length_) {}
+
+void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
+                                   RTPStream* rtpStream,
+                                   absl::string_view out_file_name,
+                                   int channels,
+                                   int file_num,
+                                   int loss_rate,
+                                   int burst_length) {
+  loss_rate_ = loss_rate;
+  burst_length_ = burst_length;
+  burst_lost_counter_ = burst_length_;  // To prevent first packet gets lost.
+  rtc::StringBuilder ss;
+  ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
+  Receiver::Setup(acm, rtpStream, ss.str(), channels, file_num);
+}
+
+bool ReceiverWithPacketLoss::IncomingPacket() {
+  if (!_rtpStream->EndOfFile()) {
+    if (packet_counter_ == 0) {
+      _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
+                                               _payloadSizeBytes, &_nextTime);
+      if (_realPayloadSizeBytes == 0) {
+        if (_rtpStream->EndOfFile()) {
+          packet_counter_ = 0;
+          return true;
+        } else {
+          return false;
+        }
+      }
+    }
+
+    if (!PacketLost()) {
+      _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpHeader);
+    }
+    packet_counter_++;
+    _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
+                                             _payloadSizeBytes, &_nextTime);
+    if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
+      packet_counter_ = 0;
+      lost_packet_counter_ = 0;
+    }
+  }
+  return true;
+}
+
+bool ReceiverWithPacketLoss::PacketLost() {
+  if (burst_lost_counter_ < burst_length_) {
+    lost_packet_counter_++;
+    burst_lost_counter_++;
+    return true;
+  }
+
+  if (lost_packet_counter_ * 100 < loss_rate_ * packet_counter_) {
+    lost_packet_counter_++;
+    burst_lost_counter_ = 1;
+    return true;
+  }
+  return false;
+}
+
+SenderWithFEC::SenderWithFEC() : expected_loss_rate_(0) {}
+
+void SenderWithFEC::Setup(AudioCodingModule* acm,
+                          RTPStream* rtpStream,
+                          absl::string_view in_file_name,
+                          int payload_type,
+                          SdpAudioFormat format,
+                          int expected_loss_rate) {
+  Sender::Setup(acm, rtpStream, in_file_name, format.clockrate_hz, payload_type,
+                format);
+  EXPECT_TRUE(SetFEC(true));
+  EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
+}
+
+bool SenderWithFEC::SetFEC(bool enable_fec) {
+  bool success = false;
+  _acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
+    if (*enc && (*enc)->SetFec(enable_fec)) {
+      success = true;
+    }
+  });
+  return success;
+}
+
+bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
+  if (_acm->SetPacketLossRate(expected_loss_rate) == 0) {
+    expected_loss_rate_ = expected_loss_rate;
+    return true;
+  }
+  return false;
+}
+
+PacketLossTest::PacketLossTest(int channels,
+                               int expected_loss_rate,
+                               int actual_loss_rate,
+                               int burst_length)
+    : channels_(channels),
+      in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz"
+                                   : "audio_coding/teststereo32kHz"),
+      sample_rate_hz_(32000),
+      expected_loss_rate_(expected_loss_rate),
+      actual_loss_rate_(actual_loss_rate),
+      burst_length_(burst_length) {}
+
+void PacketLossTest::Perform() {
+#ifndef WEBRTC_CODEC_OPUS
+  return;
+#else
+  RTPFile rtpFile;
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
+      AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+  SdpAudioFormat send_format = SdpAudioFormat("opus", 48000, 2);
+  if (channels_ == 2) {
+    send_format.parameters = {{"stereo", "1"}};
+  }
+
+  std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                    "packet_loss_test");
+  rtpFile.Open(fileName.c_str(), "wb+");
+  rtpFile.WriteHeader();
+  SenderWithFEC sender;
+  sender.Setup(acm.get(), &rtpFile, in_file_name_, 120, send_format,
+               expected_loss_rate_);
+  sender.Run();
+  sender.Teardown();
+  rtpFile.Close();
+
+  rtpFile.Open(fileName.c_str(), "rb");
+  rtpFile.ReadHeader();
+  ReceiverWithPacketLoss receiver;
+  receiver.Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 15,
+                 actual_loss_rate_, burst_length_);
+  receiver.Run();
+  receiver.Teardown();
+  rtpFile.Close();
+#endif
+}
+
+}  // namespace webrtc