Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 77 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.h b/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.h
new file mode 100644
index 0000000000..d841d65a1b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.h
@@ -0,0 +1,77 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/test/EncodeDecodeTest.h"
+
+namespace webrtc {
+
+class ReceiverWithPacketLoss : public Receiver {
+ public:
+  ReceiverWithPacketLoss();
+  void Setup(AudioCodingModule* acm,
+             RTPStream* rtpStream,
+             absl::string_view out_file_name,
+             int channels,
+             int file_num,
+             int loss_rate,
+             int burst_length);
+  bool IncomingPacket() override;
+
+ protected:
+  bool PacketLost();
+  int loss_rate_;
+  int burst_length_;
+  int packet_counter_;
+  int lost_packet_counter_;
+  int burst_lost_counter_;
+};
+
+class SenderWithFEC : public Sender {
+ public:
+  SenderWithFEC();
+  void Setup(AudioCodingModule* acm,
+             RTPStream* rtpStream,
+             absl::string_view in_file_name,
+             int payload_type,
+             SdpAudioFormat format,
+             int expected_loss_rate);
+  bool SetPacketLossRate(int expected_loss_rate);
+  bool SetFEC(bool enable_fec);
+
+ protected:
+  int expected_loss_rate_;
+};
+
+class PacketLossTest {
+ public:
+  PacketLossTest(int channels,
+                 int expected_loss_rate_,
+                 int actual_loss_rate,
+                 int burst_length);
+  void Perform();
+
+ protected:
+  int channels_;
+  std::string in_file_name_;
+  int sample_rate_hz_;
+  int expected_loss_rate_;
+  int actual_loss_rate_;
+  int burst_length_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_