Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 32 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
new file mode 100644
index 0000000000..b5b4d91577
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
@@ -0,0 +1,32 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+namespace test {
+
+class NetEqPerformanceTest {
+ public:
+  // Runs a performance test with parameters as follows:
+  //   `runtime_ms`: the simulation time, i.e., the duration of the audio data.
+  //   `lossrate`: drop one out of `lossrate` packets, e.g., one out of 10.
+  //   `drift_factor`: clock drift in [0, 1].
+  // Returns the runtime in ms.
+  static int64_t Run(int runtime_ms, int lossrate, double drift_factor);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_