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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h')
-rw-r--r--third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h62
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diff --git a/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h b/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h
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+++ b/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h
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+/*
+ * Copyright (c) 2012 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_TWOWAYCOMMUNICATION_H_
+#define MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+
+#include <memory>
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+
+namespace webrtc {
+
+class TwoWayCommunication {
+ public:
+ TwoWayCommunication();
+ ~TwoWayCommunication();
+
+ void Perform();
+
+ private:
+ void SetUpAutotest(AudioEncoderFactory* const encoder_factory,
+ const SdpAudioFormat& format1,
+ int payload_type1,
+ const SdpAudioFormat& format2,
+ int payload_type2);
+
+ std::unique_ptr<AudioCodingModule> _acmA;
+ std::unique_ptr<AudioCodingModule> _acmB;
+
+ std::unique_ptr<AudioCodingModule> _acmRefA;
+ std::unique_ptr<AudioCodingModule> _acmRefB;
+
+ Channel* _channel_A2B;
+ Channel* _channel_B2A;
+
+ Channel* _channelRef_A2B;
+ Channel* _channelRef_B2A;
+
+ PCMFile _inFileA;
+ PCMFile _inFileB;
+
+ PCMFile _outFileA;
+ PCMFile _outFileB;
+
+ PCMFile _outFileRefA;
+ PCMFile _outFileRefB;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_