From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Fri, 19 Apr 2024 02:47:55 +0200
Subject: Adding upstream version 124.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 third_party/libwebrtc/test/mock_audio_encoder.cc | 57 ++++++++++++++++++++++++
 1 file changed, 57 insertions(+)
 create mode 100644 third_party/libwebrtc/test/mock_audio_encoder.cc

(limited to 'third_party/libwebrtc/test/mock_audio_encoder.cc')

diff --git a/third_party/libwebrtc/test/mock_audio_encoder.cc b/third_party/libwebrtc/test/mock_audio_encoder.cc
new file mode 100644
index 0000000000..36615111a5
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_audio_encoder.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2016 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 "test/mock_audio_encoder.h"
+
+namespace webrtc {
+
+MockAudioEncoder::MockAudioEncoder() = default;
+MockAudioEncoder::~MockAudioEncoder() = default;
+
+MockAudioEncoder::FakeEncoding::FakeEncoding(
+    const AudioEncoder::EncodedInfo& info)
+    : info_(info) {}
+
+MockAudioEncoder::FakeEncoding::FakeEncoding(size_t encoded_bytes) {
+  info_.encoded_bytes = encoded_bytes;
+}
+
+AudioEncoder::EncodedInfo MockAudioEncoder::FakeEncoding::operator()(
+    uint32_t timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  encoded->SetSize(encoded->size() + info_.encoded_bytes);
+  return info_;
+}
+
+MockAudioEncoder::CopyEncoding::~CopyEncoding() = default;
+
+MockAudioEncoder::CopyEncoding::CopyEncoding(
+    AudioEncoder::EncodedInfo info,
+    rtc::ArrayView<const uint8_t> payload)
+    : info_(info), payload_(payload) {}
+
+MockAudioEncoder::CopyEncoding::CopyEncoding(
+    rtc::ArrayView<const uint8_t> payload)
+    : payload_(payload) {
+  info_.encoded_bytes = payload_.size();
+}
+
+AudioEncoder::EncodedInfo MockAudioEncoder::CopyEncoding::operator()(
+    uint32_t timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  RTC_CHECK(encoded);
+  RTC_CHECK_LE(info_.encoded_bytes, payload_.size());
+  encoded->AppendData(payload_.data(), info_.encoded_bytes);
+  return info_;
+}
+
+}  // namespace webrtc
-- 
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