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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/mock_audio_encoder.cc
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/mock_audio_encoder.cc')
-rw-r--r--third_party/libwebrtc/test/mock_audio_encoder.cc57
1 files changed, 57 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/mock_audio_encoder.cc b/third_party/libwebrtc/test/mock_audio_encoder.cc
new file mode 100644
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+++ 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