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Unified Diff: webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc

Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed issues from comments, rewrote MockAudioEncoderHelper Created 4 years, 10 months ago
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Index: webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc
new file mode 100644
index 0000000000000000000000000000000000000000..7ad52b1b9b1c0b588f973a0f1add24dfba44d79d
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc
@@ -0,0 +1,73 @@
+/*
+ * 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 "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+
+namespace webrtc {
+
+namespace MockAudioEncoderHelper {
+
+EncodeFunction FakeEncoding(AudioEncoder::EncodedInfo info) {
+ return [info] (uint32_t timestamp,
+ rtc::ArrayView<const int16_t> audio,
+ rtc::Buffer* encoded) {
+ encoded->SetSize(encoded->size() + info.encoded_bytes);
+ return info;
+ };
+}
+
+EncodeFunction FakeEncoding(size_t encoded_bytes) {
+ AudioEncoder::EncodedInfo info;
+ info.encoded_bytes = encoded_bytes;
+ return FakeEncoding(info);
+}
+
+EncodeFunction CopyEncoding(AudioEncoder::EncodedInfo info,
+ rtc::ArrayView<const uint8_t> payload) {
+ return [info, payload] (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;
+ };
+}
+
+EncodeFunction CopyEncoding(rtc::ArrayView<const uint8_t> payload) {
+ AudioEncoder::EncodedInfo info;
+ info.encoded_bytes = payload.size();
+ return CopyEncoding(info, payload);
+}
+
+DeprecatedEncodeFunction DEPRECATED_CopyEncoding(
+ AudioEncoder::EncodedInfo info,
+ rtc::ArrayView<const uint8_t> payload) {
+ return [info, payload] (uint32_t timestamp,
+ rtc::ArrayView<const int16_t> audio,
+ size_t max_encoded_bytes,
+ uint8_t* encoded) {
+ RTC_CHECK(encoded);
+ RTC_CHECK_LE(info.encoded_bytes, payload.size());
+ std::memcpy(encoded, payload.data(), info.encoded_bytes);
+ return info;
+ };
+}
+
+DeprecatedEncodeFunction DEPRECATED_CopyEncoding(
+ rtc::ArrayView<const uint8_t> payload) {
+ AudioEncoder::EncodedInfo info;
+ info.encoded_bytes = payload.size();
+ return DEPRECATED_CopyEncoding(info, payload);
+}
+
+} // namespace MockAudioEncoderHelper
+
+} // namespace webrtc

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