Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc

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..52849691ac6cebc67266d088ef102b3492acbe3a
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc
@@ -0,0 +1,74 @@
+/*
+ *  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 {
+
+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(
+    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_;
+}
+
+MockAudioEncoderDeprecated::CopyEncoding::CopyEncoding(
+    AudioEncoder::EncodedInfo info,
+    rtc::ArrayView<const uint8_t> payload)
+    : info_(info), payload_(payload) { }
+
+MockAudioEncoderDeprecated::CopyEncoding::CopyEncoding(
+    rtc::ArrayView<const uint8_t> payload)
+    : payload_(payload) {
+  info_.encoded_bytes = payload_.size();
+}
+
+AudioEncoder::EncodedInfo MockAudioEncoderDeprecated::CopyEncoding::operator()(
+    uint32_t timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    size_t max_bytes_encoded,
+    uint8_t* encoded) {
+  RTC_CHECK(encoded);
+  RTC_CHECK_LE(info_.encoded_bytes, payload_.size());
+  std::memcpy(encoded, payload_.data(), info_.encoded_bytes);
+  return info_;
+}
+
+}  // namespace webrtc