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

webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc

+/*
+ *  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 "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+using ::testing::Return;
+
+namespace webrtc {
+
+TEST(AudioEncoderTest, EncodeInternalRedirectsOk) {
+  const size_t kPayloadSize = 16;
+  const uint8_t payload[kPayloadSize] =
+      {0xf, 0xe, 0xd, 0xc, 0xb, 0xa, 0x9, 0x8,
+       0x7, 0x6, 0x5, 0x4, 0x3, 0x2, 0x1, 0x0};
+
+  MockAudioEncoderDeprecated old_impl;
+  MockAudioEncoder new_impl;
+  MockAudioEncoderBase* impls[] = { &old_impl, &new_impl };
+  for (auto* impl : impls) {
+    EXPECT_CALL(*impl, Die());
+    EXPECT_CALL(*impl, MaxEncodedBytes())
+        .WillRepeatedly(Return(kPayloadSize * 2));
+    EXPECT_CALL(*impl, NumChannels()).WillRepeatedly(Return(1));
+    EXPECT_CALL(*impl, SampleRateHz()).WillRepeatedly(Return(8000));
+  }
+
+  EXPECT_CALL(old_impl, EncodeInternal(_, _, _, _)).WillOnce(
+      Invoke(MockAudioEncoderDeprecated::CopyEncoding(payload)));
+
+  EXPECT_CALL(new_impl, EncodeInternal(_, _, _)).WillOnce(
+      Invoke(MockAudioEncoder::CopyEncoding(payload)));
+
+  int16_t audio[80];
+  uint8_t output_array[kPayloadSize * 2];
+  rtc::Buffer output_buffer;
+
+  AudioEncoder* old_encoder = &old_impl;
+  AudioEncoder* new_encoder = &new_impl;
+  auto old_info = old_encoder->Encode(0, audio, &output_buffer);
+  auto new_info = new_encoder->DEPRECATED_Encode(0, audio,
+                                                 kPayloadSize * 2,
+                                                 output_array);
+
+  EXPECT_EQ(old_info.encoded_bytes, kPayloadSize);
+  EXPECT_EQ(new_info.encoded_bytes, kPayloadSize);
+  EXPECT_EQ(output_buffer.size(), kPayloadSize);
+
+  for (size_t i = 0; i != kPayloadSize; ++i) {
+    EXPECT_EQ(output_buffer.data()[i], payload[i]);
+    EXPECT_EQ(output_array[i], payload[i]);
+  }
+}
+
+}  // namespace webrtc