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.h

Index: webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
index 66adde4be159c3af8fd0b404d09f80e8cc95c021..218e404de4887c2fa9c60a1340e3c0e5a424a252 100644
--- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
@@ -17,9 +17,9 @@
 
 namespace webrtc {
 
-class MockAudioEncoder final : public AudioEncoder {
+class MockAudioEncoderBase : public AudioEncoder {
  public:
-  ~MockAudioEncoder() override { Die(); }
+  ~MockAudioEncoderBase() override { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_METHOD1(Mark, void(std::string desc));
   MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
@@ -29,12 +29,6 @@ class MockAudioEncoder final : public AudioEncoder {
   MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
   MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
   MOCK_CONST_METHOD0(GetTargetBitrate, int());
-  // Note, we explicitly chose not to create a mock for the Encode method.
-  MOCK_METHOD4(EncodeInternal,
-               EncodedInfo(uint32_t timestamp,
-                           rtc::ArrayView<const int16_t> audio,
-                           size_t max_encoded_bytes,
-                           uint8_t* encoded));
   MOCK_METHOD0(Reset, void());
   MOCK_METHOD1(SetFec, bool(bool enable));
   MOCK_METHOD1(SetDtx, bool(bool enable));
@@ -46,6 +40,63 @@ class MockAudioEncoder final : public AudioEncoder {
   MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
 };
 
+class MockAudioEncoder final : public MockAudioEncoderBase {
+ public:
+  // Note, we explicitly chose not to create a mock for the Encode method.
+  MOCK_METHOD3(EncodeInternal,
+               EncodedInfo(uint32_t timestamp,
+                           rtc::ArrayView<const int16_t> audio,
+                           rtc::Buffer* encoded));
+};
+
+class MockAudioEncoderDeprecated final : public MockAudioEncoderBase {
+ public:
+  // Note, we explicitly chose not to create a mock for the Encode method.
+  MOCK_METHOD4(EncodeInternal,
+               EncodedInfo(uint32_t timestamp,
+                           rtc::ArrayView<const int16_t> audio,
+                           size_t max_encoded_bytes,
+                           uint8_t* encoded));
+};
+
+class MockAudioEncoderHelper {
+ public:
+  MockAudioEncoderHelper() : write_payload_(false), payload_(NULL) {

[kwiberg-webrtc, 2016/02/25 00:29:04]

nullptr

[ossu, 2016/02/25 10:39:51]

Acknowledged.

+    memset(&info_, 0, sizeof(info_));
+  }
+
+  AudioEncoder::EncodedInfo Encode(uint32_t timestamp,
+                                   rtc::ArrayView<const int16_t> audio,
+                                   rtc::Buffer* encoded) {
+    RTC_CHECK(encoded);
+
+    if (write_payload_)
+      encoded->AppendData(payload_, info_.encoded_bytes);
+    else
+      encoded->SetSize(encoded->size() + info_.encoded_bytes);
+
+    return info_;
+  }
+
+  AudioEncoder::EncodedInfo DEPRECATED_Encode(
+      uint32_t timestamp,
+      rtc::ArrayView<const int16_t> audio,
+      size_t max_bytes_encoded,
+      uint8_t* encoded) {
+
+    if (write_payload_) {
+      RTC_CHECK(encoded);
+      std::memcpy(encoded, payload_, info_.encoded_bytes);
+    }
+
+    return info_;
+  }
+
+  AudioEncoder::EncodedInfo info_;
+  bool write_payload_;
+  const uint8_t* payload_;
+};
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_