Let PacketSource::NextPacket() return an std::unique_ptr

webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h

 /*
  *  Copyright (c) 2014 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.
  */
 
@@ skipping 26 matching lines @@
   // Registers the send codec. Returns true on success, false otherwise.
   bool RegisterCodec(const char* payload_name,
                      int sampling_freq_hz,
                      int channels,
                      int payload_type,
                      int frame_size_samples);
 
   // Registers an external send codec. Returns true on success, false otherwise.
   bool RegisterExternalCodec(AudioEncoder* external_speech_encoder);
 
-  // Returns the next encoded packet. Returns NULL if the test duration was
-  // exceeded. Ownership of the packet is handed over to the caller.
   // Inherited from PacketSource.
-  Packet* NextPacket() override;
+  std::unique_ptr<Packet> NextPacket() override;
 
   // Inherited from AudioPacketizationCallback.
   int32_t SendData(FrameType frame_type,
                    uint8_t payload_type,
                    uint32_t timestamp,
                    const uint8_t* payload_data,
                    size_t payload_len_bytes,
                    const RTPFragmentationHeader* fragmentation) override;
 
   AudioCodingModule* acm() { return acm_.get(); }
 
  private:
   static const int kBlockSizeMs = 10;
 
   // Creates a Packet object from the last packet produced by ACM (and received
-  // through the SendData method as a callback). Ownership of the new Packet
-  // object is transferred to the caller.
-  Packet* CreatePacket();
+  // through the SendData method as a callback).
+  std::unique_ptr<Packet> CreatePacket();
 
   SimulatedClock clock_;
   std::unique_ptr<AudioCodingModule> acm_;
   InputAudioFile* audio_source_;
   int source_rate_hz_;
   const size_t input_block_size_samples_;
   AudioFrame input_frame_;
   bool codec_registered_;
   int test_duration_ms_;
   // The following member variables are set whenever SendData() is called.
   FrameType frame_type_;
   int payload_type_;
   uint32_t timestamp_;
   uint16_t sequence_number_;
   std::vector<uint8_t> last_payload_vec_;
   bool data_to_send_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_