Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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 95 matching lines @@
   }
   void SetupForMultiSendStream() {
     EXPECT_TRUE(SetupEngineWithSendStream());
     // Remove stream added in Setup.
     EXPECT_TRUE(call_.GetAudioSendStream(kSsrc1));
     EXPECT_TRUE(channel_->RemoveSendStream(kSsrc1));
     // Verify the channel does not exist.
     EXPECT_FALSE(call_.GetAudioSendStream(kSsrc1));
   }
   void DeliverPacket(const void* data, int len) {
-    rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len);
+    rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len);
     channel_->OnPacketReceived(&packet, rtc::PacketTime());
   }
   void TearDown() override {
     delete channel_;
     engine_.Terminate();
   }
 
   const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) {
     const auto* send_stream = call_.GetAudioSendStream(ssrc);
     EXPECT_TRUE(send_stream);
@@ skipping 2926 matching lines @@
   for (uint32_t ssrc : ssrcs) {
     const auto* s = call_.GetAudioReceiveStream(ssrc);
     EXPECT_NE(nullptr, s);
     EXPECT_EQ(0, s->GetConfig().rtp.extensions.size());
   }
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
   // Test that packets are forwarded to the Call when configured accordingly.
   const uint32_t kAudioSsrc = 1;
-  rtc::Buffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
+  rtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
   static const unsigned char kRtcp[] = {
     0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02,
     0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00,
     0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
     0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
   };
-  rtc::Buffer kRtcpPacket(kRtcp, sizeof(kRtcp));
+  rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp));
 
   EXPECT_TRUE(SetupEngineWithSendStream());
   cricket::WebRtcVoiceMediaChannel* media_channel =
       static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
   EXPECT_TRUE(media_channel->AddRecvStream(
       cricket::StreamParams::CreateLegacy(kAudioSsrc)));
 
   EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
   const cricket::FakeAudioReceiveStream* s =
@@ skipping 224 matching lines @@
   cricket::WebRtcVoiceEngine engine;
   EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config()));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
                                            cricket::AudioOptions(), call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }