| Index: webrtc/media/engine/webrtcvoiceengine.h
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| diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
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| index a16d0428fe71a80c0484042f8ecb8fd1faebb49a..0ccc64908dc77539a2f7139253e78d8adccfe9a7 100644
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| --- a/webrtc/media/engine/webrtcvoiceengine.h
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| +++ b/webrtc/media/engine/webrtcvoiceengine.h
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| @@ -179,9 +179,9 @@
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|    bool CanInsertDtmf() override;
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|    bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
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|  
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| -  void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
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| +  void OnPacketReceived(rtc::Buffer* packet,
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|                          const rtc::PacketTime& packet_time) override;
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| -  void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
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| +  void OnRtcpReceived(rtc::Buffer* packet,
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|                        const rtc::PacketTime& packet_time) override;
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|    void OnReadyToSend(bool ready) override {}
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|    bool GetStats(VoiceMediaInfo* info) override;
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| @@ -194,14 +194,16 @@
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|    bool SendRtp(const uint8_t* data,
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|                 size_t len,
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|                 const webrtc::PacketOptions& options) override {
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| -    rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
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| +    rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
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| +                       kMaxRtpPacketLen);
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|      rtc::PacketOptions rtc_options;
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|      rtc_options.packet_id = options.packet_id;
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|      return VoiceMediaChannel::SendPacket(&packet, rtc_options);
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|    }
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|  
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|    bool SendRtcp(const uint8_t* data, size_t len) override {
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| -    rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
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| +    rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
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| +                       kMaxRtpPacketLen);
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|      return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
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|    }
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|  
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| 
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