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