Switch voice transport to use Call and Stream instead of VoENetwork.
VoENetwork is kept for now, but is not really used anylonger.
webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.
BUG=webrtc:5079
TBR=tommi
Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index d22da74..ac49e28 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -218,12 +218,12 @@
int SetOpusDtx(bool enable_dtx);
// VoENetwork
- int32_t RegisterExternalTransport(Transport& transport);
+ int32_t RegisterExternalTransport(Transport* transport);
int32_t DeRegisterExternalTransport();
- int32_t ReceivedRTPPacket(const int8_t* data,
+ int32_t ReceivedRTPPacket(const uint8_t* received_packet,
size_t length,
const PacketTime& packet_time);
- int32_t ReceivedRTCPPacket(const int8_t* data, size_t length);
+ int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
// VoEFile
int StartPlayingFileLocally(const char* fileName,