Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc |
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc |
index 3b8b14015a203271eec9edabdee1b7c775556396..20b221bd7f77f2ca60557ca7839619f0725dda11 100644 |
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc |
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc |
@@ -611,7 +611,9 @@ int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const { |
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload, |
const size_t payload_length, |
const WebRtcRTPHeader& rtp_header) { |
- return receiver_.InsertPacket(rtp_header, incoming_payload, payload_length); |
+ return receiver_.InsertPacket( |
+ rtp_header, |
+ rtc::ArrayView<const uint8_t>(incoming_payload, payload_length)); |
} |
// Minimum playout delay (Used for lip-sync). |