| 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).
|
|
|