AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments

webrtc/modules/audio_coding/main/acm2/acm_receiver.h

Index: webrtc/modules/audio_coding/main/acm2/acm_receiver.h
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 4b080ba93ae50100fce2d41a5727a35e01e6805a..f92ff9617a6195e228766fff8fac7b18a13a5ca9 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -14,6 +14,7 @@
 #include <map>
 #include <vector>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
@@ -66,8 +67,7 @@ class AcmReceiver {
   //                           <0 if NetEq returned an error.
   //
   int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                   const uint8_t* incoming_payload,
-                   size_t length_payload);
+                   rtc::ArrayView<const uint8_t> incoming_payload);
 
   //
   // Asks NetEq for 10 milliseconds of decoded audio.
@@ -279,7 +279,7 @@ class AcmReceiver {
 
  private:
   const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
-                                    const uint8_t* payload) const
+                                    uint8_t payload_type) const
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;