1 files changed, 7 insertions, 4 deletions
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index 0a41c6ec20..d7b01fe33a 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -11,10 +11,12 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
 
+#include <string>
+
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace test {
@@ -36,8 +38,8 @@ class NetEqExternalDecoderTest {
   // |payload_size_bytes| bytes. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
-  virtual void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
-                            size_t payload_size_bytes,
+  virtual void InsertPacket(WebRtcRTPHeader rtp_header,
+                            rtc::ArrayView<const uint8_t> payload,
                             uint32_t receive_timestamp);
 
   // Get 10 ms of audio data. The data is written to |output|, which can hold
@@ -49,9 +51,10 @@ class NetEqExternalDecoderTest {
 
  private:
   NetEqDecoder codec_;
+  std::string name_ = "dummy name";
   AudioDecoder* decoder_;
   int sample_rate_hz_;
-  int channels_;
+  size_t channels_;
   rtc::scoped_ptr<NetEq> neteq_;
 };