Index: webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h |
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 c9fe11f65609175f27622c2e65aa623a3324e262..0f71e5df06b24479e1b015521a15939231132c5b 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 |
@@ -36,8 +36,8 @@ 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 |