commit 714e3cbb48c704fe696e1f73f9a2f8dc0d2c0a16
author Ali Tofigh <alito@webrtc.org> Wed Jul 20 12:53:07 2022 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Wed Jul 20 13:34:23 2022 +0000
tree 05ecc67664c0bedcd565d2ae662f82a7a80f7ff9
parent 761072f68e2cf568ef8ed5bc103eea35722f03ea

Adopt absl::string_view in modules/audio_coding/

Bug: webrtc:13579
Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37573}

modules/audio_coding/test/PacketLossTest.h

diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index 4c0dfd8..d841d65 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -13,6 +13,7 @@
 
 #include <string>
 
+#include "absl/strings/string_view.h"
 #include "modules/audio_coding/test/EncodeDecodeTest.h"
 
 namespace webrtc {
@@ -22,7 +23,7 @@
   ReceiverWithPacketLoss();
   void Setup(AudioCodingModule* acm,
              RTPStream* rtpStream,
-             std::string out_file_name,
+             absl::string_view out_file_name,
              int channels,
              int file_num,
              int loss_rate,
@@ -43,7 +44,7 @@
   SenderWithFEC();
   void Setup(AudioCodingModule* acm,
              RTPStream* rtpStream,
-             std::string in_file_name,
+             absl::string_view in_file_name,
              int payload_type,
              SdpAudioFormat format,
              int expected_loss_rate);