commit 056f9738bf7a3d16da45398239656e165c4e0851
author Fredrik Solenberg <solenberg@webrtc.org> Mon Dec 03 15:22:16 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Mon Dec 03 15:16:20 2018 +0000
tree 4dc9545e3453d52604787e4ee9fd0c749fc3a941
parent 450b548cade5498bbdef6eddce0703f52296bf65

Remove CodecInst pt.1

Update audio_coding tests to not use CodecInst.

Bug: webrtc:7626
Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
Reviewed-on: https://webrtc-review.googlesource.com/c/112594
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25879}

modules/audio_coding/test/PacketLossTest.h

diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index 6018301..b26f6ec 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -11,7 +11,6 @@
 #ifndef MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
 #define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
 
-#include <memory>
 #include <string>
 #include "modules/audio_coding/test/EncodeDecodeTest.h"
 
@@ -24,6 +23,7 @@
              RTPStream* rtpStream,
              std::string out_file_name,
              int channels,
+             int file_num,
              int loss_rate,
              int burst_length);
   bool IncomingPacket() override;
@@ -43,8 +43,8 @@
   void Setup(AudioCodingModule* acm,
              RTPStream* rtpStream,
              std::string in_file_name,
-             int sample_rate,
-             int channels,
+             int payload_type,
+             SdpAudioFormat format,
              int expected_loss_rate);
   bool SetPacketLossRate(int expected_loss_rate);
   bool SetFEC(bool enable_fec);
@@ -65,8 +65,6 @@
   int channels_;
   std::string in_file_name_;
   int sample_rate_hz_;
-  std::unique_ptr<SenderWithFEC> sender_;
-  std::unique_ptr<ReceiverWithPacketLoss> receiver_;
   int expected_loss_rate_;
   int actual_loss_rate_;
   int burst_length_;