commit ec0f45be1144078338805b8bd525b29e8e736085
author Fredrik Solenberg <solenberg@webrtc.org> Mon Dec 03 15:50:44 2018 +0000
committer Commit Bot <commit-bot@chromium.org> Mon Dec 03 15:50:51 2018 +0000
tree 63daac9a0b8ed19a2397140f6f66c2e4059dd594
parent 9d54bd889898baa02a2b3920cbaf80fa3c70e598

Revert "Remove CodecInst pt.1"

This reverts commit 056f9738bf7a3d16da45398239656e165c4e0851.

Reason for revert: breaks downstream

Original change's description:
> 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}

TBR=solenberg@webrtc.org,kwiberg@webrtc.org

Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7626
Reviewed-on: https://webrtc-review.googlesource.com/c/112906
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25881}

modules/audio_coding/test/PacketLossTest.h

diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index b26f6ec..6018301 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -11,6 +11,7 @@
 #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"
 
@@ -23,7 +24,6 @@
              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 payload_type,
-             SdpAudioFormat format,
+             int sample_rate,
+             int channels,
              int expected_loss_rate);
   bool SetPacketLossRate(int expected_loss_rate);
   bool SetFEC(bool enable_fec);
@@ -65,6 +65,8 @@
   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_;