commit 657b296ff5448e62bd4d0c3daeee1900cbb168f0
author Fredrik Solenberg <solenberg@webrtc.org> Wed Dec 05 10:30:25 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Dec 05 10:38:23 2018 +0000
tree 0d9087b305b5e5a044711f0dc9f71ad2a1d5b9e4
parent 5b1477839d8569291b88dfe950089d0ebf34bc8f

Reland "Remove CodecInst pt.1"

This is a reland of 056f9738bf7a3d16da45398239656e165c4e0851

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}

Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}

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_;