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.cc

diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index a1629fd..6f87659 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -30,6 +30,7 @@
                                    RTPStream* rtpStream,
                                    std::string out_file_name,
                                    int channels,
+                                   int file_num,
                                    int loss_rate,
                                    int burst_length) {
   loss_rate_ = loss_rate;
@@ -37,7 +38,7 @@
   burst_lost_counter_ = burst_length_;  // To prevent first packet gets lost.
   rtc::StringBuilder ss;
   ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
-  Receiver::Setup(acm, rtpStream, ss.str(), channels);
+  Receiver::Setup(acm, rtpStream, ss.str(), channels, file_num);
 }
 
 bool ReceiverWithPacketLoss::IncomingPacket() {
@@ -89,10 +90,11 @@
 void SenderWithFEC::Setup(AudioCodingModule* acm,
                           RTPStream* rtpStream,
                           std::string in_file_name,
-                          int sample_rate,
-                          int channels,
+                          int payload_type,
+                          SdpAudioFormat format,
                           int expected_loss_rate) {
-  Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
+  Sender::Setup(acm, rtpStream, in_file_name, format.clockrate_hz, payload_type,
+                format);
   EXPECT_TRUE(SetFEC(true));
   EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
 }
@@ -123,8 +125,6 @@
       in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz"
                                    : "audio_coding/teststereo32kHz"),
       sample_rate_hz_(32000),
-      sender_(new SenderWithFEC),
-      receiver_(new ReceiverWithPacketLoss),
       expected_loss_rate_(expected_loss_rate),
       actual_loss_rate_(actual_loss_rate),
       burst_length_(burst_length) {}
@@ -133,40 +133,32 @@
 #ifndef WEBRTC_CODEC_OPUS
   return;
 #else
-  AudioCodingModule::Config config;
-  config.decoder_factory = CreateBuiltinAudioDecoderFactory();
-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(config));
-
-  int codec_id = acm->Codec("opus", 48000, channels_);
-
   RTPFile rtpFile;
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
+      AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+  SdpAudioFormat send_format = SdpAudioFormat("opus", 48000, 2);
+  if (channels_ == 2) {
+    send_format.parameters = {{"stereo", "1"}};
+  }
+
   std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
                                                     "packet_loss_test");
-
-  // Encode to file
   rtpFile.Open(fileName.c_str(), "wb+");
   rtpFile.WriteHeader();
-
-  sender_->codeId = codec_id;
-
-  sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
+  SenderWithFEC sender;
+  sender.Setup(acm.get(), &rtpFile, in_file_name_, 120, send_format,
                  expected_loss_rate_);
-  if (acm->SendCodec()) {
-    sender_->Run();
-  }
-  sender_->Teardown();
+  sender.Run();
+  sender.Teardown();
   rtpFile.Close();
 
-  // Decode to file
   rtpFile.Open(fileName.c_str(), "rb");
   rtpFile.ReadHeader();
-
-  receiver_->codeId = codec_id;
-
-  receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
+  ReceiverWithPacketLoss receiver;
+  receiver.Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 15,
                    actual_loss_rate_, burst_length_);
-  receiver_->Run();
-  receiver_->Teardown();
+  receiver.Run();
+  receiver.Teardown();
   rtpFile.Close();
 #endif
 }