Making FakeNetworkPipe demux audio and video packets.

webrtc/audio/test/low_bandwidth_audio_test.cc

 /*
  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <algorithm>
 
 #include "webrtc/audio/test/low_bandwidth_audio_test.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace {
 // Wait half a second between stopping sending and stopping receiving audio.
 constexpr int kExtraRecordTimeMs = 500;
 
-// Large bitrate by default.
-const webrtc::CodecInst kDefaultCodec{120, "OPUS", 48000, 960, 2, 64000};
-
 // The best that can be done with PESQ.
 constexpr int kAudioFileBitRate = 16000;
-}
+}  // namespace
 
 namespace webrtc {
 namespace test {
 
 AudioQualityTest::AudioQualityTest()
     : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
 
 size_t AudioQualityTest::GetNumVideoStreams() const {
   return 0;
 }
@@ skipping 33 matching lines @@
 }
 
 FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() {
   return FakeNetworkPipe::Config();
 }
 
 test::PacketTransport* AudioQualityTest::CreateSendTransport(
     Call* sender_call) {
   return new test::PacketTransport(
       sender_call, this, test::PacketTransport::kSender,
-      MediaType::AUDIO,
-      GetNetworkPipeConfig());
+      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
 }
 
 test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
-  return new test::PacketTransport(nullptr, this,
-      test::PacketTransport::kReceiver, MediaType::AUDIO,
-      GetNetworkPipeConfig());
+  return new test::PacketTransport(
+      nullptr, this, test::PacketTransport::kReceiver,
+      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
 }
 
 void AudioQualityTest::ModifyAudioConfigs(
   AudioSendStream::Config* send_config,
   std::vector<AudioReceiveStream::Config>* receive_configs) {
-  send_config->send_codec_spec.codec_inst = kDefaultCodec;
+  send_config->send_codec_spec.codec_inst = webrtc::CodecInst{
+      test::CallTest::kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
 }
 
 void AudioQualityTest::PerformTest() {
   // Wait until the input audio file is done...
   send_audio_device_->WaitForRecordingEnd();
   // and some extra time to account for network delay.
   SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
 }
 
 void AudioQualityTest::OnTestFinished() {
@@ skipping 12 matching lines @@
 TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
   AudioQualityTest test;
   RunBaseTest(&test);
 }
 
 
 class Mobile2GNetworkTest : public AudioQualityTest {
   void ModifyAudioConfigs(AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override {
     send_config->send_codec_spec.codec_inst = CodecInst{
-        120,     // pltype
-        "OPUS",  // plname
-        48000,   // plfreq
-        2880,    // pacsize
-        1,       // channels
-        6000     // rate bits/sec
+        test::CallTest::kAudioSendPayloadType,  // pltype
+        "OPUS",                                 // plname
+        48000,                                  // plfreq
+        2880,                                   // pacsize
+        1,                                      // channels
+        6000                                    // rate bits/sec
     };
   }
 
   FakeNetworkPipe::Config GetNetworkPipeConfig() override {
     FakeNetworkPipe::Config pipe_config;
     pipe_config.link_capacity_kbps = 12;
     pipe_config.queue_length_packets = 1500;
     pipe_config.queue_delay_ms = 400;
     return pipe_config;
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc