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};
-

[oprypin_webrtc, 2017/04/06 11:45:32]

Why does the constant need to be moved?

[minyue-webrtc, 2017/04/06 11:56:33]

Because I do not want to use "120" any longer. (But yes, you may argue that we
can move line 92 in PS3 here. I see only one line of using the const, so I think
inline is better.)

I have a comment in PS3. The basic idea is that for tests derived from CallTest,
we try not to introduce new payload types than those defined in CallTest, so
that when one creates a Transport, he/she does not need to consider what
payload_type_map to give. Arbitrary payload types may also lead to conflicts.

 // 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};

[minyue-webrtc, 2017/04/06 11:33:38]

use the payload defined in CallTest so that
CreateReceiveTransport/CreateSendTransport can use
test::CallTest::payload_type_map_

 }
 
 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