Making FakeNetworkPipe demux audio and video packets.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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 "webrtc/video/video_quality_test.h"
@@ skipping 36 matching lines @@
 #include "webrtc/test/vcm_capturer.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
 namespace {
 
 constexpr int kSendStatsPollingIntervalMs = 1000;
 constexpr int kPayloadTypeH264 = 122;
 constexpr int kPayloadTypeVP8 = 123;
 constexpr int kPayloadTypeVP9 = 124;
+const std::map<uint8_t, webrtc::MediaType> additional_pt_map = {

[nisse-webrtc, 2017/04/06 12:23:56]

What do you think about defining *all* payload types used by tests, and the
payload type map, in a single place? Should make it easier to avoid collisions.

[minyue-webrtc, 2017/04/06 18:45:10]

sure. will try

+    {kPayloadTypeH264, webrtc::MediaType::VIDEO},
+    {kPayloadTypeVP8, webrtc::MediaType::VIDEO},
+    {kPayloadTypeVP9, webrtc::MediaType::VIDEO},
+};
+
 constexpr size_t kMaxComparisons = 10;
 constexpr char kSyncGroup[] = "av_sync";
 constexpr int kOpusMinBitrateBps = 6000;
 constexpr int kOpusBitrateFbBps = 32000;
 constexpr int kFramesSentInQuickTest = 1;
 constexpr uint32_t kThumbnailSendSsrcStart = 0xE0000;
 constexpr uint32_t kThumbnailRtxSsrcStart = 0xF0000;
 
 struct VoiceEngineState {
   VoiceEngineState()
@@ skipping 1515 matching lines @@
     RTC_CHECK(graph_data_output_file)
         << "Can't open the file " << params_.analyzer.graph_data_output_filename
         << "!";
   }
 
   webrtc::RtcEventLogNullImpl event_log;
   Call::Config call_config(&event_log_);
   call_config.bitrate_config = params.call.call_bitrate_config;
   CreateCalls(call_config, call_config);
 
+  std::map<uint8_t, MediaType> payload_type_map = payload_type_map_;
+  payload_type_map.insert(additional_pt_map.begin(), additional_pt_map.end());
+
   test::LayerFilteringTransport send_transport(
       params_.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
-      params_.video.selected_tl, params_.ss.selected_sl);
-  test::DirectTransport recv_transport(
-      params_.pipe, receiver_call_.get(), MediaType::VIDEO);
+      params_.video.selected_tl, params_.ss.selected_sl, payload_type_map);
+
+  test::DirectTransport recv_transport(params_.pipe, receiver_call_.get(),
+                                       payload_type_map);
 
   std::string graph_title = params_.analyzer.graph_title;
   if (graph_title.empty())
     graph_title = VideoQualityTest::GenerateGraphTitle();
 
   bool is_quick_test_enabled = field_trial::IsEnabled("WebRTC-QuickPerfTest");
   VideoAnalyzer analyzer(
       &send_transport, params_.analyzer.test_label,
       params_.analyzer.avg_psnr_threshold, params_.analyzer.avg_ssim_threshold,
       is_quick_test_enabled
@@ skipping 98 matching lines @@
   // adaptation.
   audio_send_config_.rtp.extensions.clear();
   if (params_.call.send_side_bwe) {
     audio_send_config_.rtp.extensions.push_back(
         webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
                              test::kTransportSequenceNumberExtensionId));
     audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps;
     audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
   }
   audio_send_config_.send_codec_spec.codec_inst =
-      CodecInst{120, "OPUS", 48000, 960, 2, 64000};
+      CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
   audio_send_config_.send_codec_spec.enable_opus_dtx = params_.audio.dtx;
   audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
 
   AudioReceiveStream::Config audio_config;
   audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
   audio_config.rtcp_send_transport = transport;
   audio_config.voe_channel_id = receive_channel_id;
   audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
   audio_config.rtp.transport_cc = params_.call.send_side_bwe;
   audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
   audio_config.decoder_factory = decoder_factory_;
+  audio_config.decoder_map = {{kAudioSendPayloadType, {"OPUS", 48000, 2}}};
   if (params_.video.enabled && params_.audio.sync_video)
     audio_config.sync_group = kSyncGroup;
 
   *audio_receive_stream = call->CreateAudioReceiveStream(audio_config);
 }
 
 void VideoQualityTest::RunWithRenderers(const Params& params) {
   params_ = params;
   CheckParams();
 
   // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
   // match the full stack tests.
   webrtc::RtcEventLogNullImpl event_log;
   Call::Config call_config(&event_log_);
   call_config.bitrate_config = params_.call.call_bitrate_config;
 
   ::VoiceEngineState voe;
   if (params_.audio.enabled) {
     CreateVoiceEngine(&voe, decoder_factory_);
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe.voice_engine;
     audio_state_config.audio_mixer = AudioMixerImpl::Create();
     call_config.audio_state = AudioState::Create(audio_state_config);
   }
 
   std::unique_ptr<Call> call(Call::Create(call_config));
 
   // TODO(minyue): consider if this is a good transport even for audio only
   // calls.
+  std::map<uint8_t, MediaType> payload_type_map = payload_type_map_;
+  payload_type_map.insert(additional_pt_map.begin(), additional_pt_map.end());
   test::LayerFilteringTransport transport(
       params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
-      params.video.selected_tl, params_.ss.selected_sl);
+      params.video.selected_tl, params_.ss.selected_sl, payload_type_map);
+
   // TODO(ivica): Use two calls to be able to merge with RunWithAnalyzer or at
   // least share as much code as possible. That way this test would also match
   // the full stack tests better.
   transport.SetReceiver(call->Receiver());
 
   VideoReceiveStream* video_receive_stream = nullptr;
   FlexfecReceiveStream* flexfec_receive_stream = nullptr;
   std::unique_ptr<test::VideoRenderer> local_preview;
   std::unique_ptr<test::VideoRenderer> loopback_video;
   if (params_.video.enabled) {
@@ skipping 115 matching lines @@
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 }  // namespace webrtc