Chromium Code Reviews| Index: webrtc/video/video_quality_test.cc |
| diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc |
| index 3fa07db7fdbcc3f5a3ce60193b019daeb5229ddf..dbfe66593bf14d88387410dab9ec25fa8f9a5a61 100644 |
| --- a/webrtc/video/video_quality_test.cc |
| +++ b/webrtc/video/video_quality_test.cc |
| @@ -54,6 +54,12 @@ 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 typ
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; |
| @@ -1589,11 +1595,15 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) { |
| 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()) |
| @@ -1712,7 +1722,7 @@ void VideoQualityTest::SetupAudio(int send_channel_id, |
| 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_); |
| @@ -1724,6 +1734,7 @@ void VideoQualityTest::SetupAudio(int send_channel_id, |
| 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; |
| @@ -1753,9 +1764,12 @@ void VideoQualityTest::RunWithRenderers(const Params& params) { |
| // 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. |