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Side by Side Diff: webrtc/video/video_quality_test.cc

Issue 2730433005: Fixed VP8 simulcast full-stack-tests to not decode non-selected streams. (Closed)
Patch Set: Created 3 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 #include "webrtc/video/video_quality_test.h" 10 #include "webrtc/video/video_quality_test.h"
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1264 1264
1265 CreateMatchingReceiveConfigs(recv_transport); 1265 CreateMatchingReceiveConfigs(recv_transport);
1266 1266
1267 for (size_t i = 0; i < num_video_streams; ++i) { 1267 for (size_t i = 0; i < num_video_streams; ++i) {
1268 video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 1268 video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
1269 video_receive_configs_[i].rtp.rtx_ssrc = kSendRtxSsrcs[i]; 1269 video_receive_configs_[i].rtp.rtx_ssrc = kSendRtxSsrcs[i];
1270 video_receive_configs_[i].rtp.rtx_payload_types[payload_type] = 1270 video_receive_configs_[i].rtp.rtx_payload_types[payload_type] =
1271 kSendRtxPayloadType; 1271 kSendRtxPayloadType;
1272 video_receive_configs_[i].rtp.transport_cc = params_.call.send_side_bwe; 1272 video_receive_configs_[i].rtp.transport_cc = params_.call.send_side_bwe;
1273 video_receive_configs_[i].rtp.remb = !params_.call.send_side_bwe; 1273 video_receive_configs_[i].rtp.remb = !params_.call.send_side_bwe;
1274 if (i != params_.ss.selected_stream) {
sprang_webrtc 2017/03/02 15:27:29 Maybe add a comment explaining what is going on he
ilnik 2017/03/02 15:37:19 Done.
1275 VideoReceiveStream::Decoder decoder;
1276 decoder.decoder = new test::FakeDecoder();
1277 decoder.payload_type = video_send_config_.encoder_settings.payload_type;
1278 decoder.payload_name = video_send_config_.encoder_settings.payload_name;
1279 video_receive_configs_[i].decoders.clear();
1280 allocated_decoders_.push_back(
1281 std::unique_ptr<VideoDecoder>(decoder.decoder));
sprang_webrtc 2017/03/02 15:27:29 nit: you can even just do allocated_decoders_.e
ilnik 2017/03/02 15:37:19 Done.
1282 video_receive_configs_[i].decoders.push_back(decoder);
1283 }
1274 } 1284 }
1275 1285
1276 if (params_.video.flexfec) { 1286 if (params_.video.flexfec) {
1277 // Override send config constructed by CreateSendConfig. 1287 // Override send config constructed by CreateSendConfig.
1278 video_send_config_.rtp.flexfec.protected_media_ssrcs = { 1288 video_send_config_.rtp.flexfec.protected_media_ssrcs = {
1279 kVideoSendSsrcs[params_.ss.selected_stream]}; 1289 kVideoSendSsrcs[params_.ss.selected_stream]};
1280 1290
1281 // The matching receive config is _not_ created by 1291 // The matching receive config is _not_ created by
1282 // CreateMatchingReceiveConfigs, since VideoQualityTest is not a BaseTest. 1292 // CreateMatchingReceiveConfigs, since VideoQualityTest is not a BaseTest.
1283 // Set up the receive config manually instead. 1293 // Set up the receive config manually instead.
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1713 std::ostringstream str; 1723 std::ostringstream str;
1714 str << receive_logs_++; 1724 str << receive_logs_++;
1715 std::string path = 1725 std::string path =
1716 params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; 1726 params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
1717 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), 1727 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
1718 10000000); 1728 10000000);
1719 } 1729 }
1720 } 1730 }
1721 1731
1722 } // namespace webrtc 1732 } // namespace webrtc
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