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

Issue 1315413002: Wire up currently-received video codec to stats. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: feedback Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 <algorithm> 10 #include <algorithm>
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2220 stats.frame_counts.delta_frames != 0; 2220 stats.frame_counts.delta_frames != 0;
2221 2221
2222 receive_stats_filled_["CName"] |= !stats.c_name.empty(); 2222 receive_stats_filled_["CName"] |= !stats.c_name.empty();
2223 2223
2224 receive_stats_filled_["RtcpPacketTypeCount"] |= 2224 receive_stats_filled_["RtcpPacketTypeCount"] |=
2225 stats.rtcp_packet_type_counts.fir_packets != 0 || 2225 stats.rtcp_packet_type_counts.fir_packets != 0 ||
2226 stats.rtcp_packet_type_counts.nack_packets != 0 || 2226 stats.rtcp_packet_type_counts.nack_packets != 0 ||
2227 stats.rtcp_packet_type_counts.pli_packets != 0 || 2227 stats.rtcp_packet_type_counts.pli_packets != 0 ||
2228 stats.rtcp_packet_type_counts.nack_requests != 0 || 2228 stats.rtcp_packet_type_counts.nack_requests != 0 ||
2229 stats.rtcp_packet_type_counts.unique_nack_requests != 0; 2229 stats.rtcp_packet_type_counts.unique_nack_requests != 0;
2230
2231 assert(stats.current_payload_type == -1 ||
2232 stats.current_payload_type == kFakeSendPayloadType);
2233 receive_stats_filled_["IncomingPayloadType"] |=
2234 stats.current_payload_type == kFakeSendPayloadType;
2230 } 2235 }
2231 2236
2232 return AllStatsFilled(receive_stats_filled_); 2237 return AllStatsFilled(receive_stats_filled_);
2233 } 2238 }
2234 2239
2235 bool CheckSendStats() { 2240 bool CheckSendStats() {
2236 DCHECK(send_stream_ != nullptr); 2241 DCHECK(send_stream_ != nullptr);
2237 VideoSendStream::Stats stats = send_stream_->GetStats(); 2242 VideoSendStream::Stats stats = send_stream_->GetStats();
2238 2243
2239 send_stats_filled_["NumStreams"] |= 2244 send_stats_filled_["NumStreams"] |=
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3051 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) 3056 EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
3052 << "Enabling RTX requires rtpmap: rtx negotiation."; 3057 << "Enabling RTX requires rtpmap: rtx negotiation.";
3053 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) 3058 EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
3054 << "Enabling RTP extensions require negotiation."; 3059 << "Enabling RTP extensions require negotiation.";
3055 3060
3056 VerifyEmptyNackConfig(default_receive_config.rtp.nack); 3061 VerifyEmptyNackConfig(default_receive_config.rtp.nack);
3057 VerifyEmptyFecConfig(default_receive_config.rtp.fec); 3062 VerifyEmptyFecConfig(default_receive_config.rtp.fec);
3058 } 3063 }
3059 3064
3060 } // namespace webrtc 3065 } // namespace webrtc
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