Expose codec implementation names in stats.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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>
@@ skipping 2321 matching lines @@
       for (size_t i = 0; i < receive_streams_.size(); ++i) {
         VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
         EXPECT_EQ(expected_receive_ssrcs_[i], stats.ssrc);
 
         // Make sure all fields have been populated.
         // TODO(pbos): Use CompoundKey if/when we ever know that all stats are
         // always filled for all receivers.
         receive_stats_filled_["IncomingRate"] |=
             stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
 
+        send_stats_filled_["DecoderImplementationName"] |=
+            stats.decoder_implementation_name ==
+            test::FakeDecoder::kImplementationName;
         receive_stats_filled_["RenderDelayAsHighAsExpected"] |=
             stats.render_delay_ms >= kExpectedRenderDelayMs;
 
         receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
 
         receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
 
         receive_stats_filled_["StatisticsUpdated"] |=
             stats.rtcp_stats.cumulative_lost != 0 ||
             stats.rtcp_stats.extended_max_sequence_number != 0 ||
@@ skipping 35 matching lines @@
     bool CheckSendStats() {
       RTC_DCHECK(send_stream_ != nullptr);
       VideoSendStream::Stats stats = send_stream_->GetStats();
 
       send_stats_filled_["NumStreams"] |=
           stats.substreams.size() == expected_send_ssrcs_.size();
 
       send_stats_filled_["CpuOveruseMetrics"] |=
           stats.avg_encode_time_ms != 0 || stats.encode_usage_percent != 0;
 
+      send_stats_filled_["EncoderImplementationName"] |=
+          stats.encoder_implementation_name ==
+          test::FakeEncoder::kImplementationName;
+
       for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
                stats.substreams.begin();
            it != stats.substreams.end(); ++it) {
         EXPECT_TRUE(expected_send_ssrcs_.find(it->first) !=
                     expected_send_ssrcs_.end());
 
         send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
             stats.input_frame_rate != 0;
 
         const VideoSendStream::StreamStats& stream_stats = it->second;
@@ skipping 791 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc