commit b0d8a3728cc8622cf39365859262b413d23ca5d9
author Per K <perkj@webrtc.org> Thu Jan 26 15:11:31 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu Jan 26 17:36:01 2023
tree 8dd830f82b9bb28d0f79a104f0d9a3c1ec58d1da
parent 6c032cb8356b0d3f717c4fcf50796241f2bba6c2

Ensure CallTest derived tests per default set min/max audio bitrate.

This ensure BWE works as intended with transport sequence numbers on
audio.

Tested with webrtc_perf_tests --gtest_filter=CallPerfTest.Min_Bitrate_VideoAndAudio
and --gtest_filter=Rampup*

Bug: webrtc:14854, webrtc:7135, b/266786240
Change-Id: I3b7a743149c22035e582a2157b5f0a93747857cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291523
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39208}

call/call_perf_tests.cc

diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index a50f5ee..f90d8ec 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -35,6 +35,7 @@
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "rtc_base/thread.h"
@@ -910,7 +911,6 @@
                                             int start_bwe,
                                             int max_bwe) {
   static const std::string kAudioTrackId = "audio_track_0";
-  static constexpr int kOpusBitrateFbBps = 32000;
   static constexpr int kBitrateStabilizationMs = 10000;
   static constexpr int kBitrateMeasurements = 10;
   static constexpr int kBitrateMeasurementMs = 1000;
@@ -989,8 +989,12 @@
         }
         avg_rtt = avg_rtt / kBitrateMeasurements;
         if (avg_rtt > kMinGoodRttMs) {
+          RTC_LOG(LS_WARNING)
+              << "Failed test bitrate: " << test_bitrate << " RTT: " << avg_rtt;
           break;
         } else {
+          RTC_LOG(LS_INFO) << "Passed test bitrate: " << test_bitrate
+                           << " RTT: " << avg_rtt;
           last_passed_test_bitrate = test_bitrate;
         }
       }
@@ -1015,13 +1019,6 @@
 
     size_t GetNumAudioStreams() const override { return 1; }
 
-    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
-                            std::vector<AudioReceiveStreamInterface::Config>*
-                                receive_configs) override {
-      send_config->send_codec_spec->target_bitrate_bps =
-          absl::optional<int>(kOpusBitrateFbBps);
-    }
-
    private:
     const int test_bitrate_from_;
     const int test_bitrate_to_;
@@ -1039,13 +1036,7 @@
   RunBaseTest(&test);
 }
 
-// TODO(bugs.webrtc.org/8878)
-#if defined(WEBRTC_MAC)
-#define MAYBE_Min_Bitrate_VideoAndAudio DISABLED_Min_Bitrate_VideoAndAudio
-#else
-#define MAYBE_Min_Bitrate_VideoAndAudio Min_Bitrate_VideoAndAudio
-#endif
-TEST_F(CallPerfTest, MAYBE_Min_Bitrate_VideoAndAudio) {
+TEST_F(CallPerfTest, Min_Bitrate_VideoAndAudio) {
   TestMinAudioVideoBitrate(110, 40, -10, 10000, 70000, 200000);
 }
 

test/call_test.cc

diff --git a/test/call_test.cc b/test/call_test.cc
index 62d1839..7eccf78 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -292,6 +292,8 @@
 
     audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
         kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
+    audio_send_config.min_bitrate_bps = 6000;
+    audio_send_config.max_bitrate_bps = 60000;
     audio_send_config.encoder_factory = audio_encoder_factory_;
     SetAudioConfig(audio_send_config);
   }