Temporary switch back to generated audio to fix test flakes
Wav file capturer won't repeat file or produce silence after file end and WebRTC pipeline will crash in such case. In future we need to make it possible to continue audio after file was ended to behalf in the same way as video files capturer. Bug: webrtc:10138 Change-Id: I35f5bd33790cd430a56002a44af0abb894a96d29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132795 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27609}
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@ -85,9 +85,6 @@ TEST(PeerConnectionE2EQualityTestSmokeTest, RunWithEmulatedNetwork) {
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alice->AddVideoConfig(std::move(video_config));
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AudioConfig audio_config;
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audio_config.stream_label = "alice-audio";
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audio_config.mode = AudioConfig::Mode::kFile;
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audio_config.input_file_name = test::ResourcePath(
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"pc_quality_smoke_test_alice_source", "wav");
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alice->SetAudioConfig(std::move(audio_config));
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});
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@ -101,9 +98,6 @@ TEST(PeerConnectionE2EQualityTestSmokeTest, RunWithEmulatedNetwork) {
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bob->AddVideoConfig(std::move(video_config));
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AudioConfig audio_config;
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audio_config.stream_label = "bob-audio";
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audio_config.mode = AudioConfig::Mode::kFile;
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audio_config.input_file_name = test::ResourcePath(
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"pc_quality_smoke_test_bob_source", "wav");
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bob->SetAudioConfig(std::move(audio_config));
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});
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