Fix CallPerfTest tests
iSAC has been removed, the tests now use Opus which requires min/max bitrate to be set. Bug: webrtc:14450 Change-Id: I872764b1ebb9115e314f146749fe710a7665ad62 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284060 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38680}
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@ -267,8 +267,11 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
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AudioSendStream::Config audio_send_config(audio_send_transport.get());
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audio_send_config.rtp.ssrc = kAudioSendSsrc;
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// TODO(bugs.webrtc.org/14683): Let the tests fail with invalid config.
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audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
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kAudioSendPayloadType, {"ISAC", 16000, 1});
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kAudioSendPayloadType, {"OPUS", 48000, 2});
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audio_send_config.min_bitrate_bps = 6000;
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audio_send_config.max_bitrate_bps = 510000;
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audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
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audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config);
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@ -290,7 +293,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
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audio_recv_config.sync_group = kSyncGroup;
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audio_recv_config.decoder_factory = audio_decoder_factory_;
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audio_recv_config.decoder_map = {
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{kAudioSendPayloadType, {"ISAC", 16000, 1}}};
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{kAudioSendPayloadType, {"OPUS", 48000, 2}}};
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if (create_first == CreateOrder::kAudioFirst) {
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audio_receive_stream =
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