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Add audio streams to CallTest and a first A/V call test.

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Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1542653002 .

Cr-Commit-Position: refs/heads/master@{#11171}
This commit is contained in:
Stefan Holmer
2016-01-07 17:43:18 +01:00
parent ecd21b481f
commit 9fea80f50d
9 changed files with 483 additions and 151 deletions
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6
webrtc/call/call_perf_tests.cc
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@ -284,7 +284,7 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
test::FakeDecoder fake_decoder;
CreateSendConfig(1, &sync_send_transport);
CreateSendConfig(1, 0, &sync_send_transport);
CreateMatchingReceiveConfigs(&sync_receive_transport);
AudioSendStream::Config audio_send_config(&audio_send_transport);
@ -318,9 +318,9 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
if (create_audio_first) {
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
CreateStreams();
CreateVideoStreams();
} else {
CreateStreams();
CreateVideoStreams();
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}