9fea80f50d
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}
31 lines
978 B
Plaintext
31 lines
978 B
Plaintext
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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{
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'conditions': [
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['OS=="linux" or OS=="mac" or OS=="win" or OS=="android"', {
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'variables': {
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'files': [
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'<(DEPTH)/resources/foreman_cif_short.yuv',
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'<(DEPTH)/resources/voice_engine/audio_long16.pcm',
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],
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},
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}],
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['OS=="linux" or OS=="mac" or OS=="win"', {
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'variables': {
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'command': [
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'<(PRODUCT_DIR)/video_engine_tests<(EXECUTABLE_SUFFIX)',
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],
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'files': [
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'<(DEPTH)/DEPS',
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'<(PRODUCT_DIR)/video_engine_tests<(EXECUTABLE_SUFFIX)',
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],
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},
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}],
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],
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}
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