c7a8b08a7c
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1397123003 Cr-Commit-Position: refs/heads/master@{#10307}
35 lines
1.1 KiB
C++
35 lines
1.1 KiB
C++
/*
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* Copyright (c) 2015 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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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/audio/audio_send_stream.h"
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namespace webrtc {
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TEST(AudioSendStreamTest, ConfigToString) {
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const int kAbsSendTimeId = 3;
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AudioSendStream::Config config(nullptr);
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config.rtp.ssrc = 1234;
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config.rtp.extensions.push_back(
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RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
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config.voe_channel_id = 1;
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config.cng_payload_type = 42;
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config.red_payload_type = 17;
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EXPECT_GT(config.ToString().size(), 0u);
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}
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TEST(AudioSendStreamTest, ConstructDestruct) {
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AudioSendStream::Config config(nullptr);
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config.voe_channel_id = 1;
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internal::AudioSendStream send_stream(config);
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}
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} // namespace webrtc
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