43e83d44f0
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.
Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1411083006
Cr-Commit-Position: refs/heads/master@{#10340}
105 lines
2.7 KiB
C++
105 lines
2.7 KiB
C++
/*
|
|
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include <list>
|
|
|
|
#include "testing/gtest/include/gtest/gtest.h"
|
|
|
|
#include "webrtc/call.h"
|
|
|
|
namespace {
|
|
|
|
struct CallHelper {
|
|
CallHelper() {
|
|
webrtc::Call::Config config;
|
|
// TODO(solenberg): Fill in with VoiceEngine* etc.
|
|
call_.reset(webrtc::Call::Create(config));
|
|
}
|
|
|
|
webrtc::Call* operator->() { return call_.get(); }
|
|
|
|
private:
|
|
rtc::scoped_ptr<webrtc::Call> call_;
|
|
};
|
|
} // namespace
|
|
|
|
namespace webrtc {
|
|
|
|
TEST(CallTest, ConstructDestruct) {
|
|
CallHelper call;
|
|
}
|
|
|
|
TEST(CallTest, CreateDestroy_AudioSendStream) {
|
|
CallHelper call;
|
|
AudioSendStream::Config config(nullptr);
|
|
config.rtp.ssrc = 42;
|
|
config.voe_channel_id = 123;
|
|
AudioSendStream* stream = call->CreateAudioSendStream(config);
|
|
EXPECT_NE(stream, nullptr);
|
|
call->DestroyAudioSendStream(stream);
|
|
}
|
|
|
|
TEST(CallTest, CreateDestroy_AudioReceiveStream) {
|
|
CallHelper call;
|
|
AudioReceiveStream::Config config;
|
|
config.rtp.remote_ssrc = 42;
|
|
config.voe_channel_id = 123;
|
|
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
|
|
EXPECT_NE(stream, nullptr);
|
|
call->DestroyAudioReceiveStream(stream);
|
|
}
|
|
|
|
TEST(CallTest, CreateDestroy_AudioSendStreams) {
|
|
CallHelper call;
|
|
AudioSendStream::Config config(nullptr);
|
|
config.voe_channel_id = 123;
|
|
std::list<AudioSendStream*> streams;
|
|
for (int i = 0; i < 2; ++i) {
|
|
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
|
|
config.rtp.ssrc = ssrc;
|
|
AudioSendStream* stream = call->CreateAudioSendStream(config);
|
|
EXPECT_NE(stream, nullptr);
|
|
if (ssrc & 1) {
|
|
streams.push_back(stream);
|
|
} else {
|
|
streams.push_front(stream);
|
|
}
|
|
}
|
|
for (auto s : streams) {
|
|
call->DestroyAudioSendStream(s);
|
|
}
|
|
streams.clear();
|
|
}
|
|
}
|
|
|
|
TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
|
|
CallHelper call;
|
|
AudioReceiveStream::Config config;
|
|
config.voe_channel_id = 123;
|
|
std::list<AudioReceiveStream*> streams;
|
|
for (int i = 0; i < 2; ++i) {
|
|
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
|
|
config.rtp.remote_ssrc = ssrc;
|
|
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
|
|
EXPECT_NE(stream, nullptr);
|
|
if (ssrc & 1) {
|
|
streams.push_back(stream);
|
|
} else {
|
|
streams.push_front(stream);
|
|
}
|
|
}
|
|
for (auto s : streams) {
|
|
call->DestroyAudioReceiveStream(s);
|
|
}
|
|
streams.clear();
|
|
}
|
|
}
|
|
} // namespace webrtc
|