webrtc/api/test/loopback_media_transport_unittest.cc
Niels Möller 2e47f7c4ee Implement test class LoopbackMediaTransport
Bug: webrtc:9719
Change-Id: I82aa962d1cb8f2c8f56f766cb12562690e595045
Reviewed-on: https://webrtc-review.googlesource.com/c/105661
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25196}
2018-10-16 09:21:28 +00:00

60 lines
2 KiB
C++

/*
* Copyright 2018 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 <vector>
#include "api/test/loopback_media_transport.h"
#include "test/gmock.h"
namespace webrtc {
namespace {
class MockMediaTransportAudioSinkInterface
: public MediaTransportAudioSinkInterface {
public:
MOCK_METHOD2(OnData, void(uint64_t, MediaTransportEncodedAudioFrame));
};
// Test only uses the sequence number.
MediaTransportEncodedAudioFrame CreateAudioFrame(int sequence_number) {
static constexpr int kSamplingRateHz = 48000;
static constexpr int kStartingSampleIndex = 0;
static constexpr int kSamplesPerChannel = 480;
static constexpr uint8_t kPayloadType = 17;
return MediaTransportEncodedAudioFrame(
kSamplingRateHz, kStartingSampleIndex, kSamplesPerChannel,
sequence_number, MediaTransportEncodedAudioFrame::FrameType::kSpeech,
kPayloadType, std::vector<uint8_t>(kSamplesPerChannel));
}
} // namespace
TEST(LoopbackMediaTransport, AudioWithNoSinkSilentlyIgnored) {
MediaTransportPair transport_pair;
transport_pair.first()->SendAudioFrame(1, CreateAudioFrame(0));
transport_pair.second()->SendAudioFrame(2, CreateAudioFrame(0));
}
TEST(LoopbackMediaTransport, AudioDeliveredToSink) {
MediaTransportPair transport_pair;
testing::StrictMock<MockMediaTransportAudioSinkInterface> sink;
EXPECT_CALL(sink,
OnData(1, testing::Property(
&MediaTransportEncodedAudioFrame::sequence_number,
testing::Eq(10))));
transport_pair.second()->SetReceiveAudioSink(&sink);
transport_pair.first()->SendAudioFrame(1, CreateAudioFrame(10));
transport_pair.second()->SetReceiveAudioSink(nullptr);
}
} // namespace webrtc