mirror of
https://github.com/mollyim/webrtc.git
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e5447fb6d1
The fake rtp transport controller is only used by CallBitrateTest, but the functionality tested in CallBitrateTest is now tested in RtpBitrateConfiguratorTest. Removing the fake rtp transport controller send reduces the complexity of refactoring the rtp transport controller send interface. Bug: webrtc:8415 Change-Id: I4673daea4e68521e7e14293514830d6e704219bc Reviewed-on: https://webrtc-review.googlesource.com/54480 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22125}
281 lines
9 KiB
C++
281 lines
9 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 <list>
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#include <map>
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#include <memory>
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#include <utility>
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/test/mock_audio_mixer.h"
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#include "audio/audio_send_stream.h"
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#include "audio/audio_receive_stream.h"
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#include "call/audio_state.h"
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#include "call/call.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "modules/audio_device/include/mock_audio_device.h"
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#include "modules/audio_processing/include/mock_audio_processing.h"
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#include "modules/pacing/mock/mock_paced_sender.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "rtc_base/ptr_util.h"
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#include "test/fake_encoder.h"
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#include "test/gtest.h"
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#include "test/mock_audio_decoder_factory.h"
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#include "test/mock_transport.h"
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namespace {
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struct CallHelper {
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CallHelper() {
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webrtc::AudioState::Config audio_state_config;
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audio_state_config.audio_mixer =
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new rtc::RefCountedObject<webrtc::test::MockAudioMixer>();
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audio_state_config.audio_processing =
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new rtc::RefCountedObject<webrtc::test::MockAudioProcessing>();
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audio_state_config.audio_device_module =
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new rtc::RefCountedObject<webrtc::test::MockAudioDeviceModule>();
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webrtc::Call::Config config(&event_log_);
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config.audio_state = webrtc::AudioState::Create(audio_state_config);
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call_.reset(webrtc::Call::Create(config));
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}
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webrtc::Call* operator->() { return call_.get(); }
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private:
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webrtc::RtcEventLogNullImpl event_log_;
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std::unique_ptr<webrtc::Call> call_;
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};
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} // namespace
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namespace webrtc {
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TEST(CallTest, ConstructDestruct) {
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CallHelper call;
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}
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TEST(CallTest, CreateDestroy_AudioSendStream) {
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CallHelper call;
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AudioSendStream::Config config(nullptr);
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config.rtp.ssrc = 42;
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AudioSendStream* stream = call->CreateAudioSendStream(config);
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EXPECT_NE(stream, nullptr);
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call->DestroyAudioSendStream(stream);
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}
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TEST(CallTest, CreateDestroy_AudioReceiveStream) {
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CallHelper call;
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AudioReceiveStream::Config config;
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config.rtp.remote_ssrc = 42;
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config.decoder_factory =
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
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AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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call->DestroyAudioReceiveStream(stream);
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}
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TEST(CallTest, CreateDestroy_AudioSendStreams) {
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CallHelper call;
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AudioSendStream::Config config(nullptr);
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std::list<AudioSendStream*> streams;
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for (int i = 0; i < 2; ++i) {
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for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
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config.rtp.ssrc = ssrc;
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AudioSendStream* stream = call->CreateAudioSendStream(config);
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EXPECT_NE(stream, nullptr);
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if (ssrc & 1) {
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streams.push_back(stream);
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} else {
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streams.push_front(stream);
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}
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}
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for (auto s : streams) {
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call->DestroyAudioSendStream(s);
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}
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streams.clear();
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}
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}
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TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
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CallHelper call;
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AudioReceiveStream::Config config;
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config.decoder_factory =
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
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std::list<AudioReceiveStream*> streams;
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for (int i = 0; i < 2; ++i) {
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for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
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config.rtp.remote_ssrc = ssrc;
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AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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if (ssrc & 1) {
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streams.push_back(stream);
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} else {
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streams.push_front(stream);
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}
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}
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for (auto s : streams) {
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call->DestroyAudioReceiveStream(s);
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}
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streams.clear();
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}
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}
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TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
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CallHelper call;
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AudioReceiveStream::Config recv_config;
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recv_config.rtp.remote_ssrc = 42;
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recv_config.rtp.local_ssrc = 777;
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recv_config.decoder_factory =
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
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AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
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EXPECT_NE(recv_stream, nullptr);
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AudioSendStream::Config send_config(nullptr);
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send_config.rtp.ssrc = 777;
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AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
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EXPECT_NE(send_stream, nullptr);
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internal::AudioReceiveStream* internal_recv_stream =
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static_cast<internal::AudioReceiveStream*>(recv_stream);
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EXPECT_EQ(send_stream,
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internal_recv_stream->GetAssociatedSendStreamForTesting());
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call->DestroyAudioSendStream(send_stream);
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EXPECT_EQ(nullptr, internal_recv_stream->GetAssociatedSendStreamForTesting());
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call->DestroyAudioReceiveStream(recv_stream);
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}
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TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
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CallHelper call;
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AudioSendStream::Config send_config(nullptr);
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send_config.rtp.ssrc = 777;
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AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
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EXPECT_NE(send_stream, nullptr);
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AudioReceiveStream::Config recv_config;
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recv_config.rtp.remote_ssrc = 42;
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recv_config.rtp.local_ssrc = 777;
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recv_config.decoder_factory =
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
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AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
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EXPECT_NE(recv_stream, nullptr);
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internal::AudioReceiveStream* internal_recv_stream =
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static_cast<internal::AudioReceiveStream*>(recv_stream);
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EXPECT_EQ(send_stream,
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internal_recv_stream->GetAssociatedSendStreamForTesting());
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call->DestroyAudioReceiveStream(recv_stream);
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call->DestroyAudioSendStream(send_stream);
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}
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TEST(CallTest, CreateDestroy_FlexfecReceiveStream) {
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CallHelper call;
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MockTransport rtcp_send_transport;
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FlexfecReceiveStream::Config config(&rtcp_send_transport);
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config.payload_type = 118;
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config.remote_ssrc = 38837212;
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config.protected_media_ssrcs = {27273};
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FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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call->DestroyFlexfecReceiveStream(stream);
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}
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TEST(CallTest, CreateDestroy_FlexfecReceiveStreams) {
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CallHelper call;
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MockTransport rtcp_send_transport;
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FlexfecReceiveStream::Config config(&rtcp_send_transport);
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config.payload_type = 118;
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std::list<FlexfecReceiveStream*> streams;
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for (int i = 0; i < 2; ++i) {
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for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
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config.remote_ssrc = ssrc;
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config.protected_media_ssrcs = {ssrc + 1};
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FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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if (ssrc & 1) {
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streams.push_back(stream);
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} else {
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streams.push_front(stream);
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}
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}
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for (auto s : streams) {
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call->DestroyFlexfecReceiveStream(s);
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}
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streams.clear();
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}
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}
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TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) {
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CallHelper call;
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MockTransport rtcp_send_transport;
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FlexfecReceiveStream::Config config(&rtcp_send_transport);
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config.payload_type = 118;
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config.protected_media_ssrcs = {1324234};
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FlexfecReceiveStream* stream;
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std::list<FlexfecReceiveStream*> streams;
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config.remote_ssrc = 838383;
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stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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streams.push_back(stream);
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config.remote_ssrc = 424993;
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stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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streams.push_back(stream);
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config.remote_ssrc = 99383;
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stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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streams.push_back(stream);
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config.remote_ssrc = 5548;
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stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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streams.push_back(stream);
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for (auto s : streams) {
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call->DestroyFlexfecReceiveStream(s);
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}
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}
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TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
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constexpr uint32_t kSSRC = 12345;
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CallHelper call;
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auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
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AudioSendStream::Config config(nullptr);
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config.rtp.ssrc = ssrc;
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AudioSendStream* stream = call->CreateAudioSendStream(config);
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const RtpState rtp_state =
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static_cast<internal::AudioSendStream*>(stream)->GetRtpState();
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call->DestroyAudioSendStream(stream);
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return rtp_state;
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};
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const RtpState rtp_state1 = create_stream_and_get_rtp_state(kSSRC);
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const RtpState rtp_state2 = create_stream_and_get_rtp_state(kSSRC);
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EXPECT_EQ(rtp_state1.sequence_number, rtp_state2.sequence_number);
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EXPECT_EQ(rtp_state1.start_timestamp, rtp_state2.start_timestamp);
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EXPECT_EQ(rtp_state1.timestamp, rtp_state2.timestamp);
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EXPECT_EQ(rtp_state1.capture_time_ms, rtp_state2.capture_time_ms);
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EXPECT_EQ(rtp_state1.last_timestamp_time_ms,
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rtp_state2.last_timestamp_time_ms);
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EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
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}
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} // namespace webrtc
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