diff options
| author | Fredrik Solenberg <solenberg@webrtc.org> | 2018-01-11 13:52:30 +0100 |
|---|---|---|
| committer | Commit Bot <commit-bot@chromium.org> | 2018-01-11 12:58:31 +0000 |
| commit | 8f5787a9191224efbf1dd1dd33a3359d0eb6264d (patch) | |
| tree | 48c9da271a6052fad6c157ef316aa3fb0d99e936 | |
| parent | aa6b24f23c155122343cbf0de08ad44758be5f65 (diff) | |
| download | webrtc-8f5787a9191224efbf1dd1dd33a3359d0eb6264d.tar.gz | |
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
* VoEBase contains only stub methods (until downstream code is
updated).
* voe::Channel and ChannelProxy classes remain, but are now created
internally to the streams. As a result,
internal::Audio[Receive|Send]Stream can have a ChannelProxy injected
for testing.
* Stream classes share Call::module_process_thread_ for their RtpRtcp
modules, rather than using a separate thread shared only among audio
streams.
* voe::Channel instances use Call::worker_queue_ for encoding packets,
rather than having a separate queue for audio (send) streams.
Bug: webrtc:4690
Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d
Reviewed-on: https://webrtc-review.googlesource.com/34640
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21578}
51 files changed, 503 insertions, 2002 deletions
diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 0fd0ac8746..f7b1cb5869 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -25,7 +25,6 @@ rtc_static_library("audio") { "conversion.h", "null_audio_poller.cc", "null_audio_poller.h", - "scoped_voe_interface.h", "time_interval.cc", "time_interval.h", ] diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 3c53818060..bfa125a734 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -25,8 +25,6 @@ #include "rtc_base/logging.h" #include "rtc_base/timeutils.h" #include "voice_engine/channel_proxy.h" -#include "voice_engine/include/voe_base.h" -#include "voice_engine/voice_engine_impl.h" namespace webrtc { @@ -53,7 +51,6 @@ std::string AudioReceiveStream::Config::ToString() const { ss << "{rtp: " << rtp.ToString(); ss << ", rtcp_send_transport: " << (rtcp_send_transport ? "(Transport)" : "null"); - ss << ", voe_channel_id: " << voe_channel_id; if (!sync_group.empty()) { ss << ", sync_group: " << sync_group; } @@ -62,32 +59,58 @@ std::string AudioReceiveStream::Config::ToString() const { } namespace internal { +namespace { +std::unique_ptr<voe::ChannelProxy> CreateChannelAndProxy( + webrtc::AudioState* audio_state, + ProcessThread* module_process_thread, + const webrtc::AudioReceiveStream::Config& config) { + RTC_DCHECK(audio_state); + internal::AudioState* internal_audio_state = + static_cast<internal::AudioState*>(audio_state); + return std::unique_ptr<voe::ChannelProxy>(new voe::ChannelProxy( + std::unique_ptr<voe::Channel>(new voe::Channel( + module_process_thread, + internal_audio_state->audio_device_module(), + config.jitter_buffer_max_packets, + config.jitter_buffer_fast_accelerate, + config.decoder_factory)))); +} +} // namespace + AudioReceiveStream::AudioReceiveStream( RtpStreamReceiverControllerInterface* receiver_controller, PacketRouter* packet_router, + ProcessThread* module_process_thread, const webrtc::AudioReceiveStream::Config& config, const rtc::scoped_refptr<webrtc::AudioState>& audio_state, webrtc::RtcEventLog* event_log) - : audio_state_(audio_state) { + : AudioReceiveStream(receiver_controller, + packet_router, + config, + audio_state, + event_log, + CreateChannelAndProxy(audio_state.get(), + module_process_thread, + config)) {} + +AudioReceiveStream::AudioReceiveStream( + RtpStreamReceiverControllerInterface* receiver_controller, + PacketRouter* packet_router, + const webrtc::AudioReceiveStream::Config& config, + const rtc::scoped_refptr<webrtc::AudioState>& audio_state, + webrtc::RtcEventLog* event_log, + std::unique_ptr<voe::ChannelProxy> channel_proxy) + : audio_state_(audio_state), + channel_proxy_(std::move(channel_proxy)) { RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.ToString(); - RTC_DCHECK_NE(config.voe_channel_id, -1); - RTC_DCHECK(audio_state_.get()); + RTC_DCHECK(receiver_controller); RTC_DCHECK(packet_router); + RTC_DCHECK(config.decoder_factory); + RTC_DCHECK(audio_state_); + RTC_DCHECK(channel_proxy_); module_process_thread_checker_.DetachFromThread(); - VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); - channel_proxy_ = voe_impl->GetChannelProxy(config.voe_channel_id); - - // TODO(ossu): This is where we'd like to set the decoder factory to - // use. However, since it needs to be included when constructing Channel, we - // cannot do that until we're able to move Channel ownership into the - // Audio{Send,Receive}Streams. The best we can do is check that we're not - // trying to use two different factories using the different interfaces. - RTC_CHECK(config.decoder_factory); - RTC_CHECK_EQ(config.decoder_factory, - channel_proxy_->GetAudioDecoderFactory()); - channel_proxy_->SetRtcEventLog(event_log); channel_proxy_->RegisterTransport(config.rtcp_send_transport); @@ -204,9 +227,9 @@ int AudioReceiveStream::GetOutputLevel() const { return channel_proxy_->GetSpeechOutputLevel(); } -void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) { +void AudioReceiveStream::SetSink(AudioSinkInterface* sink) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); - channel_proxy_->SetSink(std::move(sink)); + channel_proxy_->SetSink(sink); } void AudioReceiveStream::SetGain(float gain) { @@ -278,13 +301,11 @@ void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) { void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); if (send_stream) { - VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); - std::unique_ptr<voe::ChannelProxy> send_channel_proxy = - voe_impl->GetChannelProxy(send_stream->GetConfig().voe_channel_id); - channel_proxy_->AssociateSendChannel(*send_channel_proxy.get()); + channel_proxy_->AssociateSendChannel(send_stream->GetChannelProxy()); } else { channel_proxy_->DisassociateSendChannel(); } + associated_send_stream_ = send_stream; } void AudioReceiveStream::SignalNetworkState(NetworkState state) { @@ -312,10 +333,10 @@ const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { return config_; } -VoiceEngine* AudioReceiveStream::voice_engine() const { - auto* voice_engine = audio_state()->voice_engine(); - RTC_DCHECK(voice_engine); - return voice_engine; +const AudioSendStream* + AudioReceiveStream::GetAssociatedSendStreamForTesting() const { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + return associated_send_stream_; } internal::AudioState* AudioReceiveStream::audio_state() const { @@ -346,8 +367,6 @@ void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream, // Configuration parameters which cannot be changed. RTC_DCHECK(first_time || - old_config.voe_channel_id == new_config.voe_channel_id); - RTC_DCHECK(first_time || old_config.rtp.remote_ssrc == new_config.rtp.remote_ssrc); RTC_DCHECK(first_time || old_config.rtcp_send_transport == new_config.rtcp_send_transport); diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h index 0c843bec58..999039f21a 100644 --- a/audio/audio_receive_stream.h +++ b/audio/audio_receive_stream.h @@ -24,6 +24,7 @@ namespace webrtc { class PacketRouter; +class ProcessThread; class RtcEventLog; class RtpPacketReceived; class RtpStreamReceiverControllerInterface; @@ -42,9 +43,17 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, public: AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller, PacketRouter* packet_router, + ProcessThread* module_process_thread, const webrtc::AudioReceiveStream::Config& config, const rtc::scoped_refptr<webrtc::AudioState>& audio_state, webrtc::RtcEventLog* event_log); + // For unit tests, which need to supply a mock channel proxy. + AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller, + PacketRouter* packet_router, + const webrtc::AudioReceiveStream::Config& config, + const rtc::scoped_refptr<webrtc::AudioState>& audio_state, + webrtc::RtcEventLog* event_log, + std::unique_ptr<voe::ChannelProxy> channel_proxy); ~AudioReceiveStream() override; // webrtc::AudioReceiveStream implementation. @@ -53,7 +62,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, void Stop() override; webrtc::AudioReceiveStream::Stats GetStats() const override; int GetOutputLevel() const override; - void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; + void SetSink(AudioSinkInterface* sink) override; void SetGain(float gain) override; std::vector<webrtc::RtpSource> GetSources() const override; @@ -79,6 +88,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, void SignalNetworkState(NetworkState state); bool DeliverRtcp(const uint8_t* packet, size_t length); const webrtc::AudioReceiveStream::Config& config() const; + const AudioSendStream* GetAssociatedSendStreamForTesting() const; private: // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is @@ -94,7 +104,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, const Config& new_config, bool first_time); - VoiceEngine* voice_engine() const; AudioState* audio_state() const; rtc::ThreadChecker worker_thread_checker_; @@ -102,6 +111,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, webrtc::AudioReceiveStream::Config config_; rtc::scoped_refptr<webrtc::AudioState> audio_state_; std::unique_ptr<voe::ChannelProxy> channel_proxy_; + AudioSendStream* associated_send_stream_ = nullptr; bool playing_ RTC_ACCESS_ON(worker_thread_checker_) = false; diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc index e03fbd5baf..062e71faf9 100644 --- a/audio/audio_receive_stream_unittest.cc +++ b/audio/audio_receive_stream_unittest.cc @@ -25,7 +25,6 @@ #include "test/gtest.h" #include "test/mock_audio_decoder_factory.h" #include "test/mock_voe_channel_proxy.h" -#include "test/mock_voice_engine.h" namespace webrtc { namespace test { @@ -48,8 +47,6 @@ AudioDecodingCallStats MakeAudioDecodeStatsForTest() { return audio_decode_stats; } -const int kChannelId1 = 2; -const int kChannelId2 = 29; const uint32_t kRemoteSsrc = 1234; const uint32_t kLocalSsrc = 5678; const size_t kOneByteExtensionHeaderLength = 4; @@ -73,61 +70,47 @@ const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest(); struct ConfigHelper { ConfigHelper() - : decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>), - audio_mixer_(new rtc::RefCountedObject<MockAudioMixer>()) { + : ConfigHelper(new rtc::RefCountedObject<MockAudioMixer>()) {} + + explicit ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer) + : audio_mixer_(audio_mixer) { using testing::Invoke; AudioState::Config config; - config.voice_engine = &voice_engine_; config.audio_mixer = audio_mixer_; config.audio_processing = new rtc::RefCountedObject<MockAudioProcessing>(); config.audio_device_module = - new rtc::RefCountedObject<MockAudioDeviceModule>(); + new rtc::RefCountedObject<testing::NiceMock<MockAudioDeviceModule>>(); audio_state_ = AudioState::Create(config); - EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId1)) - .WillOnce(Invoke([this](int channel_id) { - EXPECT_FALSE(channel_proxy_); - channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); - EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1); - EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1); - EXPECT_CALL(*channel_proxy_, - SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId)) - .Times(1); - EXPECT_CALL(*channel_proxy_, - EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId)) - .Times(1); - EXPECT_CALL(*channel_proxy_, - RegisterReceiverCongestionControlObjects(&packet_router_)) - .Times(1); - EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects()) - .Times(1); - EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2); - EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory()) - .WillOnce(ReturnRef(decoder_factory_)); - testing::Expectation expect_set = - EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_)) - .Times(1); - EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) - .Times(1) - .After(expect_set); - EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1); - EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_)) - .WillRepeatedly( - Invoke([](const std::map<int, SdpAudioFormat>& codecs) { - EXPECT_THAT(codecs, testing::IsEmpty()); - })); - return channel_proxy_; - })); - EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId2)) - .WillRepeatedly(Invoke([this](int channel_id) { - testing::NiceMock<MockVoEChannelProxy>* proxy = - new testing::NiceMock<MockVoEChannelProxy>(); - EXPECT_CALL(*proxy, GetAudioDecoderFactory()) - .WillOnce(ReturnRef(decoder_factory_)); - return proxy; - })); - stream_config_.voe_channel_id = kChannelId1; + channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); + EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1); + EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1); + EXPECT_CALL(*channel_proxy_, + SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId)) + .Times(1); + EXPECT_CALL(*channel_proxy_, + EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId)) + .Times(1); + EXPECT_CALL(*channel_proxy_, + RegisterReceiverCongestionControlObjects(&packet_router_)) + .Times(1); + EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects()) + .Times(1); + EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2); + testing::Expectation expect_set = + EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_)) + .Times(1); + EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) + .Times(1) + .After(expect_set); + EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1); + EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_)) + .WillRepeatedly( + Invoke([](const std::map<int, SdpAudioFormat>& codecs) { + EXPECT_THAT(codecs, testing::IsEmpty()); + })); + stream_config_.rtp.local_ssrc = kLocalSsrc; stream_config_.rtp.remote_ssrc = kRemoteSsrc; stream_config_.rtp.nack.rtp_history_ms = 300; @@ -135,19 +118,24 @@ struct ConfigHelper { RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); stream_config_.rtp.extensions.push_back(RtpExtension( RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); - stream_config_.decoder_factory = decoder_factory_; + stream_config_.decoder_factory = + new rtc::RefCountedObject<MockAudioDecoderFactory>; + } + + std::unique_ptr<internal::AudioReceiveStream> CreateAudioReceiveStream() { + return std::unique_ptr<internal::AudioReceiveStream>( + new internal::AudioReceiveStream( + &rtp_stream_receiver_controller_, + &packet_router_, + stream_config_, + audio_state_, + &event_log_, + std::unique_ptr<voe::ChannelProxy>(channel_proxy_))); } - PacketRouter* packet_router() { return &packet_router_; } - MockRtcEventLog* event_log() { return &event_log_; } AudioReceiveStream::Config& config() { return stream_config_; } - rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; } - MockVoiceEngine& voice_engine() { return voice_engine_; } MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } - RtpStreamReceiverControllerInterface* rtp_stream_receiver_controller() { - return &rtp_stream_receiver_controller_; - } void SetupMockForGetStats() { using testing::DoAll; @@ -174,9 +162,7 @@ struct ConfigHelper { private: PacketRouter packet_router_; - rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; MockRtcEventLog event_log_; - testing::StrictMock<MockVoiceEngine> voice_engine_; rtc::scoped_refptr<AudioState> audio_state_; rtc::scoped_refptr<MockAudioMixer> audio_mixer_; AudioReceiveStream::Config stream_config_; @@ -240,32 +226,25 @@ TEST(AudioReceiveStreamTest, ConfigToString) { AudioReceiveStream::Config config; config.rtp.remote_ssrc = kRemoteSsrc; config.rtp.local_ssrc = kLocalSsrc; - config.voe_channel_id = kChannelId1; config.rtp.extensions.push_back( RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); EXPECT_EQ( "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: " "{rtp_history_ms: 0}, extensions: [{uri: " "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, " - "rtcp_send_transport: null, voe_channel_id: 2}", + "rtcp_send_transport: null}", config.ToString()); } TEST(AudioReceiveStreamTest, ConstructDestruct) { ConfigHelper helper; - internal::AudioReceiveStream recv_stream( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - helper.config(), helper.audio_state(), helper.event_log()); + auto recv_stream = helper.CreateAudioReceiveStream(); } TEST(AudioReceiveStreamTest, ReceiveRtpPacket) { ConfigHelper helper; helper.config().rtp.transport_cc = true; - internal::AudioReceiveStream recv_stream( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - helper.config(), helper.audio_state(), helper.event_log()); + auto recv_stream = helper.CreateAudioReceiveStream(); const int kTransportSequenceNumberValue = 1234; std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( kTransportSequenceNumberId, kTransportSequenceNumberValue, 2); @@ -278,32 +257,25 @@ TEST(AudioReceiveStreamTest, ReceiveRtpPacket) { EXPECT_CALL(*helper.channel_proxy(), OnRtpPacket(testing::Ref(parsed_packet))); - recv_stream.OnRtpPacket(parsed_packet); + recv_stream->OnRtpPacket(parsed_packet); } TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) { ConfigHelper helper; helper.config().rtp.transport_cc = true; - internal::AudioReceiveStream recv_stream( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - helper.config(), helper.audio_state(), helper.event_log()); - + auto recv_stream = helper.CreateAudioReceiveStream(); std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport(); EXPECT_CALL(*helper.channel_proxy(), ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size())) .WillOnce(Return(true)); - EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size())); + EXPECT_TRUE(recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size())); } TEST(AudioReceiveStreamTest, GetStats) { ConfigHelper helper; - internal::AudioReceiveStream recv_stream( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - helper.config(), helper.audio_state(), helper.event_log()); + auto recv_stream = helper.CreateAudioReceiveStream(); helper.SetupMockForGetStats(); - AudioReceiveStream::Stats stats = recv_stream.GetStats(); + AudioReceiveStream::Stats stats = recv_stream->GetStats(); EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc); EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd); EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived), @@ -354,62 +326,50 @@ TEST(AudioReceiveStreamTest, GetStats) { TEST(AudioReceiveStreamTest, SetGain) { ConfigHelper helper; - internal::AudioReceiveStream recv_stream( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - helper.config(), helper.audio_state(), helper.event_log()); + auto recv_stream = helper.CreateAudioReceiveStream(); EXPECT_CALL(*helper.channel_proxy(), SetChannelOutputVolumeScaling(FloatEq(0.765f))); - recv_stream.SetGain(0.765f); + recv_stream->SetGain(0.765f); } TEST(AudioReceiveStreamTest, StreamsShouldBeAddedToMixerOnceOnStart) { - ConfigHelper helper; - internal::AudioReceiveStream recv_stream1( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - helper.config(), helper.audio_state(), helper.event_log()); - AudioReceiveStream::Config config2 = helper.config(); - config2.voe_channel_id = kChannelId2; - internal::AudioReceiveStream recv_stream2( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - config2, helper.audio_state(), helper.event_log()); - - EXPECT_CALL(*helper.channel_proxy(), StartPlayout()).Times(1); - EXPECT_CALL(*helper.channel_proxy(), StopPlayout()).Times(1); - EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream1)) + ConfigHelper helper1; + ConfigHelper helper2(helper1.audio_mixer()); + auto recv_stream1 = helper1.CreateAudioReceiveStream(); + auto recv_stream2 = helper2.CreateAudioReceiveStream(); + + EXPECT_CALL(*helper1.channel_proxy(), StartPlayout()).Times(1); + EXPECT_CALL(*helper2.channel_proxy(), StartPlayout()).Times(1); + EXPECT_CALL(*helper1.channel_proxy(), StopPlayout()).Times(1); + EXPECT_CALL(*helper2.channel_proxy(), StopPlayout()).Times(1); + EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream1.get())) .WillOnce(Return(true)); - EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream2)) + EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream2.get())) .WillOnce(Return(true)); - EXPECT_CALL(*helper.audio_mixer(), RemoveSource(&recv_stream1)).Times(1); - EXPECT_CALL(*helper.audio_mixer(), RemoveSource(&recv_stream2)).Times(1); + EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream1.get())) + .Times(1); + EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream2.get())) + .Times(1); - recv_stream1.Start(); - recv_stream2.Start(); + recv_stream1->Start(); + recv_stream2->Start(); // One more should not result in any more mixer sources added. - recv_stream1.Start(); + recv_stream1->Start(); // Stop stream before it is being destructed. - recv_stream2.Stop(); + recv_stream2->Stop(); } TEST(AudioReceiveStreamTest, ReconfigureWithSameConfig) { ConfigHelper helper; - internal::AudioReceiveStream recv_stream( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - helper.config(), helper.audio_state(), helper.event_log()); - recv_stream.Reconfigure(helper.config()); + auto recv_stream = helper.CreateAudioReceiveStream(); + recv_stream->Reconfigure(helper.config()); } TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) { ConfigHelper helper; - internal::AudioReceiveStream recv_stream( - helper.rtp_stream_receiver_controller(), - helper.packet_router(), - helper.config(), helper.audio_state(), helper.event_log()); + auto recv_stream = helper.CreateAudioReceiveStream(); auto new_config = helper.config(); new_config.rtp.local_ssrc = kLocalSsrc + 1; @@ -433,7 +393,7 @@ TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) { EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId + 1)) .Times(1); - recv_stream.Reconfigure(new_config); + recv_stream->Reconfigure(new_config); } } // namespace test } // namespace webrtc diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index 2e5654a43f..d207dbd578 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -29,17 +29,15 @@ #include "rtc_base/timeutils.h" #include "system_wrappers/include/field_trial.h" #include "voice_engine/channel_proxy.h" -#include "voice_engine/voice_engine_impl.h" namespace webrtc { - namespace internal { +namespace { // TODO(eladalon): Subsequent CL will make these values experiment-dependent. constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000; constexpr size_t kPacketLossRateMinNumAckedPackets = 50; constexpr size_t kRecoverablePacketLossRateMinNumAckedPairs = 40; -namespace { void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy, rtc::FunctionView<void(AudioEncoder*)> lambda) { channel_proxy->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) { @@ -47,6 +45,20 @@ void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy, lambda(encoder_ptr->get()); }); } + +std::unique_ptr<voe::ChannelProxy> CreateChannelAndProxy( + webrtc::AudioState* audio_state, + rtc::TaskQueue* worker_queue, + ProcessThread* module_process_thread) { + RTC_DCHECK(audio_state); + internal::AudioState* internal_audio_state = + static_cast<internal::AudioState*>(audio_state); + return std::unique_ptr<voe::ChannelProxy>(new voe::ChannelProxy( + std::unique_ptr<voe::Channel>(new voe::Channel( + worker_queue, + module_process_thread, + internal_audio_state->audio_device_module())))); +} } // namespace // TODO(saza): Move this declaration further down when we can use @@ -77,14 +89,38 @@ AudioSendStream::AudioSendStream( const webrtc::AudioSendStream::Config& config, const rtc::scoped_refptr<webrtc::AudioState>& audio_state, rtc::TaskQueue* worker_queue, + ProcessThread* module_process_thread, RtpTransportControllerSendInterface* transport, BitrateAllocator* bitrate_allocator, RtcEventLog* event_log, RtcpRttStats* rtcp_rtt_stats, const rtc::Optional<RtpState>& suspended_rtp_state) + : AudioSendStream(config, + audio_state, + worker_queue, + transport, + bitrate_allocator, + event_log, + rtcp_rtt_stats, + suspended_rtp_state, + CreateChannelAndProxy(audio_state.get(), + worker_queue, + module_process_thread)) {} + +AudioSendStream::AudioSendStream( + const webrtc::AudioSendStream::Config& config, + const rtc::scoped_refptr<webrtc::AudioState>& audio_state, + rtc::TaskQueue* worker_queue, + RtpTransportControllerSendInterface* transport, + BitrateAllocator* bitrate_allocator, + RtcEventLog* event_log, + RtcpRttStats* rtcp_rtt_stats, + const rtc::Optional<RtpState>& suspended_rtp_state, + std::unique_ptr<voe::ChannelProxy> channel_proxy) : worker_queue_(worker_queue), config_(Config(nullptr)), audio_state_(audio_state), + channel_proxy_(std::move(channel_proxy)), event_log_(event_log), bitrate_allocator_(bitrate_allocator), transport_(transport), @@ -94,13 +130,13 @@ AudioSendStream::AudioSendStream( rtp_rtcp_module_(nullptr), suspended_rtp_state_(suspended_rtp_state) { RTC_LOG(LS_INFO) << "AudioSendStream: " << config.ToString(); - RTC_DCHECK_NE(config.voe_channel_id, -1); - RTC_DCHECK(audio_state_.get()); + RTC_DCHECK(worker_queue_); + RTC_DCHECK(audio_state_); + RTC_DCHECK(channel_proxy_); + RTC_DCHECK(bitrate_allocator_); RTC_DCHECK(transport); RTC_DCHECK(transport->send_side_cc()); - VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); - channel_proxy_ = voe_impl->GetChannelProxy(config.voe_channel_id); channel_proxy_->SetRtcEventLog(event_log_); channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats); channel_proxy_->SetRTCPStatus(true); @@ -423,6 +459,11 @@ const TimeInterval& AudioSendStream::GetActiveLifetime() const { return active_lifetime_; } +const voe::ChannelProxy& AudioSendStream::GetChannelProxy() const { + RTC_DCHECK(channel_proxy_.get()); + return *channel_proxy_.get(); +} + internal::AudioState* AudioSendStream::audio_state() { internal::AudioState* audio_state = static_cast<internal::AudioState*>(audio_state_.get()); @@ -437,14 +478,6 @@ const internal::AudioState* AudioSendStream::audio_state() const { return audio_state; } -VoiceEngine* AudioSendStream::voice_engine() const { - internal::AudioState* audio_state = - static_cast<internal::AudioState*>(audio_state_.get()); - VoiceEngine* voice_engine = audio_state->voice_engine(); - RTC_DCHECK(voice_engine); - return voice_engine; -} - void AudioSendStream::StoreEncoderProperties(int sample_rate_hz, size_t num_channels) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h index 9060e64584..bab8f80de5 100644 --- a/audio/audio_send_stream.h +++ b/audio/audio_send_stream.h @@ -25,7 +25,6 @@ #include "voice_engine/transport_feedback_packet_loss_tracker.h" namespace webrtc { -class VoiceEngine; class RtcEventLog; class RtcpBandwidthObserver; class RtcpRttStats; @@ -45,11 +44,22 @@ class AudioSendStream final : public webrtc::AudioSendStream, AudioSendStream(const webrtc::AudioSendStream::Config& config, const rtc::scoped_refptr<webrtc::AudioState>& audio_state, rtc::TaskQueue* worker_queue, + ProcessThread* module_process_thread, RtpTransportControllerSendInterface* transport, BitrateAllocator* bitrate_allocator, RtcEventLog* event_log, RtcpRttStats* rtcp_rtt_stats, const rtc::Optional<RtpState>& suspended_rtp_state); + // For unit tests, which need to supply a mock channel proxy. + AudioSendStream(const webrtc::AudioSendStream::Config& config, + const rtc::scoped_refptr<webrtc::AudioState>& audio_state, + rtc::TaskQueue* worker_queue, + RtpTransportControllerSendInterface* transport, + BitrateAllocator* bitrate_allocator, + RtcEventLog* event_log, + RtcpRttStats* rtcp_rtt_stats, + const rtc::Optional<RtpState>& suspended_rtp_state, + std::unique_ptr<voe::ChannelProxy> channel_proxy); ~AudioSendStream() override; // webrtc::AudioSendStream implementation. @@ -83,13 +93,13 @@ class AudioSendStream final : public webrtc::AudioSendStream, RtpState GetRtpState() const; const TimeInterval& GetActiveLifetime() const; + const voe::ChannelProxy& GetChannelProxy() const; private: class TimedTransport; internal::AudioState* audio_state(); const internal::AudioState* audio_state() const; - VoiceEngine* voice_engine() const; void StoreEncoderProperties(int sample_rate_hz, size_t num_channels); diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc index 5afdffe67b..4644bf471e 100644 --- a/audio/audio_send_stream_unittest.cc +++ b/audio/audio_send_stream_unittest.cc @@ -33,7 +33,6 @@ #include "test/mock_audio_encoder.h" #include "test/mock_audio_encoder_factory.h" #include "test/mock_voe_channel_proxy.h" -#include "test/mock_voice_engine.h" namespace webrtc { namespace test { @@ -46,7 +45,6 @@ using testing::Invoke; using testing::Return; using testing::StrEq; -const int kChannelId = 1; const uint32_t kSsrc = 1234; const char* kCName = "foo_name"; const int kAudioLevelId = 2; @@ -141,7 +139,6 @@ struct ConfigHelper { using testing::Invoke; AudioState::Config config; - config.voice_engine = &voice_engine_; config.audio_mixer = AudioMixerImpl::Create(); config.audio_processing = audio_processing_; config.audio_device_module = @@ -149,19 +146,12 @@ struct ConfigHelper { audio_state_ = AudioState::Create(config); SetupDefaultChannelProxy(audio_bwe_enabled); - - EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) - .WillOnce(Invoke([this](int channel_id) { - return channel_proxy_; - })); - SetupMockForSetupSendCodec(expect_set_encoder_call); - // Use ISAC as default codec so as to prevent unnecessary |voice_engine_| + // Use ISAC as default codec so as to prevent unnecessary |channel_proxy_| // calls from the default ctor behavior. stream_config_.send_codec_spec = AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat); - stream_config_.voe_channel_id = kChannelId; stream_config_.rtp.ssrc = kSsrc; stream_config_.rtp.nack.rtp_history_ms = 200; stream_config_.rtp.c_name = kCName; @@ -175,18 +165,27 @@ struct ConfigHelper { stream_config_.max_bitrate_bps = 65000; } + std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() { + return std::unique_ptr<internal::AudioSendStream>( + new internal::AudioSendStream( + stream_config_, + audio_state_, + &worker_queue_, + &fake_transport_, + &bitrate_allocator_, + &event_log_, + &rtcp_rtt_stats_, + rtc::nullopt, + std::unique_ptr<voe::ChannelProxy>(channel_proxy_))); + } + AudioSendStream::Config& config() { return stream_config_; } MockAudioEncoderFactory& mock_encoder_factory() { return *static_cast<MockAudioEncoderFactory*>( stream_config_.encoder_factory.get()); } - rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } RtpTransportControllerSendInterface* transport() { return &fake_transport_; } - BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; } - rtc::TaskQueue* worker_queue() { return &worker_queue_; } - RtcEventLog* event_log() { return &event_log_; } - MockVoiceEngine* voice_engine() { return &voice_engine_; } static void AddBweToConfig(AudioSendStream::Config* config) { config->rtp.extensions.push_back( @@ -255,8 +254,6 @@ struct ConfigHelper { })); } - RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; } - void SetupMockForSendTelephoneEvent() { EXPECT_TRUE(channel_proxy_); EXPECT_CALL(*channel_proxy_, @@ -305,7 +302,6 @@ struct ConfigHelper { } private: - testing::StrictMock<MockVoiceEngine> voice_engine_; rtc::scoped_refptr<AudioState> audio_state_; AudioSendStream::Config stream_config_; testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; @@ -332,7 +328,6 @@ TEST(AudioSendStreamTest, ConfigToString) { AudioSendStream::Config config(nullptr); config.rtp.ssrc = kSsrc; config.rtp.c_name = kCName; - config.voe_channel_id = kChannelId; config.min_bitrate_bps = 12000; config.max_bitrate_bps = 34000; config.send_codec_spec = @@ -347,7 +342,7 @@ TEST(AudioSendStreamTest, ConfigToString) { "{rtp: {ssrc: 1234, extensions: [{uri: " "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: " "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: null, " - "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, " + "min_bitrate_bps: 12000, max_bitrate_bps: 34000, " "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " "cng_payload_type: 42, payload_type: 103, " "format: {name: isac, clockrate_hz: 16000, num_channels: 1, " @@ -357,58 +352,40 @@ TEST(AudioSendStreamTest, ConfigToString) { TEST(AudioSendStreamTest, ConstructDestruct) { ConfigHelper helper(false, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); } TEST(AudioSendStreamTest, SendTelephoneEvent) { ConfigHelper helper(false, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); helper.SetupMockForSendTelephoneEvent(); - EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType, + EXPECT_TRUE(send_stream->SendTelephoneEvent(kTelephoneEventPayloadType, kTelephoneEventPayloadFrequency, kTelephoneEventCode, kTelephoneEventDuration)); } TEST(AudioSendStreamTest, SetMuted) { ConfigHelper helper(false, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true)); - send_stream.SetMuted(true); + send_stream->SetMuted(true); } TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) { ConfigHelper helper(true, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); } TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) { ConfigHelper helper(false, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); } TEST(AudioSendStreamTest, GetStats) { ConfigHelper helper(false, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); helper.SetupMockForGetStats(); - AudioSendStream::Stats stats = send_stream.GetStats(true); + AudioSendStream::Stats stats = send_stream->GetStats(true); EXPECT_EQ(kSsrc, stats.local_ssrc); EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent); EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent); @@ -441,13 +418,12 @@ TEST(AudioSendStreamTest, GetStats) { TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) { ConfigHelper helper(false, true); - auto stream_config = helper.config(); - stream_config.send_codec_spec = + helper.config().send_codec_spec = AudioSendStream::Config::SendCodecSpec(0, kOpusFormat); const std::string kAnaConfigString = "abcde"; const std::string kAnaReconfigString = "12345"; - stream_config.audio_network_adaptor_config = kAnaConfigString; + helper.config().audio_network_adaptor_config = kAnaConfigString; EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _)) .WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString]( @@ -463,25 +439,22 @@ TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) { *return_value = std::move(mock_encoder); })); - internal::AudioSendStream send_stream( - stream_config, helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); + auto stream_config = helper.config(); stream_config.audio_network_adaptor_config = kAnaReconfigString; helper.SetupMockForModifyEncoder(); - send_stream.Reconfigure(stream_config); + send_stream->Reconfigure(stream_config); } // VAD is applied when codec is mono and the CNG frequency matches the codec // clock rate. TEST(AudioSendStreamTest, SendCodecCanApplyVad) { ConfigHelper helper(false, false); - auto stream_config = helper.config(); - stream_config.send_codec_spec = + helper.config().send_codec_spec = AudioSendStream::Config::SendCodecSpec(9, kG722Format); - stream_config.send_codec_spec->cng_payload_type = 105; + helper.config().send_codec_spec->cng_payload_type = 105; using ::testing::Invoke; std::unique_ptr<AudioEncoder> stolen_encoder; EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _)) @@ -492,10 +465,7 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) { return true; })); - internal::AudioSendStream send_stream( - stream_config, helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); // We cannot truly determine if the encoder created is an AudioEncoderCng. It // is the only reasonable implementation that will return something from @@ -506,24 +476,18 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) { TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { ConfigHelper helper(false, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); EXPECT_CALL(*helper.channel_proxy(), SetBitrate(helper.config().max_bitrate_bps, _)); - send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50, - 6000); + send_stream->OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50, + 6000); } TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) { ConfigHelper helper(false, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000)); - send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000); + send_stream->OnBitrateUpdated(50000, 0.0, 50, 5000); } // Test that AudioSendStream doesn't recreate the encoder unnecessarily. @@ -537,23 +501,16 @@ TEST(AudioSendStreamTest, DontRecreateEncoder) { EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _)) .WillOnce(Return(true)); - auto stream_config = helper.config(); - stream_config.send_codec_spec = + helper.config().send_codec_spec = AudioSendStream::Config::SendCodecSpec(9, kG722Format); - stream_config.send_codec_spec->cng_payload_type = 105; - internal::AudioSendStream send_stream( - stream_config, helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); - send_stream.Reconfigure(stream_config); + helper.config().send_codec_spec->cng_payload_type = 105; + auto send_stream = helper.CreateAudioSendStream(); + send_stream->Reconfigure(helper.config()); } TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) { ConfigHelper helper(false, true); - internal::AudioSendStream send_stream( - helper.config(), helper.audio_state(), helper.worker_queue(), - helper.transport(), helper.bitrate_allocator(), helper.event_log(), - helper.rtcp_rtt_stats(), rtc::nullopt); + auto send_stream = helper.CreateAudioSendStream(); auto new_config = helper.config(); ConfigHelper::AddBweToConfig(&new_config); EXPECT_CALL(*helper.channel_proxy(), @@ -567,7 +524,7 @@ TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) { helper.transport(), Ne(nullptr))) .Times(1); } - send_stream.Reconfigure(new_config); + send_stream->Reconfigure(new_config); } } // namespace test } // namespace webrtc diff --git a/audio/audio_state.cc b/audio/audio_state.cc index 9d980e8a58..24a7ce801e 100644 --- a/audio/audio_state.cc +++ b/audio/audio_state.cc @@ -27,7 +27,6 @@ namespace internal { AudioState::AudioState(const AudioState::Config& config) : config_(config), - voe_base_(config.voice_engine), audio_transport_(config_.audio_mixer, config_.audio_processing.get(), config_.audio_device_module.get()) { @@ -42,11 +41,6 @@ AudioState::~AudioState() { RTC_DCHECK(sending_streams_.empty()); } -VoiceEngine* AudioState::voice_engine() { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - return config_.voice_engine; -} - bool AudioState::typing_noise_detected() const { RTC_DCHECK(thread_checker_.CalledOnValidThread()); return audio_transport_.typing_noise_detected(); diff --git a/audio/audio_state.h b/audio/audio_state.h index 3a92d42b42..d4e4e3fd3c 100644 --- a/audio/audio_state.h +++ b/audio/audio_state.h @@ -17,13 +17,11 @@ #include "audio/audio_transport_impl.h" #include "audio/null_audio_poller.h" -#include "audio/scoped_voe_interface.h" #include "call/audio_state.h" #include "rtc_base/constructormagic.h" #include "rtc_base/criticalsection.h" #include "rtc_base/refcount.h" #include "rtc_base/thread_checker.h" -#include "voice_engine/include/voe_base.h" namespace webrtc { @@ -51,7 +49,11 @@ class AudioState final : public webrtc::AudioState { Stats GetAudioInputStats() const override; void SetStereoChannelSwapping(bool enable) override; - VoiceEngine* voice_engine(); + AudioDeviceModule* audio_device_module() { + RTC_DCHECK(config_.audio_device_module); + return config_.audio_device_module.get(); + } + bool typing_noise_detected() const; void AddReceivingStream(webrtc::AudioReceiveStream* stream); @@ -74,9 +76,6 @@ class AudioState final : public webrtc::AudioState { bool recording_enabled_ = true; bool playout_enabled_ = true; - // We hold one interface pointer to the VoE to make sure it is kept alive. - ScopedVoEInterface<VoEBase> voe_base_; - // Reference count; implementation copied from rtc::RefCountedObject. // TODO(nisse): Use RefCountedObject or RefCountedBase instead. mutable volatile int ref_count_ = 0; diff --git a/audio/audio_state_unittest.cc b/audio/audio_state_unittest.cc index abae9857b7..b4452add9a 100644 --- a/audio/audio_state_unittest.cc +++ b/audio/audio_state_unittest.cc @@ -16,8 +16,8 @@ #include "modules/audio_device/include/mock_audio_device.h" #include "modules/audio_mixer/audio_mixer_impl.h" #include "modules/audio_processing/include/mock_audio_processing.h" +#include "rtc_base/refcountedobject.h" #include "test/gtest.h" -#include "test/mock_voice_engine.h" namespace webrtc { namespace test { @@ -28,7 +28,6 @@ constexpr int kNumberOfChannels = 1; struct ConfigHelper { ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) { - audio_state_config.voice_engine = &mock_voice_engine; audio_state_config.audio_mixer = audio_mixer; audio_state_config.audio_processing = new rtc::RefCountedObject<testing::NiceMock<MockAudioProcessing>>(); @@ -36,11 +35,9 @@ struct ConfigHelper { new rtc::RefCountedObject<MockAudioDeviceModule>(); } AudioState::Config& config() { return audio_state_config; } - MockVoiceEngine& voice_engine() { return mock_voice_engine; } rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; } private: - testing::StrictMock<MockVoiceEngine> mock_voice_engine; AudioState::Config audio_state_config; rtc::scoped_refptr<AudioMixer> audio_mixer; }; @@ -106,13 +103,6 @@ TEST(AudioStateTest, ConstructDestruct) { new internal::AudioState(helper.config())); } -TEST(AudioStateTest, GetVoiceEngine) { - ConfigHelper helper; - std::unique_ptr<internal::AudioState> audio_state( - new internal::AudioState(helper.config())); - EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine()); -} - TEST(AudioStateTest, RecordedAudioArrivesAtSingleStream) { ConfigHelper helper; std::unique_ptr<internal::AudioState> audio_state( diff --git a/audio/scoped_voe_interface.h b/audio/scoped_voe_interface.h deleted file mode 100644 index 7aa2d1dacf..0000000000 --- a/audio/scoped_voe_interface.h +++ /dev/null @@ -1,45 +0,0 @@ -/* - * 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. - */ - -#ifndef AUDIO_SCOPED_VOE_INTERFACE_H_ -#define AUDIO_SCOPED_VOE_INTERFACE_H_ - -#include "rtc_base/checks.h" - -namespace webrtc { - -class VoiceEngine; - -namespace internal { - -// Utility template for obtaining and holding a reference to a VoiceEngine -// interface and making sure it is released when this object goes out of scope. -template<class T> class ScopedVoEInterface { - public: - explicit ScopedVoEInterface(webrtc::VoiceEngine* e) - : ptr_(T::GetInterface(e)) { - RTC_DCHECK(ptr_); - } - ~ScopedVoEInterface() { - if (ptr_) { - ptr_->Release(); - } - } - T* operator->() { - RTC_DCHECK(ptr_); - return ptr_; - } - private: - T* ptr_; -}; -} // namespace internal -} // namespace webrtc - -#endif // AUDIO_SCOPED_VOE_INTERFACE_H_ diff --git a/call/BUILD.gn b/call/BUILD.gn index 90460e7670..94b0f15d15 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -208,6 +208,7 @@ if (rtc_include_tests) { "../api:libjingle_peerconnection_api", "../api:mock_audio_mixer", "../api/audio_codecs:builtin_audio_decoder_factory", + "../audio:audio", "../logging:rtc_event_log_api", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer", diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index 505fd0851b..f8d787cb50 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h @@ -29,11 +29,6 @@ namespace webrtc { class AudioSinkInterface; -// WORK IN PROGRESS -// This class is under development and is not yet intended for for use outside -// of WebRtc/Libjingle. Please use the VoiceEngine API instead. -// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 - class AudioReceiveStream { public: struct Stats { @@ -103,12 +98,13 @@ class AudioReceiveStream { Transport* rtcp_send_transport = nullptr; - // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- - // level components. - // TODO(solenberg): Remove when VoiceEngine channels are created outside - // of Call. + // TODO(solenberg): Remove once clients don't use it anymore. int voe_channel_id = -1; + // NetEq settings. + size_t jitter_buffer_max_packets = 50; + bool jitter_buffer_fast_accelerate = false; + // Identifier for an A/V synchronization group. Empty string to disable. // TODO(pbos): Synchronize streams in a sync group, not just one video // stream to one audio stream. Tracked by issue webrtc:4762. @@ -135,14 +131,13 @@ class AudioReceiveStream { virtual int GetOutputLevel() const = 0; // Sets an audio sink that receives unmixed audio from the receive stream. - // Ownership of the sink is passed to the stream and can be used by the - // caller to do lifetime management (i.e. when the sink's dtor is called). + // Ownership of the sink is managed by the caller. // Only one sink can be set and passing a null sink clears an existing one. // NOTE: Audio must still somehow be pulled through AudioTransport for audio // to stream through this sink. In practice, this happens if mixed audio // is being pulled+rendered and/or if audio is being pulled for the purposes // of feeding to the AEC. - virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; + virtual void SetSink(AudioSinkInterface* sink) = 0; // Sets playback gain of the stream, applied when mixing, and thus after it // is potentially forwarded to any attached AudioSinkInterface implementation. diff --git a/call/audio_send_stream.cc b/call/audio_send_stream.cc index d0043a2df1..274a6ebdb4 100644 --- a/call/audio_send_stream.cc +++ b/call/audio_send_stream.cc @@ -26,7 +26,6 @@ std::string AudioSendStream::Config::ToString() const { std::stringstream ss; ss << "{rtp: " << rtp.ToString(); ss << ", send_transport: " << (send_transport ? "(Transport)" : "null"); - ss << ", voe_channel_id: " << voe_channel_id; ss << ", min_bitrate_bps: " << min_bitrate_bps; ss << ", max_bitrate_bps: " << max_bitrate_bps; ss << ", send_codec_spec: " diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h index 30e3b5b3f7..03f32b79e6 100644 --- a/call/audio_send_stream.h +++ b/call/audio_send_stream.h @@ -30,11 +30,6 @@ namespace webrtc { class AudioFrame; -// WORK IN PROGRESS -// This class is under development and is not yet intended for for use outside -// of WebRtc/Libjingle. Please use the VoiceEngine API instead. -// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 - class AudioSendStream { public: struct Stats { @@ -92,10 +87,7 @@ class AudioSendStream { // the entire life of the AudioSendStream and is owned by the API client. Transport* send_transport = nullptr; - // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level - // components. - // TODO(solenberg): Remove when VoiceEngine channels are created outside - // of Call. + // TODO(solenberg): Remove once clients don't use it anymore. int voe_channel_id = -1; // Bitrate limits used for variable audio bitrate streams. Set both to -1 to diff --git a/call/audio_state.h b/call/audio_state.h index 56d5d44075..a8e57f0f3f 100644 --- a/call/audio_state.h +++ b/call/audio_state.h @@ -21,19 +21,12 @@ class AudioProcessing; class AudioTransport; class VoiceEngine; -// WORK IN PROGRESS -// This class is under development and is not yet intended for for use outside -// of WebRtc/Libjingle. Please use the VoiceEngine API instead. -// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 - // AudioState holds the state which must be shared between multiple instances of // webrtc::Call for audio processing purposes. class AudioState : public rtc::RefCountInterface { public: struct Config { - // VoiceEngine used for audio streams and audio/video synchronization. - // AudioState will tickle the VoE refcount to keep it alive for as long as - // the AudioState itself. + // TODO(solenberg): Remove once clients don't use it anymore. VoiceEngine* voice_engine = nullptr; // The audio mixer connected to active receive streams. One per diff --git a/call/call.cc b/call/call.cc index 6a97d64c64..fceea13768 100644 --- a/call/call.cc +++ b/call/call.cc @@ -20,7 +20,6 @@ #include "audio/audio_receive_stream.h" #include "audio/audio_send_stream.h" #include "audio/audio_state.h" -#include "audio/scoped_voe_interface.h" #include "audio/time_interval.h" #include "call/bitrate_allocator.h" #include "call/call.h" @@ -605,9 +604,9 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream( } AudioSendStream* send_stream = new AudioSendStream( - config, config_.audio_state, &worker_queue_, transport_send_.get(), - bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats(), - suspended_rtp_state); + config, config_.audio_state, &worker_queue_, module_process_thread_.get(), + transport_send_.get(), bitrate_allocator_.get(), event_log_, + call_stats_->rtcp_rtt_stats(), suspended_rtp_state); { WriteLockScoped write_lock(*send_crit_); RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == @@ -663,8 +662,8 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( event_log_->Log(rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>( CreateRtcLogStreamConfig(config))); AudioReceiveStream* receive_stream = new AudioReceiveStream( - &audio_receiver_controller_, transport_send_->packet_router(), config, - config_.audio_state, event_log_); + &audio_receiver_controller_, transport_send_->packet_router(), + module_process_thread_.get(), config, config_.audio_state, event_log_); { WriteLockScoped write_lock(*receive_crit_); receive_rtp_config_[config.rtp.remote_ssrc] = diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc index 47295056df..e9747a4fe6 100644 --- a/call/call_perf_tests.cc +++ b/call/call_perf_tests.cc @@ -40,7 +40,6 @@ #include "test/testsupport/fileutils.h" #include "test/testsupport/perf_test.h" #include "video/transport_adapter.h" -#include "voice_engine/include/voe_base.h" using webrtc::test::DriftingClock; using webrtc::test::FakeAudioDevice; @@ -152,15 +151,10 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, const uint32_t kAudioSendSsrc = 1234; const uint32_t kAudioRecvSsrc = 5678; - int send_channel_id; - int recv_channel_id; - FakeNetworkPipe::Config audio_net_config; audio_net_config.queue_delay_ms = 500; audio_net_config.loss_percent = 5; - VoiceEngine* voice_engine; - VoEBase* voe_base; VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), test_label); std::map<uint8_t, MediaType> audio_pt_map; @@ -176,22 +170,13 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, task_queue_.SendTask([&]() { metrics::Reset(); - voice_engine = VoiceEngine::Create(); - voe_base = VoEBase::GetInterface(voice_engine); rtc::scoped_refptr<FakeAudioDevice> fake_audio_device = new rtc::RefCountedObject<FakeAudioDevice>( FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000), FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed); EXPECT_EQ(0, fake_audio_device->Init()); - EXPECT_EQ(0, voe_base->Init(fake_audio_device.get(), nullptr, - decoder_factory_)); - VoEBase::ChannelConfig config; - config.enable_voice_pacing = true; - send_channel_id = voe_base->CreateChannel(config); - recv_channel_id = voe_base->CreateChannel(); AudioState::Config send_audio_state_config; - send_audio_state_config.voice_engine = voice_engine; send_audio_state_config.audio_mixer = AudioMixerImpl::Create(); send_audio_state_config.audio_processing = AudioProcessingBuilder().Create(); @@ -237,7 +222,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, CreateMatchingReceiveConfigs(receive_transport.get()); AudioSendStream::Config audio_send_config(audio_send_transport.get()); - audio_send_config.voe_channel_id = send_channel_id; audio_send_config.rtp.ssrc = kAudioSendSsrc; audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec( kAudioSendPayloadType, {"ISAC", 16000, 1}); @@ -258,7 +242,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, AudioReceiveStream::Config audio_recv_config; audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc; audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc; - audio_recv_config.voe_channel_id = recv_channel_id; audio_recv_config.sync_group = kSyncGroup; audio_recv_config.decoder_factory = decoder_factory_; audio_recv_config.decoder_map = { @@ -304,13 +287,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, sender_call_->DestroyAudioSendStream(audio_send_stream); receiver_call_->DestroyAudioReceiveStream(audio_receive_stream); - voe_base->DeleteChannel(send_channel_id); - voe_base->DeleteChannel(recv_channel_id); - voe_base->Release(); - DestroyCalls(); - - VoiceEngine::Delete(voice_engine); }); observer.PrintResults(); diff --git a/call/call_unittest.cc b/call/call_unittest.cc index a9bb36fff9..dc36ce2072 100644 --- a/call/call_unittest.cc +++ b/call/call_unittest.cc @@ -15,6 +15,8 @@ #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/test/mock_audio_mixer.h" +#include "audio/audio_send_stream.h" +#include "audio/audio_receive_stream.h" #include "call/audio_state.h" #include "call/call.h" #include "call/fake_rtp_transport_controller_send.h" @@ -29,16 +31,12 @@ #include "test/gtest.h" #include "test/mock_audio_decoder_factory.h" #include "test/mock_transport.h" -#include "test/mock_voice_engine.h" namespace { struct CallHelper { - explicit CallHelper( - rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) - : voice_engine_(decoder_factory) { + CallHelper() { webrtc::AudioState::Config audio_state_config; - audio_state_config.voice_engine = &voice_engine_; audio_state_config.audio_mixer = new rtc::RefCountedObject<webrtc::test::MockAudioMixer>(); audio_state_config.audio_processing = @@ -51,10 +49,8 @@ struct CallHelper { } webrtc::Call* operator->() { return call_.get(); } - webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; } private: - testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; webrtc::RtcEventLogNullImpl event_log_; std::unique_ptr<webrtc::Call> call_; }; @@ -70,20 +66,17 @@ 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) { - rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory( - new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>); - CallHelper call(decoder_factory); + CallHelper call; AudioReceiveStream::Config config; config.rtp.remote_ssrc = 42; - config.voe_channel_id = 123; - config.decoder_factory = decoder_factory; + config.decoder_factory = + new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>(); AudioReceiveStream* stream = call->CreateAudioReceiveStream(config); EXPECT_NE(stream, nullptr); call->DestroyAudioReceiveStream(stream); @@ -92,7 +85,6 @@ TEST(CallTest, CreateDestroy_AudioReceiveStream) { 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) { @@ -113,12 +105,10 @@ TEST(CallTest, CreateDestroy_AudioSendStreams) { } TEST(CallTest, CreateDestroy_AudioReceiveStreams) { - rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory( - new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>); - CallHelper call(decoder_factory); + CallHelper call; AudioReceiveStream::Config config; - config.voe_channel_id = 123; - config.decoder_factory = decoder_factory; + config.decoder_factory = + new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>(); std::list<AudioReceiveStream*> streams; for (int i = 0; i < 2; ++i) { for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { @@ -139,112 +129,51 @@ TEST(CallTest, CreateDestroy_AudioReceiveStreams) { } TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) { - rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory( - new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>); - CallHelper call(decoder_factory); - ::testing::NiceMock<MockRtpRtcp> mock_rtp_rtcp; - - constexpr int kRecvChannelId = 101; - - // Set up the mock to create a channel proxy which we know of, so that we can - // add our expectations to it. - test::MockVoEChannelProxy* recv_channel_proxy = nullptr; - EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_)) - .WillRepeatedly(testing::Invoke([&](int channel_id) { - test::MockVoEChannelProxy* channel_proxy = - new testing::NiceMock<test::MockVoEChannelProxy>(); - EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory()) - .WillRepeatedly(testing::ReturnRef(decoder_factory)); - EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_)) - .WillRepeatedly(testing::Invoke( - [](const std::map<int, SdpAudioFormat>& codecs) { - EXPECT_THAT(codecs, testing::IsEmpty()); - })); - EXPECT_CALL(*channel_proxy, GetRtpRtcp(testing::_, testing::_)) - .WillRepeatedly(testing::SetArgPointee<0>(&mock_rtp_rtcp)); - // If being called for the send channel, save a pointer to the channel - // proxy for later. - if (channel_id == kRecvChannelId) { - EXPECT_FALSE(recv_channel_proxy); - recv_channel_proxy = channel_proxy; - } - return channel_proxy; - })); - + CallHelper call; AudioReceiveStream::Config recv_config; recv_config.rtp.remote_ssrc = 42; recv_config.rtp.local_ssrc = 777; - recv_config.voe_channel_id = kRecvChannelId; - recv_config.decoder_factory = decoder_factory; + recv_config.decoder_factory = + new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>(); AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config); EXPECT_NE(recv_stream, nullptr); - EXPECT_CALL(*recv_channel_proxy, AssociateSendChannel(testing::_)).Times(1); AudioSendStream::Config send_config(nullptr); send_config.rtp.ssrc = 777; - send_config.voe_channel_id = 123; AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); EXPECT_NE(send_stream, nullptr); - EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1); + internal::AudioReceiveStream* internal_recv_stream = + static_cast<internal::AudioReceiveStream*>(recv_stream); + EXPECT_EQ(send_stream, + internal_recv_stream->GetAssociatedSendStreamForTesting()); + call->DestroyAudioSendStream(send_stream); + EXPECT_EQ(nullptr, internal_recv_stream->GetAssociatedSendStreamForTesting()); - EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1); call->DestroyAudioReceiveStream(recv_stream); } TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) { - rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory( - new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>); - CallHelper call(decoder_factory); - ::testing::NiceMock<MockRtpRtcp> mock_rtp_rtcp; - - constexpr int kRecvChannelId = 101; - - // Set up the mock to create a channel proxy which we know of, so that we can - // add our expectations to it. - test::MockVoEChannelProxy* recv_channel_proxy = nullptr; - EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_)) - .WillRepeatedly(testing::Invoke([&](int channel_id) { - test::MockVoEChannelProxy* channel_proxy = - new testing::NiceMock<test::MockVoEChannelProxy>(); - EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory()) - .WillRepeatedly(testing::ReturnRef(decoder_factory)); - EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_)) - .WillRepeatedly(testing::Invoke( - [](const std::map<int, SdpAudioFormat>& codecs) { - EXPECT_THAT(codecs, testing::IsEmpty()); - })); - EXPECT_CALL(*channel_proxy, GetRtpRtcp(testing::_, testing::_)) - .WillRepeatedly(testing::SetArgPointee<0>(&mock_rtp_rtcp)); - // If being called for the send channel, save a pointer to the channel - // proxy for later. - if (channel_id == kRecvChannelId) { - EXPECT_FALSE(recv_channel_proxy); - recv_channel_proxy = channel_proxy; - // We need to set this expectation here since the channel proxy is - // created as a side effect of CreateAudioReceiveStream(). - EXPECT_CALL(*recv_channel_proxy, - AssociateSendChannel(testing::_)).Times(1); - } - return channel_proxy; - })); - + CallHelper call; AudioSendStream::Config send_config(nullptr); send_config.rtp.ssrc = 777; - send_config.voe_channel_id = 123; AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); EXPECT_NE(send_stream, nullptr); AudioReceiveStream::Config recv_config; recv_config.rtp.remote_ssrc = 42; recv_config.rtp.local_ssrc = 777; - recv_config.voe_channel_id = kRecvChannelId; - recv_config.decoder_factory = decoder_factory; + recv_config.decoder_factory = + new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>(); AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config); EXPECT_NE(recv_stream, nullptr); - EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1); + internal::AudioReceiveStream* internal_recv_stream = + static_cast<internal::AudioReceiveStream*>(recv_stream); + EXPECT_EQ(send_stream, + internal_recv_stream->GetAssociatedSendStreamForTesting()); + call->DestroyAudioReceiveStream(recv_stream); call->DestroyAudioSendStream(send_stream); @@ -434,54 +363,15 @@ TEST(CallBitrateTest, TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) { constexpr uint32_t kSSRC = 12345; - - // There's similar functionality in cricket::VoEWrapper but it's not reachable - // from here. Since we're working on removing VoE interfaces, I doubt it's - // worth making VoEWrapper more easily available. - struct ScopedVoiceEngine { - ScopedVoiceEngine() - : voe(VoiceEngine::Create()), - base(VoEBase::GetInterface(voe)) {} - ~ScopedVoiceEngine() { - base->Release(); - EXPECT_TRUE(VoiceEngine::Delete(voe)); - } - - VoiceEngine* voe; - VoEBase* base; - }; - ScopedVoiceEngine voice_engine; - - AudioState::Config audio_state_config; - audio_state_config.voice_engine = voice_engine.voe; - audio_state_config.audio_mixer = - new rtc::RefCountedObject<test::MockAudioMixer>(); - audio_state_config.audio_processing = - new rtc::RefCountedObject<test::MockAudioProcessing>(); - audio_state_config.audio_device_module = - new rtc::RefCountedObject<test::MockAudioDeviceModule>(); - voice_engine.base->Init(audio_state_config.audio_device_module, nullptr, - CreateBuiltinAudioDecoderFactory()); - auto audio_state = AudioState::Create(audio_state_config); - - RtcEventLogNullImpl event_log; - Call::Config call_config(&event_log); - call_config.audio_state = audio_state; - std::unique_ptr<Call> call(Call::Create(call_config)); + CallHelper call; auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) { AudioSendStream::Config config(nullptr); config.rtp.ssrc = ssrc; - config.voe_channel_id = voice_engine.base->CreateChannel(); AudioSendStream* stream = call->CreateAudioSendStream(config); - VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine.voe); - auto channel_proxy = voe_impl->GetChannelProxy(config.voe_channel_id); - RtpRtcp* rtp_rtcp = nullptr; - RtpReceiver* rtp_receiver = nullptr; // Unused but required for call. - channel_proxy->GetRtpRtcp(&rtp_rtcp, &rtp_receiver); - const RtpState rtp_state = rtp_rtcp->GetRtpState(); + const RtpState rtp_state = + static_cast<internal::AudioSendStream*>(stream)->GetRtpState(); call->DestroyAudioSendStream(stream); - voice_engine.base->DeleteChannel(config.voe_channel_id); return rtp_state; }; @@ -496,6 +386,7 @@ TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) { rtp_state2.last_timestamp_time_ms); EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent); } + TEST(CallBitrateTest, BiggerMaskMinUsed) { CallBitrateHelper call; Call::Config::BitrateConfigMask mask; diff --git a/media/BUILD.gn b/media/BUILD.gn index a64f394749..54473fe86e 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -238,7 +238,6 @@ rtc_static_library("rtc_audio_video") { "engine/webrtcvideocapturerfactory.h", "engine/webrtcvideoengine.cc", "engine/webrtcvideoengine.h", - "engine/webrtcvoe.h", "engine/webrtcvoiceengine.cc", "engine/webrtcvoiceengine.h", @@ -427,7 +426,6 @@ if (rtc_include_tests) { "engine/fakewebrtcvcmfactory.h", "engine/fakewebrtcvideocapturemodule.h", "engine/fakewebrtcvideoengine.h", - "engine/fakewebrtcvoiceengine.h", ] configs += [ ":rtc_unittest_main_config" ] diff --git a/media/engine/fakewebrtccall.cc b/media/engine/fakewebrtccall.cc index edc6d8bd73..5dcfd762e9 100644 --- a/media/engine/fakewebrtccall.cc +++ b/media/engine/fakewebrtccall.cc @@ -23,7 +23,6 @@ namespace cricket { FakeAudioSendStream::FakeAudioSendStream( int id, const webrtc::AudioSendStream::Config& config) : id_(id), config_(config) { - RTC_DCHECK(config.voe_channel_id != -1); } void FakeAudioSendStream::Reconfigure( @@ -72,7 +71,6 @@ webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats( FakeAudioReceiveStream::FakeAudioReceiveStream( int id, const webrtc::AudioReceiveStream::Config& config) : id_(id), config_(config) { - RTC_DCHECK(config.voe_channel_id != -1); } const webrtc::AudioReceiveStream::Config& @@ -107,9 +105,8 @@ webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { return stats_; } -void FakeAudioReceiveStream::SetSink( - std::unique_ptr<webrtc::AudioSinkInterface> sink) { - sink_ = std::move(sink); +void FakeAudioReceiveStream::SetSink(webrtc::AudioSinkInterface* sink) { + sink_ = sink; } void FakeAudioReceiveStream::SetGain(float gain) { diff --git a/media/engine/fakewebrtccall.h b/media/engine/fakewebrtccall.h index c355055b9b..cd9f1eb2cb 100644 --- a/media/engine/fakewebrtccall.h +++ b/media/engine/fakewebrtccall.h @@ -86,7 +86,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { void SetStats(const webrtc::AudioReceiveStream::Stats& stats); int received_packets() const { return received_packets_; } bool VerifyLastPacket(const uint8_t* data, size_t length) const; - const webrtc::AudioSinkInterface* sink() const { return sink_.get(); } + const webrtc::AudioSinkInterface* sink() const { return sink_; } float gain() const { return gain_; } bool DeliverRtp(const uint8_t* packet, size_t length, @@ -101,7 +101,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { webrtc::AudioReceiveStream::Stats GetStats() const override; int GetOutputLevel() const override { return 0; } - void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; + void SetSink(webrtc::AudioSinkInterface* sink) override; void SetGain(float gain) override; std::vector<webrtc::RtpSource> GetSources() const override { return std::vector<webrtc::RtpSource>(); @@ -111,7 +111,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { webrtc::AudioReceiveStream::Config config_; webrtc::AudioReceiveStream::Stats stats_; int received_packets_ = 0; - std::unique_ptr<webrtc::AudioSinkInterface> sink_; + webrtc::AudioSinkInterface* sink_ = nullptr; float gain_ = 1.0f; rtc::Buffer last_packet_; bool started_ = false; diff --git a/media/engine/fakewebrtcvoiceengine.h b/media/engine/fakewebrtcvoiceengine.h deleted file mode 100644 index e171bbca01..0000000000 --- a/media/engine/fakewebrtcvoiceengine.h +++ /dev/null @@ -1,107 +0,0 @@ -/* - * Copyright (c) 2010 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. - */ - -#ifndef MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ -#define MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ - -#include <map> -#include <vector> - -#include "media/engine/webrtcvoe.h" -#include "rtc_base/checks.h" - -namespace cricket { - -#define WEBRTC_CHECK_CHANNEL(channel) \ - if (channels_.find(channel) == channels_.end()) return -1; - -#define WEBRTC_STUB(method, args) \ - int method args override { return 0; } - -#define WEBRTC_FUNC(method, args) int method args override - -class FakeWebRtcVoiceEngine : public webrtc::VoEBase { - public: - struct Channel { - std::vector<webrtc::CodecInst> recv_codecs; - size_t neteq_capacity = 0; - bool neteq_fast_accelerate = false; - }; - - FakeWebRtcVoiceEngine() {} - ~FakeWebRtcVoiceEngine() override { - RTC_CHECK(channels_.empty()); - } - - bool IsInited() const { return inited_; } - int GetLastChannel() const { return last_channel_; } - int GetNumChannels() const { return static_cast<int>(channels_.size()); } - void set_fail_create_channel(bool fail_create_channel) { - fail_create_channel_ = fail_create_channel; - } - - WEBRTC_STUB(Release, ()); - - // webrtc::VoEBase - WEBRTC_FUNC(Init, - (webrtc::AudioDeviceModule* adm, - webrtc::AudioProcessing* audioproc, - const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& - decoder_factory)) { - inited_ = true; - return 0; - } - void Terminate() override { - inited_ = false; - } - WEBRTC_FUNC(CreateChannel, ()) { - return CreateChannel(webrtc::VoEBase::ChannelConfig()); - } - WEBRTC_FUNC(CreateChannel, (const webrtc::VoEBase::ChannelConfig& config)) { - if (fail_create_channel_) { - return -1; - } - Channel* ch = new Channel(); - ch->neteq_capacity = config.acm_config.neteq_config.max_packets_in_buffer; - ch->neteq_fast_accelerate = - config.acm_config.neteq_config.enable_fast_accelerate; - channels_[++last_channel_] = ch; - return last_channel_; - } - WEBRTC_FUNC(DeleteChannel, (int channel)) { - WEBRTC_CHECK_CHANNEL(channel); - delete channels_[channel]; - channels_.erase(channel); - return 0; - } - - size_t GetNetEqCapacity() const { - auto ch = channels_.find(last_channel_); - RTC_DCHECK(ch != channels_.end()); - return ch->second->neteq_capacity; - } - bool GetNetEqFastAccelerate() const { - auto ch = channels_.find(last_channel_); - RTC_CHECK(ch != channels_.end()); - return ch->second->neteq_fast_accelerate; - } - - private: - bool inited_ = false; - int last_channel_ = -1; - std::map<int, Channel*> channels_; - bool fail_create_channel_ = false; - - RTC_DISALLOW_COPY_AND_ASSIGN(FakeWebRtcVoiceEngine); -}; - -} // namespace cricket - -#endif // MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ diff --git a/media/engine/webrtcmediaengine.h b/media/engine/webrtcmediaengine.h index 1b1649aec6..632e730478 100644 --- a/media/engine/webrtcmediaengine.h +++ b/media/engine/webrtcmediaengine.h @@ -17,10 +17,10 @@ #include "call/call.h" #include "media/base/mediaengine.h" +#include "modules/audio_device/include/audio_device.h" namespace webrtc { class AudioDecoderFactory; -class AudioDeviceModule; class AudioMixer; class AudioProcessing; class VideoDecoderFactory; diff --git a/media/engine/webrtcvoe.h b/media/engine/webrtcvoe.h deleted file mode 100644 index 5d5342906e..0000000000 --- a/media/engine/webrtcvoe.h +++ /dev/null @@ -1,89 +0,0 @@ -/* - * Copyright (c) 2004 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. - */ - -#ifndef MEDIA_ENGINE_WEBRTCVOE_H_ -#define MEDIA_ENGINE_WEBRTCVOE_H_ - -#include <memory> - -#include "common_types.h" // NOLINT(build/include) -#include "modules/audio_device/include/audio_device.h" -#include "voice_engine/include/voe_base.h" -#include "voice_engine/include/voe_errors.h" - -namespace cricket { -// automatically handles lifetime of WebRtc VoiceEngine -class scoped_voe_engine { - public: - explicit scoped_voe_engine(webrtc::VoiceEngine* e) : ptr(e) {} - // RTC_DCHECK, to ensure that there are no leaks at shutdown - ~scoped_voe_engine() { - if (ptr) { - const bool success = webrtc::VoiceEngine::Delete(ptr); - RTC_DCHECK(success); - } - } - // Releases the current pointer. - void reset() { - if (ptr) { - const bool success = webrtc::VoiceEngine::Delete(ptr); - RTC_DCHECK(success); - ptr = NULL; - } - } - webrtc::VoiceEngine* get() const { return ptr; } - private: - webrtc::VoiceEngine* ptr; -}; - -// unique_ptr-like class to handle obtaining and releasing WebRTC interface -// pointers. -template<class T> -class scoped_voe_ptr { - public: - explicit scoped_voe_ptr(const scoped_voe_engine& e) - : ptr(T::GetInterface(e.get())) {} - explicit scoped_voe_ptr(T* p) : ptr(p) {} - ~scoped_voe_ptr() { if (ptr) ptr->Release(); } - T* operator->() const { return ptr; } - T* get() const { return ptr; } - - // Releases the current pointer. - void reset() { - if (ptr) { - ptr->Release(); - ptr = NULL; - } - } - - private: - T* ptr; -}; - -// Utility class for aggregating the various WebRTC interface. -// Fake implementations can also be injected for testing. -class VoEWrapper { - public: - VoEWrapper() - : engine_(webrtc::VoiceEngine::Create()), base_(engine_) { - } - explicit VoEWrapper(webrtc::VoEBase* base) - : engine_(webrtc::VoiceEngine::Create()), base_(base) {} - ~VoEWrapper() {} - webrtc::VoiceEngine* engine() const { return engine_.get(); } - webrtc::VoEBase* base() const { return base_.get(); } - - private: - scoped_voe_engine engine_; - scoped_voe_ptr<webrtc::VoEBase> base_; -}; -} // namespace cricket - -#endif // MEDIA_ENGINE_WEBRTCVOE_H_ diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc index d8fd1e7f77..edd91218ba 100644 --- a/media/engine/webrtcvoiceengine.cc +++ b/media/engine/webrtcvoiceengine.cc @@ -27,7 +27,6 @@ #include "media/engine/apm_helpers.h" #include "media/engine/payload_type_mapper.h" #include "media/engine/webrtcmediaengine.h" -#include "media/engine/webrtcvoe.h" #include "modules/audio_device/audio_device_impl.h" #include "modules/audio_mixer/audio_mixer_impl.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" @@ -199,26 +198,11 @@ WebRtcVoiceEngine::WebRtcVoiceEngine( const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) - : WebRtcVoiceEngine(adm, - encoder_factory, - decoder_factory, - audio_mixer, - audio_processing, - nullptr) {} - -WebRtcVoiceEngine::WebRtcVoiceEngine( - webrtc::AudioDeviceModule* adm, - const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, - const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, - rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, - rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing, - VoEWrapper* voe_wrapper) : adm_(adm), encoder_factory_(encoder_factory), decoder_factory_(decoder_factory), audio_mixer_(audio_mixer), - apm_(audio_processing), - voe_wrapper_(voe_wrapper) { + apm_(audio_processing) { // This may be called from any thread, so detach thread checkers. worker_thread_checker_.DetachFromThread(); signal_thread_checker_.DetachFromThread(); @@ -234,7 +218,6 @@ WebRtcVoiceEngine::~WebRtcVoiceEngine() { RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; if (initialized_) { StopAecDump(); - voe_wrapper_->base()->Terminate(); // Stop AudioDevice. adm()->StopPlayout(); @@ -252,12 +235,6 @@ void WebRtcVoiceEngine::Init() { low_priority_worker_queue_.reset( new rtc::TaskQueue("rtc-low-prio", rtc::TaskQueue::Priority::LOW)); - // VoEWrapper needs to be created on the worker thread. It's expected to be - // null here unless it's being injected for testing. - if (!voe_wrapper_) { - voe_wrapper_.reset(new VoEWrapper()); - } - // Load our audio codec lists. RTC_LOG(LS_INFO) << "Supported send codecs in order of preference:"; send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders()); @@ -271,8 +248,6 @@ void WebRtcVoiceEngine::Init() { RTC_LOG(LS_INFO) << ToString(codec); } - channel_config_.enable_voice_pacing = true; - #if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE) // No ADM supplied? Create a default one. if (!adm_) { @@ -283,12 +258,10 @@ void WebRtcVoiceEngine::Init() { RTC_CHECK(adm()); webrtc::adm_helpers::Init(adm()); webrtc::apm_helpers::Init(apm()); - RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm(), nullptr, decoder_factory_)); // Set up AudioState. { webrtc::AudioState::Config config; - config.voice_engine = voe()->engine(); if (audio_mixer_) { config.audio_mixer = audio_mixer_; } else { @@ -537,13 +510,13 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { if (options.audio_jitter_buffer_max_packets) { RTC_LOG(LS_INFO) << "NetEq capacity is " << *options.audio_jitter_buffer_max_packets; - channel_config_.acm_config.neteq_config.max_packets_in_buffer = + audio_jitter_buffer_max_packets_ = std::max(20, *options.audio_jitter_buffer_max_packets); } if (options.audio_jitter_buffer_fast_accelerate) { RTC_LOG(LS_INFO) << "NetEq fast mode? " << *options.audio_jitter_buffer_fast_accelerate; - channel_config_.acm_config.neteq_config.enable_fast_accelerate = + audio_jitter_buffer_fast_accelerate_ = *options.audio_jitter_buffer_fast_accelerate; } @@ -690,11 +663,6 @@ void WebRtcVoiceEngine::StopAecDump() { apm()->DetachAecDump(); } -int WebRtcVoiceEngine::CreateVoEChannel() { - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return voe_wrapper_->base()->CreateChannel(channel_config_); -} - webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); RTC_DCHECK(adm_); @@ -793,7 +761,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream : public AudioSource::Sink { public: WebRtcAudioSendStream( - int ch, uint32_t ssrc, const std::string& c_name, const std::string track_id, @@ -811,12 +778,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")), max_send_bitrate_bps_(max_send_bitrate_bps), rtp_parameters_(CreateRtpParametersWithOneEncoding()) { - RTC_DCHECK_GE(ch, 0); RTC_DCHECK(call); RTC_DCHECK(encoder_factory); config_.rtp.ssrc = ssrc; config_.rtp.c_name = c_name; - config_.voe_channel_id = ch; config_.rtp.extensions = extensions; config_.audio_network_adaptor_config = audio_network_adaptor_config; config_.encoder_factory = encoder_factory; @@ -970,12 +935,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream UpdateSendState(); } - // Accessor to the VoE channel ID. - int channel() const { - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return config_.voe_channel_id; - } - const webrtc::RtpParameters& rtp_parameters() const { return rtp_parameters_; } @@ -1153,7 +1112,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { public: WebRtcAudioReceiveStream( - int ch, uint32_t remote_ssrc, uint32_t local_ssrc, bool use_transport_cc, @@ -1163,9 +1121,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { webrtc::Call* call, webrtc::Transport* rtcp_send_transport, const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, - const std::map<int, webrtc::SdpAudioFormat>& decoder_map) + const std::map<int, webrtc::SdpAudioFormat>& decoder_map, + size_t jitter_buffer_max_packets, + bool jitter_buffer_fast_accelerate) : call_(call), config_() { - RTC_DCHECK_GE(ch, 0); RTC_DCHECK(call); config_.rtp.remote_ssrc = remote_ssrc; config_.rtp.local_ssrc = local_ssrc; @@ -1173,7 +1132,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0; config_.rtp.extensions = extensions; config_.rtcp_send_transport = rtcp_send_transport; - config_.voe_channel_id = ch; + config_.jitter_buffer_max_packets = jitter_buffer_max_packets; + config_.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate; config_.sync_group = sync_group; config_.decoder_factory = decoder_factory; config_.decoder_map = decoder_map; @@ -1234,14 +1194,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { return stream_->GetOutputLevel(); } - int channel() const { - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return config_.voe_channel_id; - } - void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - stream_->SetSink(std::move(sink)); + raw_audio_sink_ = std::move(sink); + stream_->SetSink(raw_audio_sink_.get()); } void SetOutputVolume(double volume) { @@ -1253,10 +1209,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); RTC_DCHECK(stream_); if (playout) { - RTC_LOG(LS_INFO) << "Starting playout for channel #" << channel(); stream_->Start(); } else { - RTC_LOG(LS_INFO) << "Stopping playout for channel #" << channel(); stream_->Stop(); } playout_ = playout; @@ -1277,6 +1231,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { stream_ = call_->CreateAudioReceiveStream(config_); RTC_CHECK(stream_); SetPlayout(playout_); + stream_->SetSink(raw_audio_sink_.get()); } void ReconfigureAudioReceiveStream() { @@ -1292,6 +1247,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { // configuration changes. webrtc::AudioReceiveStream* stream_ = nullptr; bool playout_ = false; + std::unique_ptr<webrtc::AudioSinkInterface> raw_audio_sink_; RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream); }; @@ -1801,24 +1757,6 @@ bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc, return true; } -int WebRtcVoiceMediaChannel::CreateVoEChannel() { - int id = engine()->CreateVoEChannel(); - if (id == -1) { - RTC_LOG(LS_WARNING) << "CreateVoEChannel() failed."; - return -1; - } - - return id; -} - -bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) { - if (engine()->voe()->base()->DeleteChannel(channel) == -1) { - RTC_LOG(LS_WARNING) << "DeleteChannel(" << channel << ") failed."; - return false; - } - return true; -} - bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream"); RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); @@ -1827,22 +1765,23 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { uint32_t ssrc = sp.first_ssrc(); RTC_DCHECK(0 != ssrc); - if (GetSendChannelId(ssrc) != -1) { + if (send_streams_.find(ssrc) != send_streams_.end()) { RTC_LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc; return false; } - // Create a new channel for sending audio data. - int channel = CreateVoEChannel(); - if (channel == -1) { - return false; - } - rtc::Optional<std::string> audio_network_adaptor_config = GetAudioNetworkAdaptorConfig(options_); WebRtcAudioSendStream* stream = new WebRtcAudioSendStream( - channel, ssrc, sp.cname, sp.id, send_codec_spec_, send_rtp_extensions_, - max_send_bitrate_bps_, audio_network_adaptor_config, call_, this, + ssrc, + sp.cname, + sp.id, + send_codec_spec_, + send_rtp_extensions_, + max_send_bitrate_bps_, + audio_network_adaptor_config, + call_, + this, engine()->encoder_factory_); send_streams_.insert(std::make_pair(ssrc, stream)); @@ -1880,15 +1819,8 @@ bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) { // the first active send stream and use that instead, reassociating receive // streams. - // Clean up and delete the send stream+channel. - int channel = it->second->channel(); - RTC_LOG(LS_INFO) << "Removing audio send stream " << ssrc - << " with VoiceEngine channel #" << channel << "."; delete it->second; send_streams_.erase(it); - if (!DeleteVoEChannel(channel)) { - return false; - } if (send_streams_.empty()) { SetSend(false); } @@ -1917,23 +1849,27 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { return true; } - if (GetReceiveChannelId(ssrc) != -1) { + if (recv_streams_.find(ssrc) != recv_streams_.end()) { RTC_LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc; return false; } // Create a new channel for receiving audio data. - const int channel = CreateVoEChannel(); - if (channel == -1) { - return false; - } - recv_streams_.insert(std::make_pair( ssrc, new WebRtcAudioReceiveStream( - channel, ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_, - recv_nack_enabled_, sp.sync_label, recv_rtp_extensions_, call_, this, - engine()->decoder_factory_, decoder_map_))); + ssrc, + receiver_reports_ssrc_, + recv_transport_cc_enabled_, + recv_nack_enabled_, + sp.sync_label, + recv_rtp_extensions_, + call_, + this, + engine()->decoder_factory_, + decoder_map_, + engine()->audio_jitter_buffer_max_packets_, + engine()->audio_jitter_buffer_fast_accelerate_))); recv_streams_[ssrc]->SetPlayout(playout_); return true; @@ -1953,15 +1889,10 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { MaybeDeregisterUnsignaledRecvStream(ssrc); - const int channel = it->second->channel(); - - // Clean up and delete the receive stream+channel. - RTC_LOG(LS_INFO) << "Removing audio receive stream " << ssrc - << " with VoiceEngine channel #" << channel << "."; it->second->SetRawAudioSink(nullptr); delete it->second; recv_streams_.erase(it); - return DeleteVoEChannel(channel); + return true; } bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc, @@ -2327,24 +2258,6 @@ std::vector<webrtc::RtpSource> WebRtcVoiceMediaChannel::GetSources( return it->second->GetSources(); } -int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const { - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - const auto it = recv_streams_.find(ssrc); - if (it != recv_streams_.end()) { - return it->second->channel(); - } - return -1; -} - -int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const { - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - const auto it = send_streams_.find(ssrc); - if (it != send_streams_.end()) { - return it->second->channel(); - } - return -1; -} - bool WebRtcVoiceMediaChannel:: MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h index e5880e29da..47aba8a1e3 100644 --- a/media/engine/webrtcvoiceengine.h +++ b/media/engine/webrtcvoiceengine.h @@ -22,7 +22,6 @@ #include "call/call.h" #include "media/base/rtputils.h" #include "media/engine/apm_helpers.h" -#include "media/engine/webrtcvoe.h" #include "modules/audio_processing/include/audio_processing.h" #include "pc/channel.h" #include "rtc_base/buffer.h" @@ -37,7 +36,6 @@ namespace cricket { class AudioDeviceModule; class AudioMixer; class AudioSource; -class VoEWrapper; class WebRtcVoiceMediaChannel; // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. @@ -51,14 +49,6 @@ class WebRtcVoiceEngine final { const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing); - // Dependency injection for testing. - WebRtcVoiceEngine( - webrtc::AudioDeviceModule* adm, - const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, - const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, - rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, - rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing, - VoEWrapper* voe_wrapper); ~WebRtcVoiceEngine(); // Does initialization that needs to occur on the worker thread. @@ -81,8 +71,6 @@ class WebRtcVoiceEngine final { void RegisterChannel(WebRtcVoiceMediaChannel* channel); void UnregisterChannel(WebRtcVoiceMediaChannel* channel); - VoEWrapper* voe() { return voe_wrapper_.get(); } - // Starts AEC dump using an existing file. A maximum file size in bytes can be // specified. When the maximum file size is reached, logging is stopped and // the file is closed. If max_size_bytes is set to <= 0, no limit will be @@ -125,12 +113,10 @@ class WebRtcVoiceEngine final { // The audio processing module. rtc::scoped_refptr<webrtc::AudioProcessing> apm_; // The primary instance of WebRtc VoiceEngine. - std::unique_ptr<VoEWrapper> voe_wrapper_; rtc::scoped_refptr<webrtc::AudioState> audio_state_; std::vector<AudioCodec> send_codecs_; std::vector<AudioCodec> recv_codecs_; std::vector<WebRtcVoiceMediaChannel*> channels_; - webrtc::VoEBase::ChannelConfig channel_config_; bool is_dumping_aec_ = false; bool initialized_ = false; @@ -145,6 +131,9 @@ class WebRtcVoiceEngine final { rtc::Optional<bool> experimental_ns_; rtc::Optional<bool> intelligibility_enhancer_; rtc::Optional<bool> level_control_; + // Jitter buffer settings for new streams. + size_t audio_jitter_buffer_max_packets_ = 50; + bool audio_jitter_buffer_fast_accelerate_ = false; RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine); }; @@ -224,9 +213,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions()); } - int GetReceiveChannelId(uint32_t ssrc) const; - int GetSendChannelId(uint32_t ssrc) const; - private: bool SetOptions(const AudioOptions& options); bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc index ebf6e9e271..9b56dbf826 100644 --- a/media/engine/webrtcvoiceengine_unittest.cc +++ b/media/engine/webrtcvoiceengine_unittest.cc @@ -21,7 +21,6 @@ #include "media/base/fakertp.h" #include "media/base/mediaconstants.h" #include "media/engine/fakewebrtccall.h" -#include "media/engine/fakewebrtcvoiceengine.h" #include "media/engine/webrtcvoiceengine.h" #include "modules/audio_device/include/mock_audio_device.h" #include "modules/audio_processing/include/mock_audio_processing.h" @@ -133,17 +132,12 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) { EXPECT_CALL(*apm, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config)); EXPECT_CALL(*apm, SetExtraOptions(testing::_)); EXPECT_CALL(*apm, DetachAecDump()); - cricket::FakeWebRtcVoiceEngine voe; - EXPECT_FALSE(voe.IsInited()); { cricket::WebRtcVoiceEngine engine( &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), - webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, - new cricket::VoEWrapper(&voe)); + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); engine.Init(); - EXPECT_TRUE(voe.IsInited()); } - EXPECT_FALSE(voe.IsInited()); } class FakeAudioSink : public webrtc::AudioSinkInterface { @@ -167,7 +161,6 @@ class WebRtcVoiceEngineTestFake : public testing::Test { apm_ns_(*apm_->noise_suppression()), apm_vd_(*apm_->voice_detection()), call_(webrtc::Call::Config(&event_log_)), - voe_(), override_field_trials_(field_trials) { // AudioDeviceModule. AdmSetupExpectations(&adm_); @@ -198,8 +191,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test { auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); engine_.reset(new cricket::WebRtcVoiceEngine( - &adm_, encoder_factory, decoder_factory, nullptr, apm_, - new cricket::VoEWrapper(&voe_))); + &adm_, encoder_factory, decoder_factory, nullptr, apm_)); engine_->Init(); send_parameters_.codecs.push_back(kPcmuCodec); recv_parameters_.codecs.push_back(kPcmuCodec); @@ -708,7 +700,6 @@ class WebRtcVoiceEngineTestFake : public testing::Test { webrtc::test::MockVoiceDetection& apm_vd_; webrtc::RtcEventLogNullImpl event_log_; cricket::FakeCall call_; - cricket::FakeWebRtcVoiceEngine voe_; std::unique_ptr<cricket::WebRtcVoiceEngine> engine_; cricket::VoiceMediaChannel* channel_ = nullptr; cricket::AudioSendParameters send_parameters_; @@ -2644,26 +2635,10 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Recreate) { EXPECT_NE(audio_receive_stream_id, streams.front()->id()); } -// Test that we properly handle failures to add a receive stream. -TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamFail) { - EXPECT_TRUE(SetupChannel()); - voe_.set_fail_create_channel(true); - EXPECT_FALSE(AddRecvStream(2)); -} - -// Test that we properly handle failures to add a send stream. -TEST_F(WebRtcVoiceEngineTestFake, AddSendStreamFail) { - EXPECT_TRUE(SetupChannel()); - voe_.set_fail_create_channel(true); - EXPECT_FALSE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2))); -} - // Test that AddRecvStream creates new stream. TEST_F(WebRtcVoiceEngineTestFake, AddRecvStream) { EXPECT_TRUE(SetupRecvStream()); - int channel_num = voe_.GetLastChannel(); EXPECT_TRUE(AddRecvStream(1)); - EXPECT_NE(channel_num, voe_.GetLastChannel()); } // Test that after adding a recv stream, we do not decode more codecs than @@ -2687,10 +2662,13 @@ TEST_F(WebRtcVoiceEngineTestFake, StreamCleanup) { SetSendParameters(send_parameters_); EXPECT_TRUE(AddRecvStream(1)); EXPECT_TRUE(AddRecvStream(2)); - EXPECT_EQ(3, voe_.GetNumChannels()); // default channel + 2 added + + EXPECT_EQ(1, call_.GetAudioSendStreams().size()); + EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); delete channel_; channel_ = NULL; - EXPECT_EQ(0, voe_.GetNumChannels()); + EXPECT_EQ(0, call_.GetAudioSendStreams().size()); + EXPECT_EQ(0, call_.GetAudioReceiveStreams().size()); } TEST_F(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithZeroSsrc) { @@ -2698,18 +2676,10 @@ TEST_F(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithZeroSsrc) { EXPECT_FALSE(AddRecvStream(0)); } -TEST_F(WebRtcVoiceEngineTestFake, TestNoLeakingWhenAddRecvStreamFail) { +TEST_F(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithSameSsrc) { EXPECT_TRUE(SetupChannel()); EXPECT_TRUE(AddRecvStream(1)); - // Manually delete channel to simulate a failure. - int channel = voe_.GetLastChannel(); - EXPECT_EQ(0, voe_.DeleteChannel(channel)); - // Add recv stream 2 should work. - EXPECT_TRUE(AddRecvStream(2)); - int new_channel = voe_.GetLastChannel(); - EXPECT_NE(channel, new_channel); - // The last created channel is deleted too. - EXPECT_EQ(0, voe_.DeleteChannel(new_channel)); + EXPECT_FALSE(AddRecvStream(1)); } // Test the InsertDtmf on default send stream as caller. @@ -2734,6 +2704,7 @@ TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) { TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) { EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); EXPECT_CALL(adm_, BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false)); EXPECT_CALL(adm_, @@ -2741,15 +2712,15 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) { EXPECT_CALL(adm_, BuiltInNSIsAvailable()).Times(2).WillRepeatedly(Return(false)); - EXPECT_EQ(50, voe_.GetNetEqCapacity()); - EXPECT_FALSE(voe_.GetNetEqFastAccelerate()); + EXPECT_EQ(50, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); + EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate); // Nothing set in AudioOptions, so everything should be as default. send_parameters_.options = cricket::AudioOptions(); SetSendParameters(send_parameters_); EXPECT_TRUE(IsHighPassFilterEnabled()); - EXPECT_EQ(50, voe_.GetNetEqCapacity()); - EXPECT_FALSE(voe_.GetNetEqFastAccelerate()); + EXPECT_EQ(50, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); + EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate); // Turn echo cancellation off EXPECT_CALL(apm_ec_, Enable(false)).WillOnce(Return(0)); @@ -2992,36 +2963,6 @@ TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) { channel->SetInterface(nullptr); } -TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelId) { - EXPECT_TRUE(SetupChannel()); - cricket::WebRtcVoiceMediaChannel* media_channel = - static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); - EXPECT_EQ(-1, media_channel->GetReceiveChannelId(0)); - EXPECT_TRUE(AddRecvStream(kSsrcX)); - int channel_id = voe_.GetLastChannel(); - EXPECT_EQ(channel_id, media_channel->GetReceiveChannelId(kSsrcX)); - EXPECT_EQ(-1, media_channel->GetReceiveChannelId(kSsrcY)); - EXPECT_TRUE(AddRecvStream(kSsrcY)); - int channel_id2 = voe_.GetLastChannel(); - EXPECT_EQ(channel_id2, media_channel->GetReceiveChannelId(kSsrcY)); -} - -TEST_F(WebRtcVoiceEngineTestFake, TestGetSendChannelId) { - EXPECT_TRUE(SetupChannel()); - cricket::WebRtcVoiceMediaChannel* media_channel = - static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); - EXPECT_EQ(-1, media_channel->GetSendChannelId(0)); - EXPECT_TRUE(channel_->AddSendStream( - cricket::StreamParams::CreateLegacy(kSsrcX))); - int channel_id = voe_.GetLastChannel(); - EXPECT_EQ(channel_id, media_channel->GetSendChannelId(kSsrcX)); - EXPECT_EQ(-1, media_channel->GetSendChannelId(kSsrcY)); - EXPECT_TRUE(channel_->AddSendStream( - cricket::StreamParams::CreateLegacy(kSsrcY))); - int channel_id2 = voe_.GetLastChannel(); - EXPECT_EQ(channel_id2, media_channel->GetSendChannelId(kSsrcY)); -} - TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolume) { EXPECT_TRUE(SetupChannel()); EXPECT_FALSE(channel_->SetOutputVolume(kSsrcY, 0.5)); @@ -3353,9 +3294,9 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) { // Tests that reference counting on the external ADM is correct. TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) { testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm; - EXPECT_CALL(adm, AddRef()).Times(5); + EXPECT_CALL(adm, AddRef()).Times(3); EXPECT_CALL(adm, Release()) - .Times(5) + .Times(3) .WillRepeatedly(Return(rtc::RefCountReleaseStatus::kDroppedLastRef)); { rtc::scoped_refptr<webrtc::AudioProcessing> apm = diff --git a/test/BUILD.gn b/test/BUILD.gn index 00a7ae5656..c058b5c023 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -568,7 +568,6 @@ rtc_source_set("test_common") { "layer_filtering_transport.h", "mock_transport.h", "mock_voe_channel_proxy.h", - "mock_voice_engine.h", "null_transport.cc", "null_transport.h", "rtp_rtcp_observer.h", diff --git a/test/call_test.cc b/test/call_test.cc index 6f02149abc..5fbba88a4f 100644 --- a/test/call_test.cc +++ b/test/call_test.cc @@ -21,7 +21,6 @@ #include "rtc_base/event.h" #include "rtc_base/ptr_util.h" #include "test/testsupport/fileutils.h" -#include "voice_engine/include/voe_base.h" namespace webrtc { namespace test { @@ -67,9 +66,9 @@ void CallTest::RunBaseTest(BaseTest* test) { fake_recv_audio_device_.get()); apm_send_ = AudioProcessingBuilder().Create(); apm_recv_ = AudioProcessingBuilder().Create(); - CreateVoiceEngines(); + EXPECT_EQ(0, fake_send_audio_device_->Init()); + EXPECT_EQ(0, fake_recv_audio_device_->Init()); AudioState::Config audio_state_config; - audio_state_config.voice_engine = voe_send_.voice_engine; audio_state_config.audio_mixer = AudioMixerImpl::Create(); audio_state_config.audio_processing = apm_send_; audio_state_config.audio_device_module = fake_send_audio_device_; @@ -86,7 +85,6 @@ void CallTest::RunBaseTest(BaseTest* test) { Call::Config recv_config(test->GetReceiverCallConfig()); if (num_audio_streams_ > 0) { AudioState::Config audio_state_config; - audio_state_config.voice_engine = voe_recv_.voice_engine; audio_state_config.audio_mixer = AudioMixerImpl::Create(); audio_state_config.audio_processing = apm_recv_; audio_state_config.audio_device_module = fake_recv_audio_device_; @@ -163,8 +161,6 @@ void CallTest::RunBaseTest(BaseTest* test) { send_transport_.reset(); receive_transport_.reset(); DestroyCalls(); - if (num_audio_streams_ > 0) - DestroyVoiceEngines(); }); } @@ -219,10 +215,8 @@ void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams, Transport* send_transport) { RTC_DCHECK_LE(num_audio_streams, 1); RTC_DCHECK_LE(num_flexfec_streams, 1); - RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); if (num_audio_streams > 0) { audio_send_config_ = AudioSendStream::Config(send_transport); - audio_send_config_.voe_channel_id = voe_send_.channel_id; audio_send_config_.rtp.ssrc = kAudioSendSsrc; audio_send_config_.send_codec_spec = rtc::Optional<AudioSendStream::Config::SendCodecSpec>( @@ -281,11 +275,9 @@ void CallTest::CreateMatchingAudioAndFecConfigs( Transport* rtcp_send_transport) { RTC_DCHECK_GE(1, num_audio_streams_); if (num_audio_streams_ == 1) { - RTC_DCHECK_LE(0, voe_send_.channel_id); AudioReceiveStream::Config audio_config; audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; audio_config.rtcp_send_transport = rtcp_send_transport; - audio_config.voe_channel_id = voe_recv_.channel_id; audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; audio_config.decoder_factory = decoder_factory_; audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}}; @@ -457,43 +449,6 @@ void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) { frame_generator_capturer_->SetFakeRotation(rotation); } -void CallTest::CreateVoiceEngines() { - voe_send_.voice_engine = VoiceEngine::Create(); - voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine); - EXPECT_EQ(0, fake_send_audio_device_->Init()); - EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), - nullptr, decoder_factory_)); - VoEBase::ChannelConfig config; - config.enable_voice_pacing = true; - voe_send_.channel_id = voe_send_.base->CreateChannel(config); - EXPECT_GE(voe_send_.channel_id, 0); - - voe_recv_.voice_engine = VoiceEngine::Create(); - voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine); - EXPECT_EQ(0, fake_recv_audio_device_->Init()); - EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), - nullptr, decoder_factory_)); - voe_recv_.channel_id = voe_recv_.base->CreateChannel(); - EXPECT_GE(voe_recv_.channel_id, 0); -} - -void CallTest::DestroyVoiceEngines() { - voe_recv_.base->DeleteChannel(voe_recv_.channel_id); - voe_recv_.channel_id = -1; - voe_recv_.base->Release(); - voe_recv_.base = nullptr; - - voe_send_.base->DeleteChannel(voe_send_.channel_id); - voe_send_.channel_id = -1; - voe_send_.base->Release(); - voe_send_.base = nullptr; - - VoiceEngine::Delete(voe_send_.voice_engine); - voe_send_.voice_engine = nullptr; - VoiceEngine::Delete(voe_recv_.voice_engine); - voe_recv_.voice_engine = nullptr; -} - constexpr size_t CallTest::kNumSsrcs; const int CallTest::kDefaultWidth; const int CallTest::kDefaultHeight; diff --git a/test/call_test.h b/test/call_test.h index 11ac0e670a..c86e9b589b 100644 --- a/test/call_test.h +++ b/test/call_test.h @@ -26,9 +26,6 @@ #include "test/single_threaded_task_queue.h" namespace webrtc { - -class VoEBase; - namespace test { class BaseTest; @@ -67,9 +64,8 @@ class CallTest : public ::testing::Test { static const std::map<uint8_t, MediaType> payload_type_map_; protected: - // RunBaseTest overwrites the audio_state and the voice_engine of the send and - // receive Call configs to simplify test code and avoid having old VoiceEngine - // APIs in the tests. + // RunBaseTest overwrites the audio_state of the send and receive Call configs + // to simplify test code. void RunBaseTest(BaseTest* test); void CreateCalls(const Call::Config& sender_config, @@ -152,25 +148,6 @@ class CallTest : public ::testing::Test { SingleThreadedTaskQueueForTesting task_queue_; private: - // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API. - // These methods are used to set up legacy voice engines and channels which is - // necessary while voice engine is being refactored to the new stream API. - struct VoiceEngineState { - VoiceEngineState() - : voice_engine(nullptr), - base(nullptr), - channel_id(-1) {} - - VoiceEngine* voice_engine; - VoEBase* base; - int channel_id; - }; - - void CreateVoiceEngines(); - void DestroyVoiceEngines(); - - VoiceEngineState voe_send_; - VoiceEngineState voe_recv_; rtc::scoped_refptr<AudioProcessing> apm_send_; rtc::scoped_refptr<AudioProcessing> apm_recv_; rtc::scoped_refptr<test::FakeAudioDevice> fake_send_audio_device_; diff --git a/test/mock_voe_channel_proxy.h b/test/mock_voe_channel_proxy.h index 5760781229..4b3d30b246 100644 --- a/test/mock_voe_channel_proxy.h +++ b/test/mock_voe_channel_proxy.h @@ -62,14 +62,11 @@ class MockVoEChannelProxy : public voe::ChannelProxy { int payload_frequency)); MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms)); MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms)); - // TODO(solenberg): Talk the compiler into accepting this mock method: - // MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink)); + MOCK_METHOD1(SetSink, void(AudioSinkInterface* sink)); MOCK_METHOD1(SetInputMute, void(bool muted)); MOCK_METHOD1(RegisterTransport, void(Transport* transport)); MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet)); MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length)); - MOCK_CONST_METHOD0(GetAudioDecoderFactory, - const rtc::scoped_refptr<AudioDecoderFactory>&()); MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling)); MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log)); MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats)); diff --git a/test/mock_voice_engine.h b/test/mock_voice_engine.h deleted file mode 100644 index 0bbf5ee8f0..0000000000 --- a/test/mock_voice_engine.h +++ /dev/null @@ -1,108 +0,0 @@ -/* - * 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. - */ - -#ifndef AUDIO_MOCK_VOICE_ENGINE_H_ -#define AUDIO_MOCK_VOICE_ENGINE_H_ - -#include <memory> - -#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" -#include "test/gmock.h" -#include "test/mock_voe_channel_proxy.h" -#include "voice_engine/voice_engine_impl.h" - -namespace webrtc { -namespace test { - -// NOTE: This class inherits from VoiceEngineImpl so that its clients will be -// able to get the various interfaces as usual, via T::GetInterface(). -class MockVoiceEngine : public VoiceEngineImpl { - public: - // TODO(nisse): Valid overrides commented out, because the gmock - // methods don't use any override declarations, and we want to avoid - // warnings from -Winconsistent-missing-override. See - // http://crbug.com/428099. - MockVoiceEngine( - rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr) - : decoder_factory_(decoder_factory) { - // Increase ref count so this object isn't automatically deleted whenever - // interfaces are Release():d. - ++_ref_count; - // We add this default behavior to make the mock easier to use in tests. It - // will create a NiceMock of a voe::ChannelProxy. - // TODO(ossu): As long as AudioReceiveStream is implemented as a wrapper - // around Channel, we need to make sure ChannelProxy returns the same - // decoder factory as the one passed in when creating an AudioReceiveStream. - ON_CALL(*this, ChannelProxyFactory(testing::_)) - .WillByDefault(testing::Invoke([this](int channel_id) { - auto* proxy = - new testing::NiceMock<webrtc::test::MockVoEChannelProxy>(); - EXPECT_CALL(*proxy, GetAudioDecoderFactory()) - .WillRepeatedly(testing::ReturnRef(decoder_factory_)); - EXPECT_CALL(*proxy, SetReceiveCodecs(testing::_)) - .WillRepeatedly(testing::Invoke( - [](const std::map<int, SdpAudioFormat>& codecs) { - EXPECT_THAT(codecs, testing::IsEmpty()); - })); - EXPECT_CALL(*proxy, GetRtpRtcp(testing::_, testing::_)) - .WillRepeatedly( - testing::SetArgPointee<0>(GetMockRtpRtcp(channel_id))); - return proxy; - })); - } - virtual ~MockVoiceEngine() /* override */ { - // Decrease ref count before base class d-tor is called; otherwise it will - // trigger an assertion. - --_ref_count; - } - - // These need to be the same each call to channel_id and must not leak. - MockRtpRtcp* GetMockRtpRtcp(int channel_id) { - if (mock_rtp_rtcps_.find(channel_id) == mock_rtp_rtcps_.end()) { - mock_rtp_rtcps_[channel_id].reset(new ::testing::NiceMock<MockRtpRtcp>); - } - return mock_rtp_rtcps_[channel_id].get(); - } - - // Allows injecting a ChannelProxy factory. - MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy*(int channel_id)); - - // VoiceEngineImpl - virtual std::unique_ptr<voe::ChannelProxy> GetChannelProxy( - int channel_id) /* override */ { - return std::unique_ptr<voe::ChannelProxy>(ChannelProxyFactory(channel_id)); - } - - // VoEBase - MOCK_METHOD3( - Init, - int(AudioDeviceModule* external_adm, - AudioProcessing* external_apm, - const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)); - MOCK_METHOD0(Terminate, void()); - MOCK_METHOD0(CreateChannel, int()); - MOCK_METHOD1(CreateChannel, int(const ChannelConfig& config)); - MOCK_METHOD1(DeleteChannel, int(int channel)); - - private: - // TODO(ossu): I'm not particularly happy about keeping the decoder factory - // here, but due to how gmock is implemented, I cannot just keep it in the - // functor implementing the default version of ChannelProxyFactory, above. - // GMock creates an unfortunate copy of the functor, which would cause us to - // return a dangling reference. Fortunately, this should go away once - // voe::Channel does. - rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; - - std::map<int, std::unique_ptr<MockRtpRtcp>> mock_rtp_rtcps_; -}; -} // namespace test -} // namespace webrtc - -#endif // AUDIO_MOCK_VOICE_ENGINE_H_ diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc index ca4317c5ef..ec110ff694 100644 --- a/video/video_quality_test.cc +++ b/video/video_quality_test.cc @@ -41,7 +41,6 @@ #include "test/testsupport/test_artifacts.h" #include "test/vcm_capturer.h" #include "test/video_renderer.h" -#include "voice_engine/include/voe_base.h" DEFINE_bool(save_worst_frame, false, @@ -62,47 +61,6 @@ constexpr uint32_t kThumbnailRtxSsrcStart = 0xF0000; constexpr int kDefaultMaxQp = cricket::WebRtcVideoChannel::kDefaultQpMax; -struct VoiceEngineState { - VoiceEngineState() - : voice_engine(nullptr), - base(nullptr), - send_channel_id(-1), - receive_channel_id(-1) {} - - webrtc::VoiceEngine* voice_engine; - webrtc::VoEBase* base; - int send_channel_id; - int receive_channel_id; -}; - -void CreateVoiceEngine( - VoiceEngineState* voe, - webrtc::AudioDeviceModule* adm, - rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory) { - voe->voice_engine = webrtc::VoiceEngine::Create(); - voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine); - EXPECT_EQ(0, adm->Init()); - EXPECT_EQ(0, voe->base->Init(adm, nullptr, decoder_factory)); - webrtc::VoEBase::ChannelConfig config; - config.enable_voice_pacing = true; - voe->send_channel_id = voe->base->CreateChannel(config); - EXPECT_GE(voe->send_channel_id, 0); - voe->receive_channel_id = voe->base->CreateChannel(); - EXPECT_GE(voe->receive_channel_id, 0); -} - -void DestroyVoiceEngine(VoiceEngineState* voe) { - voe->base->DeleteChannel(voe->send_channel_id); - voe->send_channel_id = -1; - voe->base->DeleteChannel(voe->receive_channel_id); - voe->receive_channel_id = -1; - voe->base->Release(); - voe->base = nullptr; - - webrtc::VoiceEngine::Delete(voe->voice_engine); - voe->voice_engine = nullptr; -} - class VideoStreamFactory : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface { public: @@ -2023,12 +1981,9 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) { }); } -void VideoQualityTest::SetupAudio(int send_channel_id, - int receive_channel_id, - Transport* transport, +void VideoQualityTest::SetupAudio(Transport* transport, AudioReceiveStream** audio_receive_stream) { audio_send_config_ = AudioSendStream::Config(transport); - audio_send_config_.voe_channel_id = send_channel_id; audio_send_config_.rtp.ssrc = kAudioSendSsrc; // Add extension to enable audio send side BWE, and allow audio bit rate @@ -2052,7 +2007,6 @@ void VideoQualityTest::SetupAudio(int send_channel_id, AudioReceiveStream::Config audio_config; audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; audio_config.rtcp_send_transport = transport; - audio_config.voe_channel_id = receive_channel_id; audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; audio_config.rtp.transport_cc = params_.call.send_side_bwe; audio_config.rtp.extensions = audio_send_config_.rtp.extensions; @@ -2069,7 +2023,6 @@ void VideoQualityTest::RunWithRenderers(const Params& params) { num_video_streams_ = params.call.dual_video ? 2 : 1; std::unique_ptr<test::LayerFilteringTransport> send_transport; std::unique_ptr<test::DirectTransport> recv_transport; - ::VoiceEngineState voe; std::unique_ptr<test::VideoRenderer> local_preview; std::vector<std::unique_ptr<test::VideoRenderer>> loopback_renderers; AudioReceiveStream* audio_receive_stream = nullptr; @@ -2090,9 +2043,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) { 1.f); if (params_.audio.enabled) { - CreateVoiceEngine(&voe, fake_audio_device.get(), decoder_factory_); AudioState::Config audio_state_config; - audio_state_config.voice_engine = voe.voice_engine; audio_state_config.audio_mixer = AudioMixerImpl::Create(); audio_state_config.audio_processing = AudioProcessingBuilder().Create(); audio_state_config.audio_device_module = fake_audio_device; @@ -2168,8 +2119,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) { } if (params_.audio.enabled) { - SetupAudio(voe.send_channel_id, voe.receive_channel_id, - send_transport.get(), &audio_receive_stream); + SetupAudio(send_transport.get(), &audio_receive_stream); } for (VideoReceiveStream* receive_stream : video_receive_streams_) @@ -2219,9 +2169,6 @@ void VideoQualityTest::RunWithRenderers(const Params& params) { send_transport.reset(); recv_transport.reset(); - if (params_.audio.enabled) - DestroyVoiceEngine(&voe); - local_preview.reset(); loopback_renderers.clear(); diff --git a/video/video_quality_test.h b/video/video_quality_test.h index f86b20095f..20f899d124 100644 --- a/video/video_quality_test.h +++ b/video/video_quality_test.h @@ -134,9 +134,7 @@ class VideoQualityTest : public test::CallTest { void SetupVideo(Transport* send_transport, Transport* recv_transport); void SetupThumbnails(Transport* send_transport, Transport* recv_transport); void DestroyThumbnailStreams(); - void SetupAudio(int send_channel_id, - int receive_channel_id, - Transport* transport, + void SetupAudio(Transport* transport, AudioReceiveStream** audio_receive_stream); void StartEncodedFrameLogs(VideoSendStream* stream); diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn index bb9253fd95..701c3ae529 100644 --- a/voice_engine/BUILD.gn +++ b/voice_engine/BUILD.gn @@ -15,14 +15,10 @@ rtc_static_library("voice_engine") { sources = [ "channel.cc", "channel.h", - "channel_manager.cc", - "channel_manager.h", "channel_proxy.cc", "channel_proxy.h", "include/voe_base.h", "include/voe_errors.h", - "shared_data.cc", - "shared_data.h", "transport_feedback_packet_loss_tracker.cc", "transport_feedback_packet_loss_tracker.h", "utility.cc", @@ -134,10 +130,8 @@ if (rtc_include_tests) { } sources = [ - "channel_unittest.cc", "transport_feedback_packet_loss_tracker_unittest.cc", "utility_unittest.cc", - "voe_base_unittest.cc", ] data = [ diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc index c77b38b972..ef076ba4e1 100644 --- a/voice_engine/channel.cc +++ b/voice_engine/channel.cc @@ -574,24 +574,24 @@ int Channel::PreferredSampleRate() const { audio_coding_->PlayoutFrequency()); } -int32_t Channel::CreateChannel(Channel*& channel, - int32_t channelId, - uint32_t instanceId, - const VoEBase::ChannelConfig& config) { - channel = new Channel(channelId, instanceId, config); - if (channel == NULL) { - RTC_LOG(LS_ERROR) << "unable to allocate memory for new channel"; - return -1; - } - return 0; +Channel::Channel(rtc::TaskQueue* encoder_queue, + ProcessThread* module_process_thread, + AudioDeviceModule* audio_device_module) + : Channel(module_process_thread, + audio_device_module, + 0, + false, + rtc::scoped_refptr<AudioDecoderFactory>()) { + RTC_DCHECK(encoder_queue); + encoder_queue_ = encoder_queue; } -Channel::Channel(int32_t channelId, - uint32_t instanceId, - const VoEBase::ChannelConfig& config) - : _instanceId(instanceId), - _channelId(channelId), - event_log_proxy_(new RtcEventLogProxy()), +Channel::Channel(ProcessThread* module_process_thread, + AudioDeviceModule* audio_device_module, + size_t jitter_buffer_max_packets, + bool jitter_buffer_fast_playout, + rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) + : event_log_proxy_(new RtcEventLogProxy()), rtcp_rtt_stats_proxy_(new RtcpRttStatsProxy()), rtp_header_parser_(RtpHeaderParser::Create()), rtp_payload_registry_(new RTPPayloadRegistry()), @@ -613,8 +613,8 @@ Channel::Channel(int32_t channelId, rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()), capture_start_rtp_time_stamp_(-1), capture_start_ntp_time_ms_(-1), - _moduleProcessThreadPtr(NULL), - _audioDeviceModulePtr(NULL), + _moduleProcessThreadPtr(module_process_thread), + _audioDeviceModulePtr(audio_device_module), _transportPtr(NULL), input_mute_(false), previous_frame_muted_(false), @@ -623,17 +623,20 @@ Channel::Channel(int32_t channelId, transport_overhead_per_packet_(0), rtp_overhead_per_packet_(0), rtcp_observer_(new VoERtcpObserver(this)), - associate_send_channel_(ChannelOwner(nullptr)), - pacing_enabled_(config.enable_voice_pacing), + associated_send_channel_(nullptr), feedback_observer_proxy_(new TransportFeedbackProxy()), seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), kMaxRetransmissionWindowMs)), - decoder_factory_(config.acm_config.decoder_factory), use_twcc_plr_for_ana_( webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") { - AudioCodingModule::Config acm_config(config.acm_config); + RTC_DCHECK(module_process_thread); + RTC_DCHECK(audio_device_module); + AudioCodingModule::Config acm_config; + acm_config.decoder_factory = decoder_factory; + acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets; + acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout; acm_config.neteq_config.enable_muted_state = true; audio_coding_.reset(AudioCodingModule::Create(acm_config)); @@ -658,36 +661,25 @@ Channel::Channel(int32_t channelId, _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration)); _rtpRtcpModule->SetSendingMediaStatus(false); + + Init(); } Channel::~Channel() { + Terminate(); RTC_DCHECK(!channel_state_.Get().sending); RTC_DCHECK(!channel_state_.Get().playing); } -int32_t Channel::Init() { - RTC_DCHECK(construction_thread_.CalledOnValidThread()); - +void Channel::Init() { channel_state_.Reset(); - // --- Initial sanity - - if (_moduleProcessThreadPtr == NULL) { - RTC_LOG(LS_ERROR) - << "Channel::Init() must call SetEngineInformation() first"; - return -1; - } - // --- Add modules to process thread (for periodic schedulation) - _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE); // --- ACM initialization - - if (audio_coding_->InitializeReceiver() == -1) { - RTC_LOG(LS_ERROR) << "Channel::Init() unable to initialize the ACM - 1"; - return -1; - } + int error = audio_coding_->InitializeReceiver(); + RTC_DCHECK_EQ(0, error); // --- RTP/RTCP module initialization @@ -699,13 +691,10 @@ int32_t Channel::Init() { telephone_event_handler_->SetTelephoneEventForwardToDecoder(true); // RTCP is enabled by default. _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound); - // --- Register all permanent callbacks - if (audio_coding_->RegisterTransportCallback(this) == -1) { - RTC_LOG(LS_ERROR) << "Channel::Init() callbacks not registered"; - return -1; - } - return 0; + // --- Register all permanent callbacks + error = audio_coding_->RegisterTransportCallback(this); + RTC_DCHECK_EQ(0, error); } void Channel::Terminate() { @@ -720,11 +709,8 @@ void Channel::Terminate() { // 1. De-register callbacks in modules // 2. De-register modules in process thread // 3. Destroy modules - if (audio_coding_->RegisterTransportCallback(NULL) == -1) { - RTC_LOG(LS_WARNING) - << "Terminate() failed to de-register transport callback" - << " (Audio coding module)"; - } + int error = audio_coding_->RegisterTransportCallback(NULL); + RTC_DCHECK_EQ(0, error); // De-register modules in process thread if (_moduleProcessThreadPtr) @@ -733,25 +719,9 @@ void Channel::Terminate() { // End of modules shutdown } -int32_t Channel::SetEngineInformation(ProcessThread& moduleProcessThread, - AudioDeviceModule& audioDeviceModule, - rtc::TaskQueue* encoder_queue) { - RTC_DCHECK(encoder_queue); - RTC_DCHECK(!encoder_queue_); - _moduleProcessThreadPtr = &moduleProcessThread; - _audioDeviceModulePtr = &audioDeviceModule; - encoder_queue_ = encoder_queue; - return 0; -} - -void Channel::SetSink(std::unique_ptr<AudioSinkInterface> sink) { +void Channel::SetSink(AudioSinkInterface* sink) { rtc::CritScope cs(&_callbackCritSect); - audio_sink_ = std::move(sink); -} - -const rtc::scoped_refptr<AudioDecoderFactory>& -Channel::GetAudioDecoderFactory() const { - return decoder_factory_; + audio_sink_ = sink; } int32_t Channel::StartPlayout() { @@ -1387,29 +1357,17 @@ void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) { // is done and payload is ready for packetization and transmission. // Otherwise, it will return without invoking the callback. if (audio_coding_->Add10MsData(*audio_input) < 0) { - RTC_LOG(LS_ERROR) << "ACM::Add10MsData() failed for channel " << _channelId; + RTC_LOG(LS_ERROR) << "ACM::Add10MsData() failed."; return; } _timeStamp += static_cast<uint32_t>(audio_input->samples_per_channel_); } -void Channel::set_associate_send_channel(const ChannelOwner& channel) { - RTC_DCHECK(!channel.channel() || - channel.channel()->ChannelId() != _channelId); +void Channel::SetAssociatedSendChannel(Channel* channel) { + RTC_DCHECK_NE(this, channel); rtc::CritScope lock(&assoc_send_channel_lock_); - associate_send_channel_ = channel; -} - -void Channel::DisassociateSendChannel(int channel_id) { - rtc::CritScope lock(&assoc_send_channel_lock_); - Channel* channel = associate_send_channel_.channel(); - if (channel && channel->ChannelId() == channel_id) { - // If this channel is associated with a send channel of the specified - // Channel ID, disassociate with it. - ChannelOwner ref(NULL); - associate_send_channel_ = ref; - } + associated_send_channel_ = channel; } void Channel::SetRtcEventLog(RtcEventLog* event_log) { @@ -1526,25 +1484,6 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) { } } -void Channel::RegisterReceiveCodecsToRTPModule() { - // TODO(kwiberg): Iterate over the factory's supported codecs instead? - const int nSupportedCodecs = AudioCodingModule::NumberOfCodecs(); - for (int idx = 0; idx < nSupportedCodecs; idx++) { - CodecInst codec; - if (audio_coding_->Codec(idx, &codec) == -1) { - RTC_LOG(LS_WARNING) << "Unable to register codec #" << idx - << " for RTP/RTCP receiver."; - continue; - } - const SdpAudioFormat format = CodecInstToSdp(codec); - if (!decoder_factory_->IsSupportedDecoder(format) || - rtp_receiver_->RegisterReceivePayload(codec.pltype, format) == -1) { - RTC_LOG(LS_WARNING) << "Unable to register " << format - << " for RTP/RTCP receiver."; - } - } -} - int Channel::SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, unsigned char id) { @@ -1579,14 +1518,13 @@ int64_t Channel::GetRTT(bool allow_associate_channel) const { if (report_blocks.empty()) { if (allow_associate_channel) { rtc::CritScope lock(&assoc_send_channel_lock_); - Channel* channel = associate_send_channel_.channel(); // Tries to get RTT from an associated channel. This is important for // receive-only channels. - if (channel) { + if (associated_send_channel_) { // To prevent infinite recursion and deadlock, calling GetRTT of // associate channel should always use "false" for argument: // |allow_associate_channel|. - rtt = channel->GetRTT(false); + rtt = associated_send_channel_->GetRTT(false); } } return rtt; diff --git a/voice_engine/channel.h b/voice_engine/channel.h index 9018f9f7a2..e501c6d4d6 100644 --- a/voice_engine/channel.h +++ b/voice_engine/channel.h @@ -27,10 +27,9 @@ #include "modules/rtp_rtcp/include/rtp_rtcp.h" #include "rtc_base/criticalsection.h" #include "rtc_base/event.h" +#include "rtc_base/task_queue.h" #include "rtc_base/thread_checker.h" #include "voice_engine/audio_level.h" -#include "voice_engine/include/voe_base.h" -#include "voice_engine/shared_data.h" namespace rtc { class TimestampWrapAroundHandler; @@ -142,27 +141,19 @@ class Channel enum { KNumSocketThreads = 1 }; enum { KNumberOfSocketBuffers = 8 }; + // Used for send streams. + Channel(rtc::TaskQueue* encoder_queue, + ProcessThread* module_process_thread, + AudioDeviceModule* audio_device_module); + // Used for receive streams. + Channel(ProcessThread* module_process_thread, + AudioDeviceModule* audio_device_module, + size_t jitter_buffer_max_packets, + bool jitter_buffer_fast_playout, + rtc::scoped_refptr<AudioDecoderFactory> decoder_factory); virtual ~Channel(); - static int32_t CreateChannel(Channel*& channel, - int32_t channelId, - uint32_t instanceId, - const VoEBase::ChannelConfig& config); - Channel(int32_t channelId, - uint32_t instanceId, - const VoEBase::ChannelConfig& config); - int32_t Init(); - void Terminate(); - int32_t SetEngineInformation(ProcessThread& moduleProcessThread, - AudioDeviceModule& audioDeviceModule, - rtc::TaskQueue* encoder_queue); - - void SetSink(std::unique_ptr<AudioSinkInterface> sink); - // TODO(ossu): Don't use! It's only here to confirm that the decoder factory - // passed into AudioReceiveStream is the same as the one set when creating the - // ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can - // go. - const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const; + void SetSink(AudioSinkInterface* sink); void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs); @@ -270,8 +261,6 @@ class Channel int PreferredSampleRate() const; - uint32_t InstanceId() const { return _instanceId; } - int32_t ChannelId() const { return _channelId; } bool Playing() const { return channel_state_.Get().playing; } bool Sending() const { return channel_state_.Get().sending; } RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); } @@ -290,9 +279,7 @@ class Channel // Associate to a send channel. // Used for obtaining RTT for a receive-only channel. - void set_associate_send_channel(const ChannelOwner& channel); - // Disassociate a send channel if it was associated. - void DisassociateSendChannel(int channel_id); + void SetAssociatedSendChannel(Channel* channel); // Set a RtcEventLog logging object. void SetRtcEventLog(RtcEventLog* event_log); @@ -318,6 +305,9 @@ class Channel private: class ProcessAndEncodeAudioTask; + void Init(); + void Terminate(); + int GetRemoteSSRC(unsigned int& ssrc); void OnUplinkPacketLossRate(float packet_loss_rate); bool InputMute() const; @@ -330,7 +320,6 @@ class Channel bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; int ResendPackets(const uint16_t* sequence_numbers, int length); void UpdatePlayoutTimestamp(bool rtcp); - void RegisterReceiveCodecsToRTPModule(); int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, @@ -346,9 +335,6 @@ class Channel // for encoding. void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input); - uint32_t _instanceId; - int32_t _channelId; - rtc::CriticalSection _callbackCritSect; rtc::CriticalSection volume_settings_critsect_; @@ -364,7 +350,7 @@ class Channel TelephoneEventHandler* telephone_event_handler_; std::unique_ptr<RtpRtcp> _rtpRtcpModule; std::unique_ptr<AudioCodingModule> audio_coding_; - std::unique_ptr<AudioSinkInterface> audio_sink_; + AudioSinkInterface* audio_sink_ = nullptr; AudioLevel _outputAudioLevel; uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_); @@ -407,26 +393,21 @@ class Channel std::unique_ptr<VoERtcpObserver> rtcp_observer_; // An associated send channel. rtc::CriticalSection assoc_send_channel_lock_; - ChannelOwner associate_send_channel_ RTC_GUARDED_BY(assoc_send_channel_lock_); + Channel* associated_send_channel_ RTC_GUARDED_BY(assoc_send_channel_lock_); - bool pacing_enabled_; + bool pacing_enabled_ = true; PacketRouter* packet_router_ = nullptr; std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; std::unique_ptr<RateLimiter> retransmission_rate_limiter_; - // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. - rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; - rtc::ThreadChecker construction_thread_; const bool use_twcc_plr_for_ana_; rtc::CriticalSection encoder_queue_lock_; - bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false; - rtc::TaskQueue* encoder_queue_ = nullptr; }; diff --git a/voice_engine/channel_manager.cc b/voice_engine/channel_manager.cc deleted file mode 100644 index 9a82d2445f..0000000000 --- a/voice_engine/channel_manager.cc +++ /dev/null @@ -1,134 +0,0 @@ -/* - * Copyright (c) 2011 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 "voice_engine/channel_manager.h" - -#include "rtc_base/timeutils.h" -#include "voice_engine/channel.h" - -namespace webrtc { -namespace voe { - -ChannelOwner::ChannelOwner(class Channel* channel) - : channel_ref_(new ChannelRef(channel)) {} - -ChannelOwner::ChannelRef::ChannelRef(class Channel* channel) - : channel(channel) {} - -ChannelManager::ChannelManager(uint32_t instance_id) - : instance_id_(instance_id), - last_channel_id_(-1), - random_(rtc::TimeNanos()) {} - -ChannelOwner ChannelManager::CreateChannel( - const VoEBase::ChannelConfig& config) { - Channel* channel; - Channel::CreateChannel(channel, ++last_channel_id_, instance_id_, config); - // TODO(solenberg): Delete this, users should configure ssrc - // explicitly. - channel->SetLocalSSRC(random_.Rand<uint32_t>()); - - ChannelOwner channel_owner(channel); - - rtc::CritScope crit(&lock_); - - channels_.push_back(channel_owner); - - return channel_owner; -} - -ChannelOwner ChannelManager::GetChannel(int32_t channel_id) { - rtc::CritScope crit(&lock_); - - for (size_t i = 0; i < channels_.size(); ++i) { - if (channels_[i].channel()->ChannelId() == channel_id) - return channels_[i]; - } - return ChannelOwner(NULL); -} - -void ChannelManager::GetAllChannels(std::vector<ChannelOwner>* channels) { - rtc::CritScope crit(&lock_); - - *channels = channels_; -} - -void ChannelManager::DestroyChannel(int32_t channel_id) { - assert(channel_id >= 0); - // Holds a reference to a channel, this is used so that we never delete - // Channels while holding a lock, but rather when the method returns. - ChannelOwner reference(NULL); - { - rtc::CritScope crit(&lock_); - std::vector<ChannelOwner>::iterator to_delete = channels_.end(); - for (auto it = channels_.begin(); it != channels_.end(); ++it) { - Channel* channel = it->channel(); - // For channels associated with the channel to be deleted, disassociate - // with that channel. - channel->DisassociateSendChannel(channel_id); - - if (channel->ChannelId() == channel_id) { - to_delete = it; - } - } - if (to_delete != channels_.end()) { - reference = *to_delete; - channels_.erase(to_delete); - } - } - if (reference.channel()) { - // Ensure the channel is torn down now, on this thread, since a reference - // may still be held on a different thread (e.g. in the audio capture - // thread). - reference.channel()->Terminate(); - } -} - -void ChannelManager::DestroyAllChannels() { - // Holds references so that Channels are not destroyed while holding this - // lock, but rather when the method returns. - std::vector<ChannelOwner> references; - { - rtc::CritScope crit(&lock_); - references = channels_; - channels_.clear(); - } - for (auto& owner : references) { - if (owner.channel()) - owner.channel()->Terminate(); - } -} - -size_t ChannelManager::NumOfChannels() const { - rtc::CritScope crit(&lock_); - return channels_.size(); -} - -ChannelManager::Iterator::Iterator(ChannelManager* channel_manager) - : iterator_pos_(0) { - channel_manager->GetAllChannels(&channels_); -} - -Channel* ChannelManager::Iterator::GetChannel() { - if (iterator_pos_ < channels_.size()) - return channels_[iterator_pos_].channel(); - return NULL; -} - -bool ChannelManager::Iterator::IsValid() { - return iterator_pos_ < channels_.size(); -} - -void ChannelManager::Iterator::Increment() { - ++iterator_pos_; -} - -} // namespace voe -} // namespace webrtc diff --git a/voice_engine/channel_manager.h b/voice_engine/channel_manager.h deleted file mode 100644 index f7cf5f4dac..0000000000 --- a/voice_engine/channel_manager.h +++ /dev/null @@ -1,127 +0,0 @@ -/* - * Copyright (c) 2011 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. - */ - -#ifndef VOICE_ENGINE_CHANNEL_MANAGER_H_ -#define VOICE_ENGINE_CHANNEL_MANAGER_H_ - -#include <memory> -#include <vector> - -#include "api/refcountedbase.h" -#include "rtc_base/constructormagic.h" -#include "rtc_base/criticalsection.h" -#include "rtc_base/random.h" -#include "rtc_base/scoped_ref_ptr.h" -#include "system_wrappers/include/atomic32.h" -#include "typedefs.h" // NOLINT(build/include) -#include "voice_engine/include/voe_base.h" - -namespace webrtc { - -class AudioDecoderFactory; - -namespace voe { - -class Channel; - -// Shared-pointer implementation for keeping track of Channels. The underlying -// shared instance will be dropped when no more ChannelOwners point to it. -// -// One common source of ChannelOwner instances are -// ChannelManager::CreateChannel() and ChannelManager::GetChannel(...). -// It has a similar use case to shared_ptr in C++11. Should this move to C++11 -// in the future, this class should be replaced by exactly that. -// -// To access the underlying Channel, use .channel(). -// IsValid() implements a convenience method as an alternative for checking -// whether the underlying pointer is NULL or not. -// -// Channel channel_owner = channel_manager.GetChannel(channel_id); -// if (channel_owner.IsValid()) -// channel_owner.channel()->...; -// -class ChannelOwner { - public: - explicit ChannelOwner(Channel* channel); - ChannelOwner(const ChannelOwner& channel_owner) = default; - - ~ChannelOwner() = default; - - ChannelOwner& operator=(const ChannelOwner& other) = default; - - Channel* channel() const { return channel_ref_->channel.get(); } - bool IsValid() { return channel_ref_->channel.get() != NULL; } - private: - // Shared instance of a Channel. Copying ChannelOwners increase the reference - // count and destroying ChannelOwners decrease references. Channels are - // deleted when no references to them are held. - struct ChannelRef : public rtc::RefCountedBase { - ChannelRef(Channel* channel); - const std::unique_ptr<Channel> channel; - }; - - rtc::scoped_refptr<ChannelRef> channel_ref_; -}; - -class ChannelManager { - public: - ChannelManager(uint32_t instance_id); - - // Upon construction of an Iterator it will grab a copy of the channel list of - // the ChannelManager. The iteration will then occur over this state, not the - // current one of the ChannelManager. As the Iterator holds its own references - // to the Channels, they will remain valid even if they are removed from the - // ChannelManager. - class Iterator { - public: - explicit Iterator(ChannelManager* channel_manager); - - Channel* GetChannel(); - bool IsValid(); - - void Increment(); - - private: - size_t iterator_pos_; - std::vector<ChannelOwner> channels_; - - RTC_DISALLOW_COPY_AND_ASSIGN(Iterator); - }; - - // CreateChannel will always return a valid ChannelOwner instance. - ChannelOwner CreateChannel(const VoEBase::ChannelConfig& config); - - // ChannelOwner.channel() will be NULL if channel_id is invalid or no longer - // exists. This should be checked with ChannelOwner::IsValid(). - ChannelOwner GetChannel(int32_t channel_id); - void GetAllChannels(std::vector<ChannelOwner>* channels); - - void DestroyChannel(int32_t channel_id); - void DestroyAllChannels(); - - size_t NumOfChannels() const; - - private: - uint32_t instance_id_; - - Atomic32 last_channel_id_; - - rtc::CriticalSection lock_; - std::vector<ChannelOwner> channels_; - - // For generation of random ssrc:s. - webrtc::Random random_; - - RTC_DISALLOW_COPY_AND_ASSIGN(ChannelManager); -}; -} // namespace voe -} // namespace webrtc - -#endif // VOICE_ENGINE_CHANNEL_MANAGER_H_ diff --git a/voice_engine/channel_proxy.cc b/voice_engine/channel_proxy.cc index 08233ae37d..6442f2b1cf 100644 --- a/voice_engine/channel_proxy.cc +++ b/voice_engine/channel_proxy.cc @@ -17,15 +17,14 @@ #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_minmax.h" -#include "voice_engine/channel.h" namespace webrtc { namespace voe { -ChannelProxy::ChannelProxy() : channel_owner_(nullptr) {} +ChannelProxy::ChannelProxy() {} -ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) : - channel_owner_(channel_owner) { - RTC_CHECK(channel_owner_.channel()); +ChannelProxy::ChannelProxy(std::unique_ptr<Channel> channel) : + channel_(std::move(channel)) { + RTC_DCHECK(channel_); module_process_thread_checker_.DetachFromThread(); } @@ -34,23 +33,23 @@ ChannelProxy::~ChannelProxy() {} bool ChannelProxy::SetEncoder(int payload_type, std::unique_ptr<AudioEncoder> encoder) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->SetEncoder(payload_type, std::move(encoder)); + return channel_->SetEncoder(payload_type, std::move(encoder)); } void ChannelProxy::ModifyEncoder( rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->ModifyEncoder(modifier); + channel_->ModifyEncoder(modifier); } void ChannelProxy::SetRTCPStatus(bool enable) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetRTCPStatus(enable); + channel_->SetRTCPStatus(enable); } void ChannelProxy::SetLocalSSRC(uint32_t ssrc) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - int error = channel()->SetLocalSSRC(ssrc); + int error = channel_->SetLocalSSRC(ssrc); RTC_DCHECK_EQ(0, error); } @@ -58,65 +57,65 @@ void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); // Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array. std::string c_name_limited = c_name.substr(0, 255); - int error = channel()->SetRTCP_CNAME(c_name_limited.c_str()); + int error = channel_->SetRTCP_CNAME(c_name_limited.c_str()); RTC_DCHECK_EQ(0, error); } void ChannelProxy::SetNACKStatus(bool enable, int max_packets) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetNACKStatus(enable, max_packets); + channel_->SetNACKStatus(enable, max_packets); } void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - int error = channel()->SetSendAudioLevelIndicationStatus(enable, id); + int error = channel_->SetSendAudioLevelIndicationStatus(enable, id); RTC_DCHECK_EQ(0, error); } void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id); + int error = channel_->SetReceiveAudioLevelIndicationStatus(enable, id); RTC_DCHECK_EQ(0, error); } void ChannelProxy::EnableSendTransportSequenceNumber(int id) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->EnableSendTransportSequenceNumber(id); + channel_->EnableSendTransportSequenceNumber(id); } void ChannelProxy::EnableReceiveTransportSequenceNumber(int id) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->EnableReceiveTransportSequenceNumber(id); + channel_->EnableReceiveTransportSequenceNumber(id); } void ChannelProxy::RegisterSenderCongestionControlObjects( RtpTransportControllerSendInterface* transport, RtcpBandwidthObserver* bandwidth_observer) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->RegisterSenderCongestionControlObjects(transport, + channel_->RegisterSenderCongestionControlObjects(transport, bandwidth_observer); } void ChannelProxy::RegisterReceiverCongestionControlObjects( PacketRouter* packet_router) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->RegisterReceiverCongestionControlObjects(packet_router); + channel_->RegisterReceiverCongestionControlObjects(packet_router); } void ChannelProxy::ResetSenderCongestionControlObjects() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->ResetSenderCongestionControlObjects(); + channel_->ResetSenderCongestionControlObjects(); } void ChannelProxy::ResetReceiverCongestionControlObjects() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->ResetReceiverCongestionControlObjects(); + channel_->ResetReceiverCongestionControlObjects(); } CallStatistics ChannelProxy::GetRTCPStatistics() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); CallStatistics stats = {0}; - int error = channel()->GetRTPStatistics(stats); + int error = channel_->GetRTPStatistics(stats); RTC_DCHECK_EQ(0, error); return stats; } @@ -124,7 +123,7 @@ CallStatistics ChannelProxy::GetRTCPStatistics() const { std::vector<ReportBlock> ChannelProxy::GetRemoteRTCPReportBlocks() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); std::vector<webrtc::ReportBlock> blocks; - int error = channel()->GetRemoteRTCPReportBlocks(&blocks); + int error = channel_->GetRemoteRTCPReportBlocks(&blocks); RTC_DCHECK_EQ(0, error); return blocks; } @@ -132,7 +131,7 @@ std::vector<ReportBlock> ChannelProxy::GetRemoteRTCPReportBlocks() const { NetworkStatistics ChannelProxy::GetNetworkStatistics() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); NetworkStatistics stats = {0}; - int error = channel()->GetNetworkStatistics(stats); + int error = channel_->GetNetworkStatistics(stats); RTC_DCHECK_EQ(0, error); return stats; } @@ -140,51 +139,51 @@ NetworkStatistics ChannelProxy::GetNetworkStatistics() const { AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); AudioDecodingCallStats stats; - channel()->GetDecodingCallStatistics(&stats); + channel_->GetDecodingCallStatistics(&stats); return stats; } ANAStats ChannelProxy::GetANAStatistics() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->GetANAStatistics(); + return channel_->GetANAStatistics(); } int ChannelProxy::GetSpeechOutputLevel() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->GetSpeechOutputLevel(); + return channel_->GetSpeechOutputLevel(); } int ChannelProxy::GetSpeechOutputLevelFullRange() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->GetSpeechOutputLevelFullRange(); + return channel_->GetSpeechOutputLevelFullRange(); } double ChannelProxy::GetTotalOutputEnergy() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->GetTotalOutputEnergy(); + return channel_->GetTotalOutputEnergy(); } double ChannelProxy::GetTotalOutputDuration() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->GetTotalOutputDuration(); + return channel_->GetTotalOutputDuration(); } uint32_t ChannelProxy::GetDelayEstimate() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() || module_process_thread_checker_.CalledOnValidThread()); - return channel()->GetDelayEstimate(); + return channel_->GetDelayEstimate(); } bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->SetSendTelephoneEventPayloadType(payload_type, + return channel_->SetSendTelephoneEventPayloadType(payload_type, payload_frequency) == 0; } bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->SendTelephoneEventOutband(event, duration_ms) == 0; + return channel_->SendTelephoneEventOutband(event, duration_ms) == 0; } void ChannelProxy::SetBitrate(int bitrate_bps, int64_t probing_interval_ms) { @@ -194,88 +193,82 @@ void ChannelProxy::SetBitrate(int bitrate_bps, int64_t probing_interval_ms) { // rules. // RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() || // module_process_thread_checker_.CalledOnValidThread()); - channel()->SetBitRate(bitrate_bps, probing_interval_ms); + channel_->SetBitRate(bitrate_bps, probing_interval_ms); } void ChannelProxy::SetReceiveCodecs( const std::map<int, SdpAudioFormat>& codecs) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetReceiveCodecs(codecs); + channel_->SetReceiveCodecs(codecs); } -void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) { +void ChannelProxy::SetSink(AudioSinkInterface* sink) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetSink(std::move(sink)); + channel_->SetSink(sink); } void ChannelProxy::SetInputMute(bool muted) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetInputMute(muted); + channel_->SetInputMute(muted); } void ChannelProxy::RegisterTransport(Transport* transport) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->RegisterTransport(transport); + channel_->RegisterTransport(transport); } void ChannelProxy::OnRtpPacket(const RtpPacketReceived& packet) { // May be called on either worker thread or network thread. - channel()->OnRtpPacket(packet); + channel_->OnRtpPacket(packet); } bool ChannelProxy::ReceivedRTCPPacket(const uint8_t* packet, size_t length) { // May be called on either worker thread or network thread. - return channel()->ReceivedRTCPPacket(packet, length) == 0; -} - -const rtc::scoped_refptr<AudioDecoderFactory>& - ChannelProxy::GetAudioDecoderFactory() const { - RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->GetAudioDecoderFactory(); + return channel_->ReceivedRTCPPacket(packet, length) == 0; } void ChannelProxy::SetChannelOutputVolumeScaling(float scaling) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetChannelOutputVolumeScaling(scaling); + channel_->SetChannelOutputVolumeScaling(scaling); } void ChannelProxy::SetRtcEventLog(RtcEventLog* event_log) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetRtcEventLog(event_log); + channel_->SetRtcEventLog(event_log); } AudioMixer::Source::AudioFrameInfo ChannelProxy::GetAudioFrameWithInfo( int sample_rate_hz, AudioFrame* audio_frame) { RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); - return channel()->GetAudioFrameWithInfo(sample_rate_hz, audio_frame); + return channel_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame); } int ChannelProxy::PreferredSampleRate() const { RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); - return channel()->PreferredSampleRate(); + return channel_->PreferredSampleRate(); } void ChannelProxy::ProcessAndEncodeAudio( std::unique_ptr<AudioFrame> audio_frame) { RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); - return channel()->ProcessAndEncodeAudio(std::move(audio_frame)); + return channel_->ProcessAndEncodeAudio(std::move(audio_frame)); } void ChannelProxy::SetTransportOverhead(int transport_overhead_per_packet) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetTransportOverhead(transport_overhead_per_packet); + channel_->SetTransportOverhead(transport_overhead_per_packet); } void ChannelProxy::AssociateSendChannel( const ChannelProxy& send_channel_proxy) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->set_associate_send_channel(send_channel_proxy.channel_owner_); + channel_->SetAssociatedSendChannel(send_channel_proxy.channel_.get()); } void ChannelProxy::DisassociateSendChannel() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->set_associate_send_channel(ChannelOwner(nullptr)); + channel_->SetAssociatedSendChannel(nullptr); } void ChannelProxy::GetRtpRtcp(RtpRtcp** rtp_rtcp, @@ -283,14 +276,14 @@ void ChannelProxy::GetRtpRtcp(RtpRtcp** rtp_rtcp, RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread()); RTC_DCHECK(rtp_rtcp); RTC_DCHECK(rtp_receiver); - int error = channel()->GetRtpRtcp(rtp_rtcp, rtp_receiver); + int error = channel_->GetRtpRtcp(rtp_rtcp, rtp_receiver); RTC_DCHECK_EQ(0, error); } uint32_t ChannelProxy::GetPlayoutTimestamp() const { RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_); unsigned int timestamp = 0; - int error = channel()->GetPlayoutTimestamp(timestamp); + int error = channel_->GetPlayoutTimestamp(timestamp); RTC_DCHECK(!error || timestamp == 0); return timestamp; } @@ -300,7 +293,7 @@ void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) { // Limit to range accepted by both VoE and ACM, so we're at least getting as // close as possible, instead of failing. delay_ms = rtc::SafeClamp(delay_ms, 0, 10000); - int error = channel()->SetMinimumPlayoutDelay(delay_ms); + int error = channel_->SetMinimumPlayoutDelay(delay_ms); if (0 != error) { RTC_LOG(LS_WARNING) << "Error setting minimum playout delay."; } @@ -308,57 +301,51 @@ void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) { void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->SetRtcpRttStats(rtcp_rtt_stats); + channel_->SetRtcpRttStats(rtcp_rtt_stats); } bool ChannelProxy::GetRecCodec(CodecInst* codec_inst) const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->GetRecCodec(*codec_inst) == 0; + return channel_->GetRecCodec(*codec_inst) == 0; } void ChannelProxy::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->OnTwccBasedUplinkPacketLossRate(packet_loss_rate); + channel_->OnTwccBasedUplinkPacketLossRate(packet_loss_rate); } void ChannelProxy::OnRecoverableUplinkPacketLossRate( float recoverable_packet_loss_rate) { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate); + channel_->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate); } std::vector<RtpSource> ChannelProxy::GetSources() const { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - return channel()->GetSources(); + return channel_->GetSources(); } void ChannelProxy::StartSend() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - int error = channel()->StartSend(); + int error = channel_->StartSend(); RTC_DCHECK_EQ(0, error); } void ChannelProxy::StopSend() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - channel()->StopSend(); + channel_->StopSend(); } void ChannelProxy::StartPlayout() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - int error = channel()->StartPlayout(); + int error = channel_->StartPlayout(); RTC_DCHECK_EQ(0, error); } void ChannelProxy::StopPlayout() { RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); - int error = channel()->StopPlayout(); + int error = channel_->StopPlayout(); RTC_DCHECK_EQ(0, error); } - -Channel* ChannelProxy::channel() const { - RTC_DCHECK(channel_owner_.channel()); - return channel_owner_.channel(); -} - } // namespace voe } // namespace webrtc diff --git a/voice_engine/channel_proxy.h b/voice_engine/channel_proxy.h index d4fa048a78..1d2c8ce043 100644 --- a/voice_engine/channel_proxy.h +++ b/voice_engine/channel_proxy.h @@ -11,6 +11,10 @@ #ifndef VOICE_ENGINE_CHANNEL_PROXY_H_ #define VOICE_ENGINE_CHANNEL_PROXY_H_ +#include <memory> +#include <string> +#include <vector> + #include "api/audio/audio_mixer.h" #include "api/audio_codecs/audio_encoder.h" #include "api/rtpreceiverinterface.h" @@ -19,11 +23,6 @@ #include "rtc_base/race_checker.h" #include "rtc_base/thread_checker.h" #include "voice_engine/channel.h" -#include "voice_engine/channel_manager.h" - -#include <memory> -#include <string> -#include <vector> namespace webrtc { @@ -52,7 +51,7 @@ namespace voe { class ChannelProxy : public RtpPacketSinkInterface { public: ChannelProxy(); - explicit ChannelProxy(const ChannelOwner& channel_owner); + explicit ChannelProxy(std::unique_ptr<Channel> channel); virtual ~ChannelProxy(); virtual bool SetEncoder(int payload_type, @@ -92,15 +91,13 @@ class ChannelProxy : public RtpPacketSinkInterface { virtual bool SendTelephoneEventOutband(int event, int duration_ms); virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms); virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs); - virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink); + virtual void SetSink(AudioSinkInterface* sink); virtual void SetInputMute(bool muted); virtual void RegisterTransport(Transport* transport); // Implements RtpPacketSinkInterface void OnRtpPacket(const RtpPacketReceived& packet) override; virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length); - virtual const rtc::scoped_refptr<AudioDecoderFactory>& - GetAudioDecoderFactory() const; virtual void SetChannelOutputVolumeScaling(float scaling); virtual void SetRtcEventLog(RtcEventLog* event_log); virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( @@ -127,8 +124,6 @@ class ChannelProxy : public RtpPacketSinkInterface { virtual void StopPlayout(); private: - Channel* channel() const; - // Thread checkers document and lock usage of some methods on voe::Channel to // specific threads we know about. The goal is to eventually split up // voe::Channel into parts with single-threaded semantics, and thereby reduce @@ -141,7 +136,7 @@ class ChannelProxy : public RtpPacketSinkInterface { // audio thread to another, but access is still sequential. rtc::RaceChecker audio_thread_race_checker_; rtc::RaceChecker video_capture_thread_race_checker_; - ChannelOwner channel_owner_; + std::unique_ptr<Channel> channel_; RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy); }; diff --git a/voice_engine/channel_unittest.cc b/voice_engine/channel_unittest.cc deleted file mode 100644 index 37dd9b54dd..0000000000 --- a/voice_engine/channel_unittest.cc +++ /dev/null @@ -1,15 +0,0 @@ -/* - * Copyright (c) 2011 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 "test/gtest.h" -#include "voice_engine/channel.h" - -// Empty test just to get coverage metrics. -TEST(ChannelTest, EmptyTestToGetCodeCoverage) {} diff --git a/voice_engine/include/voe_base.h b/voice_engine/include/voe_base.h index a76d2bc943..8924fae2e3 100644 --- a/voice_engine/include/voe_base.h +++ b/voice_engine/include/voe_base.h @@ -102,7 +102,6 @@ class WEBRTC_DLLEXPORT VoEBase { // decoder_factory member of the ACM config will be ignored (the decoder // factory set through Init() will always be used). // Returns channel ID or -1 in case of an error. - virtual int CreateChannel() = 0; virtual int CreateChannel(const ChannelConfig& config) = 0; // Deletes an existing channel and releases the utilized resources. diff --git a/voice_engine/shared_data.cc b/voice_engine/shared_data.cc deleted file mode 100644 index 420a7da2cf..0000000000 --- a/voice_engine/shared_data.cc +++ /dev/null @@ -1,48 +0,0 @@ -/* - * Copyright (c) 2012 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 "voice_engine/shared_data.h" - -#include "voice_engine/channel.h" - -namespace webrtc { - -namespace voe { - -static int32_t _gInstanceCounter = 0; - -SharedData::SharedData() - : _instanceId(++_gInstanceCounter), - _channelManager(_gInstanceCounter), - _audioDevicePtr(NULL), - _moduleProcessThreadPtr(ProcessThread::Create("VoiceProcessThread")), - encoder_queue_("AudioEncoderQueue") { -} - -SharedData::~SharedData() -{ - if (_audioDevicePtr) { - _audioDevicePtr->Release(); - } - _moduleProcessThreadPtr->Stop(); -} - -rtc::TaskQueue* SharedData::encoder_queue() { - RTC_DCHECK_RUN_ON(&construction_thread_); - return &encoder_queue_; -} - -void SharedData::set_audio_device( - const rtc::scoped_refptr<AudioDeviceModule>& audio_device) { - _audioDevicePtr = audio_device; -} -} // namespace voe - -} // namespace webrtc diff --git a/voice_engine/shared_data.h b/voice_engine/shared_data.h deleted file mode 100644 index f1364b9667..0000000000 --- a/voice_engine/shared_data.h +++ /dev/null @@ -1,60 +0,0 @@ -/* - * Copyright (c) 2012 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. - */ - -#ifndef VOICE_ENGINE_SHARED_DATA_H_ -#define VOICE_ENGINE_SHARED_DATA_H_ - -#include <memory> - -#include "modules/audio_device/include/audio_device.h" -#include "modules/utility/include/process_thread.h" -#include "rtc_base/criticalsection.h" -#include "rtc_base/scoped_ref_ptr.h" -#include "rtc_base/task_queue.h" -#include "rtc_base/thread_annotations.h" -#include "rtc_base/thread_checker.h" -#include "voice_engine/channel_manager.h" - -class ProcessThread; - -namespace webrtc { -namespace voe { - -class SharedData -{ -public: - // Public accessors. - uint32_t instance_id() const { return _instanceId; } - ChannelManager& channel_manager() { return _channelManager; } - AudioDeviceModule* audio_device() { return _audioDevicePtr.get(); } - void set_audio_device( - const rtc::scoped_refptr<AudioDeviceModule>& audio_device); - rtc::CriticalSection* crit_sec() { return &_apiCritPtr; } - ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); } - rtc::TaskQueue* encoder_queue(); - -protected: - rtc::ThreadChecker construction_thread_; - const uint32_t _instanceId; - rtc::CriticalSection _apiCritPtr; - ChannelManager _channelManager; - rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr; - std::unique_ptr<ProcessThread> _moduleProcessThreadPtr; - // |encoder_queue| is defined last to ensure all pending tasks are cancelled - // and deleted before any other members. - rtc::TaskQueue encoder_queue_ RTC_ACCESS_ON(construction_thread_); - - SharedData(); - virtual ~SharedData(); -}; - -} // namespace voe -} // namespace webrtc -#endif // VOICE_ENGINE_SHARED_DATA_H_ diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc index 88371eb240..a7eaf74ada 100644 --- a/voice_engine/voe_base_impl.cc +++ b/voice_engine/voe_base_impl.cc @@ -10,14 +10,6 @@ #include "voice_engine/voe_base_impl.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/include/audio_coding_module.h" -#include "modules/audio_device/audio_device_impl.h" -#include "rtc_base/format_macros.h" -#include "rtc_base/location.h" -#include "rtc_base/logging.h" -#include "voice_engine/channel.h" -#include "voice_engine/include/voe_errors.h" #include "voice_engine/voice_engine_impl.h" namespace webrtc { @@ -31,93 +23,7 @@ VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine) { return s; } -VoEBaseImpl::VoEBaseImpl(voe::SharedData* shared) - : shared_(shared) {} +VoEBaseImpl::VoEBaseImpl() {} -VoEBaseImpl::~VoEBaseImpl() { - TerminateInternal(); -} - -int VoEBaseImpl::Init( - AudioDeviceModule* audio_device, - AudioProcessing* audio_processing, - const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) { - RTC_DCHECK(audio_device); - rtc::CritScope cs(shared_->crit_sec()); - if (shared_->process_thread()) { - shared_->process_thread()->Start(); - } - - shared_->set_audio_device(audio_device); - - RTC_DCHECK(decoder_factory); - decoder_factory_ = decoder_factory; - - return 0; -} - -void VoEBaseImpl::Terminate() { - rtc::CritScope cs(shared_->crit_sec()); - TerminateInternal(); -} - -int VoEBaseImpl::CreateChannel() { - return CreateChannel(ChannelConfig()); -} - -int VoEBaseImpl::CreateChannel(const ChannelConfig& config) { - rtc::CritScope cs(shared_->crit_sec()); - ChannelConfig config_copy(config); - config_copy.acm_config.decoder_factory = decoder_factory_; - voe::ChannelOwner channel_owner = - shared_->channel_manager().CreateChannel(config_copy); - return InitializeChannel(&channel_owner); -} - -int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) { - if (channel_owner->channel()->SetEngineInformation( - *shared_->process_thread(), *shared_->audio_device(), - shared_->encoder_queue()) != 0) { - RTC_LOG(LS_ERROR) - << "CreateChannel() failed to associate engine and channel." - " Destroying channel."; - shared_->channel_manager().DestroyChannel( - channel_owner->channel()->ChannelId()); - return -1; - } else if (channel_owner->channel()->Init() != 0) { - RTC_LOG(LS_ERROR) - << "CreateChannel() failed to initialize channel. Destroying" - " channel."; - shared_->channel_manager().DestroyChannel( - channel_owner->channel()->ChannelId()); - return -1; - } - return channel_owner->channel()->ChannelId(); -} - -int VoEBaseImpl::DeleteChannel(int channel) { - rtc::CritScope cs(shared_->crit_sec()); - { - voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel); - voe::Channel* channelPtr = ch.channel(); - if (channelPtr == nullptr) { - RTC_LOG(LS_ERROR) << "DeleteChannel() failed to locate channel"; - return -1; - } - } - - shared_->channel_manager().DestroyChannel(channel); - return 0; -} - -void VoEBaseImpl::TerminateInternal() { - // Delete any remaining channel objects - shared_->channel_manager().DestroyAllChannels(); - - if (shared_->process_thread()) { - shared_->process_thread()->Stop(); - } - - shared_->set_audio_device(nullptr); -} +VoEBaseImpl::~VoEBaseImpl() {} } // namespace webrtc diff --git a/voice_engine/voe_base_impl.h b/voice_engine/voe_base_impl.h index a56e520cbb..37867766d1 100644 --- a/voice_engine/voe_base_impl.h +++ b/voice_engine/voe_base_impl.h @@ -13,38 +13,24 @@ #include "voice_engine/include/voe_base.h" -#include "modules/include/module_common_types.h" -#include "rtc_base/criticalsection.h" -#include "voice_engine/shared_data.h" - namespace webrtc { -class ProcessThread; - class VoEBaseImpl : public VoEBase { public: int Init( AudioDeviceModule* audio_device, AudioProcessing* audio_processing, - const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) override; - void Terminate() override; + const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) override { + return 0; + } + void Terminate() override {} - int CreateChannel() override; - int CreateChannel(const ChannelConfig& config) override; - int DeleteChannel(int channel) override; + int CreateChannel(const ChannelConfig& config) override { return 1; } + int DeleteChannel(int channel) override { return 0; } protected: - VoEBaseImpl(voe::SharedData* shared); + VoEBaseImpl(); ~VoEBaseImpl() override; - - private: - void TerminateInternal(); - - // Initialize channel by setting Engine Information then initializing - // channel. - int InitializeChannel(voe::ChannelOwner* channel_owner); - rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; - voe::SharedData* shared_; }; } // namespace webrtc diff --git a/voice_engine/voe_base_unittest.cc b/voice_engine/voe_base_unittest.cc deleted file mode 100644 index 2642a100d6..0000000000 --- a/voice_engine/voe_base_unittest.cc +++ /dev/null @@ -1,51 +0,0 @@ -/* - * Copyright (c) 2013 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 "voice_engine/include/voe_base.h" - -#include "api/audio_codecs/builtin_audio_decoder_factory.h" -#include "modules/audio_device/include/fake_audio_device.h" -#include "test/gtest.h" - -namespace webrtc { - -class VoEBaseTest : public ::testing::Test { - protected: - VoEBaseTest() - : voe_(VoiceEngine::Create()), - base_(VoEBase::GetInterface(voe_)) { - EXPECT_NE(nullptr, base_); - } - - ~VoEBaseTest() { - base_->Terminate(); - EXPECT_EQ(1, base_->Release()); - EXPECT_TRUE(VoiceEngine::Delete(voe_)); - } - - VoiceEngine* voe_; - VoEBase* base_; - FakeAudioDeviceModule adm_; -}; - -TEST_F(VoEBaseTest, InitWithExternalAudioDevice) { - EXPECT_EQ(0, - base_->Init(&adm_, nullptr, CreateBuiltinAudioDecoderFactory())); -} - -TEST_F(VoEBaseTest, CreateChannelAfterInit) { - EXPECT_EQ(0, - base_->Init(&adm_, nullptr, CreateBuiltinAudioDecoderFactory())); - int channelID = base_->CreateChannel(); - EXPECT_NE(channelID, -1); - EXPECT_EQ(0, base_->DeleteChannel(channelID)); -} - -} // namespace webrtc diff --git a/voice_engine/voice_engine_impl.cc b/voice_engine/voice_engine_impl.cc index b0ff0849ba..f47b4d7dbd 100644 --- a/voice_engine/voice_engine_impl.cc +++ b/voice_engine/voice_engine_impl.cc @@ -7,16 +7,6 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ - -#if defined(WEBRTC_ANDROID) -#include "modules/audio_device/android/audio_device_template.h" -#include "modules/audio_device/android/audio_record_jni.h" -#include "modules/audio_device/android/audio_track_jni.h" -#endif - -#include "modules/audio_coding/include/audio_coding_module.h" -#include "rtc_base/checks.h" -#include "voice_engine/channel_proxy.h" #include "voice_engine/voice_engine_impl.h" namespace webrtc { @@ -57,14 +47,6 @@ int VoiceEngineImpl::Release() { return new_ref; } -std::unique_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy( - int channel_id) { - RTC_DCHECK(channel_id >= 0); - rtc::CritScope cs(crit_sec()); - return std::unique_ptr<voe::ChannelProxy>( - new voe::ChannelProxy(channel_manager().GetChannel(channel_id))); -} - VoiceEngine* VoiceEngine::Create() { return GetVoiceEngine(); } diff --git a/voice_engine/voice_engine_impl.h b/voice_engine/voice_engine_impl.h index 917cc1aa68..41b72824bd 100644 --- a/voice_engine/voice_engine_impl.h +++ b/voice_engine/voice_engine_impl.h @@ -18,17 +18,12 @@ #include "voice_engine/voe_base_impl.h" namespace webrtc { -namespace voe { -class ChannelProxy; -} // namespace voe -class VoiceEngineImpl : public voe::SharedData, // Must be the first base class - public VoiceEngine, +class VoiceEngineImpl : public VoiceEngine, public VoEBaseImpl { public: VoiceEngineImpl() - : SharedData(), - VoEBaseImpl(this), + : VoEBaseImpl(), _ref_count(0) {} ~VoiceEngineImpl() override { assert(_ref_count.Value() == 0); } @@ -37,10 +32,6 @@ class VoiceEngineImpl : public voe::SharedData, // Must be the first base class // This implements the Release() method for all the inherited interfaces. int Release() override; - // Backdoor to access a voe::Channel object without a channel ID. This is only - // to be used while refactoring the VoE API! - virtual std::unique_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id); - // This is *protected* so that FakeVoiceEngine can inherit from the class and // manipulate the reference count. See: fake_voice_engine.h. protected: |