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}diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 0fd0ac8..f7b1cb5 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -25,7 +25,6 @@
"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 3c53818..bfa125a 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 @@
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 @@
}
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)
+ : 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)
- : audio_state_(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 @@
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::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 @@
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 @@
// 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 0c843be..999039f 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 @@
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 @@
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 @@
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 @@
const Config& new_config,
bool first_time);
- VoiceEngine* voice_engine() const;
AudioState* audio_state() const;
rtc::ThreadChecker worker_thread_checker_;
@@ -102,6 +111,7 @@
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 e03fbd5..062e71f 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 @@
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 @@
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;
@@ -135,20 +118,25 @@
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>;
}
- 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_;
+ 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_)));
}
+ AudioReceiveStream::Config& config() { return stream_config_; }
+ rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
+ MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
+
void SetupMockForGetStats() {
using testing::DoAll;
using testing::SetArgPointee;
@@ -174,9 +162,7 @@
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 @@
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 @@
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, 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());
+ ConfigHelper helper1;
+ ConfigHelper helper2(helper1.audio_mixer());
+ auto recv_stream1 = helper1.CreateAudioReceiveStream();
+ auto recv_stream2 = helper2.CreateAudioReceiveStream();
- EXPECT_CALL(*helper.channel_proxy(), StartPlayout()).Times(1);
- EXPECT_CALL(*helper.channel_proxy(), StopPlayout()).Times(1);
- EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream1))
+ 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 @@
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 2e5654a..d207dbd 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 @@
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 @@
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 @@
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 @@
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 @@
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 9060e64..bab8f80 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 @@
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 @@
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 5afdffe..4644bf4 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::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 @@
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 @@
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 @@
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 @@
}));
}
- RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; }
-
void SetupMockForSendTelephoneEvent() {
EXPECT_TRUE(channel_proxy_);
EXPECT_CALL(*channel_proxy_,
@@ -305,7 +302,6 @@
}
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 @@
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 @@
"{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, 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, 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 @@
*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 @@
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, 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 @@
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 @@
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 9d980e8..24a7ce8 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -27,7 +27,6 @@
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 @@
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 3a92d42..d4e4e3f 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 @@
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 @@
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 abae985..b4452ad 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 @@
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 @@
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 @@
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 7aa2d1d..0000000
--- 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_