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/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