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APM: Add build flag to allow building WebRTC without APM

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This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.

Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
This commit is contained in:
Per Åhgren
2020-04-26 23:56:17 +02:00
committed by Commit Bot
parent 86bd33a1e7
commit cc73ed3e70
31 changed files with 1887 additions and 1403 deletions
Download Patch File Download Diff File Expand all files Collapse all files

373
call/call_unittest.cc
View File

@ -35,13 +35,15 @@
namespace {
struct CallHelper {
CallHelper() {
explicit CallHelper(bool use_null_audio_processing) {
task_queue_factory_ = webrtc::CreateDefaultTaskQueueFactory();
webrtc::AudioState::Config audio_state_config;
audio_state_config.audio_mixer =
new rtc::RefCountedObject<webrtc::test::MockAudioMixer>();
audio_state_config.audio_processing =
new rtc::RefCountedObject<webrtc::test::MockAudioProcessing>();
use_null_audio_processing
? nullptr
: new rtc::RefCountedObject<webrtc::test::MockAudioProcessing>();
audio_state_config.audio_device_module =
new rtc::RefCountedObject<webrtc::test::MockAudioDeviceModule>();
webrtc::Call::Config config(&event_log_);
@ -64,236 +66,261 @@ struct CallHelper {
namespace webrtc {
TEST(CallTest, ConstructDestruct) {
CallHelper call;
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
}
}
TEST(CallTest, CreateDestroy_AudioSendStream) {
CallHelper call;
MockTransport send_transport;
AudioSendStream::Config config(&send_transport);
config.rtp.ssrc = 42;
AudioSendStream* stream = call->CreateAudioSendStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyAudioSendStream(stream);
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
MockTransport send_transport;
AudioSendStream::Config config(&send_transport);
config.rtp.ssrc = 42;
AudioSendStream* stream = call->CreateAudioSendStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyAudioSendStream(stream);
}
}
TEST(CallTest, CreateDestroy_AudioReceiveStream) {
CallHelper call;
AudioReceiveStream::Config config;
MockTransport rtcp_send_transport;
config.rtp.remote_ssrc = 42;
config.rtcp_send_transport = &rtcp_send_transport;
config.decoder_factory =
new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyAudioReceiveStream(stream);
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
AudioReceiveStream::Config config;
MockTransport rtcp_send_transport;
config.rtp.remote_ssrc = 42;
config.rtcp_send_transport = &rtcp_send_transport;
config.decoder_factory =
new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyAudioReceiveStream(stream);
}
}
TEST(CallTest, CreateDestroy_AudioSendStreams) {
CallHelper call;
MockTransport send_transport;
AudioSendStream::Config config(&send_transport);
std::list<AudioSendStream*> streams;
for (int i = 0; i < 2; ++i) {
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
config.rtp.ssrc = ssrc;
AudioSendStream* stream = call->CreateAudioSendStream(config);
EXPECT_NE(stream, nullptr);
if (ssrc & 1) {
streams.push_back(stream);
} else {
streams.push_front(stream);
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
MockTransport send_transport;
AudioSendStream::Config config(&send_transport);
std::list<AudioSendStream*> streams;
for (int i = 0; i < 2; ++i) {
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
config.rtp.ssrc = ssrc;
AudioSendStream* stream = call->CreateAudioSendStream(config);
EXPECT_NE(stream, nullptr);
if (ssrc & 1) {
streams.push_back(stream);
} else {
streams.push_front(stream);
}
}
for (auto s : streams) {
call->DestroyAudioSendStream(s);
}
streams.clear();
}
for (auto s : streams) {
call->DestroyAudioSendStream(s);
}
streams.clear();
}
}
TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
CallHelper call;
AudioReceiveStream::Config config;
MockTransport rtcp_send_transport;
config.rtcp_send_transport = &rtcp_send_transport;
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) {
config.rtp.remote_ssrc = ssrc;
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
EXPECT_NE(stream, nullptr);
if (ssrc & 1) {
streams.push_back(stream);
} else {
streams.push_front(stream);
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
AudioReceiveStream::Config config;
MockTransport rtcp_send_transport;
config.rtcp_send_transport = &rtcp_send_transport;
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) {
config.rtp.remote_ssrc = ssrc;
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
EXPECT_NE(stream, nullptr);
if (ssrc & 1) {
streams.push_back(stream);
} else {
streams.push_front(stream);
}
}
for (auto s : streams) {
call->DestroyAudioReceiveStream(s);
}
streams.clear();
}
for (auto s : streams) {
call->DestroyAudioReceiveStream(s);
}
streams.clear();
}
}
TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
CallHelper call;
AudioReceiveStream::Config recv_config;
MockTransport rtcp_send_transport;
recv_config.rtp.remote_ssrc = 42;
recv_config.rtp.local_ssrc = 777;
recv_config.rtcp_send_transport = &rtcp_send_transport;
recv_config.decoder_factory =
new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
EXPECT_NE(recv_stream, nullptr);
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
AudioReceiveStream::Config recv_config;
MockTransport rtcp_send_transport;
recv_config.rtp.remote_ssrc = 42;
recv_config.rtp.local_ssrc = 777;
recv_config.rtcp_send_transport = &rtcp_send_transport;
recv_config.decoder_factory =
new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* recv_stream =
call->CreateAudioReceiveStream(recv_config);
EXPECT_NE(recv_stream, nullptr);
MockTransport send_transport;
AudioSendStream::Config send_config(&send_transport);
send_config.rtp.ssrc = 777;
AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
EXPECT_NE(send_stream, nullptr);
MockTransport send_transport;
AudioSendStream::Config send_config(&send_transport);
send_config.rtp.ssrc = 777;
AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
EXPECT_NE(send_stream, nullptr);
internal::AudioReceiveStream* internal_recv_stream =
static_cast<internal::AudioReceiveStream*>(recv_stream);
EXPECT_EQ(send_stream,
internal_recv_stream->GetAssociatedSendStreamForTesting());
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());
call->DestroyAudioSendStream(send_stream);
EXPECT_EQ(nullptr,
internal_recv_stream->GetAssociatedSendStreamForTesting());
call->DestroyAudioReceiveStream(recv_stream);
call->DestroyAudioReceiveStream(recv_stream);
}
}
TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
CallHelper call;
MockTransport send_transport;
AudioSendStream::Config send_config(&send_transport);
send_config.rtp.ssrc = 777;
AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
EXPECT_NE(send_stream, nullptr);
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
MockTransport send_transport;
AudioSendStream::Config send_config(&send_transport);
send_config.rtp.ssrc = 777;
AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
EXPECT_NE(send_stream, nullptr);
AudioReceiveStream::Config recv_config;
MockTransport rtcp_send_transport;
recv_config.rtp.remote_ssrc = 42;
recv_config.rtp.local_ssrc = 777;
recv_config.rtcp_send_transport = &rtcp_send_transport;
recv_config.decoder_factory =
new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
EXPECT_NE(recv_stream, nullptr);
AudioReceiveStream::Config recv_config;
MockTransport rtcp_send_transport;
recv_config.rtp.remote_ssrc = 42;
recv_config.rtp.local_ssrc = 777;
recv_config.rtcp_send_transport = &rtcp_send_transport;
recv_config.decoder_factory =
new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* recv_stream =
call->CreateAudioReceiveStream(recv_config);
EXPECT_NE(recv_stream, nullptr);
internal::AudioReceiveStream* internal_recv_stream =
static_cast<internal::AudioReceiveStream*>(recv_stream);
EXPECT_EQ(send_stream,
internal_recv_stream->GetAssociatedSendStreamForTesting());
internal::AudioReceiveStream* internal_recv_stream =
static_cast<internal::AudioReceiveStream*>(recv_stream);
EXPECT_EQ(send_stream,
internal_recv_stream->GetAssociatedSendStreamForTesting());
call->DestroyAudioReceiveStream(recv_stream);
call->DestroyAudioReceiveStream(recv_stream);
call->DestroyAudioSendStream(send_stream);
call->DestroyAudioSendStream(send_stream);
}
}
TEST(CallTest, CreateDestroy_FlexfecReceiveStream) {
CallHelper call;
MockTransport rtcp_send_transport;
FlexfecReceiveStream::Config config(&rtcp_send_transport);
config.payload_type = 118;
config.remote_ssrc = 38837212;
config.protected_media_ssrcs = {27273};
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
MockTransport rtcp_send_transport;
FlexfecReceiveStream::Config config(&rtcp_send_transport);
config.payload_type = 118;
config.remote_ssrc = 38837212;
config.protected_media_ssrcs = {27273};
FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyFlexfecReceiveStream(stream);
FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyFlexfecReceiveStream(stream);
}
}
TEST(CallTest, CreateDestroy_FlexfecReceiveStreams) {
CallHelper call;
MockTransport rtcp_send_transport;
FlexfecReceiveStream::Config config(&rtcp_send_transport);
config.payload_type = 118;
std::list<FlexfecReceiveStream*> streams;
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
MockTransport rtcp_send_transport;
FlexfecReceiveStream::Config config(&rtcp_send_transport);
config.payload_type = 118;
std::list<FlexfecReceiveStream*> streams;
for (int i = 0; i < 2; ++i) {
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
config.remote_ssrc = ssrc;
config.protected_media_ssrcs = {ssrc + 1};
FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
if (ssrc & 1) {
streams.push_back(stream);
} else {
streams.push_front(stream);
for (int i = 0; i < 2; ++i) {
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
config.remote_ssrc = ssrc;
config.protected_media_ssrcs = {ssrc + 1};
FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
if (ssrc & 1) {
streams.push_back(stream);
} else {
streams.push_front(stream);
}
}
for (auto s : streams) {
call->DestroyFlexfecReceiveStream(s);
}
streams.clear();
}
for (auto s : streams) {
call->DestroyFlexfecReceiveStream(s);
}
streams.clear();
}
}
TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) {
CallHelper call;
MockTransport rtcp_send_transport;
FlexfecReceiveStream::Config config(&rtcp_send_transport);
config.payload_type = 118;
config.protected_media_ssrcs = {1324234};
FlexfecReceiveStream* stream;
std::list<FlexfecReceiveStream*> streams;
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
MockTransport rtcp_send_transport;
FlexfecReceiveStream::Config config(&rtcp_send_transport);
config.payload_type = 118;
config.protected_media_ssrcs = {1324234};
FlexfecReceiveStream* stream;
std::list<FlexfecReceiveStream*> streams;
config.remote_ssrc = 838383;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
config.remote_ssrc = 838383;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
config.remote_ssrc = 424993;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
config.remote_ssrc = 424993;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
config.remote_ssrc = 99383;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
config.remote_ssrc = 99383;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
config.remote_ssrc = 5548;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
config.remote_ssrc = 5548;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
for (auto s : streams) {
call->DestroyFlexfecReceiveStream(s);
for (auto s : streams) {
call->DestroyFlexfecReceiveStream(s);
}
}
}
TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
constexpr uint32_t kSSRC = 12345;
CallHelper call;
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
MockTransport send_transport;
AudioSendStream::Config config(&send_transport);
config.rtp.ssrc = ssrc;
AudioSendStream* stream = call->CreateAudioSendStream(config);
const RtpState rtp_state =
static_cast<internal::AudioSendStream*>(stream)->GetRtpState();
call->DestroyAudioSendStream(stream);
return rtp_state;
};
auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
MockTransport send_transport;
AudioSendStream::Config config(&send_transport);
config.rtp.ssrc = ssrc;
AudioSendStream* stream = call->CreateAudioSendStream(config);
const RtpState rtp_state =
static_cast<internal::AudioSendStream*>(stream)->GetRtpState();
call->DestroyAudioSendStream(stream);
return rtp_state;
};
const RtpState rtp_state1 = create_stream_and_get_rtp_state(kSSRC);
const RtpState rtp_state2 = create_stream_and_get_rtp_state(kSSRC);
const RtpState rtp_state1 = create_stream_and_get_rtp_state(kSSRC);
const RtpState rtp_state2 = create_stream_and_get_rtp_state(kSSRC);
EXPECT_EQ(rtp_state1.sequence_number, rtp_state2.sequence_number);
EXPECT_EQ(rtp_state1.start_timestamp, rtp_state2.start_timestamp);
EXPECT_EQ(rtp_state1.timestamp, rtp_state2.timestamp);
EXPECT_EQ(rtp_state1.capture_time_ms, rtp_state2.capture_time_ms);
EXPECT_EQ(rtp_state1.last_timestamp_time_ms,
rtp_state2.last_timestamp_time_ms);
EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
EXPECT_EQ(rtp_state1.sequence_number, rtp_state2.sequence_number);
EXPECT_EQ(rtp_state1.start_timestamp, rtp_state2.start_timestamp);
EXPECT_EQ(rtp_state1.timestamp, rtp_state2.timestamp);
EXPECT_EQ(rtp_state1.capture_time_ms, rtp_state2.capture_time_ms);
EXPECT_EQ(rtp_state1.last_timestamp_time_ms,
rtp_state2.last_timestamp_time_ms);
EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
}
}
} // namespace webrtc