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webrtc_m130/call/call_unittest.cc
Tommi 553c869c58 Start consolidating management/querying of stats on the Call thread. 
Call is instantiated on what we traditionally call the 'worker thread'
in PeerConnection terms. Call statistics are however gathered, processed
and reported in a number of different ways, which results in a lot of
locking, which is also unpredictable due to the those actions themselves
contending with other parts of the system.

Designating the worker thread as the general owner of the stats, helps
us keeps things regular and avoids loading unrelated task queues/threads
with reporting things like histograms or locking up due to a call to
GetStats().

This is a reland of remaining changes from https://webrtc-review.googlesource.com/c/src/+/172847:
This applies the changes from the above CL to the forked files and
switches call.cc over to using the forked implementation.

Bug: webrtc:11489
Change-Id: I93ad560500806ddd0e6df1448b1bcf5a1aae7583
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174000
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31186}
2020-05-08 07:24:39 +00:00

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/*
* 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.
*/
#include "call/call.h"
#include <list>
#include <map>
#include <memory>
#include <utility>
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/mock_audio_mixer.h"
#include "api/transport/field_trial_based_config.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
#include "call/audio_state.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "test/fake_encoder.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
#include "test/mock_transport.h"
#include "test/run_loop.h"
namespace {
struct 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 =
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_);
config.audio_state = webrtc::AudioState::Create(audio_state_config);
config.task_queue_factory = task_queue_factory_.get();
config.trials = &field_trials_;
call_.reset(webrtc::Call::Create(config));
}
webrtc::Call* operator->() { return call_.get(); }
private:
webrtc::test::RunLoop loop_;
webrtc::RtcEventLogNull event_log_;
webrtc::FieldTrialBasedConfig field_trials_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
std::unique_ptr<webrtc::Call> call_;
};
} // namespace
namespace webrtc {
TEST(CallTest, ConstructDestruct) {
for (bool use_null_audio_processing : {false, true}) {
CallHelper call(use_null_audio_processing);
}
}
TEST(CallTest, CreateDestroy_AudioSendStream) {
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) {
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) {
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();
}
}
}
TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
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();
}
}
}
TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
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);
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->DestroyAudioReceiveStream(recv_stream);
}
}
TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
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);
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);
}
}
TEST(CallTest, CreateDestroy_FlexfecReceiveStream) {
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);
}
}
TEST(CallTest, CreateDestroy_FlexfecReceiveStreams) {
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 (auto s : streams) {
call->DestroyFlexfecReceiveStream(s);
}
streams.clear();
}
}
}
TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) {
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 = 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 = 5548;
stream = call->CreateFlexfecReceiveStream(config);
EXPECT_NE(stream, nullptr);
streams.push_back(stream);
for (auto s : streams) {
call->DestroyFlexfecReceiveStream(s);
}
}
}
TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
constexpr uint32_t kSSRC = 12345;
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;
};
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);
}
}
} // namespace webrtc
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