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Bug 786234 - Part 5: More detailed test-cases. r=ethanhugg

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This commit is contained in:
Byron Campen [:bwc] 2014-01-17 17:11:00 -08:00
parent 90a124346a
commit 3d95737fdf
1 changed files with 116 additions and 38 deletions
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154
media/webrtc/signaling/test/mediapipeline_unittest.cpp
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@ -114,7 +114,7 @@ class TransportInfo {
dtls_ = nullptr;
}
void Stop() {
void Shutdown() {
if (loopback_) {
loopback_->Disconnect();
}
@ -168,11 +168,6 @@ class TestAgent {
void StopInt() {
audio_->GetStream()->Stop();
audio_rtp_transport_.Stop();
audio_rtcp_transport_.Stop();
bundle_transport_.Stop();
if (audio_pipeline_)
audio_pipeline_->ShutdownTransport_s();
}
void Stop() {
@ -184,14 +179,60 @@ class TestAgent {
mozilla::SyncRunnable::DispatchToThread(
test_utils->sts_target(),
WrapRunnable(this, &TestAgent::StopInt));
}
audio_pipeline_ = nullptr;
void Shutdown_s() {
audio_rtp_transport_.Shutdown();
audio_rtcp_transport_.Shutdown();
bundle_transport_.Shutdown();
if (audio_pipeline_)
audio_pipeline_->ShutdownTransport_s();
}
void Shutdown() {
if (audio_pipeline_)
audio_pipeline_->ShutdownMedia_m();
mozilla::SyncRunnable::DispatchToThread(
test_utils->sts_target(),
WrapRunnable(this, &TestAgent::Shutdown_s));
}
uint32_t GetRemoteSSRC() {
uint32_t res = 0;
audio_conduit_->GetRemoteSSRC(&res);
return res;
}
uint32_t GetLocalSSRC() {
uint32_t res = 0;
audio_conduit_->GetLocalSSRC(&res);
return res;
}
int GetAudioRtpCountSent() {
return audio_pipeline_->rtp_packets_sent();
}
int GetAudioRtpCountReceived() {
return audio_pipeline_->rtp_packets_received();
}
int GetAudioRtcpCountSent() {
return audio_pipeline_->rtcp_packets_sent();
}
int GetAudioRtcpCountReceived() {
return audio_pipeline_->rtcp_packets_received();
}
protected:
mozilla::AudioCodecConfig audio_config_;
mozilla::RefPtr<mozilla::MediaSessionConduit> audio_conduit_;
nsRefPtr<DOMMediaStream> audio_;
// TODO(bcampen@mozilla.com): Right now this does not let us test RTCP in
// both directions; only the sender's RTCP is sent, but the receiver should
// be sending it too.
mozilla::RefPtr<mozilla::MediaPipeline> audio_pipeline_;
TransportInfo audio_rtp_transport_;
TransportInfo audio_rtcp_transport_;
@ -238,14 +279,6 @@ class TestAgentSend : public TestAgent {
audio_pipeline_->Init();
}
int GetAudioRtpCount() {
return audio_pipeline_->rtp_packets_sent();
}
int GetAudioRtcpCount() {
return audio_pipeline_->rtcp_packets_received();
}
void SetUsingBundle(bool use_bundle) {
use_bundle_ = use_bundle;
}
@ -286,6 +319,7 @@ class TestAgentReceive : public TestAgent {
RefPtr<TransportFlow> bundle_transport;
if (bundle_filter_) {
bundle_transport = bundle_transport_.flow_;
bundle_filter_->AddLocalSSRC(GetLocalSSRC());
}
audio_pipeline_ = new mozilla::MediaPipelineReceiveAudio(
@ -303,14 +337,6 @@ class TestAgentReceive : public TestAgent {
audio_pipeline_->Init();
}
int GetAudioRtpCount() {
return audio_pipeline_->rtp_packets_received();
}
int GetAudioRtcpCount() {
return audio_pipeline_->rtcp_packets_sent();
}
void SetBundleFilter(nsAutoPtr<MediaPipelineFilter> filter) {
bundle_filter_ = filter;
}
@ -323,6 +349,7 @@ class TestAgentReceive : public TestAgent {
nsAutoPtr<MediaPipelineFilter> filter) {
audio_pipeline_->UpdateFilterFromRemoteDescription_s(filter);
}
private:
nsAutoPtr<MediaPipelineFilter> bundle_filter_;
};
@ -333,6 +360,8 @@ class MediaPipelineTest : public ::testing::Test {
~MediaPipelineTest() {
p1_.Stop();
p2_.Stop();
p1_.Shutdown();
p2_.Shutdown();
}
// Setup transport.
@ -362,7 +391,9 @@ class MediaPipelineTest : public ::testing::Test {
nsAutoPtr<MediaPipelineFilter> localFilter =
nsAutoPtr<MediaPipelineFilter>(nullptr),
nsAutoPtr<MediaPipelineFilter> remoteFilter =
nsAutoPtr<MediaPipelineFilter>(nullptr)) {
nsAutoPtr<MediaPipelineFilter>(nullptr),
unsigned int ms_until_answer = 500,
unsigned int ms_of_traffic_after_answer = 10000) {
// We do not support testing bundle without rtcp mux, since that doesn't
// make any sense.
@ -386,16 +417,20 @@ class MediaPipelineTest : public ::testing::Test {
p1_.Start();
// Simulate pre-answer traffic
PR_Sleep(500);
PR_Sleep(ms_until_answer);
mozilla::SyncRunnable::DispatchToThread(
test_utils->sts_target(),
WrapRunnable(&p2_, &TestAgentReceive::SetUsingBundle_s, bundle));
if (bundle) {
// Leaving remoteFilter not set implies we want to test sunny-day
if (!remoteFilter) {
remoteFilter = new MediaPipelineFilter;
// Might not be safe, strictly speaking.
remoteFilter->AddRemoteSSRC(p1_.GetLocalSSRC());
}
mozilla::SyncRunnable::DispatchToThread(
test_utils->sts_target(),
WrapRunnable(&p2_,
@ -405,26 +440,43 @@ class MediaPipelineTest : public ::testing::Test {
// wait for some RTP/RTCP tx and rx to happen
PR_Sleep(10000);
if (bundle) {
// Filter should have eaten everything
ASSERT_EQ(0, p2_.GetAudioRtpCount());
} else {
ASSERT_GE(p1_.GetAudioRtpCount(), 40);
ASSERT_GE(p2_.GetAudioRtpCount(), 40);
ASSERT_GE(p2_.GetAudioRtcpCount(), 1);
}
PR_Sleep(ms_of_traffic_after_answer);
p1_.Stop();
p2_.Stop();
// wait for any packets in flight to arrive
PR_Sleep(100);
p1_.Shutdown();
p2_.Shutdown();
if (!bundle) {
// If we are doing bundle, allow the test-case to do this checking.
ASSERT_GE(p1_.GetAudioRtpCountSent(), 40);
ASSERT_EQ(p1_.GetAudioRtpCountReceived(), p2_.GetAudioRtpCountSent());
ASSERT_EQ(p1_.GetAudioRtpCountSent(), p2_.GetAudioRtpCountReceived());
// Calling ShutdownMedia_m on both pipelines does not stop the flow of
// RTCP. So, we might be off by one here.
ASSERT_LE(p2_.GetAudioRtcpCountReceived(), p1_.GetAudioRtcpCountSent());
ASSERT_GE(p2_.GetAudioRtcpCountReceived() + 1, p1_.GetAudioRtcpCountSent());
}
}
void TestAudioReceiverOffersBundle(bool bundle_accepted,
nsAutoPtr<MediaPipelineFilter> localFilter,
nsAutoPtr<MediaPipelineFilter> remoteFilter =
nsAutoPtr<MediaPipelineFilter>(nullptr)) {
TestAudioSend(true, bundle_accepted, localFilter, remoteFilter);
nsAutoPtr<MediaPipelineFilter>(nullptr),
unsigned int ms_until_answer = 500,
unsigned int ms_of_traffic_after_answer = 10000) {
TestAudioSend(true,
bundle_accepted,
localFilter,
remoteFilter,
ms_until_answer,
ms_of_traffic_after_answer);
}
protected:
TestAgentSend p1_;
@ -829,7 +881,33 @@ TEST_F(MediaPipelineTest, TestAudioSendBundleOfferedAndDeclined) {
TEST_F(MediaPipelineTest, TestAudioSendBundleOfferedAndAccepted) {
nsAutoPtr<MediaPipelineFilter> filter(new MediaPipelineFilter);
TestAudioReceiverOffersBundle(true, filter);
// These durations have to be _extremely_ long to have any assurance that
// some RTCP will be sent at all. This is because the first RTCP packet
// is sometimes sent before the transports are ready, which causes it to
// be dropped.
TestAudioReceiverOffersBundle(true,
filter,
// We do not specify the filter for the remote description, so it will be
// set to something sane after a short time.
nsAutoPtr<MediaPipelineFilter>(),
10000,
10000);
// Some packets should have been dropped, but not all
ASSERT_GT(p1_.GetAudioRtpCountSent(), p2_.GetAudioRtpCountReceived());
ASSERT_GT(p2_.GetAudioRtpCountReceived(), 40);
ASSERT_GT(p1_.GetAudioRtcpCountSent(), 1);
ASSERT_GT(p1_.GetAudioRtcpCountSent(), p2_.GetAudioRtcpCountReceived());
ASSERT_GT(p2_.GetAudioRtcpCountReceived(), 0);
}
TEST_F(MediaPipelineTest, TestAudioSendBundleOfferedAndAcceptedEmptyFilter) {
nsAutoPtr<MediaPipelineFilter> filter(new MediaPipelineFilter);
nsAutoPtr<MediaPipelineFilter> bad_answer_filter(new MediaPipelineFilter);
TestAudioReceiverOffersBundle(true, filter, bad_answer_filter);
// Filter is empty, so should drop everything.
ASSERT_EQ(0, p2_.GetAudioRtpCountReceived());
ASSERT_EQ(0, p2_.GetAudioRtcpCountReceived());
}
} // end namespace