diff --git a/media/webrtc/signaling/test/FakeMediaStreams.h b/media/webrtc/signaling/test/FakeMediaStreams.h
index f15656a1d876da351ebf932a2b1d4290e1103619..2f1ec88be035c09a814961c704a26f06f72c8083 100644
--- a/media/webrtc/signaling/test/FakeMediaStreams.h
+++ b/media/webrtc/signaling/test/FakeMediaStreams.h
@@ -95,13 +95,46 @@ class Fake_SourceMediaStream : public Fake_MediaStream {
public:
Fake_SourceMediaStream() : mSegmentsAdded(0),
mPullEnabled(false),
+ mStop(false),
mPeriodic(new Fake_MediaPeriodic(this)) {}
void AddTrack(mozilla::TrackID aID, mozilla::TrackRate aRate, mozilla::TrackTicks aStart,
mozilla::MediaSegment* aSegment) {}
void AppendToTrack(mozilla::TrackID aID, mozilla::MediaSegment* aSegment) {
- ++mSegmentsAdded;
+ bool nonZeroSample = false;
+ MOZ_ASSERT(aSegment);
+ if(aSegment->GetType() == mozilla::MediaSegment::AUDIO) {
+ //On audio segment append, we verify for validity
+ //of the audio samples.
+ mozilla::AudioSegment* audio =
+ static_cast<mozilla::AudioSegment*>(aSegment);
+ mozilla::AudioSegment::ChunkIterator iter(*audio);
+ while(!iter.IsEnded()) {
+ mozilla::AudioChunk& chunk = *(iter);
+ MOZ_ASSERT(chunk.mBuffer);
+ const int16_t* buf =
+ static_cast<const int16_t*>(chunk.mBuffer->Data());
+ for(int i=0; i<chunk.mDuration; i++) {
+ if(buf[i]) {
+ //atleast one non-zero sample found.
+ nonZeroSample = true;
+ break;
+ }
+ }
+ //process next chunk
+ iter.Next();
+ }
+ if(nonZeroSample) {
+ //we increment segments count if
+ //atleast one non-zero samples was found.
+ ++mSegmentsAdded;
+ }
+ } else {
+ //in the case of video segment appended, we just increase the
+ //segment count.
+ ++mSegmentsAdded;
+ }
}
void AdvanceKnownTracksTime(mozilla::StreamTime aKnownTime) {}
@@ -109,6 +142,10 @@ class Fake_SourceMediaStream : public Fake_MediaStream {
void SetPullEnabled(bool aEnabled) {
mPullEnabled = aEnabled;
}
+ //Don't pull anymore data,if mStop is true.
+ void StopStream() {
+ mStop = true;
+ }
virtual Fake_SourceMediaStream *AsSourceStream() { return this; }
@@ -124,6 +161,7 @@ class Fake_SourceMediaStream : public Fake_MediaStream {
protected:
int mSegmentsAdded;
bool mPullEnabled;
+ bool mStop;
nsRefPtr<Fake_MediaPeriodic> mPeriodic;
nsCOMPtr<nsITimer> mTimer;
};
@@ -202,9 +240,17 @@ class Fake_MediaStreamBase : public Fake_MediaStream {
class Fake_AudioStreamSource : public Fake_MediaStreamBase {
public:
- Fake_AudioStreamSource() : Fake_MediaStreamBase() {}
-
+ Fake_AudioStreamSource() : Fake_MediaStreamBase(),
+ mCount(0),
+ mStop(false) {}
+ //Signaling Agent indicates us to stop generating
+ //further audio.
+ void StopStream() {
+ mStop = true;
+ }
virtual void Periodic();
+ int mCount;
+ bool mStop;
};
class Fake_VideoStreamSource : public Fake_MediaStreamBase {
diff --git a/media/webrtc/signaling/test/FakeMediaStreamsImpl.h b/media/webrtc/signaling/test/FakeMediaStreamsImpl.h
index e48047c6b6cbe87e51e0e3df91c5f147176c8152..a6187ee0849c9e44c955f1a901921f0271a094c1 100644
--- a/media/webrtc/signaling/test/FakeMediaStreamsImpl.h
+++ b/media/webrtc/signaling/test/FakeMediaStreamsImpl.h
@@ -10,6 +10,9 @@
#include "nspr.h"
#include "nsError.h"
+static const int AUDIO_BUFFER_SIZE = 1600;
+static const int NUM_CHANNELS = 2;
+
NS_IMPL_THREADSAFE_ISUPPORTS1(Fake_nsDOMMediaStream, nsIDOMMediaStream)
// DOM Media stream
@@ -42,7 +45,10 @@ nsresult Fake_SourceMediaStream::Stop() {
}
void Fake_SourceMediaStream::Periodic() {
- if (mPullEnabled) {
+ // Pull more audio-samples iff pulling is enabled
+ // and we are not asked by the signaling agent to stop
+ //pulling data.
+ if (mPullEnabled && !mStop) {
for (std::set<Fake_MediaStreamListener *>::iterator it =
mListeners.begin(); it != mListeners.end(); ++it) {
(*it)->NotifyPull(NULL, mozilla::MillisecondsToMediaTime(10));
@@ -50,7 +56,6 @@ void Fake_SourceMediaStream::Periodic() {
}
}
-
// Fake_MediaStreamBase
nsresult Fake_MediaStreamBase::Start() {
mTimer = do_CreateInstance(NS_TIMER_CONTRACTID);
@@ -71,11 +76,29 @@ nsresult Fake_MediaStreamBase::Stop() {
// Fake_AudioStreamSource
void Fake_AudioStreamSource::Periodic() {
+ //Are we asked to stop pumping audio samples ?
+ if(mStop) {
+ return;
+ }
+ //Generate Signed 16 Bit Audio samples
+ nsRefPtr<mozilla::SharedBuffer> samples =
+ mozilla::SharedBuffer::Create(AUDIO_BUFFER_SIZE * NUM_CHANNELS * sizeof(int16_t));
+ for(int i=0; i<(1600*2); i++) {
+ //saw tooth audio sample
+ reinterpret_cast<int16_t *>(samples->Data())[i] =
+ ((mCount % 8) * 4000) - (7*4000)/2;
+ mCount++;
+ }
+
mozilla::AudioSegment segment;
segment.Init(1);
- segment.InsertNullDataAtStart(160);
+ segment.AppendFrames(samples.forget(),
+ AUDIO_BUFFER_SIZE,
+ 0,
+ AUDIO_BUFFER_SIZE,
+ mozilla::AUDIO_FORMAT_S16);
- for (std::set<Fake_MediaStreamListener *>::iterator it = mListeners.begin();
+ for(std::set<Fake_MediaStreamListener *>::iterator it = mListeners.begin();
it != mListeners.end(); ++it) {
(*it)->NotifyQueuedTrackChanges(NULL, // Graph
0, // TrackID
diff --git a/media/webrtc/signaling/test/signaling_unittests.cpp b/media/webrtc/signaling/test/signaling_unittests.cpp
index d053a184388f4dea084afb79002c0c297fc02b59..2348158537739cc387392eb32c64b67fa51f044f 100644
--- a/media/webrtc/signaling/test/signaling_unittests.cpp
+++ b/media/webrtc/signaling/test/signaling_unittests.cpp
@@ -677,6 +677,22 @@ void CreateAnswer(sipcc::MediaConstraints& constraints, std::string offer,
domMediaStream_->GetStream())->GetSegmentsAdded();
}
+ //Stops generating new audio data for transmission.
+ //Should be called before Cleanup of the peer connection.
+ void CloseSendStreams() {
+ static_cast<Fake_AudioStreamSource*>(
+ domMediaStream_->GetStream())->StopStream();
+ }
+
+ //Stops pulling audio data off the receivers.
+ //Should be called before Cleanup of the peer connection.
+ void CloseReceiveStreams() {
+ std::vector<nsDOMMediaStream *> streams =
+ pObserver->GetStreams();
+ for(int i=0; i < streams.size(); i++) {
+ streams[i]->GetStream()->AsSourceStream()->StopStream();
+ }
+ }
public:
mozilla::RefPtr<sipcc::PeerConnectionImpl> pc;
@@ -1189,6 +1205,8 @@ TEST_F(SignalingTest, OfferModifiedAnswer)
OfferModifiedAnswer(constraints, constraints, SHOULD_SENDRECV_AV,
SHOULD_SENDRECV_AV);
PR_Sleep(kDefaultTimeout * 2); // Wait for completion
+ a1_.CloseSendStreams();
+ a2_.CloseReceiveStreams();
}
TEST_F(SignalingTest, FullCall)
@@ -1199,6 +1217,8 @@ TEST_F(SignalingTest, FullCall)
PR_Sleep(kDefaultTimeout * 2); // Wait for some data to get written
+ a1_.CloseSendStreams();
+ a2_.CloseReceiveStreams();
// Check that we wrote a bunch of data
ASSERT_GE(a1_.GetPacketsSent(0), 40);
//ASSERT_GE(a2_.GetPacketsSent(0), 40);
@@ -1214,6 +1234,8 @@ TEST_F(SignalingTest, FullCallTrickle)
PR_Sleep(kDefaultTimeout * 2); // Wait for some data to get written
+ a1_.CloseSendStreams();
+ a2_.CloseReceiveStreams();
ASSERT_GE(a1_.GetPacketsSent(0), 40);
ASSERT_GE(a2_.GetPacketsReceived(0), 40);
}