mirror of
https://gitlab.winehq.org/wine/wine-gecko.git
synced 2024-09-13 09:24:08 -07:00
Bug 786307: Implement RTCP MUX in MediaPipeline r=ekr
This commit is contained in:
Suhas Nandakumar
parent
160161e0a6
commit
b10a1e2005
@ -84,25 +84,28 @@ nsresult MediaPipeline::Init_s() {
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MOZ_MTLOG(ML_ERROR, "RTP transport is already in error state");
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TransportFailed_s(rtp_transport_);
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return NS_ERROR_FAILURE;
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} else {
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if (!muxed_) {
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rtcp_transport_->SignalStateChange.connect(this,
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&MediaPipeline::StateChange);
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}
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if (rtcp_transport_->state() == TransportLayer::TS_OPEN) {
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res = TransportReady_s(rtcp_transport_);
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if (NS_FAILED(res)) {
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MOZ_MTLOG(ML_ERROR, "Error calling TransportReady(); res="
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<< static_cast<uint32_t>(res) << " in " << __FUNCTION__);
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return res;
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}
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} else if (rtcp_transport_->state() == TransportLayer::TS_ERROR) {
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MOZ_MTLOG(ML_ERROR, "RTCP transport is already in error state");
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TransportFailed_s(rtcp_transport_);
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return NS_ERROR_FAILURE;
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// If rtcp_transport_ is the same as rtp_transport_ then we are muxing.
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// Otherwise, set it up separately.
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if (rtcp_transport_ != rtp_transport_) {
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rtcp_transport_->SignalStateChange.connect(this,
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&MediaPipeline::StateChange);
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if (rtcp_transport_->state() == TransportLayer::TS_OPEN) {
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res = TransportReady_s(rtcp_transport_);
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if (NS_FAILED(res)) {
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MOZ_MTLOG(ML_ERROR, "Error calling TransportReady(); res="
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<< static_cast<uint32_t>(res) << " in " << __FUNCTION__);
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return res;
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}
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} else if (rtcp_transport_->state() == TransportLayer::TS_ERROR) {
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MOZ_MTLOG(ML_ERROR, "RTCP transport is already in error state");
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TransportFailed_s(rtcp_transport_);
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return NS_ERROR_FAILURE;
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}
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}
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return NS_OK;
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}
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@ -121,6 +124,12 @@ void MediaPipeline::ShutdownTransport_s() {
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}
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void MediaPipeline::StateChange(TransportFlow *flow, TransportLayer::State state) {
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// If rtcp_transport_ is the same as rtp_transport_ then we are muxing.
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// So the only flow should be the RTP flow.
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if (rtcp_transport_ == rtp_transport_) {
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MOZ_ASSERT(flow == rtp_transport_);
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}
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if (state == TransportLayer::TS_OPEN) {
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MOZ_MTLOG(ML_DEBUG, "Flow is ready");
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TransportReady_s(flow);
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@ -216,7 +225,8 @@ nsresult MediaPipeline::TransportReady_s(TransportFlow *flow) {
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}
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// Start listening
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if (muxed_) {
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// If rtcp_transport_ is the same as rtp_transport_ then we are muxing
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if (rtcp_transport_ == rtp_transport_) {
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MOZ_ASSERT(!rtcp_send_srtp_ && !rtcp_recv_srtp_);
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rtcp_send_srtp_ = rtp_send_srtp_;
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rtcp_recv_srtp_ = rtp_recv_srtp_;
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@ -227,6 +237,7 @@ nsresult MediaPipeline::TransportReady_s(TransportFlow *flow) {
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dtls->downward()->SignalPacketReceived.connect(this,
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&MediaPipeline::
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PacketReceived);
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rtcp_state_ = MP_OPEN;
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} else {
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MOZ_MTLOG(ML_DEBUG, "Listening for RTP packets received on " <<
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static_cast<void *>(dtls->downward()));
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@ -269,6 +280,13 @@ nsresult MediaPipeline::TransportFailed_s(TransportFlow *flow) {
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*state = MP_CLOSED;
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// If rtcp_transport_ is the same as rtp_transport_ then we are muxing
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if(rtcp_transport_ == rtp_transport_) {
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MOZ_ASSERT(state != &rtcp_state_);
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rtcp_state_ = MP_CLOSED;
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}
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MOZ_MTLOG(ML_DEBUG, "Transport closed for flow " << (rtcp ? "rtcp" : "rtp"));
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NS_WARNING(
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@ -353,7 +371,7 @@ void MediaPipeline::RtpPacketReceived(TransportLayer *layer,
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const unsigned char *data,
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size_t len) {
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if (!transport_->pipeline()) {
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MOZ_MTLOG(ML_DEBUG, "Discarding incoming packet; transport disconnected");
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MOZ_MTLOG(ML_ERROR, "Discarding incoming packet; transport disconnected");
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return;
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}
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@ -363,7 +381,7 @@ void MediaPipeline::RtpPacketReceived(TransportLayer *layer,
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}
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if (rtp_state_ != MP_OPEN) {
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MOZ_MTLOG(ML_DEBUG, "Discarding incoming packet; pipeline not open");
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MOZ_MTLOG(ML_ERROR, "Discarding incoming packet; pipeline not open");
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return;
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}
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@ -92,9 +92,16 @@ class MediaPipeline : public sigslot::has_slots<> {
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rtcp_packets_sent_(0),
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rtp_packets_received_(0),
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rtcp_packets_received_(0),
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muxed_((rtcp_transport_ == NULL) || (rtp_transport_ == rtcp_transport_)),
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pc_(pc),
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description_() {
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// To indicate rtcp-mux rtcp_transport should be NULL.
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// Therefore it's an error to send in the same flow for
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// both rtp and rtcp.
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MOZ_ASSERT(rtp_transport_ != rtcp_transport_);
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if (!rtcp_transport_) {
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rtcp_transport_ = rtp_transport;
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}
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}
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virtual ~MediaPipeline();
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@ -116,9 +123,9 @@ class MediaPipeline : public sigslot::has_slots<> {
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virtual Direction direction() const { return direction_; }
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int rtp_packets_sent() const { return rtp_packets_sent_; }
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int rtcp_packets_sent() const { return rtp_packets_sent_; }
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int rtcp_packets_sent() const { return rtcp_packets_sent_; }
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int rtp_packets_received() const { return rtp_packets_received_; }
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int rtcp_packets_received() const { return rtp_packets_received_; }
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int rtcp_packets_received() const { return rtcp_packets_received_; }
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MediaSessionConduit *Conduit() { return conduit_; }
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@ -209,7 +216,6 @@ class MediaPipeline : public sigslot::has_slots<> {
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int rtcp_packets_received_;
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// Written on Init. Read on STS thread.
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bool muxed_;
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std::string pc_;
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std::string description_;
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@ -290,6 +296,7 @@ private:
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// and transmitting to the network.
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class MediaPipelineTransmit : public MediaPipeline {
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public:
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// Set rtcp_transport to NULL to use rtcp-mux
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MediaPipelineTransmit(const std::string& pc,
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nsCOMPtr<nsIEventTarget> main_thread,
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nsCOMPtr<nsIEventTarget> sts_thread,
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@ -384,6 +391,7 @@ class MediaPipelineTransmit : public MediaPipeline {
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// rendering video.
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class MediaPipelineReceive : public MediaPipeline {
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public:
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// Set rtcp_transport to NULL to use rtcp-mux
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MediaPipelineReceive(const std::string& pc,
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nsCOMPtr<nsIEventTarget> main_thread,
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nsCOMPtr<nsIEventTarget> sts_thread,
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@ -42,47 +42,94 @@ MOZ_MTLOG_MODULE("mediapipeline")
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MtransportTestUtils *test_utils;
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namespace {
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class TestAgent {
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class TransportInfo {
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public:
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TestAgent() :
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audio_flow_(new TransportFlow()),
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audio_prsock_(new TransportLayerPrsock()),
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audio_dtls_(new TransportLayerDtls()),
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audio_config_(109, "opus", 48000, 960, 2, 64000),
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audio_conduit_(mozilla::AudioSessionConduit::Create(NULL)),
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audio_(),
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audio_pipeline_(),
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video_flow_(new TransportFlow()),
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video_prsock_(new TransportLayerPrsock()),
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video_config_(120, "VP8"),
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video_conduit_(mozilla::VideoSessionConduit::Create()),
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video_(),
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video_pipeline_() {
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}
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TransportInfo() :
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flow_(NULL),
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prsock_(NULL),
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dtls_(NULL) {}
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void ConnectSocket(PRFileDesc *fd, bool client) {
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void Init(bool client) {
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nsresult res;
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res = audio_prsock_->Init();
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flow_ = new TransportFlow();
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prsock_ = new TransportLayerPrsock();
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dtls_ = new TransportLayerDtls();
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res = prsock_->Init();
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if (res != NS_OK) {
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FreeLayers();
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}
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ASSERT_EQ((nsresult)NS_OK, res);
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mozilla::SyncRunnable::DispatchToThread(
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test_utils->sts_target(),
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WrapRunnable(audio_prsock_, &TransportLayerPrsock::Import, fd, &res));
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ASSERT_TRUE(NS_SUCCEEDED(res));
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ASSERT_EQ((nsresult)NS_OK, audio_flow_->PushLayer(audio_prsock_));
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std::vector<uint16_t> ciphers;
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ciphers.push_back(SRTP_AES128_CM_HMAC_SHA1_80);
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audio_dtls_->SetSrtpCiphers(ciphers);
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audio_dtls_->SetIdentity(DtlsIdentity::Generate());
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audio_dtls_->SetRole(client ? TransportLayerDtls::CLIENT :
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TransportLayerDtls::SERVER);
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audio_flow_->PushLayer(audio_dtls_);
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dtls_->SetSrtpCiphers(ciphers);
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dtls_->SetIdentity(DtlsIdentity::Generate());
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dtls_->SetRole(client ? TransportLayerDtls::CLIENT :
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TransportLayerDtls::SERVER);
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dtls_->SetVerificationAllowAll();
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}
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virtual void CreatePipelines_s() = 0;
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void PushLayers() {
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nsresult res;
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nsAutoPtr<std::queue<TransportLayer *> > layers(
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new std::queue<TransportLayer *>);
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layers->push(prsock_);
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layers->push(dtls_);
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res = flow_->PushLayers(layers);
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if (res != NS_OK) {
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FreeLayers();
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}
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ASSERT_EQ((nsresult)NS_OK, res);
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}
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// Free the memory allocated at the beginning of Init
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// if failure occurs before layers setup.
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void FreeLayers() {
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delete prsock_;
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delete dtls_;
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}
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void Stop() {
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flow_ = NULL;
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}
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mozilla::RefPtr<TransportFlow> flow_;
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TransportLayerPrsock *prsock_;
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TransportLayerDtls *dtls_;
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};
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class TestAgent {
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public:
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TestAgent() :
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audio_config_(109, "opus", 48000, 960, 2, 64000),
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audio_conduit_(mozilla::AudioSessionConduit::Create(NULL)),
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audio_(),
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audio_pipeline_() {
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}
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void ConnectSocket(PRFileDesc *fd, bool client, bool isRtcp) {
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nsresult res;
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TransportInfo *transport = isRtcp ?
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&audio_rtcp_transport_ : &audio_rtp_transport_;
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transport->Init(client);
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mozilla::SyncRunnable::DispatchToThread(
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test_utils->sts_target(),
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WrapRunnable(transport->prsock_, &TransportLayerPrsock::Import, fd, &res));
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if (!NS_SUCCEEDED(res)) {
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transport->FreeLayers();
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}
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ASSERT_TRUE(NS_SUCCEEDED(res));
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transport->PushLayers();
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}
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virtual void CreatePipelines_s(bool aIsRtcpMux) = 0;
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void Start() {
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nsresult ret;
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@ -98,12 +145,10 @@ class TestAgent {
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void StopInt() {
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audio_->GetStream()->Stop();
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audio_flow_ = NULL;
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video_flow_ = NULL;
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audio_rtp_transport_.Stop();
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audio_rtcp_transport_.Stop();
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if (audio_pipeline_)
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audio_pipeline_->ShutdownTransport_s();
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if (video_pipeline_)
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video_pipeline_->ShutdownTransport_s();
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}
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void Stop() {
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@ -111,39 +156,28 @@ class TestAgent {
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if (audio_pipeline_)
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audio_pipeline_->ShutdownMedia_m();
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if (video_pipeline_)
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video_pipeline_->ShutdownMedia_m();
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mozilla::SyncRunnable::DispatchToThread(
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test_utils->sts_target(),
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WrapRunnable(this, &TestAgent::StopInt));
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audio_pipeline_ = NULL;
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video_pipeline_ = NULL;
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PR_Sleep(1000); // Deal with race condition
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}
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protected:
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mozilla::RefPtr<TransportFlow> audio_flow_;
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TransportLayerPrsock *audio_prsock_;
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TransportLayerDtls *audio_dtls_;
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mozilla::AudioCodecConfig audio_config_;
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mozilla::RefPtr<mozilla::MediaSessionConduit> audio_conduit_;
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nsRefPtr<DOMMediaStream> audio_;
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mozilla::RefPtr<mozilla::MediaPipeline> audio_pipeline_;
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mozilla::RefPtr<TransportFlow> video_flow_;
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TransportLayerPrsock *video_prsock_;
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mozilla::VideoCodecConfig video_config_;
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mozilla::RefPtr<mozilla::MediaSessionConduit> video_conduit_;
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nsRefPtr<DOMMediaStream> video_;
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mozilla::RefPtr<mozilla::MediaPipeline> video_pipeline_;
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TransportInfo audio_rtp_transport_;
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TransportInfo audio_rtcp_transport_;
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};
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class TestAgentSend : public TestAgent {
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public:
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virtual void CreatePipelines_s() {
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virtual void CreatePipelines_s(bool aIsRtcpMux) {
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audio_ = new Fake_DOMMediaStream(new Fake_AudioStreamSource());
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mozilla::MediaConduitErrorCode err =
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@ -153,16 +187,29 @@ class TestAgentSend : public TestAgent {
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std::string test_pc("PC");
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if (aIsRtcpMux) {
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ASSERT_FALSE(audio_rtcp_transport_.flow_);
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}
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audio_pipeline_ = new mozilla::MediaPipelineTransmit(
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test_pc,
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NULL,
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test_utils->sts_target(),
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audio_->GetStream(), 1, audio_conduit_, audio_flow_, NULL);
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audio_->GetStream(),
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1,
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audio_conduit_,
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audio_rtp_transport_.flow_,
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audio_rtcp_transport_.flow_);
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audio_pipeline_->Init();
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}
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// video_ = new Fake_DOMMediaStream(new Fake_VideoStreamSource());
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// video_pipeline_ = new mozilla::MediaPipelineTransmit(video_, video_conduit_, &video_flow_, &video_flow_);
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int GetAudioRtpCount() {
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return audio_pipeline_->rtp_packets_sent();
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}
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int GetAudioRtcpCount() {
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return audio_pipeline_->rtcp_packets_received();
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}
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private:
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@ -171,7 +218,7 @@ class TestAgentSend : public TestAgent {
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class TestAgentReceive : public TestAgent {
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public:
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virtual void CreatePipelines_s() {
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virtual void CreatePipelines_s(bool aIsRtcpMux) {
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mozilla::SourceMediaStream *audio = new Fake_SourceMediaStream();
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audio->SetPullEnabled(true);
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@ -190,17 +237,30 @@ class TestAgentReceive : public TestAgent {
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EXPECT_EQ(mozilla::kMediaConduitNoError, err);
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std::string test_pc("PC");
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if (aIsRtcpMux) {
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ASSERT_FALSE(audio_rtcp_transport_.flow_);
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}
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audio_pipeline_ = new mozilla::MediaPipelineReceiveAudio(
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test_pc,
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NULL,
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test_utils->sts_target(),
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audio_->GetStream(), 1,
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static_cast<mozilla::AudioSessionConduit *>(audio_conduit_.get()),
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audio_flow_, NULL);
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audio_rtp_transport_.flow_, audio_rtcp_transport_.flow_);
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audio_pipeline_->Init();
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}
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int GetAudioRtpCount() {
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return audio_pipeline_->rtp_packets_received();
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}
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int GetAudioRtcpCount() {
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return audio_pipeline_->rtcp_packets_sent();
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}
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private:
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};
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@ -208,44 +268,85 @@ class TestAgentReceive : public TestAgent {
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class MediaPipelineTest : public ::testing::Test {
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public:
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MediaPipelineTest() : p1_() {
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fds_[0] = fds_[1] = NULL;
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rtp_fds_[0] = rtp_fds_[1] = NULL;
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rtcp_fds_[0] = rtcp_fds_[1] = NULL;
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}
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void SetUp() {
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PRStatus status = PR_NewTCPSocketPair(fds_);
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// Setup transport.
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void InitTransports(bool aIsRtcpMux) {
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// Create RTP related transport.
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PRStatus status = PR_NewTCPSocketPair(rtp_fds_);
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ASSERT_EQ(status, PR_SUCCESS);
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// RTP, DTLS server
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mozilla::SyncRunnable::DispatchToThread(
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test_utils->sts_target(),
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WrapRunnable(&p1_, &TestAgent::ConnectSocket, fds_[0], false));
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WrapRunnable(&p1_, &TestAgent::ConnectSocket, rtp_fds_[0], false, false));
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// RTP, DTLS client
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mozilla::SyncRunnable::DispatchToThread(
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test_utils->sts_target(),
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WrapRunnable(&p2_, &TestAgent::ConnectSocket, fds_[1], false));
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WrapRunnable(&p2_, &TestAgent::ConnectSocket, rtp_fds_[1], true, false));
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mozilla::SyncRunnable::DispatchToThread(
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test_utils->sts_target(),
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WrapRunnable(&p1_, &TestAgent::CreatePipelines_s));
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// Create RTCP flows separately if we are not muxing them.
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if(!aIsRtcpMux) {
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status = PR_NewTCPSocketPair(rtcp_fds_);
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ASSERT_EQ(status, PR_SUCCESS);
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mozilla::SyncRunnable::DispatchToThread(
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test_utils->sts_target(),
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WrapRunnable(&p2_, &TestAgent::CreatePipelines_s));
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// RTCP, DTLS server
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mozilla::SyncRunnable::DispatchToThread(
|
||||
test_utils->sts_target(),
|
||||
WrapRunnable(&p1_, &TestAgent::ConnectSocket, rtcp_fds_[0], false, true));
|
||||
|
||||
// RTCP, DTLS client
|
||||
mozilla::SyncRunnable::DispatchToThread(
|
||||
test_utils->sts_target(),
|
||||
WrapRunnable(&p2_, &TestAgent::ConnectSocket, rtcp_fds_[1], true, true));
|
||||
}
|
||||
}
|
||||
|
||||
protected:
|
||||
PRFileDesc *fds_[2];
|
||||
// Verify RTP and RTCP
|
||||
void TestAudioSend(bool aIsRtcpMux) {
|
||||
// Setup transport flows
|
||||
InitTransports(aIsRtcpMux);
|
||||
|
||||
mozilla::SyncRunnable::DispatchToThread(
|
||||
test_utils->sts_target(),
|
||||
WrapRunnable(&p1_, &TestAgent::CreatePipelines_s, aIsRtcpMux));
|
||||
|
||||
mozilla::SyncRunnable::DispatchToThread(
|
||||
test_utils->sts_target(),
|
||||
WrapRunnable(&p2_, &TestAgent::CreatePipelines_s, aIsRtcpMux));
|
||||
|
||||
p2_.Start();
|
||||
p1_.Start();
|
||||
|
||||
// wait for some RTP/RTCP tx and rx to happen
|
||||
PR_Sleep(10000);
|
||||
|
||||
ASSERT_GE(p1_.GetAudioRtpCount(), 40);
|
||||
ASSERT_GE(p2_.GetAudioRtpCount(), 40);
|
||||
ASSERT_GE(p1_.GetAudioRtcpCount(), 1);
|
||||
ASSERT_GE(p2_.GetAudioRtcpCount(), 1);
|
||||
|
||||
p1_.Stop();
|
||||
p2_.Stop();
|
||||
}
|
||||
|
||||
protected:
|
||||
PRFileDesc *rtp_fds_[2];
|
||||
PRFileDesc *rtcp_fds_[2];
|
||||
TestAgentSend p1_;
|
||||
TestAgentReceive p2_;
|
||||
};
|
||||
|
||||
TEST_F(MediaPipelineTest, AudioSend) {
|
||||
p2_.Start();
|
||||
p1_.Start();
|
||||
PR_Sleep(1000);
|
||||
p1_.Stop();
|
||||
p2_.Stop();
|
||||
TEST_F(MediaPipelineTest, TestAudioSendNoMux) {
|
||||
TestAudioSend(false);
|
||||
}
|
||||
|
||||
TEST_F(MediaPipelineTest, TestAudioSendMux) {
|
||||
TestAudioSend(true);
|
||||
}
|
||||
|
||||
} // end namespace
|
||||
|
||||
|
||||
Loading…
Reference in New Issue
Block a user