diff options
Diffstat (limited to 'dom/media/webrtc/libwebrtcglue/AudioConduit.cpp')
| -rw-r--r-- | dom/media/webrtc/libwebrtcglue/AudioConduit.cpp | 975 |
1 files changed, 975 insertions, 0 deletions
diff --git a/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp new file mode 100644 index 0000000000..d8492f779a --- /dev/null +++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp @@ -0,0 +1,975 @@ +/* This Source Code Form is subject to the terms of the Mozilla Public + * License, v. 2.0. If a copy of the MPL was not distributed with this file, + * You can obtain one at http://mozilla.org/MPL/2.0/. */ + +#include "AudioConduit.h" + +#include "common/browser_logging/CSFLog.h" +#include "MediaConduitControl.h" +#include "mozilla/media/MediaUtils.h" +#include "mozilla/Telemetry.h" +#include "transport/runnable_utils.h" +#include "transport/SrtpFlow.h" // For SRTP_MAX_EXPANSION +#include "WebrtcCallWrapper.h" + +// libwebrtc includes +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "audio/audio_receive_stream.h" +#include "media/base/media_constants.h" + +// for ntohs +#ifdef HAVE_NETINET_IN_H +# include <netinet/in.h> +#elif defined XP_WIN +# include <winsock2.h> +#endif + +#ifdef MOZ_WIDGET_ANDROID +# include "AndroidBridge.h" +#endif + +namespace mozilla { + +namespace { + +static const char* acLogTag = "WebrtcAudioSessionConduit"; +#ifdef LOGTAG +# undef LOGTAG +#endif +#define LOGTAG acLogTag + +using namespace cricket; +using LocalDirection = MediaSessionConduitLocalDirection; + +const char kCodecParamCbr[] = "cbr"; + +} // namespace + +/** + * Factory Method for AudioConduit + */ +RefPtr<AudioSessionConduit> AudioSessionConduit::Create( + RefPtr<WebrtcCallWrapper> aCall, + nsCOMPtr<nsISerialEventTarget> aStsThread) { + CSFLogDebug(LOGTAG, "%s ", __FUNCTION__); + MOZ_ASSERT(NS_IsMainThread()); + + return MakeRefPtr<WebrtcAudioConduit>(std::move(aCall), + std::move(aStsThread)); +} + +#define INIT_MIRROR(name, val) \ + name(aCallThread, val, "WebrtcAudioConduit::Control::" #name " (Mirror)") +WebrtcAudioConduit::Control::Control(const RefPtr<AbstractThread>& aCallThread) + : INIT_MIRROR(mReceiving, false), + INIT_MIRROR(mTransmitting, false), + INIT_MIRROR(mLocalSsrcs, Ssrcs()), + INIT_MIRROR(mLocalCname, std::string()), + INIT_MIRROR(mMid, std::string()), + INIT_MIRROR(mRemoteSsrc, 0), + INIT_MIRROR(mSyncGroup, std::string()), + INIT_MIRROR(mLocalRecvRtpExtensions, RtpExtList()), + INIT_MIRROR(mLocalSendRtpExtensions, RtpExtList()), + INIT_MIRROR(mSendCodec, Nothing()), + INIT_MIRROR(mRecvCodecs, std::vector<AudioCodecConfig>()) {} +#undef INIT_MIRROR + +RefPtr<GenericPromise> WebrtcAudioConduit::Shutdown() { + MOZ_ASSERT(NS_IsMainThread()); + + return InvokeAsync(mCallThread, "WebrtcAudioConduit::Shutdown (main thread)", + [this, self = RefPtr<WebrtcAudioConduit>(this)] { + mControl.mReceiving.DisconnectIfConnected(); + mControl.mTransmitting.DisconnectIfConnected(); + mControl.mLocalSsrcs.DisconnectIfConnected(); + mControl.mLocalCname.DisconnectIfConnected(); + mControl.mMid.DisconnectIfConnected(); + mControl.mRemoteSsrc.DisconnectIfConnected(); + mControl.mSyncGroup.DisconnectIfConnected(); + mControl.mLocalRecvRtpExtensions.DisconnectIfConnected(); + mControl.mLocalSendRtpExtensions.DisconnectIfConnected(); + mControl.mSendCodec.DisconnectIfConnected(); + mControl.mRecvCodecs.DisconnectIfConnected(); + mControl.mOnDtmfEventListener.DisconnectIfExists(); + mWatchManager.Shutdown(); + + { + AutoWriteLock lock(mLock); + DeleteSendStream(); + DeleteRecvStream(); + } + + return GenericPromise::CreateAndResolve( + true, "WebrtcAudioConduit::Shutdown (call thread)"); + }); +} + +WebrtcAudioConduit::WebrtcAudioConduit( + RefPtr<WebrtcCallWrapper> aCall, nsCOMPtr<nsISerialEventTarget> aStsThread) + : mCall(std::move(aCall)), + mSendTransport(this), + mRecvTransport(this), + mRecvStreamConfig(), + mRecvStream(nullptr), + mSendStreamConfig(&mSendTransport), + mSendStream(nullptr), + mSendStreamRunning(false), + mRecvStreamRunning(false), + mDtmfEnabled(false), + mLock("WebrtcAudioConduit::mLock"), + mCallThread(std::move(mCall->mCallThread)), + mStsThread(std::move(aStsThread)), + mControl(mCall->mCallThread), + mWatchManager(this, mCall->mCallThread) { + mRecvStreamConfig.rtcp_send_transport = &mRecvTransport; + mRecvStreamConfig.rtp.rtcp_event_observer = this; +} + +/** + * Destruction defines for our super-classes + */ +WebrtcAudioConduit::~WebrtcAudioConduit() { + CSFLogDebug(LOGTAG, "%s ", __FUNCTION__); + MOZ_ASSERT(!mSendStream && !mRecvStream, + "Call DeleteStreams prior to ~WebrtcAudioConduit."); +} + +#define CONNECT(aCanonical, aMirror) \ + do { \ + (aMirror).Connect(aCanonical); \ + mWatchManager.Watch(aMirror, &WebrtcAudioConduit::OnControlConfigChange); \ + } while (0) + +void WebrtcAudioConduit::InitControl(AudioConduitControlInterface* aControl) { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + + CONNECT(aControl->CanonicalReceiving(), mControl.mReceiving); + CONNECT(aControl->CanonicalTransmitting(), mControl.mTransmitting); + CONNECT(aControl->CanonicalLocalSsrcs(), mControl.mLocalSsrcs); + CONNECT(aControl->CanonicalLocalCname(), mControl.mLocalCname); + CONNECT(aControl->CanonicalMid(), mControl.mMid); + CONNECT(aControl->CanonicalRemoteSsrc(), mControl.mRemoteSsrc); + CONNECT(aControl->CanonicalSyncGroup(), mControl.mSyncGroup); + CONNECT(aControl->CanonicalLocalRecvRtpExtensions(), + mControl.mLocalRecvRtpExtensions); + CONNECT(aControl->CanonicalLocalSendRtpExtensions(), + mControl.mLocalSendRtpExtensions); + CONNECT(aControl->CanonicalAudioSendCodec(), mControl.mSendCodec); + CONNECT(aControl->CanonicalAudioRecvCodecs(), mControl.mRecvCodecs); + mControl.mOnDtmfEventListener = aControl->OnDtmfEvent().Connect( + mCall->mCallThread, this, &WebrtcAudioConduit::OnDtmfEvent); +} + +#undef CONNECT + +void WebrtcAudioConduit::OnDtmfEvent(const DtmfEvent& aEvent) { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(mSendStream); + MOZ_ASSERT(mDtmfEnabled); + mSendStream->SendTelephoneEvent(aEvent.mPayloadType, aEvent.mPayloadFrequency, + aEvent.mEventCode, aEvent.mLengthMs); +} + +void WebrtcAudioConduit::OnControlConfigChange() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + + bool recvStreamReconfigureNeeded = false; + bool sendStreamReconfigureNeeded = false; + bool recvStreamRecreationNeeded = false; + bool sendStreamRecreationNeeded = false; + + if (!mControl.mLocalSsrcs.Ref().empty()) { + if (mControl.mLocalSsrcs.Ref()[0] != mSendStreamConfig.rtp.ssrc) { + sendStreamRecreationNeeded = true; + + // For now... + recvStreamRecreationNeeded = true; + } + mRecvStreamConfig.rtp.local_ssrc = mControl.mLocalSsrcs.Ref()[0]; + mSendStreamConfig.rtp.ssrc = mControl.mLocalSsrcs.Ref()[0]; + + // In the future we can do this instead of recreating the recv stream: + // if (mRecvStream) { + // mCall->Call()->OnLocalSsrcUpdated(mRecvStream, + // mControl.mLocalSsrcs.Ref()[0]); + // } + } + + if (mControl.mLocalCname.Ref() != mSendStreamConfig.rtp.c_name) { + mSendStreamConfig.rtp.c_name = mControl.mLocalCname.Ref(); + sendStreamReconfigureNeeded = true; + } + + if (mControl.mMid.Ref() != mSendStreamConfig.rtp.mid) { + mSendStreamConfig.rtp.mid = mControl.mMid.Ref(); + sendStreamReconfigureNeeded = true; + } + + if (mControl.mRemoteSsrc.Ref() != mControl.mConfiguredRemoteSsrc) { + mRecvStreamConfig.rtp.remote_ssrc = mControl.mConfiguredRemoteSsrc = + mControl.mRemoteSsrc.Ref(); + recvStreamRecreationNeeded = true; + } + + if (mControl.mSyncGroup.Ref() != mRecvStreamConfig.sync_group) { + mRecvStreamConfig.sync_group = mControl.mSyncGroup.Ref(); + // For now... + recvStreamRecreationNeeded = true; + // In the future we can do this instead of recreating the recv stream: + // if (mRecvStream) { + // mCall->Call()->OnUpdateSyncGroup(mRecvStream, + // mRecvStreamConfig.sync_group); + // } + } + + if (auto filteredExtensions = FilterExtensions( + LocalDirection::kRecv, mControl.mLocalRecvRtpExtensions); + filteredExtensions != mRecvStreamConfig.rtp.extensions) { + mRecvStreamConfig.rtp.extensions = std::move(filteredExtensions); + // For now... + recvStreamRecreationNeeded = true; + // In the future we can do this instead of recreating the recv stream: + // if (mRecvStream) { + // mRecvStream->SetRtpExtensions(mRecvStreamConfig.rtp.extensions); + //} + } + + if (auto filteredExtensions = FilterExtensions( + LocalDirection::kSend, mControl.mLocalSendRtpExtensions); + filteredExtensions != mSendStreamConfig.rtp.extensions) { + // At the very least, we need a reconfigure. Recreation needed if the + // extmap for any extension has changed, but not for adding/removing + // extensions. + sendStreamReconfigureNeeded = true; + + for (const auto& newExt : filteredExtensions) { + if (sendStreamRecreationNeeded) { + break; + } + for (const auto& oldExt : mSendStreamConfig.rtp.extensions) { + if (newExt.uri == oldExt.uri) { + if (newExt.id != oldExt.id) { + sendStreamRecreationNeeded = true; + } + // We're done handling newExt, one way or another + break; + } + } + } + + mSendStreamConfig.rtp.extensions = std::move(filteredExtensions); + } + + mControl.mSendCodec.Ref().apply([&](const auto& aConfig) { + if (mControl.mConfiguredSendCodec != mControl.mSendCodec.Ref()) { + mControl.mConfiguredSendCodec = mControl.mSendCodec; + if (ValidateCodecConfig(aConfig, true) == kMediaConduitNoError) { + mSendStreamConfig.encoder_factory = + webrtc::CreateBuiltinAudioEncoderFactory(); + + webrtc::AudioSendStream::Config::SendCodecSpec spec( + aConfig.mType, CodecConfigToLibwebrtcFormat(aConfig)); + mSendStreamConfig.send_codec_spec = spec; + + mDtmfEnabled = aConfig.mDtmfEnabled; + sendStreamReconfigureNeeded = true; + } + } + }); + + if (mControl.mConfiguredRecvCodecs != mControl.mRecvCodecs.Ref()) { + mControl.mConfiguredRecvCodecs = mControl.mRecvCodecs; + mRecvStreamConfig.decoder_factory = mCall->mAudioDecoderFactory; + mRecvStreamConfig.decoder_map.clear(); + + for (const auto& codec : mControl.mRecvCodecs.Ref()) { + if (ValidateCodecConfig(codec, false) != kMediaConduitNoError) { + continue; + } + mRecvStreamConfig.decoder_map.emplace( + codec.mType, CodecConfigToLibwebrtcFormat(codec)); + } + + recvStreamReconfigureNeeded = true; + } + + if (!recvStreamReconfigureNeeded && !sendStreamReconfigureNeeded && + !recvStreamRecreationNeeded && !sendStreamRecreationNeeded && + mControl.mReceiving == mRecvStreamRunning && + mControl.mTransmitting == mSendStreamRunning) { + // No changes applied -- no need to lock. + return; + } + + if (recvStreamRecreationNeeded) { + recvStreamReconfigureNeeded = false; + } + if (sendStreamRecreationNeeded) { + sendStreamReconfigureNeeded = false; + } + + { + AutoWriteLock lock(mLock); + // Recreate/Stop/Start streams as needed. + if (recvStreamRecreationNeeded) { + DeleteRecvStream(); + } + if (mControl.mReceiving) { + CreateRecvStream(); + } + if (sendStreamRecreationNeeded) { + DeleteSendStream(); + } + if (mControl.mTransmitting) { + CreateSendStream(); + } + } + + // We make sure to not hold the lock while stopping/starting/reconfiguring + // streams, so as to not cause deadlocks. These methods can cause our platform + // codecs to dispatch sync runnables to main, and main may grab the lock. + + if (mRecvStream && recvStreamReconfigureNeeded) { + MOZ_ASSERT(!recvStreamRecreationNeeded); + mRecvStream->SetDecoderMap(mRecvStreamConfig.decoder_map); + } + + if (mSendStream && sendStreamReconfigureNeeded) { + MOZ_ASSERT(!sendStreamRecreationNeeded); + // TODO: Pass a callback here, so we can react to RTCErrors thrown by + // libwebrtc. + mSendStream->Reconfigure(mSendStreamConfig, nullptr); + } + + if (!mControl.mReceiving) { + StopReceiving(); + } + if (!mControl.mTransmitting) { + StopTransmitting(); + } + + if (mControl.mReceiving) { + StartReceiving(); + } + if (mControl.mTransmitting) { + StartTransmitting(); + } +} + +std::vector<uint32_t> WebrtcAudioConduit::GetLocalSSRCs() const { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + return std::vector<uint32_t>(1, mRecvStreamConfig.rtp.local_ssrc); +} + +bool WebrtcAudioConduit::OverrideRemoteSSRC(uint32_t aSsrc) { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + + if (mRecvStreamConfig.rtp.remote_ssrc == aSsrc) { + return true; + } + mRecvStreamConfig.rtp.remote_ssrc = aSsrc; + + const bool wasReceiving = mRecvStreamRunning; + const bool hadRecvStream = mRecvStream; + + StopReceiving(); + + if (hadRecvStream) { + AutoWriteLock lock(mLock); + DeleteRecvStream(); + CreateRecvStream(); + } + + if (wasReceiving) { + StartReceiving(); + } + return true; +} + +Maybe<Ssrc> WebrtcAudioConduit::GetRemoteSSRC() const { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + // libwebrtc uses 0 to mean a lack of SSRC. That is not to spec. + return mRecvStreamConfig.rtp.remote_ssrc == 0 + ? Nothing() + : Some(mRecvStreamConfig.rtp.remote_ssrc); +} + +Maybe<webrtc::AudioReceiveStreamInterface::Stats> +WebrtcAudioConduit::GetReceiverStats() const { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + if (!mRecvStream) { + return Nothing(); + } + return Some(mRecvStream->GetStats()); +} + +Maybe<webrtc::AudioSendStream::Stats> WebrtcAudioConduit::GetSenderStats() + const { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + if (!mSendStream) { + return Nothing(); + } + return Some(mSendStream->GetStats()); +} + +Maybe<webrtc::CallBasicStats> WebrtcAudioConduit::GetCallStats() const { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + if (!mCall->Call()) { + return Nothing(); + } + return Some(mCall->Call()->GetStats()); +} + +void WebrtcAudioConduit::OnRtcpBye() { mRtcpByeEvent.Notify(); } + +void WebrtcAudioConduit::OnRtcpTimeout() { mRtcpTimeoutEvent.Notify(); } + +void WebrtcAudioConduit::SetTransportActive(bool aActive) { + MOZ_ASSERT(mStsThread->IsOnCurrentThread()); + if (mTransportActive == aActive) { + return; + } + + // If false, This stops us from sending + mTransportActive = aActive; + + // We queue this because there might be notifications to these listeners + // pending, and we don't want to drop them by letting this jump ahead of + // those notifications. We move the listeners into the lambda in case the + // transport comes back up before we disconnect them. (The Connect calls + // happen in MediaPipeline) + // We retain a strong reference to ourself, because the listeners are holding + // a non-refcounted reference to us, and moving them into the lambda could + // conceivably allow them to outlive us. + if (!aActive) { + MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction( + __func__, + [self = RefPtr<WebrtcAudioConduit>(this), + recvRtpListener = std::move(mReceiverRtpEventListener), + recvRtcpListener = std::move(mReceiverRtcpEventListener), + sendRtcpListener = std::move(mSenderRtcpEventListener)]() mutable { + recvRtpListener.DisconnectIfExists(); + recvRtcpListener.DisconnectIfExists(); + sendRtcpListener.DisconnectIfExists(); + }))); + } +} + +// AudioSessionConduit Implementation +MediaConduitErrorCode WebrtcAudioConduit::SendAudioFrame( + std::unique_ptr<webrtc::AudioFrame> frame) { + CSFLogDebug(LOGTAG, "%s ", __FUNCTION__); + // Following checks need to be performed + // 1. Non null audio buffer pointer, and + // 2. Valid sample rate, and + // 3. Appropriate Sample Length for 10 ms audio-frame. This represents the + // block size used upstream for processing. + // Ex: for 16000 sample rate , valid block-length is 160. + // Similarly for 32000 sample rate, valid block length is 320. + + if (!frame->data() || + (IsSamplingFreqSupported(frame->sample_rate_hz()) == false) || + ((frame->samples_per_channel() % (frame->sample_rate_hz() / 100) != 0))) { + CSFLogError(LOGTAG, "%s Invalid Parameters ", __FUNCTION__); + MOZ_ASSERT(PR_FALSE); + return kMediaConduitMalformedArgument; + } + + // This is the AudioProxyThread, blocking it for a bit is fine. + AutoReadLock lock(mLock); + if (!mSendStreamRunning) { + CSFLogError(LOGTAG, "%s Engine not transmitting ", __FUNCTION__); + return kMediaConduitSessionNotInited; + } + + mSendStream->SendAudioData(std::move(frame)); + return kMediaConduitNoError; +} + +MediaConduitErrorCode WebrtcAudioConduit::GetAudioFrame( + int32_t samplingFreqHz, webrtc::AudioFrame* frame) { + CSFLogDebug(LOGTAG, "%s ", __FUNCTION__); + + // validate params + if (!frame) { + CSFLogError(LOGTAG, "%s Null Audio Buffer Pointer", __FUNCTION__); + MOZ_ASSERT(PR_FALSE); + return kMediaConduitMalformedArgument; + } + + // Validate sample length + if (GetNum10msSamplesForFrequency(samplingFreqHz) == 0) { + CSFLogError(LOGTAG, "%s Invalid Sampling Frequency ", __FUNCTION__); + MOZ_ASSERT(PR_FALSE); + return kMediaConduitMalformedArgument; + } + + // If the lock is taken, skip this chunk to avoid blocking the audio thread. + AutoTryReadLock tryLock(mLock); + if (!tryLock) { + CSFLogError(LOGTAG, "%s Conduit going through negotiation ", __FUNCTION__); + return kMediaConduitPlayoutError; + } + + // Conduit should have reception enabled before we ask for decoded + // samples + if (!mRecvStreamRunning) { + CSFLogError(LOGTAG, "%s Engine not Receiving ", __FUNCTION__); + return kMediaConduitSessionNotInited; + } + + // Unfortunate to have to cast to an internal class, but that looks like the + // only way short of interfacing with a layer above (which mixes all streams, + // which we don't want) or a layer below (which we try to avoid because it is + // less stable). + auto info = static_cast<webrtc::AudioReceiveStreamImpl*>(mRecvStream) + ->GetAudioFrameWithInfo(samplingFreqHz, frame); + + if (info == webrtc::AudioMixer::Source::AudioFrameInfo::kError) { + CSFLogError(LOGTAG, "%s Getting audio frame failed", __FUNCTION__); + return kMediaConduitPlayoutError; + } + + CSFLogDebug(LOGTAG, "%s Got %zu channels of %zu samples", __FUNCTION__, + frame->num_channels(), frame->samples_per_channel()); + return kMediaConduitNoError; +} + +// Transport Layer Callbacks +void WebrtcAudioConduit::OnRtpReceived(webrtc::RtpPacketReceived&& aPacket, + webrtc::RTPHeader&& aHeader) { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + + if (mAllowSsrcChange && mRecvStreamConfig.rtp.remote_ssrc != aHeader.ssrc) { + CSFLogDebug(LOGTAG, "%s: switching from SSRC %u to %u", __FUNCTION__, + mRecvStreamConfig.rtp.remote_ssrc, aHeader.ssrc); + OverrideRemoteSSRC(aHeader.ssrc); + } + + CSFLogVerbose(LOGTAG, "%s: seq# %u, Len %zu, SSRC %u (0x%x) ", __FUNCTION__, + aPacket.SequenceNumber(), aPacket.size(), aPacket.Ssrc(), + aPacket.Ssrc()); + + mRtpPacketEvent.Notify(); + if (mCall->Call()) { + mCall->Call()->Receiver()->DeliverRtpPacket( + webrtc::MediaType::AUDIO, std::move(aPacket), + [self = RefPtr<WebrtcAudioConduit>(this)]( + const webrtc::RtpPacketReceived& packet) { + CSFLogVerbose( + LOGTAG, + "AudioConduit %p: failed demuxing packet, ssrc: %u seq: %u", + self.get(), packet.Ssrc(), packet.SequenceNumber()); + return false; + }); + } +} + +void WebrtcAudioConduit::OnRtcpReceived(MediaPacket&& aPacket) { + CSFLogDebug(LOGTAG, "%s", __FUNCTION__); + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + + if (mCall->Call()) { + mCall->Call()->Receiver()->DeliverRtcpPacket( + rtc::CopyOnWriteBuffer(aPacket.data(), aPacket.len())); + } +} + +Maybe<uint16_t> WebrtcAudioConduit::RtpSendBaseSeqFor(uint32_t aSsrc) const { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + auto it = mRtpSendBaseSeqs.find(aSsrc); + if (it == mRtpSendBaseSeqs.end()) { + return Nothing(); + } + return Some(it->second); +} + +const dom::RTCStatsTimestampMaker& WebrtcAudioConduit::GetTimestampMaker() + const { + return mCall->GetTimestampMaker(); +} + +void WebrtcAudioConduit::StopTransmitting() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(!mLock.LockedForWritingByCurrentThread()); + + if (!mSendStreamRunning) { + return; + } + + if (mSendStream) { + CSFLogDebug(LOGTAG, "%s Stopping send stream", __FUNCTION__); + mSendStream->Stop(); + } + + mSendStreamRunning = false; +} + +void WebrtcAudioConduit::StartTransmitting() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(mSendStream); + MOZ_ASSERT(!mLock.LockedForWritingByCurrentThread()); + + if (mSendStreamRunning) { + return; + } + + CSFLogDebug(LOGTAG, "%s Starting send stream", __FUNCTION__); + + mCall->Call()->SignalChannelNetworkState(webrtc::MediaType::AUDIO, + webrtc::kNetworkUp); + mSendStream->Start(); + mSendStreamRunning = true; +} + +void WebrtcAudioConduit::StopReceiving() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(!mLock.LockedForWritingByCurrentThread()); + + if (!mRecvStreamRunning) { + return; + } + + if (mRecvStream) { + CSFLogDebug(LOGTAG, "%s Stopping recv stream", __FUNCTION__); + mRecvStream->Stop(); + } + + mRecvStreamRunning = false; +} + +void WebrtcAudioConduit::StartReceiving() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(mRecvStream); + MOZ_ASSERT(!mLock.LockedForWritingByCurrentThread()); + + if (mRecvStreamRunning) { + return; + } + + CSFLogDebug(LOGTAG, "%s Starting receive stream (SSRC %u (0x%x))", + __FUNCTION__, mRecvStreamConfig.rtp.remote_ssrc, + mRecvStreamConfig.rtp.remote_ssrc); + + mCall->Call()->SignalChannelNetworkState(webrtc::MediaType::AUDIO, + webrtc::kNetworkUp); + mRecvStream->Start(); + mRecvStreamRunning = true; +} + +bool WebrtcAudioConduit::SendRtp(const uint8_t* aData, size_t aLength, + const webrtc::PacketOptions& aOptions) { + MOZ_ASSERT(aLength >= 12); + const uint16_t seqno = ntohs(*((uint16_t*)&aData[2])); + const uint32_t ssrc = ntohl(*((uint32_t*)&aData[8])); + + CSFLogVerbose( + LOGTAG, + "AudioConduit %p: Sending RTP Packet seq# %u, len %zu, SSRC %u (0x%x)", + this, seqno, aLength, ssrc, ssrc); + + if (!mTransportActive) { + CSFLogError(LOGTAG, "AudioConduit %p: RTP Packet Send Failed ", this); + return false; + } + + MediaPacket packet; + packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION); + packet.SetType(MediaPacket::RTP); + mSenderRtpSendEvent.Notify(std::move(packet)); + + // Parse the sequence number of the first rtp packet as base_seq. + const auto inserted = mRtpSendBaseSeqs_n.insert({ssrc, seqno}).second; + + if (inserted || aOptions.packet_id >= 0) { + int64_t now_ms = PR_Now() / 1000; + MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction( + __func__, [this, self = RefPtr<WebrtcAudioConduit>(this), + packet_id = aOptions.packet_id, now_ms, ssrc, seqno] { + mRtpSendBaseSeqs.insert({ssrc, seqno}); + if (packet_id >= 0) { + if (mCall->Call()) { + // TODO: This notification should ideally happen after the + // transport layer has sent the packet on the wire. + mCall->Call()->OnSentPacket({packet_id, now_ms}); + } + } + }))); + } + return true; +} + +bool WebrtcAudioConduit::SendSenderRtcp(const uint8_t* aData, size_t aLength) { + CSFLogVerbose( + LOGTAG, + "AudioConduit %p: Sending RTCP SR Packet, len %zu, SSRC %u (0x%x)", this, + aLength, (uint32_t)ntohl(*((uint32_t*)&aData[4])), + (uint32_t)ntohl(*((uint32_t*)&aData[4]))); + + if (!mTransportActive) { + CSFLogError(LOGTAG, "%s RTCP SR Packet Send Failed ", __FUNCTION__); + return false; + } + + MediaPacket packet; + packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION); + packet.SetType(MediaPacket::RTCP); + mSenderRtcpSendEvent.Notify(std::move(packet)); + return true; +} + +bool WebrtcAudioConduit::SendReceiverRtcp(const uint8_t* aData, + size_t aLength) { + CSFLogVerbose( + LOGTAG, + "AudioConduit %p: Sending RTCP RR Packet, len %zu, SSRC %u (0x%x)", this, + aLength, (uint32_t)ntohl(*((uint32_t*)&aData[4])), + (uint32_t)ntohl(*((uint32_t*)&aData[4]))); + + if (!mTransportActive) { + CSFLogError(LOGTAG, "AudioConduit %p: RTCP RR Packet Send Failed", this); + return false; + } + + MediaPacket packet; + packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION); + packet.SetType(MediaPacket::RTCP); + mReceiverRtcpSendEvent.Notify(std::move(packet)); + return true; +} + +/** + * Supported Sampling Frequencies. + */ +bool WebrtcAudioConduit::IsSamplingFreqSupported(int freq) const { + return GetNum10msSamplesForFrequency(freq) != 0; +} + +std::vector<webrtc::RtpSource> WebrtcAudioConduit::GetUpstreamRtpSources() + const { + MOZ_ASSERT(NS_IsMainThread()); + std::vector<webrtc::RtpSource> sources; + { + AutoReadLock lock(mLock); + if (mRecvStream) { + sources = mRecvStream->GetSources(); + } + } + return sources; +} + +/* Return block-length of 10 ms audio frame in number of samples */ +unsigned int WebrtcAudioConduit::GetNum10msSamplesForFrequency( + int samplingFreqHz) const { + switch (samplingFreqHz) { + case 16000: + return 160; // 160 samples + case 32000: + return 320; // 320 samples + case 44100: + return 441; // 441 samples + case 48000: + return 480; // 480 samples + default: + return 0; // invalid or unsupported + } +} + +/** + * Perform validation on the codecConfig to be applied. + * Verifies if the codec is already applied. + */ +MediaConduitErrorCode WebrtcAudioConduit::ValidateCodecConfig( + const AudioCodecConfig& codecInfo, bool send) { + if (codecInfo.mName.empty()) { + CSFLogError(LOGTAG, "%s Empty Payload Name ", __FUNCTION__); + return kMediaConduitMalformedArgument; + } + + // Only mono or stereo channels supported + if ((codecInfo.mChannels != 1) && (codecInfo.mChannels != 2)) { + CSFLogError(LOGTAG, "%s Channel Unsupported ", __FUNCTION__); + return kMediaConduitMalformedArgument; + } + + return kMediaConduitNoError; +} + +RtpExtList WebrtcAudioConduit::FilterExtensions(LocalDirection aDirection, + const RtpExtList& aExtensions) { + const bool isSend = aDirection == LocalDirection::kSend; + RtpExtList filteredExtensions; + + for (const auto& extension : aExtensions) { + // ssrc-audio-level RTP header extension + if (extension.uri == webrtc::RtpExtension::kAudioLevelUri) { + filteredExtensions.push_back( + webrtc::RtpExtension(extension.uri, extension.id)); + } + + // csrc-audio-level RTP header extension + if (extension.uri == webrtc::RtpExtension::kCsrcAudioLevelsUri) { + if (isSend) { + continue; + } + filteredExtensions.push_back( + webrtc::RtpExtension(extension.uri, extension.id)); + } + + // MID RTP header extension + if (extension.uri == webrtc::RtpExtension::kMidUri) { + if (!isSend) { + // TODO: recv mid support, see also bug 1727211 + continue; + } + filteredExtensions.push_back( + webrtc::RtpExtension(extension.uri, extension.id)); + } + } + + return filteredExtensions; +} + +webrtc::SdpAudioFormat WebrtcAudioConduit::CodecConfigToLibwebrtcFormat( + const AudioCodecConfig& aConfig) { + webrtc::SdpAudioFormat::Parameters parameters; + if (aConfig.mName == kOpusCodecName) { + if (aConfig.mChannels == 2) { + parameters[kCodecParamStereo] = kParamValueTrue; + } + if (aConfig.mFECEnabled) { + parameters[kCodecParamUseInbandFec] = kParamValueTrue; + } + if (aConfig.mDTXEnabled) { + parameters[kCodecParamUseDtx] = kParamValueTrue; + } + if (aConfig.mMaxPlaybackRate) { + parameters[kCodecParamMaxPlaybackRate] = + std::to_string(aConfig.mMaxPlaybackRate); + } + if (aConfig.mMaxAverageBitrate) { + parameters[kCodecParamMaxAverageBitrate] = + std::to_string(aConfig.mMaxAverageBitrate); + } + if (aConfig.mFrameSizeMs) { + parameters[kCodecParamPTime] = std::to_string(aConfig.mFrameSizeMs); + } + if (aConfig.mMinFrameSizeMs) { + parameters[kCodecParamMinPTime] = std::to_string(aConfig.mMinFrameSizeMs); + } + if (aConfig.mMaxFrameSizeMs) { + parameters[kCodecParamMaxPTime] = std::to_string(aConfig.mMaxFrameSizeMs); + } + if (aConfig.mCbrEnabled) { + parameters[kCodecParamCbr] = kParamValueTrue; + } + } + + return webrtc::SdpAudioFormat(aConfig.mName, aConfig.mFreq, aConfig.mChannels, + parameters); +} + +void WebrtcAudioConduit::DeleteSendStream() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(mLock.LockedForWritingByCurrentThread()); + + if (!mSendStream) { + return; + } + + mCall->Call()->DestroyAudioSendStream(mSendStream); + mSendStreamRunning = false; + mSendStream = nullptr; + + // Reset base_seqs in case ssrcs get re-used. + mRtpSendBaseSeqs.clear(); +} + +void WebrtcAudioConduit::CreateSendStream() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(mLock.LockedForWritingByCurrentThread()); + + if (mSendStream) { + return; + } + + mSendStream = mCall->Call()->CreateAudioSendStream(mSendStreamConfig); +} + +void WebrtcAudioConduit::DeleteRecvStream() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(mLock.LockedForWritingByCurrentThread()); + + if (!mRecvStream) { + return; + } + + mCall->Call()->DestroyAudioReceiveStream(mRecvStream); + mRecvStreamRunning = false; + mRecvStream = nullptr; +} + +void WebrtcAudioConduit::CreateRecvStream() { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + MOZ_ASSERT(mLock.LockedForWritingByCurrentThread()); + + if (mRecvStream) { + return; + } + + mRecvStream = mCall->Call()->CreateAudioReceiveStream(mRecvStreamConfig); + // Ensure that we set the jitter buffer target on this stream. + mRecvStream->SetBaseMinimumPlayoutDelayMs(mJitterBufferTargetMs); +} + +void WebrtcAudioConduit::SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) { + MOZ_RELEASE_ASSERT(aTargetMs <= std::numeric_limits<uint16_t>::max()); + MOZ_RELEASE_ASSERT(aTargetMs >= 0); + + MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction( + __func__, + [this, self = RefPtr<WebrtcAudioConduit>(this), targetMs = aTargetMs] { + mJitterBufferTargetMs = static_cast<uint16_t>(targetMs); + if (mRecvStream) { + mRecvStream->SetBaseMinimumPlayoutDelayMs(targetMs); + } + }))); +} + +void WebrtcAudioConduit::DeliverPacket(rtc::CopyOnWriteBuffer packet, + PacketType type) { + // Currently unused. + MOZ_ASSERT(false); +} + +Maybe<int> WebrtcAudioConduit::ActiveSendPayloadType() const { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + + auto stats = GetSenderStats(); + if (!stats) { + return Nothing(); + } + + if (!stats->codec_payload_type) { + return Nothing(); + } + + return Some(*stats->codec_payload_type); +} + +Maybe<int> WebrtcAudioConduit::ActiveRecvPayloadType() const { + MOZ_ASSERT(mCallThread->IsOnCurrentThread()); + + auto stats = GetReceiverStats(); + if (!stats) { + return Nothing(); + } + + if (!stats->codec_payload_type) { + return Nothing(); + } + + return Some(*stats->codec_payload_type); +} + +} // namespace mozilla |