blob: d4494d0d6ce5bda54c6241c0703cf1faeb970e1f [file] [log] [blame]
* Copyright 2017 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 <memory>
#include <set>
#include <string>
#include <vector>
#include "webrtc/api/ortc/ortcrtpreceiverinterface.h"
#include "webrtc/api/ortc/ortcrtpsenderinterface.h"
#include "webrtc/api/ortc/rtptransportcontrollerinterface.h"
#include "webrtc/api/ortc/srtptransportinterface.h"
#include "webrtc/call/call.h"
#include "webrtc/call/rtp_transport_controller_send.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/mediachannel.h" // For MediaConfig.
#include "webrtc/pc/channelmanager.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/sigslot.h"
#include "webrtc/rtc_base/thread.h"
namespace webrtc {
class RtpTransportAdapter;
class OrtcRtpSenderAdapter;
class OrtcRtpReceiverAdapter;
// Implementation of RtpTransportControllerInterface. Wraps a Call,
// a VoiceChannel and VideoChannel, and maintains a list of dependent RTP
// transports.
// When used along with an RtpSenderAdapter or RtpReceiverAdapter, the
// sender/receiver passes its parameters along to this class, which turns them
// into cricket:: media descriptions (the interface used by BaseChannel).
// Due to the fact that BaseChannel has different subclasses for audio/video,
// the actual BaseChannel object is not created until an RtpSender/RtpReceiver
// needs them.
// All methods should be called on the signaling thread.
// TODO(deadbeef): When BaseChannel is split apart into separate
// "RtpSender"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this adapter
// object can be replaced by a "real" one.
class RtpTransportControllerAdapter : public RtpTransportControllerInterface,
public sigslot::has_slots<> {
// Creates a proxy that will call "public interface" methods on the correct
// thread.
// Doesn't take ownership of any objects passed in.
// |channel_manager| must not be null.
static std::unique_ptr<RtpTransportControllerInterface> CreateProxied(
const cricket::MediaConfig& config,
cricket::ChannelManager* channel_manager,
webrtc::RtcEventLog* event_log,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread);
~RtpTransportControllerAdapter() override;
// RtpTransportControllerInterface implementation.
std::vector<RtpTransportInterface*> GetTransports() const override;
// These methods are used by OrtcFactory to create RtpTransports, RtpSenders
// and RtpReceivers using this controller. Called "CreateProxied" because
// these methods return proxies that will safely call methods on the correct
// thread.
RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateProxiedRtpTransport(
const RtpTransportParameters& rtcp_parameters,
PacketTransportInterface* rtp,
PacketTransportInterface* rtcp);
CreateProxiedSrtpTransport(const RtpTransportParameters& rtcp_parameters,
PacketTransportInterface* rtp,
PacketTransportInterface* rtcp);
// |transport_proxy| needs to be a proxy to a transport because the
// application may call GetTransport() on the returned sender or receiver,
// and expects it to return a thread-safe transport proxy.
RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateProxiedRtpSender(
cricket::MediaType kind,
RtpTransportInterface* transport_proxy);
CreateProxiedRtpReceiver(cricket::MediaType kind,
RtpTransportInterface* transport_proxy);
// Methods used internally by other "adapter" classes.
rtc::Thread* signaling_thread() const { return signaling_thread_; }
rtc::Thread* worker_thread() const { return worker_thread_; }
// |parameters.keepalive| will be set for ALL RTP transports in the call.
RTCError SetRtpTransportParameters(const RtpTransportParameters& parameters,
RtpTransportInterface* inner_transport);
void SetRtpTransportParameters_w(const RtpTransportParameters& parameters);
cricket::VoiceChannel* voice_channel() { return voice_channel_; }
cricket::VideoChannel* video_channel() { return video_channel_; }
// |primary_ssrc| out parameter is filled with either
// |parameters.encodings[0].ssrc|, or a generated SSRC if that's left unset.
RTCError ValidateAndApplyAudioSenderParameters(
const RtpParameters& parameters,
uint32_t* primary_ssrc);
RTCError ValidateAndApplyVideoSenderParameters(
const RtpParameters& parameters,
uint32_t* primary_ssrc);
RTCError ValidateAndApplyAudioReceiverParameters(
const RtpParameters& parameters);
RTCError ValidateAndApplyVideoReceiverParameters(
const RtpParameters& parameters);
RtpTransportControllerAdapter* GetInternal() override { return this; }
// Only expected to be called by RtpTransportControllerAdapter::CreateProxied.
RtpTransportControllerAdapter(const cricket::MediaConfig& config,
cricket::ChannelManager* channel_manager,
webrtc::RtcEventLog* event_log,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread);
void Init_w();
void Close_w();
// These return an error if another of the same type of object is already
// attached, or if |transport_proxy| can't be used with the sender/receiver
// due to the limitation that the sender/receiver of the same media type must
// use the same transport.
RTCError AttachAudioSender(OrtcRtpSenderAdapter* sender,
RtpTransportInterface* inner_transport);
RTCError AttachVideoSender(OrtcRtpSenderAdapter* sender,
RtpTransportInterface* inner_transport);
RTCError AttachAudioReceiver(OrtcRtpReceiverAdapter* receiver,
RtpTransportInterface* inner_transport);
RTCError AttachVideoReceiver(OrtcRtpReceiverAdapter* receiver,
RtpTransportInterface* inner_transport);
void OnRtpTransportDestroyed(RtpTransportAdapter* transport);
void OnAudioSenderDestroyed();
void OnVideoSenderDestroyed();
void OnAudioReceiverDestroyed();
void OnVideoReceiverDestroyed();
void CreateVoiceChannel();
void CreateVideoChannel();
void DestroyVoiceChannel();
void DestroyVideoChannel();
void CopyRtcpParametersToDescriptions(
const RtcpParameters& params,
cricket::MediaContentDescription* local,
cricket::MediaContentDescription* remote);
// Helper function to generate an SSRC that doesn't match one in any of the
// "content description" structs, or in |new_ssrcs| (which is needed since
// multiple SSRCs may be generated in one go).
uint32_t GenerateUnusedSsrc(std::set<uint32_t>* new_ssrcs) const;
// |description| is the matching description where existing SSRCs can be
// found.
// This is a member function because it may need to generate SSRCs that don't
// match existing ones, which is more than ToStreamParamsVec does.
RTCErrorOr<cricket::StreamParamsVec> MakeSendStreamParamsVec(
std::vector<RtpEncodingParameters> encodings,
const std::string& cname,
const cricket::MediaContentDescription& description) const;
// If the |rtp_transport| is a SrtpTransport, set the cryptos of the
// audio/video content descriptions.
RTCError MaybeSetCryptos(
RtpTransportInterface* rtp_transport,
cricket::MediaContentDescription* local_description,
cricket::MediaContentDescription* remote_description);
rtc::Thread* signaling_thread_;
rtc::Thread* worker_thread_;
// |transport_proxies_| and |inner_audio_transport_|/|inner_audio_transport_|
// are somewhat redundant, but the latter are only set when
// RtpSenders/RtpReceivers are attached to the transport.
std::vector<RtpTransportInterface*> transport_proxies_;
RtpTransportInterface* inner_audio_transport_ = nullptr;
RtpTransportInterface* inner_video_transport_ = nullptr;
const cricket::MediaConfig media_config_;
RtpKeepAliveConfig keepalive_;
cricket::ChannelManager* channel_manager_;
webrtc::RtcEventLog* event_log_;
std::unique_ptr<Call> call_;
webrtc::RtpTransportControllerSend* call_send_rtp_transport_controller_;
// BaseChannel takes content descriptions as input, so we store them here
// such that they can be updated when a new RtpSenderAdapter/
// RtpReceiverAdapter attaches itself.
cricket::AudioContentDescription local_audio_description_;
cricket::AudioContentDescription remote_audio_description_;
cricket::VideoContentDescription local_video_description_;
cricket::VideoContentDescription remote_video_description_;
cricket::VoiceChannel* voice_channel_ = nullptr;
cricket::VideoChannel* video_channel_ = nullptr;
bool have_audio_sender_ = false;
bool have_video_sender_ = false;
bool have_audio_receiver_ = false;
bool have_video_receiver_ = false;
} // namespace webrtc