Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 491 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/channel.h b/third_party/libwebrtc/pc/channel.h
new file mode 100644
index 0000000000..b3020e377c
--- /dev/null
+++ b/third_party/libwebrtc/pc/channel.h
@@ -0,0 +1,491 @@
+/*
+ *  Copyright 2004 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.
+ */
+
+#ifndef PC_CHANNEL_H_
+#define PC_CHANNEL_H_
+
+#include <stdint.h>
+
+#include <functional>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/crypto/crypto_options.h"
+#include "api/jsep.h"
+#include "api/media_types.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_transceiver_direction.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "call/rtp_demuxer.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "media/base/media_channel.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/stream_params.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "pc/channel_interface.h"
+#include "pc/rtp_transport_internal.h"
+#include "pc/session_description.h"
+#include "rtc_base/async_packet_socket.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/containers/flat_set.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/socket.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/unique_id_generator.h"
+
+namespace cricket {
+
+// BaseChannel contains logic common to voice and video, including enable,
+// marshaling calls to a worker and network threads, and connection and media
+// monitors.
+//
+// BaseChannel assumes signaling and other threads are allowed to make
+// synchronous calls to the worker thread, the worker thread makes synchronous
+// calls only to the network thread, and the network thread can't be blocked by
+// other threads.
+// All methods with _n suffix must be called on network thread,
+//     methods with _w suffix on worker thread
+// and methods with _s suffix on signaling thread.
+// Network and worker threads may be the same thread.
+//
+class VideoChannel;
+class VoiceChannel;
+
+class BaseChannel : public ChannelInterface,
+                    // TODO(tommi): Remove has_slots inheritance.
+                    public sigslot::has_slots<>,
+                    // TODO(tommi): Consider implementing these interfaces
+                    // via composition.
+                    public MediaChannelNetworkInterface,
+                    public webrtc::RtpPacketSinkInterface {
+ public:
+  // If `srtp_required` is true, the channel will not send or receive any
+  // RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
+  // The BaseChannel does not own the UniqueRandomIdGenerator so it is the
+  // responsibility of the user to ensure it outlives this object.
+  // TODO(zhihuang:) Create a BaseChannel::Config struct for the parameter lists
+  // which will make it easier to change the constructor.
+  BaseChannel(rtc::Thread* worker_thread,
+              rtc::Thread* network_thread,
+              rtc::Thread* signaling_thread,
+              std::unique_ptr<MediaChannel> media_channel,
+              absl::string_view mid,
+              bool srtp_required,
+              webrtc::CryptoOptions crypto_options,
+              rtc::UniqueRandomIdGenerator* ssrc_generator);
+  virtual ~BaseChannel();
+
+  rtc::Thread* worker_thread() const { return worker_thread_; }
+  rtc::Thread* network_thread() const { return network_thread_; }
+  const std::string& mid() const override { return demuxer_criteria_.mid(); }
+  // TODO(deadbeef): This is redundant; remove this.
+  absl::string_view transport_name() const override {
+    RTC_DCHECK_RUN_ON(network_thread());
+    if (rtp_transport_)
+      return rtp_transport_->transport_name();
+    return "";
+  }
+
+  // This function returns true if using SRTP (DTLS-based keying or SDES).
+  bool srtp_active() const {
+    RTC_DCHECK_RUN_ON(network_thread());
+    return rtp_transport_ && rtp_transport_->IsSrtpActive();
+  }
+
+  // Set an RTP level transport which could be an RtpTransport without
+  // encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP.
+  // This can be called from any thread and it hops to the network thread
+  // internally. It would replace the `SetTransports` and its variants.
+  bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) override;
+
+  webrtc::RtpTransportInternal* rtp_transport() const {
+    RTC_DCHECK_RUN_ON(network_thread());
+    return rtp_transport_;
+  }
+
+  // Channel control
+  bool SetLocalContent(const MediaContentDescription* content,
+                       webrtc::SdpType type,
+                       std::string& error_desc) override;
+  bool SetRemoteContent(const MediaContentDescription* content,
+                        webrtc::SdpType type,
+                        std::string& error_desc) override;
+  // Controls whether this channel will receive packets on the basis of
+  // matching payload type alone. This is needed for legacy endpoints that
+  // don't signal SSRCs or use MID/RID, but doesn't make sense if there is
+  // more than channel of specific media type, As that creates an ambiguity.
+  //
+  // This method will also remove any existing streams that were bound to this
+  // channel on the basis of payload type, since one of these streams might
+  // actually belong to a new channel. See: crbug.com/webrtc/11477
+  bool SetPayloadTypeDemuxingEnabled(bool enabled) override;
+
+  void Enable(bool enable) override;
+
+  const std::vector<StreamParams>& local_streams() const override {
+    return local_streams_;
+  }
+  const std::vector<StreamParams>& remote_streams() const override {
+    return remote_streams_;
+  }
+
+  // Used for latency measurements.
+  void SetFirstPacketReceivedCallback(std::function<void()> callback) override;
+
+  // From RtpTransport - public for testing only
+  void OnTransportReadyToSend(bool ready);
+
+  // Only public for unit tests.  Otherwise, consider protected.
+  int SetOption(SocketType type, rtc::Socket::Option o, int val) override;
+
+  // RtpPacketSinkInterface overrides.
+  void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
+
+  MediaChannel* media_channel() override { return media_channel_.get(); }
+
+  VideoMediaSendChannelInterface* video_media_send_channel() override {
+    RTC_CHECK(false) << "Attempt to fetch video channel from non-video";
+    return nullptr;
+  }
+  VoiceMediaSendChannelInterface* voice_media_send_channel() override {
+    RTC_CHECK(false) << "Attempt to fetch voice channel from non-voice";
+    return nullptr;
+  }
+  VideoMediaReceiveChannelInterface* video_media_receive_channel() override {
+    RTC_CHECK(false) << "Attempt to fetch video channel from non-video";
+    return nullptr;
+  }
+  VoiceMediaReceiveChannelInterface* voice_media_receive_channel() override {
+    RTC_CHECK(false) << "Attempt to fetch voice channel from non-voice";
+    return nullptr;
+  }
+
+ protected:
+  void set_local_content_direction(webrtc::RtpTransceiverDirection direction)
+      RTC_RUN_ON(worker_thread()) {
+    local_content_direction_ = direction;
+  }
+
+  webrtc::RtpTransceiverDirection local_content_direction() const
+      RTC_RUN_ON(worker_thread()) {
+    return local_content_direction_;
+  }
+
+  void set_remote_content_direction(webrtc::RtpTransceiverDirection direction)
+      RTC_RUN_ON(worker_thread()) {
+    remote_content_direction_ = direction;
+  }
+
+  webrtc::RtpTransceiverDirection remote_content_direction() const
+      RTC_RUN_ON(worker_thread()) {
+    return remote_content_direction_;
+  }
+
+  webrtc::RtpExtension::Filter extensions_filter() const {
+    return extensions_filter_;
+  }
+
+  bool network_initialized() RTC_RUN_ON(network_thread()) {
+    return media_channel_->HasNetworkInterface();
+  }
+
+  bool enabled() const RTC_RUN_ON(worker_thread()) { return enabled_; }
+  rtc::Thread* signaling_thread() const { return signaling_thread_; }
+
+  // Call to verify that:
+  // * The required content description directions have been set.
+  // * The channel is enabled.
+  // * The SRTP filter is active if it's needed.
+  // * The transport has been writable before, meaning it should be at least
+  //   possible to succeed in sending a packet.
+  //
+  // When any of these properties change, UpdateMediaSendRecvState_w should be
+  // called.
+  bool IsReadyToSendMedia_w() const RTC_RUN_ON(worker_thread());
+
+  // NetworkInterface implementation, called by MediaEngine
+  bool SendPacket(rtc::CopyOnWriteBuffer* packet,
+                  const rtc::PacketOptions& options) override;
+  bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
+                const rtc::PacketOptions& options) override;
+
+  // From RtpTransportInternal
+  void OnWritableState(bool writable);
+
+  void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
+
+  bool SendPacket(bool rtcp,
+                  rtc::CopyOnWriteBuffer* packet,
+                  const rtc::PacketOptions& options);
+
+  void EnableMedia_w() RTC_RUN_ON(worker_thread());
+  void DisableMedia_w() RTC_RUN_ON(worker_thread());
+
+  // Performs actions if the RTP/RTCP writable state changed. This should
+  // be called whenever a channel's writable state changes or when RTCP muxing
+  // becomes active/inactive.
+  void UpdateWritableState_n() RTC_RUN_ON(network_thread());
+  void ChannelWritable_n() RTC_RUN_ON(network_thread());
+  void ChannelNotWritable_n() RTC_RUN_ON(network_thread());
+
+  bool SetPayloadTypeDemuxingEnabled_w(bool enabled)
+      RTC_RUN_ON(worker_thread());
+
+  // Should be called whenever the conditions for
+  // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied).
+  // Updates the send/recv state of the media channel.
+  virtual void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) = 0;
+
+  bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
+                            webrtc::SdpType type,
+                            std::string& error_desc)
+      RTC_RUN_ON(worker_thread());
+  bool UpdateRemoteStreams_w(const MediaContentDescription* content,
+                             webrtc::SdpType type,
+                             std::string& error_desc)
+      RTC_RUN_ON(worker_thread());
+  virtual bool SetLocalContent_w(const MediaContentDescription* content,
+                                 webrtc::SdpType type,
+                                 std::string& error_desc)
+      RTC_RUN_ON(worker_thread()) = 0;
+  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
+                                  webrtc::SdpType type,
+                                  std::string& error_desc)
+      RTC_RUN_ON(worker_thread()) = 0;
+
+  // Returns a list of RTP header extensions where any extension URI is unique.
+  // Encrypted extensions will be either preferred or discarded, depending on
+  // the current crypto_options_.
+  RtpHeaderExtensions GetDeduplicatedRtpHeaderExtensions(
+      const RtpHeaderExtensions& extensions);
+
+  // Add `payload_type` to `demuxer_criteria_` if payload type demuxing is
+  // enabled.
+  // Returns true if the demuxer payload type changed and a re-registration
+  // is needed.
+  bool MaybeAddHandledPayloadType(int payload_type) RTC_RUN_ON(worker_thread());
+
+  // Returns true if the demuxer payload type criteria was non-empty before
+  // clearing.
+  bool ClearHandledPayloadTypes() RTC_RUN_ON(worker_thread());
+
+  // Hops to the network thread to update the transport if an update is
+  // requested. If `update_demuxer` is false and `extensions` is not set, the
+  // function simply returns. If either of these is set, the function updates
+  // the transport with either or both of the demuxer criteria and the supplied
+  // rtp header extensions.
+  // Returns `true` if either an update wasn't needed or one was successfully
+  // applied. If the return value is `false`, then updating the demuxer criteria
+  // failed, which needs to be treated as an error.
+  bool MaybeUpdateDemuxerAndRtpExtensions_w(
+      bool update_demuxer,
+      absl::optional<RtpHeaderExtensions> extensions,
+      std::string& error_desc) RTC_RUN_ON(worker_thread());
+
+  bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread());
+
+  // Return description of media channel to facilitate logging
+  std::string ToString() const;
+
+ private:
+  bool ConnectToRtpTransport_n() RTC_RUN_ON(network_thread());
+  void DisconnectFromRtpTransport_n() RTC_RUN_ON(network_thread());
+  void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
+
+  rtc::Thread* const worker_thread_;
+  rtc::Thread* const network_thread_;
+  rtc::Thread* const signaling_thread_;
+  rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> alive_;
+
+  std::function<void()> on_first_packet_received_
+      RTC_GUARDED_BY(network_thread());
+
+  webrtc::RtpTransportInternal* rtp_transport_
+      RTC_GUARDED_BY(network_thread()) = nullptr;
+
+  std::vector<std::pair<rtc::Socket::Option, int> > socket_options_
+      RTC_GUARDED_BY(network_thread());
+  std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_
+      RTC_GUARDED_BY(network_thread());
+  bool writable_ RTC_GUARDED_BY(network_thread()) = false;
+  bool was_ever_writable_n_ RTC_GUARDED_BY(network_thread()) = false;
+  bool was_ever_writable_ RTC_GUARDED_BY(worker_thread()) = false;
+  const bool srtp_required_ = true;
+
+  // Set to either kPreferEncryptedExtension or kDiscardEncryptedExtension
+  // based on the supplied CryptoOptions.
+  const webrtc::RtpExtension::Filter extensions_filter_;
+
+  // MediaChannel related members that should be accessed from the worker
+  // thread.
+  const std::unique_ptr<MediaChannel> media_channel_;
+  // Currently the `enabled_` flag is accessed from the signaling thread as
+  // well, but it can be changed only when signaling thread does a synchronous
+  // call to the worker thread, so it should be safe.
+  bool enabled_ RTC_GUARDED_BY(worker_thread()) = false;
+  bool enabled_s_ RTC_GUARDED_BY(signaling_thread()) = false;
+  bool payload_type_demuxing_enabled_ RTC_GUARDED_BY(worker_thread()) = true;
+  std::vector<StreamParams> local_streams_ RTC_GUARDED_BY(worker_thread());
+  std::vector<StreamParams> remote_streams_ RTC_GUARDED_BY(worker_thread());
+  webrtc::RtpTransceiverDirection local_content_direction_ RTC_GUARDED_BY(
+      worker_thread()) = webrtc::RtpTransceiverDirection::kInactive;
+  webrtc::RtpTransceiverDirection remote_content_direction_ RTC_GUARDED_BY(
+      worker_thread()) = webrtc::RtpTransceiverDirection::kInactive;
+
+  // Cached list of payload types, used if payload type demuxing is re-enabled.
+  webrtc::flat_set<uint8_t> payload_types_ RTC_GUARDED_BY(worker_thread());
+  // A stored copy of the rtp header extensions as applied to the transport.
+  RtpHeaderExtensions rtp_header_extensions_ RTC_GUARDED_BY(worker_thread());
+  // TODO(bugs.webrtc.org/12239): Modified on worker thread, accessed
+  // on network thread in RegisterRtpDemuxerSink_n (called from Init_w)
+  webrtc::RtpDemuxerCriteria demuxer_criteria_;
+  // This generator is used to generate SSRCs for local streams.
+  // This is needed in cases where SSRCs are not negotiated or set explicitly
+  // like in Simulcast.
+  // This object is not owned by the channel so it must outlive it.
+  rtc::UniqueRandomIdGenerator* const ssrc_generator_;
+};
+
+// VoiceChannel is a specialization that adds support for early media, DTMF,
+// and input/output level monitoring.
+class VoiceChannel : public BaseChannel {
+ public:
+  VoiceChannel(rtc::Thread* worker_thread,
+               rtc::Thread* network_thread,
+               rtc::Thread* signaling_thread,
+               std::unique_ptr<VoiceMediaChannel> channel,
+               absl::string_view mid,
+               bool srtp_required,
+               webrtc::CryptoOptions crypto_options,
+               rtc::UniqueRandomIdGenerator* ssrc_generator);
+  ~VoiceChannel();
+
+  VideoChannel* AsVideoChannel() override {
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+  VoiceChannel* AsVoiceChannel() override { return this; }
+
+  VoiceMediaSendChannelInterface* media_send_channel() override {
+    return &send_channel_;
+  }
+
+  VoiceMediaSendChannelInterface* voice_media_send_channel() override {
+    return &send_channel_;
+  }
+
+  VoiceMediaReceiveChannelInterface* media_receive_channel() override {
+    return &receive_channel_;
+  }
+
+  VoiceMediaReceiveChannelInterface* voice_media_receive_channel() override {
+    return &receive_channel_;
+  }
+
+  cricket::MediaType media_type() const override {
+    return cricket::MEDIA_TYPE_AUDIO;
+  }
+
+ private:
+  // overrides from BaseChannel
+  void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override;
+  bool SetLocalContent_w(const MediaContentDescription* content,
+                         webrtc::SdpType type,
+                         std::string& error_desc)
+      RTC_RUN_ON(worker_thread()) override;
+  bool SetRemoteContent_w(const MediaContentDescription* content,
+                          webrtc::SdpType type,
+                          std::string& error_desc)
+      RTC_RUN_ON(worker_thread()) override;
+
+  VoiceMediaSendChannel send_channel_ RTC_GUARDED_BY(worker_thread());
+  VoiceMediaReceiveChannel receive_channel_ RTC_GUARDED_BY(worker_thread());
+  // Last AudioSendParameters sent down to the media_channel() via
+  // SetSendParameters.
+  AudioSendParameters last_send_params_ RTC_GUARDED_BY(worker_thread());
+  // Last AudioRecvParameters sent down to the media_channel() via
+  // SetRecvParameters.
+  AudioRecvParameters last_recv_params_ RTC_GUARDED_BY(worker_thread());
+};
+
+// VideoChannel is a specialization for video.
+class VideoChannel : public BaseChannel {
+ public:
+  VideoChannel(rtc::Thread* worker_thread,
+               rtc::Thread* network_thread,
+               rtc::Thread* signaling_thread,
+               std::unique_ptr<VideoMediaChannel> media_channel,
+               absl::string_view mid,
+               bool srtp_required,
+               webrtc::CryptoOptions crypto_options,
+               rtc::UniqueRandomIdGenerator* ssrc_generator);
+  ~VideoChannel();
+
+  VideoChannel* AsVideoChannel() override { return this; }
+  VoiceChannel* AsVoiceChannel() override {
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  VideoMediaSendChannelInterface* media_send_channel() override {
+    return &send_channel_;
+  }
+
+  VideoMediaSendChannelInterface* video_media_send_channel() override {
+    return &send_channel_;
+  }
+
+  VideoMediaReceiveChannelInterface* media_receive_channel() override {
+    return &receive_channel_;
+  }
+
+  VideoMediaReceiveChannelInterface* video_media_receive_channel() override {
+    return &receive_channel_;
+  }
+
+  cricket::MediaType media_type() const override {
+    return cricket::MEDIA_TYPE_VIDEO;
+  }
+
+ private:
+  // overrides from BaseChannel
+  void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override;
+  bool SetLocalContent_w(const MediaContentDescription* content,
+                         webrtc::SdpType type,
+                         std::string& error_desc)
+      RTC_RUN_ON(worker_thread()) override;
+  bool SetRemoteContent_w(const MediaContentDescription* content,
+                          webrtc::SdpType type,
+                          std::string& error_desc)
+      RTC_RUN_ON(worker_thread()) override;
+
+  VideoMediaSendChannel send_channel_ RTC_GUARDED_BY(worker_thread());
+  VideoMediaReceiveChannel receive_channel_ RTC_GUARDED_BY(worker_thread());
+  // Last VideoSendParameters sent down to the media_channel() via
+  // SetSendParameters.
+  VideoSendParameters last_send_params_ RTC_GUARDED_BY(worker_thread());
+  // Last VideoRecvParameters sent down to the media_channel() via
+  // SetRecvParameters.
+  VideoRecvParameters last_recv_params_ RTC_GUARDED_BY(worker_thread());
+};
+
+}  // namespace cricket
+
+#endif  // PC_CHANNEL_H_