Implement Cast Streaming mirroring, DLNA casting, daemon+GUI, and breadd integration

Builds out the full v1 scope: a vendored+patched openscreen subset for
low-latency Cast Streaming (Mirroring receiver 0F5096E8) alongside the
existing Cast V2/HLS and new DLNA/AVTransport casting paths, breadcastd's
Idle/Casting state machine with a private IPC socket, the breadcast GTK4
popup as a thin IPC client, and bread.cast.*/bread.command.cast.* breadd
integration (device discovery, start/stop, mirroring lifecycle events).
Also adds bakery/systemd/Forgejo CI packaging.

Validated end-to-end against a real Chromecast/Google TV: negotiated
Cast Streaming session, live pipeline playback, and daemon+GUI click-to-cast/
stop through the actual popup.

breadcast-caststream-sys/vendor/openscreen/cast/streaming/sender_packet_router.h

@@ -0,0 +1,203 @@
+// Copyright 2020 The Chromium Authors
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef CAST_STREAMING_SENDER_PACKET_ROUTER_H_
+#define CAST_STREAMING_SENDER_PACKET_ROUTER_H_
+
+#include <stdint.h>
+
+#include <chrono>
+#include <memory>
+#include <vector>
+
+#include "cast/streaming/impl/bandwidth_estimator.h"
+#include "cast/streaming/public/constants.h"
+#include "cast/streaming/public/environment.h"
+#include "cast/streaming/ssrc.h"
+#include "platform/api/time.h"
+#include "platform/base/span.h"
+#include "util/alarm.h"
+#include "util/raw_ptr.h"
+#include "util/raw_ref.h"
+
+namespace openscreen::cast {
+
+// Manages network packet transmission for one or more Senders, directing each
+// inbound packet to a specific Sender instance, pacing the transmission of
+// outbound packets, and employing network bandwidth/availability monitoring and
+// congestion control.
+//
+// Instead of just sending packets whenever they want, Senders must request
+// transmission from the SenderPacketRouter. The router then calls-back to each
+// Sender, in the near future, when it has allocated an available time slice for
+// transmission. The Sender is allowed to decide, at that exact moment, which
+// packet most needs to be sent.
+//
+// Pacing strategy: Packets are sent in bursts. This allows the platform
+// (operating system) to collect many small packets into a short-term buffer,
+// which allows for optimizations at the link layer. For example, multiple
+// packets can be sent together as one larger transmission unit, and this can be
+// critical for good performance over shared-medium networks (such as 802.11
+// WiFi). https://en.wikipedia.org/wiki/Frame-bursting
+class SenderPacketRouter : public BandwidthEstimator,
+                           public Environment::PacketConsumer {
+ public:
+  class Sender {
+   public:
+    // Called to provide the Sender with what looks like a RTCP packet meant for
+    // it specifically (among other Senders) to process. `arrival_time`
+    // indicates when the packet arrived (i.e., when it was received from the
+    // platform).
+    virtual void OnReceivedRtcpPacket(Clock::time_point arrival_time,
+                                      ByteView packet) = 0;
+
+    // Populates the given `buffer` with a RTCP/RTP packet that will be sent
+    // immediately. Returns the portion of `buffer` contaning the packet, or an
+    // empty Span if nothing is ready to send.
+    virtual ByteBuffer GetRtcpPacketForImmediateSend(
+        Clock::time_point send_time,
+        ByteBuffer buffer) = 0;
+    virtual ByteBuffer GetRtpPacketForImmediateSend(Clock::time_point send_time,
+                                                    ByteBuffer buffer) = 0;
+
+    // Returns the point-in-time at which RTP sending should resume, or kNever
+    // if it should be suspended until an explicit call to RequestRtpSend(). The
+    // implementation may return a value on or before "now" to indicate an
+    // immediate resume is desired.
+    virtual Clock::time_point GetRtpResumeTime() = 0;
+
+    // Returns the last logged RTP timestamp, for use in expanding truncated
+    // packet RTP timestamps for metrics purposes.
+    virtual RtpTimeTicks GetLastRtpTimestamp() const = 0;
+
+    // Returns the type of stream that this sender is providing.
+    virtual StreamType GetStreamType() const = 0;
+
+   protected:
+    virtual ~Sender();
+  };
+
+  // Constructs an instance with default burst parameters appropriate for the
+  // given `max_burst_bitrate`.
+  explicit SenderPacketRouter(Environment& environment,
+                              int max_burst_bitrate = kDefaultMaxBurstBitrate);
+
+  // Constructs an instance with specific burst parameters. The maximum bitrate
+  // will be computed based on these (and Environment::GetMaxPacketSize()).
+  SenderPacketRouter(Environment& environment,
+                     int max_packets_per_burst,
+                     std::chrono::milliseconds burst_interval);
+
+  ~SenderPacketRouter();
+
+  int max_packet_size() const { return packet_buffer_size_; }
+  int max_burst_bitrate() const { return max_burst_bitrate_; }
+
+  // Called from a Sender constructor/destructor to register/deregister a Sender
+  // instance that processes RTP/RTCP packets from a Receiver having the given
+  // SSRC.
+  void OnSenderCreated(Ssrc receiver_ssrc, Sender* client);
+  void OnSenderDestroyed(Ssrc receiver_ssrc);
+
+  // Requests an immediate send of a RTCP packet, and then RTCP sending will
+  // repeat at regular intervals (see kRtcpSendInterval) until the Sender is
+  // de-registered.
+  void RequestRtcpSend(Ssrc receiver_ssrc);
+
+  // Requests an immediate send of a RTP packet. RTP sending will continue until
+  // the Sender stops providing packet data.
+  //
+  // See also: Sender::GetRtpResumeTime().
+  void RequestRtpSend(Ssrc receiver_ssrc);
+
+  // A reasonable default maximum bitrate for bursting. Congestion control
+  // should always be employed to limit the Senders' sustained/average outbound
+  // data volume for "fair" use of the network.
+  static constexpr int kDefaultMaxBurstBitrate = 24 << 20;  // 24 megabits/sec
+
+  // The minimum amount of time between burst-sends. The methodology by which
+  // this value was determined is lost knowledge, but is likely the result of
+  // experimentation with various network and operating system configurations.
+  // This value came from the original Chrome Cast Streaming implementation.
+  static constexpr std::chrono::milliseconds kDefaultBurstInterval{10};
+
+  // A special time_point value representing "never."
+  static constexpr Clock::time_point kNever = Clock::time_point::max();
+
+ private:
+  struct SenderEntry {
+    Ssrc receiver_ssrc;
+    raw_ptr<Sender> sender;
+    Clock::time_point next_rtcp_send_time;
+    Clock::time_point next_rtp_send_time;
+
+    // Entries are ordered by the transmission priority (high→low), as implied
+    // by their SSRC. See ssrc.h for details.
+    bool operator<(const SenderEntry& other) const {
+      return ComparePriority(receiver_ssrc, other.receiver_ssrc) < 0;
+    }
+  };
+
+  using SenderEntries = std::vector<SenderEntry>;
+
+  // Environment::PacketConsumer implementation.
+  void OnReceivedPacket(const IPEndpoint& source,
+                        Clock::time_point arrival_time,
+                        std::vector<uint8_t> packet) final;
+
+  // Helper to return an iterator pointing to the entry corresponding to the
+  // given `receiver_ssrc`, or "end" if not found.
+  SenderEntries::iterator FindEntry(Ssrc receiver_ssrc);
+
+  // Examine the next send time for all Senders, and decide whether to schedule
+  // a burst-send.
+  void ScheduleNextBurst();
+
+  // Performs a burst-send of packets. This is called whenever the Alarm fires.
+  void SendBurstOfPackets();
+
+  // Send an RTCP packet from each Sender that has one ready, and return the
+  // number of packets sent.
+  int SendJustTheRtcpPackets(Clock::time_point send_time);
+
+  // Send zero or more RTP packets from each Sender, up to a maximum of
+  // `num_packets_to_send`, and return the number of packets sent.
+  int SendJustTheRtpPackets(Clock::time_point send_time,
+                            int num_packets_to_send);
+
+  // Returns the maximum number of packets to send in one burst, based on the
+  // given parameters.
+  static int ComputeMaxPacketsPerBurst(
+      int max_burst_bitrate,
+      int packet_size,
+      std::chrono::milliseconds burst_interval);
+
+  // Returns the maximum bitrate inferred by the given parameters.
+  static int ComputeMaxBurstBitrate(int packet_size,
+                                    int max_packets_per_burst,
+                                    std::chrono::milliseconds burst_interval);
+
+  const raw_ref<Environment> environment_;
+  const int packet_buffer_size_;
+  const std::unique_ptr<uint8_t[]> packet_buffer_;
+  const int max_packets_per_burst_;
+  const std::chrono::milliseconds burst_interval_;
+  const int max_burst_bitrate_;
+
+  // Schedules the task that calls back into this SenderPacketRouter at a later
+  // time to send the next burst of packets.
+  Alarm alarm_;
+
+  // The current list of Senders and their timing information. This is
+  // maintained in order of the priority implied by the Sender SSRC's.
+  SenderEntries senders_;
+
+  // The last time a burst of packets was sent. This is used to determine the
+  // next burst time.
+  Clock::time_point last_burst_time_ = Clock::time_point::min();
+};
+
+}  // namespace openscreen::cast
+
+#endif  // CAST_STREAMING_SENDER_PACKET_ROUTER_H_