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/impl/rtcp_common.h

@@ -0,0 +1,203 @@
+// Copyright 2019 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_IMPL_RTCP_COMMON_H_
+#define CAST_STREAMING_IMPL_RTCP_COMMON_H_
+
+#include <stdint.h>
+
+#include <optional>
+#include <tuple>
+#include <vector>
+
+#include "cast/streaming/impl/ntp_time.h"
+#include "cast/streaming/impl/rtp_defines.h"
+#include "cast/streaming/impl/statistics_common.h"
+#include "cast/streaming/public/frame_id.h"
+#include "cast/streaming/rtp_time.h"
+#include "cast/streaming/ssrc.h"
+#include "platform/base/span.h"
+
+namespace openscreen::cast {
+
+struct RtcpCommonHeader {
+  RtcpCommonHeader();
+  ~RtcpCommonHeader();
+
+  RtcpPacketType packet_type = RtcpPacketType::kNull;
+
+  union {
+    // The number of report blocks if `packet_type` is kSenderReport or
+    // kReceiverReport.
+    int report_count;
+
+    // Indicates the type of an application-defined message if `packet_type` is
+    // kApplicationDefined or kPayloadSpecific.
+    RtcpSubtype subtype;
+
+    // Otherwise, not used.
+  } with{0};
+
+  // The size (in bytes) of the RTCP packet, not including the header.
+  int payload_size = 0;
+
+  // Serializes this header into the first `kRtcpCommonHeaderSize` bytes of the
+  // given `buffer` and adjusts `buffer` to point to the first byte after it.
+  void AppendFields(ByteBuffer& buffer) const;
+
+  // Parse from the 4-byte wire format in `buffer`. Returns nullopt if the data
+  // is corrupt.
+  static std::optional<RtcpCommonHeader> Parse(ByteView buffer);
+};
+
+// The middle 32-bits of the 64-bit NtpTimestamp field from the Sender Reports.
+// This is used as an opaque identifier that the Receiver will use in its
+// reports to refer to specific previous Sender Reports.
+using StatusReportId = uint32_t;
+constexpr StatusReportId ToStatusReportId(NtpTimestamp ntp_timestamp) {
+  return static_cast<uint32_t>(ntp_timestamp >> 16);
+}
+
+// One of these is optionally included with a Sender Report or a Receiver
+// Report. See: https://tools.ietf.org/html/rfc3550#section-6.4.1
+struct RtcpReportBlock {
+  RtcpReportBlock();
+  ~RtcpReportBlock();
+
+  // The intended recipient of this report block.
+  Ssrc ssrc = 0;
+
+  // The fraction of RTP packets lost since the last report, specified as a
+  // variable numerator and fixed denominator. The numerator will always be in
+  // the range [0,255] since, semantically:
+  //
+  //   a. Negative values are impossible.
+  //   b. Values greater than 255 would indicate 100% packet loss, and so a
+  //      report block would not be generated in the first place.
+  int packet_fraction_lost_numerator = 0;
+  static constexpr int kPacketFractionLostDenominator = 256;
+
+  // The total number of RTP packets lost since the start of the session. This
+  // value will always be in the range [0,2^24-1], as the wire format only
+  // provides 24 bits; so, wrap-around is possible.
+  int cumulative_packets_lost = 0;
+
+  // The highest sequence number received in any RTP packet. Wrap-around is
+  // possible.
+  uint32_t extended_high_sequence_number = 0;
+
+  // An estimate of the recent variance in RTP packet arrival times.
+  RtpTimeDelta jitter;
+
+  // The last Status Report received.
+  StatusReportId last_status_report_id{};
+
+  // The delay between when the peer received the most-recent Status Report and
+  // when this report was sent. The timebase is 65536 ticks per second and,
+  // because of the wire format, this value will always be in the range
+  // [0,65536) seconds.
+  using Delay = std::chrono::duration<int64_t, std::ratio<1, 65536>>;
+  Delay delay_since_last_report{};
+
+  // Convenience helper to compute/assign the `packet_fraction_lost_numerator`,
+  // based on the `num_apparently_sent` and `num_received` packet counts since
+  // the last report was sent.
+  void SetPacketFractionLostNumerator(int64_t num_apparently_sent,
+                                      int64_t num_received);
+
+  // Convenience helper to compute/assign the `cumulative_packets_lost`, based
+  // on the `num_apparently_sent` and `num_received` packet counts since the
+  // start of the entire session.
+  void SetCumulativePacketsLost(int64_t num_apparently_sent,
+                                int64_t num_received);
+
+  // Convenience helper to convert the given `local_clock_delay` to the
+  // RtcpReportBlock::Delay timebase, then clamp and assign it to
+  // `delay_since_last_report`.
+  void SetDelaySinceLastReport(Clock::duration local_clock_delay);
+
+  // Serializes this report block in the first `kRtcpReportBlockSize` bytes of
+  // the given `buffer` and adjusts `buffer` to point to the first byte after
+  // it.
+  void AppendFields(ByteBuffer& buffer) const;
+
+  // Scans the wire-format report blocks in `buffer`, searching for one with the
+  // matching `ssrc` and, if found, returns the parse result. Returns nullopt if
+  // the data is corrupt or no report block with the matching SSRC was found.
+  static std::optional<RtcpReportBlock> ParseOne(ByteView buffer,
+                                                 int report_count,
+                                                 Ssrc ssrc);
+};
+
+struct RtcpSenderReport {
+  RtcpSenderReport();
+  ~RtcpSenderReport();
+
+  // The point-in-time at which this report was sent, according to both: 1) the
+  // common reference clock shared by all RTP streams; 2) the RTP timestamp on
+  // the media capture/playout timeline. Together, these are used by a Receiver
+  // to achieve A/V synchronization across RTP streams for playout.
+  Clock::time_point reference_time{};
+  RtpTimeTicks rtp_timestamp;
+
+  // The total number of RTP packets transmitted since the start of the session
+  // (wrap-around is possible).
+  uint32_t send_packet_count = 0;
+
+  // The total number of payload bytes transmitted in RTP packets since the
+  // start of the session (wrap-around is possible).
+  uint32_t send_octet_count = 0;
+
+  // The report block, if present. While the RTCP spec allows for zero or
+  // multiple reports, Cast Streaming only uses zero or one.
+  std::optional<RtcpReportBlock> report_block;
+};
+
+// A pair of IDs that refers to a specific missing packet within a frame. If
+// `packet_id` is kAllPacketsLost, then it represents all the packets of a
+// frame.
+struct PacketNack {
+  FrameId frame_id;
+  FramePacketId packet_id;
+
+  constexpr bool operator==(const PacketNack& other) const {
+    return frame_id == other.frame_id && packet_id == other.packet_id;
+  }
+  constexpr bool operator!=(const PacketNack& other) const {
+    return frame_id != other.frame_id || packet_id != other.packet_id;
+  }
+  constexpr bool operator<(const PacketNack& other) const {
+    return (frame_id < other.frame_id) ||
+           (frame_id == other.frame_id && packet_id < other.packet_id);
+  }
+};
+
+// Statistics events sent from the receiver over RTCP.
+struct RtcpReceiverEventLogMessage {
+  // The statistics event type, may be either a receiver side frame event or
+  // packet event.
+  StatisticsEvent::Type type;
+
+  // The time at which this event occurred.
+  Clock::time_point timestamp;
+
+  // Only set for frame played out events.
+  // If this value is zero the frame is rendered on time.
+  // If this value is positive it means the frame is rendered late.
+  // If this value is negative it means the frame is rendered early.
+  Clock::duration delay;
+
+  // Only set for packet events.
+  // The ID of the packet associated with this event.
+  FramePacketId packet_id;
+};
+
+struct RtcpReceiverFrameLogMessage {
+  RtpTimeTicks rtp_timestamp;
+  std::vector<RtcpReceiverEventLogMessage> messages;
+};
+
+}  // namespace openscreen::cast
+
+#endif  // CAST_STREAMING_IMPL_RTCP_COMMON_H_