Implement Cast Streaming mirroring, DLNA casting, daemon+GUI, and breadd integration
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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.
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breadcast-caststream-sys/vendor/openscreen/cast/streaming/impl/rtp_packetizer.cc
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breadcast-caststream-sys/vendor/openscreen/cast/streaming/impl/rtp_packetizer.cc
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// Copyright 2019 The Chromium Authors
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// Use of this source code is governed by a BSD-style license that can be
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// found in the LICENSE file.
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#include "cast/streaming/impl/rtp_packetizer.h"
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#include <algorithm>
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#include <limits>
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#include <random>
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#include "cast/streaming/impl/packet_util.h"
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#include "platform/api/time.h"
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#include "util/big_endian.h"
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#include "util/integer_division.h"
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#include "util/osp_logging.h"
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namespace openscreen::cast {
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namespace {
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// Returns a random sequence number to start with. The reason for using a random
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// number instead of zero is unclear, but this has existed both in several
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// versions of the Cast Streaming spec and in other implementations for many
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// years.
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uint16_t GenerateRandomSequenceNumberStart() {
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// Use a statically-allocated generator, instantiated upon first use, and
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// seeded with the current time tick count. This generator was chosen because
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// it is light-weight and does not need to produce unguessable (nor
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// crypto-secure) values.
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static std::minstd_rand generator(static_cast<std::minstd_rand::result_type>(
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Clock::now().time_since_epoch().count()));
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return std::uniform_int_distribution<uint16_t>()(generator);
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}
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} // namespace
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RtpPacketizer::RtpPacketizer(RtpPayloadType payload_type,
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Ssrc sender_ssrc,
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int max_packet_size)
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: payload_type_7bits_(static_cast<uint8_t>(payload_type)),
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sender_ssrc_(sender_ssrc),
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max_packet_size_(max_packet_size),
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sequence_number_(GenerateRandomSequenceNumberStart()) {
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OSP_CHECK(IsRtpPayloadType(payload_type_7bits_));
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OSP_CHECK_GT(max_packet_size_, kMaxRtpHeaderSize);
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}
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RtpPacketizer::~RtpPacketizer() = default;
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ByteBuffer RtpPacketizer::GeneratePacket(const EncryptedFrame& frame,
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FramePacketId packet_id,
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ByteBuffer buffer) {
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OSP_CHECK_GE(static_cast<int>(buffer.size()), max_packet_size_);
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const int num_packets = ComputeNumberOfPackets(frame);
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OSP_CHECK_GT(num_packets, 0);
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OSP_CHECK_LT(int{packet_id}, num_packets);
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const bool is_last_packet = int{packet_id} == (num_packets - 1);
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// Compute the size of this packet, which is the number of bytes of header
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// plus the number of bytes of payload. Note that the optional Adaptive
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// Latency information is only added to the first packet.
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int packet_size = kBaseRtpHeaderSize;
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const bool include_adaptive_latency_change =
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(packet_id == 0 &&
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frame.new_playout_delay > std::chrono::milliseconds(0));
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if (include_adaptive_latency_change) {
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OSP_CHECK_LE(frame.new_playout_delay.count(),
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int{std::numeric_limits<uint16_t>::max()});
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packet_size += kAdaptiveLatencyHeaderSize;
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}
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int data_chunk_size = max_payload_size();
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const int data_chunk_start = data_chunk_size * int{packet_id};
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if (is_last_packet) {
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data_chunk_size = static_cast<int>(frame.data.size()) - data_chunk_start;
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}
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packet_size += data_chunk_size;
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OSP_CHECK_LE(packet_size, max_packet_size_);
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const ByteBuffer packet(buffer.data(), packet_size);
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// RTP Header.
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AppendField<uint8_t>(kRtpRequiredFirstByte, buffer);
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AppendField<uint8_t>(
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(is_last_packet ? kRtpMarkerBitMask : 0) | payload_type_7bits_, buffer);
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AppendField<uint16_t>(sequence_number_++, buffer);
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AppendField<uint32_t>(frame.rtp_timestamp.lower_32_bits(), buffer);
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AppendField<uint32_t>(sender_ssrc_, buffer);
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// Cast Header.
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AppendField<uint8_t>(
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((frame.dependency == EncodedFrame::Dependency::kKeyFrame)
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? kRtpKeyFrameBitMask
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: 0) |
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kRtpHasReferenceFrameIdBitMask |
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(include_adaptive_latency_change ? 1 : 0),
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buffer);
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AppendField<uint8_t>(frame.frame_id.lower_8_bits(), buffer);
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AppendField<uint16_t>(packet_id, buffer);
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AppendField<uint16_t>(num_packets - 1, buffer);
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AppendField<uint8_t>(frame.referenced_frame_id.lower_8_bits(), buffer);
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// Extension of Cast Header for Adaptive Latency change.
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if (include_adaptive_latency_change) {
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AppendField<uint16_t>(
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(kAdaptiveLatencyRtpExtensionType << kNumExtensionDataSizeFieldBits) |
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sizeof(uint16_t),
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buffer);
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AppendField<uint16_t>(frame.new_playout_delay.count(), buffer);
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}
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// Copy the encrypted payload data into the packet.
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auto data_chunk = frame.data.subspan(data_chunk_start, data_chunk_size);
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std::copy(data_chunk.begin(), data_chunk.end(), buffer.data());
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return packet;
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}
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int RtpPacketizer::ComputeNumberOfPackets(const EncryptedFrame& frame) const {
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// The total number of packets is computed by assuming the payload will be
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// split-up across as few packets as possible.
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int num_packets = DividePositivesRoundingUp(
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static_cast<int>(frame.data.size()), max_payload_size());
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// Edge case: There must always be at least one packet, even when there are no
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// payload bytes. Some audio codecs, for example, use zero bytes to represent
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// a period of silence.
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num_packets = std::max(1, num_packets);
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// Ensure that the entire range of FramePacketIds can be represented.
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return num_packets <= int{kMaxAllowedFramePacketId} ? num_packets : -1;
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}
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} // namespace openscreen::cast
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