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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.
79 lines
3 KiB
C++
79 lines
3 KiB
C++
// 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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#ifndef CAST_STREAMING_IMPL_RTP_PACKETIZER_H_
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#define CAST_STREAMING_IMPL_RTP_PACKETIZER_H_
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#include <stdint.h>
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#include "cast/streaming/impl/frame_crypto.h"
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#include "cast/streaming/impl/rtp_defines.h"
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#include "cast/streaming/ssrc.h"
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#include "platform/base/span.h"
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namespace openscreen::cast {
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// Transforms a logical sequence of EncryptedFrames into RTP packets for
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// transmission. A single instance of RtpPacketizer should be used for all the
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// frames in a Cast RTP stream having the same SSRC.
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class RtpPacketizer {
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public:
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// `payload_type` describes the type of the media content for the RTP stream
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// from the sender having the given `sender_ssrc`.
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//
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// The `max_packet_size` argument depends on the optimal over-the-wire size of
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// packets for the network medium being used. See discussion in rtp_defines.h
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// for further info.
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RtpPacketizer(RtpPayloadType payload_type,
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Ssrc sender_ssrc,
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int max_packet_size);
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~RtpPacketizer();
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// Wire-format one of the RTP packets for the given frame, which must only be
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// transmitted once. This method should be called in the same sequence that
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// packets will be transmitted. This also means that, if a packet needs to be
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// re-transmitted, this method should be called to generate it again. Returns
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// the subspan of `buffer` that contains the packet. `buffer` must be at least
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// as large as the `max_packet_size` passed to the constructor.
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ByteBuffer GeneratePacket(const EncryptedFrame& frame,
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FramePacketId packet_id,
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ByteBuffer buffer);
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// Given `frame`, compute the total number of packets over which the whole
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// frame will be split-up. Returns -1 if the frame is too large and cannot be
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// packetized.
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int ComputeNumberOfPackets(const EncryptedFrame& frame) const;
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// See rtp_defines.h for wire-format diagram.
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static constexpr int kBaseRtpHeaderSize =
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// Plus one byte, because this implementation always includes the 8-bit
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// Reference Frame ID field.
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kRtpPacketMinValidSize + 1;
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static constexpr int kAdaptiveLatencyHeaderSize = 4;
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static constexpr int kMaxRtpHeaderSize =
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kBaseRtpHeaderSize + kAdaptiveLatencyHeaderSize;
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private:
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int max_payload_size() const {
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// Start with the configured max packet size, then subtract reserved space
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// for packet header fields. The rest can be allocated to the payload.
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return max_packet_size_ - kMaxRtpHeaderSize;
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}
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// The validated ctor RtpPayloadType arg, in wire-format form.
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const uint8_t payload_type_7bits_;
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const Ssrc sender_ssrc_;
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const int max_packet_size_;
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// Incremented each time GeneratePacket() is called. Every packet, even those
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// re-transmitted, must have different sequence numbers (within wrap-around
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// concerns) per the RTP spec.
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uint16_t sequence_number_;
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};
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} // namespace openscreen::cast
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#endif // CAST_STREAMING_IMPL_RTP_PACKETIZER_H_
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