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/*
* Copyright (c) 2014 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 MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
#include <stddef.h>
#include <stdint.h>
#include <deque>
#include <queue>
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
namespace webrtc {
// Bit masks for NAL (F, NRI, Type) indicators.
constexpr uint8_t kH264FBit = 0x80;
constexpr uint8_t kH264NriMask = 0x60;
constexpr uint8_t kH264TypeMask = 0x1F;
// Bit masks for FU (A and B) headers.
constexpr uint8_t kH264SBit = 0x80;
constexpr uint8_t kH264EBit = 0x40;
constexpr uint8_t kH264RBit = 0x20;
class RtpPacketizerH264 : public RtpPacketizer {
public:
// Initialize with payload from encoder.
// The payload_data must be exactly one encoded H264 frame.
RtpPacketizerH264(ArrayView<const uint8_t> payload,
PayloadSizeLimits limits,
H264PacketizationMode packetization_mode);
~RtpPacketizerH264() override;
RtpPacketizerH264(const RtpPacketizerH264&) = delete;
RtpPacketizerH264& operator=(const RtpPacketizerH264&) = delete;
size_t NumPackets() const override;
// Get the next payload with H264 payload header.
// Write payload and set marker bit of the `packet`.
// Returns true on success, false otherwise.
bool NextPacket(RtpPacketToSend* rtp_packet) override;
private:
// A packet unit (H264 packet), to be put into an RTP packet:
// If a NAL unit is too large for an RTP packet, this packet unit will
// represent a FU-A packet of a single fragment of the NAL unit.
// If a NAL unit is small enough to fit within a single RTP packet, this
// packet unit may represent a single NAL unit or a STAP-A packet, of which
// there may be multiple in a single RTP packet (if so, aggregated = true).
struct PacketUnit {
PacketUnit(ArrayView<const uint8_t> source_fragment,
bool first_fragment,
bool last_fragment,
bool aggregated,
uint8_t header)
: source_fragment(source_fragment),
first_fragment(first_fragment),
last_fragment(last_fragment),
aggregated(aggregated),
header(header) {}
ArrayView<const uint8_t> source_fragment;
bool first_fragment;
bool last_fragment;
bool aggregated;
uint8_t header;
};
bool GeneratePackets(H264PacketizationMode packetization_mode);
bool PacketizeFuA(size_t fragment_index);
size_t PacketizeStapA(size_t fragment_index);
bool PacketizeSingleNalu(size_t fragment_index);
void NextAggregatePacket(RtpPacketToSend* rtp_packet);
void NextFragmentPacket(RtpPacketToSend* rtp_packet);
const PayloadSizeLimits limits_;
size_t num_packets_left_;
std::deque<ArrayView<const uint8_t>> input_fragments_;
std::queue<PacketUnit> packets_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
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