blob: efa9b97fb85abe4f2053c1134602adde4fcbcb5c [file] [log] [blame]
* 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.
#include <deque>
#include <memory>
#include <queue>
#include <string>
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
class RtpPacketizerH264 : public RtpPacketizer {
// Initialize with payload from encoder.
// The payload_data must be exactly one encoded H264 frame.
RtpPacketizerH264(size_t max_payload_len,
size_t last_packet_reduction_len,
H264PacketizationMode packetization_mode);
virtual ~RtpPacketizerH264();
size_t SetPayloadData(const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) 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;
std::string ToString() override;
// Input fragments (NAL units), with an optionally owned temporary buffer,
// used in case the fragment gets modified.
struct Fragment {
Fragment(const uint8_t* buffer, size_t length);
explicit Fragment(const Fragment& fragment);
const uint8_t* buffer = nullptr;
size_t length = 0;
std::unique_ptr<rtc::Buffer> tmp_buffer;
// 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(const Fragment& source_fragment,
bool first_fragment,
bool last_fragment,
bool aggregated,
uint8_t header)
: source_fragment(source_fragment),
header(header) {}
const Fragment source_fragment;
bool first_fragment;
bool last_fragment;
bool aggregated;
uint8_t header;
void GeneratePackets();
void PacketizeFuA(size_t fragment_index);
size_t PacketizeStapA(size_t fragment_index);
void PacketizeSingleNalu(size_t fragment_index);
void NextAggregatePacket(RtpPacketToSend* rtp_packet, bool last);
void NextFragmentPacket(RtpPacketToSend* rtp_packet);
const size_t max_payload_len_;
const size_t last_packet_reduction_len_;
size_t num_packets_left_;
const H264PacketizationMode packetization_mode_;
std::deque<Fragment> input_fragments_;
std::queue<PacketUnit> packets_;
// Depacketizer for H264.
class RtpDepacketizerH264 : public RtpDepacketizer {
virtual ~RtpDepacketizerH264();
bool Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) override;
bool ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
const uint8_t* payload_data);
bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
const uint8_t* payload_data);
size_t offset_;
size_t length_;
std::unique_ptr<rtc::Buffer> modified_buffer_;
} // namespace webrtc