| /* |
| * Copyright (c) 2016 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 "modules/video_coding/packet_buffer.h" |
| |
| #include <string.h> |
| |
| #include <algorithm> |
| #include <cstdint> |
| #include <limits> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/types/variant.h" |
| #include "api/array_view.h" |
| #include "api/rtp_packet_info.h" |
| #include "api/video/video_frame_type.h" |
| #include "common_video/h264/h264_common.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "modules/rtp_rtcp/source/rtp_video_header.h" |
| #include "modules/video_coding/codecs/h264/include/h264_globals.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/mod_ops.h" |
| |
| namespace webrtc { |
| namespace video_coding { |
| |
| PacketBuffer::Packet::Packet(const RtpPacketReceived& rtp_packet, |
| const RTPVideoHeader& video_header) |
| : marker_bit(rtp_packet.Marker()), |
| payload_type(rtp_packet.PayloadType()), |
| seq_num(rtp_packet.SequenceNumber()), |
| timestamp(rtp_packet.Timestamp()), |
| times_nacked(-1), |
| video_header(video_header) {} |
| |
| PacketBuffer::PacketBuffer(size_t start_buffer_size, size_t max_buffer_size) |
| : max_size_(max_buffer_size), |
| first_seq_num_(0), |
| first_packet_received_(false), |
| is_cleared_to_first_seq_num_(false), |
| buffer_(start_buffer_size), |
| sps_pps_idr_is_h264_keyframe_(false) { |
| RTC_DCHECK_LE(start_buffer_size, max_buffer_size); |
| // Buffer size must always be a power of 2. |
| RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0); |
| RTC_DCHECK((max_buffer_size & (max_buffer_size - 1)) == 0); |
| } |
| |
| PacketBuffer::~PacketBuffer() { |
| Clear(); |
| } |
| |
| PacketBuffer::InsertResult PacketBuffer::InsertPacket( |
| std::unique_ptr<PacketBuffer::Packet> packet) { |
| PacketBuffer::InsertResult result; |
| |
| uint16_t seq_num = packet->seq_num; |
| size_t index = seq_num % buffer_.size(); |
| |
| if (!first_packet_received_) { |
| first_seq_num_ = seq_num; |
| first_packet_received_ = true; |
| } else if (AheadOf(first_seq_num_, seq_num)) { |
| // If we have explicitly cleared past this packet then it's old, |
| // don't insert it, just silently ignore it. |
| if (is_cleared_to_first_seq_num_) { |
| return result; |
| } |
| |
| first_seq_num_ = seq_num; |
| } |
| |
| if (buffer_[index] != nullptr) { |
| // Duplicate packet, just delete the payload. |
| if (buffer_[index]->seq_num == packet->seq_num) { |
| return result; |
| } |
| |
| // The packet buffer is full, try to expand the buffer. |
| while (ExpandBufferSize() && buffer_[seq_num % buffer_.size()] != nullptr) { |
| } |
| index = seq_num % buffer_.size(); |
| |
| // Packet buffer is still full since we were unable to expand the buffer. |
| if (buffer_[index] != nullptr) { |
| // Clear the buffer, delete payload, and return false to signal that a |
| // new keyframe is needed. |
| RTC_LOG(LS_WARNING) << "Clear PacketBuffer and request key frame."; |
| ClearInternal(); |
| result.buffer_cleared = true; |
| return result; |
| } |
| } |
| |
| packet->continuous = false; |
| buffer_[index] = std::move(packet); |
| |
| UpdateMissingPackets(seq_num); |
| |
| result.packets = FindFrames(seq_num); |
| return result; |
| } |
| |
| void PacketBuffer::ClearTo(uint16_t seq_num) { |
| // We have already cleared past this sequence number, no need to do anything. |
| if (is_cleared_to_first_seq_num_ && |
| AheadOf<uint16_t>(first_seq_num_, seq_num)) { |
| return; |
| } |
| |
| // If the packet buffer was cleared between a frame was created and returned. |
| if (!first_packet_received_) |
| return; |
| |
| // Avoid iterating over the buffer more than once by capping the number of |
| // iterations to the |size_| of the buffer. |
| ++seq_num; |
| size_t diff = ForwardDiff<uint16_t>(first_seq_num_, seq_num); |
| size_t iterations = std::min(diff, buffer_.size()); |
| for (size_t i = 0; i < iterations; ++i) { |
| auto& stored = buffer_[first_seq_num_ % buffer_.size()]; |
| if (stored != nullptr && AheadOf<uint16_t>(seq_num, stored->seq_num)) { |
| stored = nullptr; |
| } |
| ++first_seq_num_; |
| } |
| |
| // If |diff| is larger than |iterations| it means that we don't increment |
| // |first_seq_num_| until we reach |seq_num|, so we set it here. |
| first_seq_num_ = seq_num; |
| |
| is_cleared_to_first_seq_num_ = true; |
| auto clear_to_it = missing_packets_.upper_bound(seq_num); |
| if (clear_to_it != missing_packets_.begin()) { |
| --clear_to_it; |
| missing_packets_.erase(missing_packets_.begin(), clear_to_it); |
| } |
| } |
| |
| void PacketBuffer::Clear() { |
| ClearInternal(); |
| } |
| |
| PacketBuffer::InsertResult PacketBuffer::InsertPadding(uint16_t seq_num) { |
| PacketBuffer::InsertResult result; |
| UpdateMissingPackets(seq_num); |
| result.packets = FindFrames(static_cast<uint16_t>(seq_num + 1)); |
| return result; |
| } |
| |
| void PacketBuffer::ForceSpsPpsIdrIsH264Keyframe() { |
| sps_pps_idr_is_h264_keyframe_ = true; |
| } |
| |
| void PacketBuffer::ClearInternal() { |
| for (auto& entry : buffer_) { |
| entry = nullptr; |
| } |
| |
| first_packet_received_ = false; |
| is_cleared_to_first_seq_num_ = false; |
| newest_inserted_seq_num_.reset(); |
| missing_packets_.clear(); |
| } |
| |
| bool PacketBuffer::ExpandBufferSize() { |
| if (buffer_.size() == max_size_) { |
| RTC_LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_ |
| << "), failed to increase size."; |
| return false; |
| } |
| |
| size_t new_size = std::min(max_size_, 2 * buffer_.size()); |
| std::vector<std::unique_ptr<Packet>> new_buffer(new_size); |
| for (std::unique_ptr<Packet>& entry : buffer_) { |
| if (entry != nullptr) { |
| new_buffer[entry->seq_num % new_size] = std::move(entry); |
| } |
| } |
| buffer_ = std::move(new_buffer); |
| RTC_LOG(LS_INFO) << "PacketBuffer size expanded to " << new_size; |
| return true; |
| } |
| |
| bool PacketBuffer::PotentialNewFrame(uint16_t seq_num) const { |
| size_t index = seq_num % buffer_.size(); |
| int prev_index = index > 0 ? index - 1 : buffer_.size() - 1; |
| const auto& entry = buffer_[index]; |
| const auto& prev_entry = buffer_[prev_index]; |
| |
| if (entry == nullptr) |
| return false; |
| if (entry->seq_num != seq_num) |
| return false; |
| if (entry->is_first_packet_in_frame()) |
| return true; |
| if (prev_entry == nullptr) |
| return false; |
| if (prev_entry->seq_num != static_cast<uint16_t>(entry->seq_num - 1)) |
| return false; |
| if (prev_entry->timestamp != entry->timestamp) |
| return false; |
| if (prev_entry->continuous) |
| return true; |
| |
| return false; |
| } |
| |
| std::vector<std::unique_ptr<PacketBuffer::Packet>> PacketBuffer::FindFrames( |
| uint16_t seq_num) { |
| std::vector<std::unique_ptr<PacketBuffer::Packet>> found_frames; |
| for (size_t i = 0; i < buffer_.size() && PotentialNewFrame(seq_num); ++i) { |
| size_t index = seq_num % buffer_.size(); |
| buffer_[index]->continuous = true; |
| |
| // If all packets of the frame is continuous, find the first packet of the |
| // frame and add all packets of the frame to the returned packets. |
| if (buffer_[index]->is_last_packet_in_frame()) { |
| uint16_t start_seq_num = seq_num; |
| |
| // Find the start index by searching backward until the packet with |
| // the |frame_begin| flag is set. |
| int start_index = index; |
| size_t tested_packets = 0; |
| int64_t frame_timestamp = buffer_[start_index]->timestamp; |
| |
| // Identify H.264 keyframes by means of SPS, PPS, and IDR. |
| bool is_h264 = buffer_[start_index]->codec() == kVideoCodecH264; |
| bool has_h264_sps = false; |
| bool has_h264_pps = false; |
| bool has_h264_idr = false; |
| bool is_h264_keyframe = false; |
| int idr_width = -1; |
| int idr_height = -1; |
| while (true) { |
| ++tested_packets; |
| |
| if (!is_h264 && buffer_[start_index]->is_first_packet_in_frame()) |
| break; |
| |
| if (is_h264) { |
| const auto* h264_header = absl::get_if<RTPVideoHeaderH264>( |
| &buffer_[start_index]->video_header.video_type_header); |
| if (!h264_header || h264_header->nalus_length >= kMaxNalusPerPacket) |
| return found_frames; |
| |
| for (size_t j = 0; j < h264_header->nalus_length; ++j) { |
| if (h264_header->nalus[j].type == H264::NaluType::kSps) { |
| has_h264_sps = true; |
| } else if (h264_header->nalus[j].type == H264::NaluType::kPps) { |
| has_h264_pps = true; |
| } else if (h264_header->nalus[j].type == H264::NaluType::kIdr) { |
| has_h264_idr = true; |
| } |
| } |
| if ((sps_pps_idr_is_h264_keyframe_ && has_h264_idr && has_h264_sps && |
| has_h264_pps) || |
| (!sps_pps_idr_is_h264_keyframe_ && has_h264_idr)) { |
| is_h264_keyframe = true; |
| // Store the resolution of key frame which is the packet with |
| // smallest index and valid resolution; typically its IDR or SPS |
| // packet; there may be packet preceeding this packet, IDR's |
| // resolution will be applied to them. |
| if (buffer_[start_index]->width() > 0 && |
| buffer_[start_index]->height() > 0) { |
| idr_width = buffer_[start_index]->width(); |
| idr_height = buffer_[start_index]->height(); |
| } |
| } |
| } |
| |
| if (tested_packets == buffer_.size()) |
| break; |
| |
| start_index = start_index > 0 ? start_index - 1 : buffer_.size() - 1; |
| |
| // In the case of H264 we don't have a frame_begin bit (yes, |
| // |frame_begin| might be set to true but that is a lie). So instead |
| // we traverese backwards as long as we have a previous packet and |
| // the timestamp of that packet is the same as this one. This may cause |
| // the PacketBuffer to hand out incomplete frames. |
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7106 |
| if (is_h264 && (buffer_[start_index] == nullptr || |
| buffer_[start_index]->timestamp != frame_timestamp)) { |
| break; |
| } |
| |
| --start_seq_num; |
| } |
| |
| if (is_h264) { |
| // Warn if this is an unsafe frame. |
| if (has_h264_idr && (!has_h264_sps || !has_h264_pps)) { |
| RTC_LOG(LS_WARNING) |
| << "Received H.264-IDR frame " |
| "(SPS: " |
| << has_h264_sps << ", PPS: " << has_h264_pps << "). Treating as " |
| << (sps_pps_idr_is_h264_keyframe_ ? "delta" : "key") |
| << " frame since WebRTC-SpsPpsIdrIsH264Keyframe is " |
| << (sps_pps_idr_is_h264_keyframe_ ? "enabled." : "disabled"); |
| } |
| |
| // Now that we have decided whether to treat this frame as a key frame |
| // or delta frame in the frame buffer, we update the field that |
| // determines if the RtpFrameObject is a key frame or delta frame. |
| const size_t first_packet_index = start_seq_num % buffer_.size(); |
| if (is_h264_keyframe) { |
| buffer_[first_packet_index]->video_header.frame_type = |
| VideoFrameType::kVideoFrameKey; |
| if (idr_width > 0 && idr_height > 0) { |
| // IDR frame was finalized and we have the correct resolution for |
| // IDR; update first packet to have same resolution as IDR. |
| buffer_[first_packet_index]->video_header.width = idr_width; |
| buffer_[first_packet_index]->video_header.height = idr_height; |
| } |
| } else { |
| buffer_[first_packet_index]->video_header.frame_type = |
| VideoFrameType::kVideoFrameDelta; |
| } |
| |
| // If this is not a keyframe, make sure there are no gaps in the packet |
| // sequence numbers up until this point. |
| if (!is_h264_keyframe && missing_packets_.upper_bound(start_seq_num) != |
| missing_packets_.begin()) { |
| return found_frames; |
| } |
| } |
| |
| const uint16_t end_seq_num = seq_num + 1; |
| // Use uint16_t type to handle sequence number wrap around case. |
| uint16_t num_packets = end_seq_num - start_seq_num; |
| found_frames.reserve(found_frames.size() + num_packets); |
| for (uint16_t i = start_seq_num; i != end_seq_num; ++i) { |
| std::unique_ptr<Packet>& packet = buffer_[i % buffer_.size()]; |
| RTC_DCHECK(packet); |
| RTC_DCHECK_EQ(i, packet->seq_num); |
| // Ensure frame boundary flags are properly set. |
| packet->video_header.is_first_packet_in_frame = (i == start_seq_num); |
| packet->video_header.is_last_packet_in_frame = (i == seq_num); |
| found_frames.push_back(std::move(packet)); |
| } |
| |
| missing_packets_.erase(missing_packets_.begin(), |
| missing_packets_.upper_bound(seq_num)); |
| } |
| ++seq_num; |
| } |
| return found_frames; |
| } |
| |
| void PacketBuffer::UpdateMissingPackets(uint16_t seq_num) { |
| if (!newest_inserted_seq_num_) |
| newest_inserted_seq_num_ = seq_num; |
| |
| const int kMaxPaddingAge = 1000; |
| if (AheadOf(seq_num, *newest_inserted_seq_num_)) { |
| uint16_t old_seq_num = seq_num - kMaxPaddingAge; |
| auto erase_to = missing_packets_.lower_bound(old_seq_num); |
| missing_packets_.erase(missing_packets_.begin(), erase_to); |
| |
| // Guard against inserting a large amount of missing packets if there is a |
| // jump in the sequence number. |
| if (AheadOf(old_seq_num, *newest_inserted_seq_num_)) |
| *newest_inserted_seq_num_ = old_seq_num; |
| |
| ++*newest_inserted_seq_num_; |
| while (AheadOf(seq_num, *newest_inserted_seq_num_)) { |
| missing_packets_.insert(*newest_inserted_seq_num_); |
| ++*newest_inserted_seq_num_; |
| } |
| } else { |
| missing_packets_.erase(seq_num); |
| } |
| } |
| |
| } // namespace video_coding |
| } // namespace webrtc |