| /* |
| * 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 <algorithm> |
| #include <limits> |
| #include <sstream> |
| #include <utility> |
| |
| #include "common_video/h264/h264_common.h" |
| #include "modules/video_coding/frame_object.h" |
| #include "rtc_base/atomicops.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "system_wrappers/include/clock.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| namespace video_coding { |
| |
| rtc::scoped_refptr<PacketBuffer> PacketBuffer::Create( |
| Clock* clock, |
| size_t start_buffer_size, |
| size_t max_buffer_size, |
| OnReceivedFrameCallback* received_frame_callback) { |
| return rtc::scoped_refptr<PacketBuffer>(new PacketBuffer( |
| clock, start_buffer_size, max_buffer_size, received_frame_callback)); |
| } |
| |
| PacketBuffer::PacketBuffer(Clock* clock, |
| size_t start_buffer_size, |
| size_t max_buffer_size, |
| OnReceivedFrameCallback* received_frame_callback) |
| : clock_(clock), |
| size_(start_buffer_size), |
| max_size_(max_buffer_size), |
| first_seq_num_(0), |
| first_packet_received_(false), |
| is_cleared_to_first_seq_num_(false), |
| data_buffer_(start_buffer_size), |
| sequence_buffer_(start_buffer_size), |
| received_frame_callback_(received_frame_callback), |
| unique_frames_seen_(0), |
| sps_pps_idr_is_h264_keyframe_( |
| field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) { |
| 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(); |
| } |
| |
| bool PacketBuffer::InsertPacket(VCMPacket* packet) { |
| std::vector<std::unique_ptr<RtpFrameObject>> found_frames; |
| { |
| rtc::CritScope lock(&crit_); |
| |
| OnTimestampReceived(packet->timestamp); |
| |
| uint16_t seq_num = packet->seqNum; |
| size_t index = seq_num % 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. |
| if (is_cleared_to_first_seq_num_) { |
| delete[] packet->dataPtr; |
| packet->dataPtr = nullptr; |
| return false; |
| } |
| |
| first_seq_num_ = seq_num; |
| } |
| |
| if (sequence_buffer_[index].used) { |
| // Duplicate packet, just delete the payload. |
| if (data_buffer_[index].seqNum == packet->seqNum) { |
| delete[] packet->dataPtr; |
| packet->dataPtr = nullptr; |
| return true; |
| } |
| |
| // The packet buffer is full, try to expand the buffer. |
| while (ExpandBufferSize() && sequence_buffer_[seq_num % size_].used) { |
| } |
| index = seq_num % size_; |
| |
| // Packet buffer is still full. |
| if (sequence_buffer_[index].used) { |
| delete[] packet->dataPtr; |
| packet->dataPtr = nullptr; |
| return false; |
| } |
| } |
| |
| sequence_buffer_[index].frame_begin = packet->is_first_packet_in_frame; |
| sequence_buffer_[index].frame_end = packet->markerBit; |
| sequence_buffer_[index].seq_num = packet->seqNum; |
| sequence_buffer_[index].continuous = false; |
| sequence_buffer_[index].frame_created = false; |
| sequence_buffer_[index].used = true; |
| data_buffer_[index] = *packet; |
| packet->dataPtr = nullptr; |
| |
| UpdateMissingPackets(packet->seqNum); |
| |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| last_received_packet_ms_ = now_ms; |
| if (packet->frameType == kVideoFrameKey) |
| last_received_keyframe_packet_ms_ = now_ms; |
| |
| found_frames = FindFrames(seq_num); |
| } |
| |
| for (std::unique_ptr<RtpFrameObject>& frame : found_frames) |
| received_frame_callback_->OnReceivedFrame(std::move(frame)); |
| |
| return true; |
| } |
| |
| void PacketBuffer::ClearTo(uint16_t seq_num) { |
| rtc::CritScope lock(&crit_); |
| // 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, size_); |
| for (size_t i = 0; i < iterations; ++i) { |
| size_t index = first_seq_num_ % size_; |
| RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num); |
| if (AheadOf<uint16_t>(seq_num, sequence_buffer_[index].seq_num)) { |
| delete[] data_buffer_[index].dataPtr; |
| data_buffer_[index].dataPtr = nullptr; |
| sequence_buffer_[index].used = false; |
| } |
| ++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() { |
| rtc::CritScope lock(&crit_); |
| for (size_t i = 0; i < size_; ++i) { |
| delete[] data_buffer_[i].dataPtr; |
| data_buffer_[i].dataPtr = nullptr; |
| sequence_buffer_[i].used = false; |
| } |
| |
| first_packet_received_ = false; |
| is_cleared_to_first_seq_num_ = false; |
| last_received_packet_ms_.reset(); |
| last_received_keyframe_packet_ms_.reset(); |
| newest_inserted_seq_num_.reset(); |
| missing_packets_.clear(); |
| } |
| |
| void PacketBuffer::PaddingReceived(uint16_t seq_num) { |
| std::vector<std::unique_ptr<RtpFrameObject>> found_frames; |
| { |
| rtc::CritScope lock(&crit_); |
| UpdateMissingPackets(seq_num); |
| found_frames = FindFrames(static_cast<uint16_t>(seq_num + 1)); |
| } |
| |
| for (std::unique_ptr<RtpFrameObject>& frame : found_frames) |
| received_frame_callback_->OnReceivedFrame(std::move(frame)); |
| } |
| |
| rtc::Optional<int64_t> PacketBuffer::LastReceivedPacketMs() const { |
| rtc::CritScope lock(&crit_); |
| return last_received_packet_ms_; |
| } |
| |
| rtc::Optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const { |
| rtc::CritScope lock(&crit_); |
| return last_received_keyframe_packet_ms_; |
| } |
| |
| int PacketBuffer::GetUniqueFramesSeen() const { |
| rtc::CritScope lock(&crit_); |
| return unique_frames_seen_; |
| } |
| |
| bool PacketBuffer::ExpandBufferSize() { |
| if (size_ == max_size_) { |
| RTC_LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_ |
| << "), failed to increase size. Clearing PacketBuffer."; |
| Clear(); |
| return false; |
| } |
| |
| size_t new_size = std::min(max_size_, 2 * size_); |
| std::vector<VCMPacket> new_data_buffer(new_size); |
| std::vector<ContinuityInfo> new_sequence_buffer(new_size); |
| for (size_t i = 0; i < size_; ++i) { |
| if (sequence_buffer_[i].used) { |
| size_t index = sequence_buffer_[i].seq_num % new_size; |
| new_sequence_buffer[index] = sequence_buffer_[i]; |
| new_data_buffer[index] = data_buffer_[i]; |
| } |
| } |
| size_ = new_size; |
| sequence_buffer_ = std::move(new_sequence_buffer); |
| data_buffer_ = std::move(new_data_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 % size_; |
| int prev_index = index > 0 ? index - 1 : size_ - 1; |
| |
| if (!sequence_buffer_[index].used) |
| return false; |
| if (sequence_buffer_[index].seq_num != seq_num) |
| return false; |
| if (sequence_buffer_[index].frame_created) |
| return false; |
| if (sequence_buffer_[index].frame_begin) |
| return true; |
| if (!sequence_buffer_[prev_index].used) |
| return false; |
| if (sequence_buffer_[prev_index].frame_created) |
| return false; |
| if (sequence_buffer_[prev_index].seq_num != |
| static_cast<uint16_t>(sequence_buffer_[index].seq_num - 1)) { |
| return false; |
| } |
| if (sequence_buffer_[prev_index].continuous) |
| return true; |
| |
| return false; |
| } |
| |
| std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames( |
| uint16_t seq_num) { |
| std::vector<std::unique_ptr<RtpFrameObject>> found_frames; |
| for (size_t i = 0; i < size_ && PotentialNewFrame(seq_num); ++i) { |
| size_t index = seq_num % size_; |
| sequence_buffer_[index].continuous = true; |
| |
| // If all packets of the frame is continuous, find the first packet of the |
| // frame and create an RtpFrameObject. |
| if (sequence_buffer_[index].frame_end) { |
| size_t frame_size = 0; |
| int max_nack_count = -1; |
| 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 = data_buffer_[start_index].timestamp; |
| |
| // Identify H.264 keyframes by means of SPS, PPS, and IDR. |
| bool is_h264 = data_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; |
| |
| while (true) { |
| ++tested_packets; |
| frame_size += data_buffer_[start_index].sizeBytes; |
| max_nack_count = |
| std::max(max_nack_count, data_buffer_[start_index].timesNacked); |
| sequence_buffer_[start_index].frame_created = true; |
| |
| if (!is_h264 && sequence_buffer_[start_index].frame_begin) |
| break; |
| |
| if (is_h264 && !is_h264_keyframe) { |
| const RTPVideoHeaderH264& header = |
| data_buffer_[start_index].video_header.codecHeader.H264; |
| for (size_t j = 0; j < header.nalus_length; ++j) { |
| if (header.nalus[j].type == H264::NaluType::kSps) { |
| has_h264_sps = true; |
| } else if (header.nalus[j].type == H264::NaluType::kPps) { |
| has_h264_pps = true; |
| } else if (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; |
| } |
| } |
| |
| if (tested_packets == size_) |
| break; |
| |
| start_index = start_index > 0 ? start_index - 1 : 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 && |
| (!sequence_buffer_[start_index].used || |
| data_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)) { |
| std::stringstream ss; |
| ss << "Received H.264-IDR frame " |
| << "(SPS: " << has_h264_sps << ", PPS: " << has_h264_pps << "). "; |
| if (sps_pps_idr_is_h264_keyframe_) { |
| ss << "Treating as delta frame since " |
| "WebRTC-SpsPpsIdrIsH264Keyframe is enabled."; |
| } else { |
| ss << "Treating as key frame since " |
| "WebRTC-SpsPpsIdrIsH264Keyframe is disabled."; |
| } |
| RTC_LOG(LS_WARNING) << ss.str(); |
| } |
| |
| // 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 % size_; |
| RTC_CHECK_LT(first_packet_index, size_); |
| if (is_h264_keyframe) { |
| data_buffer_[first_packet_index].frameType = kVideoFrameKey; |
| } else { |
| data_buffer_[first_packet_index].frameType = 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()) { |
| uint16_t stop_index = (index + 1) % size_; |
| while (start_index != stop_index) { |
| sequence_buffer_[start_index].frame_created = false; |
| start_index = (start_index + 1) % size_; |
| } |
| |
| return found_frames; |
| } |
| } |
| |
| missing_packets_.erase(missing_packets_.begin(), |
| missing_packets_.upper_bound(seq_num)); |
| |
| found_frames.emplace_back( |
| new RtpFrameObject(this, start_seq_num, seq_num, frame_size, |
| max_nack_count, clock_->TimeInMilliseconds())); |
| } |
| ++seq_num; |
| } |
| return found_frames; |
| } |
| |
| void PacketBuffer::ReturnFrame(RtpFrameObject* frame) { |
| rtc::CritScope lock(&crit_); |
| size_t index = frame->first_seq_num() % size_; |
| size_t end = (frame->last_seq_num() + 1) % size_; |
| uint16_t seq_num = frame->first_seq_num(); |
| while (index != end) { |
| if (sequence_buffer_[index].seq_num == seq_num) { |
| delete[] data_buffer_[index].dataPtr; |
| data_buffer_[index].dataPtr = nullptr; |
| sequence_buffer_[index].used = false; |
| } |
| |
| index = (index + 1) % size_; |
| ++seq_num; |
| } |
| } |
| |
| bool PacketBuffer::GetBitstream(const RtpFrameObject& frame, |
| uint8_t* destination) { |
| rtc::CritScope lock(&crit_); |
| |
| size_t index = frame.first_seq_num() % size_; |
| size_t end = (frame.last_seq_num() + 1) % size_; |
| uint16_t seq_num = frame.first_seq_num(); |
| uint8_t* destination_end = destination + frame.size(); |
| |
| do { |
| if (!sequence_buffer_[index].used || |
| sequence_buffer_[index].seq_num != seq_num) { |
| return false; |
| } |
| |
| RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num); |
| size_t length = data_buffer_[index].sizeBytes; |
| if (destination + length > destination_end) { |
| RTC_LOG(LS_WARNING) << "Frame (" << frame.id.picture_id << ":" |
| << static_cast<int>(frame.id.spatial_layer) << ")" |
| << " bitstream buffer is not large enough."; |
| return false; |
| } |
| |
| const uint8_t* source = data_buffer_[index].dataPtr; |
| memcpy(destination, source, length); |
| destination += length; |
| index = (index + 1) % size_; |
| ++seq_num; |
| } while (index != end); |
| |
| return true; |
| } |
| |
| VCMPacket* PacketBuffer::GetPacket(uint16_t seq_num) { |
| size_t index = seq_num % size_; |
| if (!sequence_buffer_[index].used || |
| seq_num != sequence_buffer_[index].seq_num) { |
| return nullptr; |
| } |
| return &data_buffer_[index]; |
| } |
| |
| int PacketBuffer::AddRef() const { |
| return rtc::AtomicOps::Increment(&ref_count_); |
| } |
| |
| int PacketBuffer::Release() const { |
| int count = rtc::AtomicOps::Decrement(&ref_count_); |
| if (!count) { |
| delete this; |
| } |
| return count; |
| } |
| |
| 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); |
| } |
| } |
| |
| void PacketBuffer::OnTimestampReceived(uint32_t rtp_timestamp) { |
| const size_t kMaxTimestampsHistory = 1000; |
| if (rtp_timestamps_history_set_.insert(rtp_timestamp).second) { |
| rtp_timestamps_history_queue_.push(rtp_timestamp); |
| ++unique_frames_seen_; |
| if (rtp_timestamps_history_set_.size() > kMaxTimestampsHistory) { |
| uint32_t discarded_timestamp = rtp_timestamps_history_queue_.front(); |
| rtp_timestamps_history_set_.erase(discarded_timestamp); |
| rtp_timestamps_history_queue_.pop(); |
| } |
| } |
| } |
| |
| } // namespace video_coding |
| } // namespace webrtc |