blob: 675002c380801e46bfbd5713758ab7808eb22bc4 [file] [log] [blame]
/*
* 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/encoded_frame.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/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
#include "modules/video_coding/frame_object.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/mod_ops.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace video_coding {
PacketBuffer::Packet::Packet(const RtpPacketReceived& rtp_packet,
const RTPVideoHeader& video_header,
int64_t ntp_time_ms,
int64_t receive_time_ms)
: marker_bit(rtp_packet.Marker()),
payload_type(rtp_packet.PayloadType()),
seq_num(rtp_packet.SequenceNumber()),
timestamp(rtp_packet.Timestamp()),
ntp_time_ms(ntp_time_ms),
times_nacked(-1),
video_header(video_header),
packet_info(rtp_packet.Ssrc(),
rtp_packet.Csrcs(),
rtp_packet.Timestamp(),
/*audio_level=*/absl::nullopt,
rtp_packet.GetExtension<AbsoluteCaptureTimeExtension>(),
receive_time_ms) {}
PacketBuffer::PacketBuffer(Clock* clock,
size_t start_buffer_size,
size_t max_buffer_size)
: clock_(clock),
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_(
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();
}
PacketBuffer::InsertResult PacketBuffer::InsertPacket(
PacketBuffer::Packet* packet) {
PacketBuffer::InsertResult result;
rtc::CritScope lock(&crit_);
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].used) {
// 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()].used) {
}
index = seq_num % buffer_.size();
// Packet buffer is still full since we were unable to expand the buffer.
if (buffer_[index].used) {
// 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.";
Clear();
result.buffer_cleared = true;
return result;
}
}
int64_t now_ms = clock_->TimeInMilliseconds();
last_received_packet_ms_ = now_ms;
if (packet->video_header.frame_type == VideoFrameType::kVideoFrameKey ||
last_received_keyframe_rtp_timestamp_ == packet->timestamp) {
last_received_keyframe_packet_ms_ = now_ms;
last_received_keyframe_rtp_timestamp_ = packet->timestamp;
}
StoredPacket& new_entry = buffer_[index];
new_entry.continuous = false;
new_entry.used = true;
new_entry.data = std::move(*packet);
UpdateMissingPackets(seq_num);
result.frames = FindFrames(seq_num);
return result;
}
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, buffer_.size());
for (size_t i = 0; i < iterations; ++i) {
size_t index = first_seq_num_ % buffer_.size();
if (AheadOf<uint16_t>(seq_num, buffer_[index].seq_num())) {
buffer_[index].data.video_payload = {};
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::ClearInterval(uint16_t start_seq_num,
uint16_t stop_seq_num) {
size_t iterations = ForwardDiff<uint16_t>(start_seq_num, stop_seq_num + 1);
RTC_DCHECK_LE(iterations, buffer_.size());
uint16_t seq_num = start_seq_num;
for (size_t i = 0; i < iterations; ++i) {
size_t index = seq_num % buffer_.size();
RTC_DCHECK_EQ(buffer_[index].seq_num(), seq_num);
buffer_[index].data.video_payload = {};
buffer_[index].used = false;
++seq_num;
}
}
void PacketBuffer::Clear() {
rtc::CritScope lock(&crit_);
for (StoredPacket& entry : buffer_) {
entry.data.video_payload = {};
entry.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();
}
PacketBuffer::InsertResult PacketBuffer::InsertPadding(uint16_t seq_num) {
PacketBuffer::InsertResult result;
rtc::CritScope lock(&crit_);
UpdateMissingPackets(seq_num);
result.frames = FindFrames(static_cast<uint16_t>(seq_num + 1));
return result;
}
absl::optional<int64_t> PacketBuffer::LastReceivedPacketMs() const {
rtc::CritScope lock(&crit_);
return last_received_packet_ms_;
}
absl::optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
rtc::CritScope lock(&crit_);
return last_received_keyframe_packet_ms_;
}
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<StoredPacket> new_buffer(new_size);
for (StoredPacket& entry : buffer_) {
if (entry.used) {
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 StoredPacket& entry = buffer_[index];
const StoredPacket& prev_entry = buffer_[prev_index];
if (!entry.used)
return false;
if (entry.seq_num() != seq_num)
return false;
if (entry.frame_begin())
return true;
if (!prev_entry.used)
return false;
if (prev_entry.seq_num() != static_cast<uint16_t>(entry.seq_num() - 1))
return false;
if (prev_entry.data.timestamp != entry.data.timestamp)
return false;
if (prev_entry.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 < 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 create an RtpFrameObject.
if (buffer_[index].frame_end()) {
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].data.timestamp;
// Identify H.264 keyframes by means of SPS, PPS, and IDR.
bool is_h264 = buffer_[start_index].data.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].frame_begin())
break;
if (is_h264) {
const auto* h264_header = absl::get_if<RTPVideoHeaderH264>(
&buffer_[start_index].data.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].data.width() > 0 &&
buffer_[start_index].data.height() > 0) {
idr_width = buffer_[start_index].data.width();
idr_height = buffer_[start_index].data.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].used ||
buffer_[start_index].data.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].data.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].data.video_header.width = idr_width;
buffer_[first_packet_index].data.video_header.height = idr_height;
}
} else {
buffer_[first_packet_index].data.video_header.frame_type =
VideoFrameType::kVideoFrameDelta;
}
// With IPPP, if this is not a keyframe, make sure there are no gaps
// in the packet sequence numbers up until this point.
const uint8_t h264tid =
buffer_[start_index].data.video_header.frame_marking.temporal_id;
if (h264tid == kNoTemporalIdx && !is_h264_keyframe &&
missing_packets_.upper_bound(start_seq_num) !=
missing_packets_.begin()) {
return found_frames;
}
}
if (auto frame = AssembleFrame(start_seq_num, seq_num)) {
found_frames.push_back(std::move(frame));
} else {
RTC_LOG(LS_ERROR) << "Failed to assemble frame from packets "
<< start_seq_num << "-" << seq_num;
}
missing_packets_.erase(missing_packets_.begin(),
missing_packets_.upper_bound(seq_num));
ClearInterval(start_seq_num, seq_num);
}
++seq_num;
}
return found_frames;
}
std::unique_ptr<RtpFrameObject> PacketBuffer::AssembleFrame(
uint16_t first_seq_num,
uint16_t last_seq_num) {
const uint16_t end_seq_num = last_seq_num + 1;
const uint16_t num_packets = end_seq_num - first_seq_num;
int max_nack_count = -1;
int64_t min_recv_time = std::numeric_limits<int64_t>::max();
int64_t max_recv_time = std::numeric_limits<int64_t>::min();
size_t frame_size = 0;
std::vector<rtc::ArrayView<const uint8_t>> payloads;
RtpPacketInfos::vector_type packet_infos;
payloads.reserve(num_packets);
packet_infos.reserve(num_packets);
for (uint16_t seq_num = first_seq_num; seq_num != end_seq_num; ++seq_num) {
const Packet& packet = GetPacket(seq_num);
max_nack_count = std::max(max_nack_count, packet.times_nacked);
min_recv_time =
std::min(min_recv_time, packet.packet_info.receive_time_ms());
max_recv_time =
std::max(max_recv_time, packet.packet_info.receive_time_ms());
frame_size += packet.video_payload.size();
payloads.emplace_back(packet.video_payload);
packet_infos.push_back(packet.packet_info);
}
const Packet& first_packet = GetPacket(first_seq_num);
rtc::scoped_refptr<EncodedImageBuffer> bitstream;
// TODO(danilchap): Hide codec-specific code paths behind an interface.
if (first_packet.codec() == VideoCodecType::kVideoCodecAV1) {
bitstream = VideoRtpDepacketizerAv1::AssembleFrame(payloads);
if (!bitstream) {
// Failed to assemble a frame. Discard and continue.
return nullptr;
}
} else {
bitstream = EncodedImageBuffer::Create(frame_size);
uint8_t* write_at = bitstream->data();
for (rtc::ArrayView<const uint8_t> payload : payloads) {
memcpy(write_at, payload.data(), payload.size());
write_at += payload.size();
}
RTC_DCHECK_EQ(write_at - bitstream->data(), bitstream->size());
}
const Packet& last_packet = GetPacket(last_seq_num);
return std::make_unique<RtpFrameObject>(
first_seq_num, //
last_seq_num, //
last_packet.marker_bit, //
max_nack_count, //
min_recv_time, //
max_recv_time, //
first_packet.timestamp, //
first_packet.ntp_time_ms, //
last_packet.video_header.video_timing, //
first_packet.payload_type, //
first_packet.codec(), //
last_packet.video_header.rotation, //
last_packet.video_header.content_type, //
first_packet.video_header, //
last_packet.video_header.color_space, //
first_packet.generic_descriptor, //
RtpPacketInfos(std::move(packet_infos)), //
std::move(bitstream));
}
const PacketBuffer::Packet& PacketBuffer::GetPacket(uint16_t seq_num) const {
const StoredPacket& entry = buffer_[seq_num % buffer_.size()];
RTC_DCHECK(entry.used);
RTC_DCHECK_EQ(seq_num, entry.seq_num());
return entry.data;
}
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