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/*
* 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 <utility>
#include "absl/types/variant.h"
#include "api/video/encoded_frame.h"
#include "common_video/h264/h264_common.h"
#include "modules/rtp_rtcp/source/rtp_video_header.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::PacketBuffer(Clock* clock,
size_t start_buffer_size,
size_t max_buffer_size,
OnAssembledFrameCallback* assembled_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),
assembled_frame_callback_(assembled_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, just silently ignore it.
if (is_cleared_to_first_seq_num_) {
delete[] packet->dataPtr;
packet->dataPtr = nullptr;
return true;
}
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 since we were unable to expand the buffer.
if (sequence_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();
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->is_last_packet_in_frame();
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->video_header.frame_type == VideoFrameType::kVideoFrameKey)
last_received_keyframe_packet_ms_ = now_ms;
found_frames = FindFrames(seq_num);
}
for (std::unique_ptr<RtpFrameObject>& frame : found_frames)
assembled_frame_callback_->OnAssembledFrame(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::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, size_);
uint16_t seq_num = start_seq_num;
for (size_t i = 0; i < iterations; ++i) {
size_t index = seq_num % size_;
RTC_DCHECK_EQ(sequence_buffer_[index].seq_num, seq_num);
RTC_DCHECK_EQ(sequence_buffer_[index].seq_num, data_buffer_[index].seqNum);
delete[] data_buffer_[index].dataPtr;
data_buffer_[index].dataPtr = nullptr;
sequence_buffer_[index].used = false;
++seq_num;
}
}
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)
assembled_frame_callback_->OnAssembledFrame(std::move(frame));
}
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_;
}
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.";
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 (data_buffer_[prev_index].timestamp != data_buffer_[index].timestamp)
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;
int64_t min_recv_time = data_buffer_[index].packet_info.receive_time_ms();
int64_t max_recv_time = data_buffer_[index].packet_info.receive_time_ms();
RtpPacketInfos::vector_type packet_infos;
// 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;
min_recv_time =
std::min(min_recv_time,
data_buffer_[start_index].packet_info.receive_time_ms());
max_recv_time =
std::max(max_recv_time,
data_buffer_[start_index].packet_info.receive_time_ms());
// Should use |push_front()| since the loop traverses backwards. But
// it's too inefficient to do so on a vector so we'll instead fix the
// order afterwards.
packet_infos.push_back(data_buffer_[start_index].packet_info);
if (!is_h264 && sequence_buffer_[start_index].frame_begin)
break;
if (is_h264) {
const auto* h264_header = absl::get_if<RTPVideoHeaderH264>(
&data_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;
}
}
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;
}
// Fix the order since the packet-finding loop traverses backwards.
std::reverse(packet_infos.begin(), packet_infos.end());
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 % size_;
RTC_CHECK_LT(first_packet_index, size_);
if (is_h264_keyframe) {
data_buffer_[first_packet_index].video_header.frame_type =
VideoFrameType::kVideoFrameKey;
} else {
data_buffer_[first_packet_index].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 =
data_buffer_[start_index].video_header.frame_marking.temporal_id;
if (h264tid == kNoTemporalIdx && !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));
const VCMPacket* first_packet = GetPacket(start_seq_num);
const VCMPacket* last_packet = GetPacket(seq_num);
auto frame = std::make_unique<RtpFrameObject>(
start_seq_num, seq_num, last_packet->markerBit, 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->payloadType, 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)),
GetEncodedImageBuffer(frame_size, start_seq_num, seq_num));
found_frames.emplace_back(std::move(frame));
ClearInterval(start_seq_num, seq_num);
}
++seq_num;
}
return found_frames;
}
rtc::scoped_refptr<EncodedImageBuffer> PacketBuffer::GetEncodedImageBuffer(
size_t frame_size,
uint16_t first_seq_num,
uint16_t last_seq_num) {
size_t index = first_seq_num % size_;
size_t end = (last_seq_num + 1) % size_;
auto buffer = EncodedImageBuffer::Create(frame_size);
size_t offset = 0;
do {
RTC_DCHECK(sequence_buffer_[index].used);
size_t length = data_buffer_[index].sizeBytes;
RTC_CHECK_LE(offset + length, buffer->size());
memcpy(buffer->data() + offset, data_buffer_[index].dataPtr, length);
offset += length;
index = (index + 1) % size_;
} while (index != end);
return buffer;
}
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];
}
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