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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/frame_object.h"
#include <sstream>
#include "common_video/h264/h264_common.h"
#include "modules/video_coding/packet_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace video_coding {
FrameObject::FrameObject()
: picture_id(0),
spatial_layer(0),
timestamp(0),
num_references(0),
inter_layer_predicted(false) {}
RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
uint16_t first_seq_num,
uint16_t last_seq_num,
size_t frame_size,
int times_nacked,
int64_t received_time)
: packet_buffer_(packet_buffer),
first_seq_num_(first_seq_num),
last_seq_num_(last_seq_num),
timestamp_(0),
received_time_(received_time),
times_nacked_(times_nacked) {
VCMPacket* first_packet = packet_buffer_->GetPacket(first_seq_num);
RTC_CHECK(first_packet);
// RtpFrameObject members
frame_type_ = first_packet->frameType;
codec_type_ = first_packet->codec;
// TODO(philipel): Remove when encoded image is replaced by FrameObject.
// VCMEncodedFrame members
CopyCodecSpecific(&first_packet->video_header);
_completeFrame = true;
_payloadType = first_packet->payloadType;
_timeStamp = first_packet->timestamp;
ntp_time_ms_ = first_packet->ntp_time_ms_;
// Setting frame's playout delays to the same values
// as of the first packet's.
SetPlayoutDelay(first_packet->video_header.playout_delay);
// Since FFmpeg use an optimized bitstream reader that reads in chunks of
// 32/64 bits we have to add at least that much padding to the buffer
// to make sure the decoder doesn't read out of bounds.
// NOTE! EncodedImage::_size is the size of the buffer (think capacity of
// an std::vector) and EncodedImage::_length is the actual size of
// the bitstream (think size of an std::vector).
if (codec_type_ == kVideoCodecH264)
_size = frame_size + EncodedImage::kBufferPaddingBytesH264;
else
_size = frame_size;
_buffer = new uint8_t[_size];
_length = frame_size;
// For H264 frames we can't determine the frame type by just looking at the
// first packet. Instead we consider the frame to be a keyframe if it contains
// an IDR, and SPS/PPS if the field trial is set.
if (codec_type_ == kVideoCodecH264) {
_frameType = kVideoFrameDelta;
frame_type_ = kVideoFrameDelta;
bool contains_sps = false;
bool contains_pps = false;
bool contains_idr = false;
for (uint16_t seq_num = first_seq_num;
seq_num != static_cast<uint16_t>(last_seq_num + 1) &&
_frameType == kVideoFrameDelta;
++seq_num) {
VCMPacket* packet = packet_buffer_->GetPacket(seq_num);
RTC_CHECK(packet);
const RTPVideoHeaderH264& header = packet->video_header.codecHeader.H264;
for (size_t i = 0; i < header.nalus_length; ++i) {
if (header.nalus[i].type == H264::NaluType::kSps) {
contains_sps = true;
} else if (header.nalus[i].type == H264::NaluType::kPps) {
contains_pps = true;
} else if (header.nalus[i].type == H264::NaluType::kIdr) {
contains_idr = true;
}
}
}
const bool sps_pps_idr_is_keyframe =
field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe");
if ((sps_pps_idr_is_keyframe && contains_idr && contains_sps &&
contains_pps) ||
(!sps_pps_idr_is_keyframe && contains_idr)) {
_frameType = kVideoFrameKey;
frame_type_ = kVideoFrameKey;
}
if (contains_idr && (!contains_sps || !contains_pps)) {
std::stringstream ss;
ss << "Received H.264-IDR frame "
<< "(SPS: " << contains_sps << ", PPS: " << contains_pps << "). ";
if (sps_pps_idr_is_keyframe) {
ss << "Treating as delta frame since WebRTC-SpsPpsIdrIsH264Keyframe is "
"enabled.";
} else {
ss << "Treating as key frame since WebRTC-SpsPpsIdrIsH264Keyframe is "
"disabled.";
}
LOG(LS_WARNING) << ss.str();
}
} else {
_frameType = first_packet->frameType;
frame_type_ = first_packet->frameType;
}
bool bitstream_copied = GetBitstream(_buffer);
RTC_DCHECK(bitstream_copied);
_encodedWidth = first_packet->width;
_encodedHeight = first_packet->height;
// FrameObject members
timestamp = first_packet->timestamp;
VCMPacket* last_packet = packet_buffer_->GetPacket(last_seq_num);
RTC_CHECK(last_packet);
RTC_CHECK(last_packet->markerBit);
// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
// ts_126114v120700p.pdf Section 7.4.5.
// The MTSI client shall add the payload bytes as defined in this clause
// onto the last RTP packet in each group of packets which make up a key
// frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
// (HEVC)).
rotation_ = last_packet->video_header.rotation;
_rotation_set = true;
content_type_ = last_packet->video_header.content_type;
if (last_packet->video_header.video_timing.flags !=
TimingFrameFlags::kInvalid) {
// ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
// as this will be dealt with at the time of reporting.
timing_.encode_start_ms =
ntp_time_ms_ +
last_packet->video_header.video_timing.encode_start_delta_ms;
timing_.encode_finish_ms =
ntp_time_ms_ +
last_packet->video_header.video_timing.encode_finish_delta_ms;
timing_.packetization_finish_ms =
ntp_time_ms_ +
last_packet->video_header.video_timing.packetization_finish_delta_ms;
timing_.pacer_exit_ms =
ntp_time_ms_ +
last_packet->video_header.video_timing.pacer_exit_delta_ms;
timing_.network_timestamp_ms =
ntp_time_ms_ +
last_packet->video_header.video_timing.network_timstamp_delta_ms;
timing_.network2_timestamp_ms =
ntp_time_ms_ +
last_packet->video_header.video_timing.network2_timstamp_delta_ms;
timing_.receive_start_ms = first_packet->receive_time_ms;
timing_.receive_finish_ms = last_packet->receive_time_ms;
}
timing_.flags = last_packet->video_header.video_timing.flags;
}
RtpFrameObject::~RtpFrameObject() {
packet_buffer_->ReturnFrame(this);
}
uint16_t RtpFrameObject::first_seq_num() const {
return first_seq_num_;
}
uint16_t RtpFrameObject::last_seq_num() const {
return last_seq_num_;
}
int RtpFrameObject::times_nacked() const {
return times_nacked_;
}
FrameType RtpFrameObject::frame_type() const {
return frame_type_;
}
VideoCodecType RtpFrameObject::codec_type() const {
return codec_type_;
}
bool RtpFrameObject::GetBitstream(uint8_t* destination) const {
return packet_buffer_->GetBitstream(*this, destination);
}
uint32_t RtpFrameObject::Timestamp() const {
return timestamp_;
}
int64_t RtpFrameObject::ReceivedTime() const {
return received_time_;
}
int64_t RtpFrameObject::RenderTime() const {
return _renderTimeMs;
}
bool RtpFrameObject::delayed_by_retransmission() const {
return times_nacked() > 0;
}
rtc::Optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
rtc::CritScope lock(&packet_buffer_->crit_);
VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
if (!packet)
return rtc::Optional<RTPVideoTypeHeader>();
return rtc::Optional<RTPVideoTypeHeader>(packet->video_header.codecHeader);
}
} // namespace video_coding
} // namespace webrtc