blob: 57c60ca44100d86b4d3e5ecbf66286450740fa56 [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/frame_object.h"
#include "common_video/h264/h264_common.h"
#include "modules/video_coding/packet_buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace video_coding {
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),
received_time_(received_time),
times_nacked_(times_nacked) {
VCMPacket* first_packet = packet_buffer_->GetPacket(first_seq_num);
RTC_CHECK(first_packet);
// EncodedFrame members
frame_type_ = first_packet->frameType;
codec_type_ = first_packet->codec;
// TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
// VCMEncodedFrame members
CopyCodecSpecific(&first_packet->video_header);
_completeFrame = true;
_payloadType = first_packet->payloadType;
SetTimestamp(first_packet->timestamp);
ntp_time_ms_ = first_packet->ntp_time_ms_;
_frameType = first_packet->frameType;
// 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;
bool bitstream_copied = GetBitstream(_buffer);
RTC_DCHECK(bitstream_copied);
_encodedWidth = first_packet->width;
_encodedHeight = first_packet->height;
// EncodedFrame members
SetTimestamp(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 !=
VideoSendTiming::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_timestamp_delta_ms;
timing_.network2_timestamp_ms =
ntp_time_ms_ +
last_packet->video_header.video_timing.network2_timestamp_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);
}
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;
}
absl::optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
rtc::CritScope lock(&packet_buffer_->crit_);
VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
if (!packet)
return absl::nullopt;
return packet->video_header.video_type_header;
}
} // namespace video_coding
} // namespace webrtc