modules/video_coding/frame_object.cc

/*
 *  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"
#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;
    }
  } 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