text: /*
   *  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 <string.h>
  
  #include "api/video/encoded_image.h"
  #include "api/video/video_timing.h"
  #include "modules/video_coding/packet.h"
  #include "modules/video_coding/packet_buffer.h"
  #include "rtc_base/checks.h"
  #include "rtc_base/criticalsection.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);
  
    AllocateBitstreamBuffer(frame_size);
    bool bitstream_copied = packet_buffer_->GetBitstream(*this, _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->is_last_packet_in_frame);
    // 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_;
  }
  
  int64_t RtpFrameObject::ReceivedTime() const {
    return received_time_;
  }
  
  int64_t RtpFrameObject::RenderTime() const {
    return _renderTimeMs;
  }
  
  void RtpFrameObject::SetSize(size_t size) {
    RTC_DCHECK_LE(size, _size);
    _length = size;
  }
  
  bool RtpFrameObject::delayed_by_retransmission() const {
    return times_nacked() > 0;
  }
  
  absl::optional<RTPVideoHeader> RtpFrameObject::GetRtpVideoHeader() const {
    rtc::CritScope lock(&packet_buffer_->crit_);
    VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
    if (!packet)
      return absl::nullopt;
    return packet->video_header;
  }
  
  absl::optional<RtpGenericFrameDescriptor>
  RtpFrameObject::GetGenericFrameDescriptor() const {
    rtc::CritScope lock(&packet_buffer_->crit_);
    VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
    if (!packet)
      return absl::nullopt;
    return packet->generic_descriptor;
  }
  
  absl::optional<FrameMarking> RtpFrameObject::GetFrameMarking() const {
    rtc::CritScope lock(&packet_buffer_->crit_);
    VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
    if (!packet)
      return absl::nullopt;
    return packet->video_header.frame_marking;
  }
  
  void RtpFrameObject::AllocateBitstreamBuffer(size_t frame_size) {
    // 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).
    size_t new_size = frame_size + (codec_type_ == kVideoCodecH264
                                        ? EncodedImage::kBufferPaddingBytesH264
                                        : 0);
    if (_size < new_size) {
      delete[] _buffer;
      _buffer = new uint8_t[new_size];
      _size = new_size;
    }
  
    _length = frame_size;
  }
  
  }  // namespace video_coding
  }  // namespace webrtc
  