/* * 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 "webrtc/modules/video_coding/frame_object.h" #include "webrtc/common_video/h264/h264_common.h" #include "webrtc/modules/video_coding/packet_buffer.h" #include "webrtc/rtc_base/checks.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 NALU. if (codec_type_ == kVideoCodecH264) { _frameType = kVideoFrameDelta; frame_type_ = kVideoFrameDelta; for (uint16_t seq_num = first_seq_num; seq_num != static_cast(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::kIdr) { _frameType = kVideoFrameKey; frame_type_ = kVideoFrameKey; break; } } } } 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 RtpFrameObject::GetCodecHeader() const { rtc::CritScope lock(&packet_buffer_->crit_); VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_); if (!packet) return rtc::Optional(); return rtc::Optional(packet->video_header.codecHeader); } } // namespace video_coding } // namespace webrtc