/* * 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 #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; SetColorSpace(last_packet->video_header.color_space ? &last_packet->video_header.color_space.value() : nullptr); _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_.flags = last_packet->video_header.video_timing.flags; timing_.receive_start_ms = first_packet->receive_time_ms; timing_.receive_finish_ms = last_packet->receive_time_ms; is_last_spatial_layer = last_packet->markerBit; } 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, capacity()); _length = size; } bool RtpFrameObject::delayed_by_retransmission() const { return times_nacked() > 0; } absl::optional 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 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 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 (capacity() < new_size) { delete[] _buffer; set_buffer(new uint8_t[new_size], new_size); } _length = frame_size; } } // namespace video_coding } // namespace webrtc