
This change is based on a discussion for integrating a new statistic that measures the delay between the first frame being received and the first frame being decoded. To enable this in the context of FrameEncryption it makes sense for packet receive timestamps to be unconditionally recorded. Bug: webrtc:10105 Change-Id: I6b3b0118121db1fe5d4a4fb16cf5d94341cd2b1b Reviewed-on: https://webrtc-review.googlesource.com/c/113487 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25931}
196 lines
6.6 KiB
C++
196 lines
6.6 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/video_coding/frame_object.h"
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#include <string.h>
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#include "api/video/encoded_image.h"
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#include "api/video/video_timing.h"
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#include "modules/video_coding/packet.h"
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#include "modules/video_coding/packet_buffer.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/criticalsection.h"
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namespace webrtc {
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namespace video_coding {
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RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
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uint16_t first_seq_num,
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uint16_t last_seq_num,
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size_t frame_size,
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int times_nacked,
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int64_t received_time)
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: packet_buffer_(packet_buffer),
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first_seq_num_(first_seq_num),
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last_seq_num_(last_seq_num),
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received_time_(received_time),
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times_nacked_(times_nacked) {
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VCMPacket* first_packet = packet_buffer_->GetPacket(first_seq_num);
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RTC_CHECK(first_packet);
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// EncodedFrame members
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frame_type_ = first_packet->frameType;
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codec_type_ = first_packet->codec;
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// TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
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// VCMEncodedFrame members
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CopyCodecSpecific(&first_packet->video_header);
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_completeFrame = true;
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_payloadType = first_packet->payloadType;
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SetTimestamp(first_packet->timestamp);
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ntp_time_ms_ = first_packet->ntp_time_ms_;
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_frameType = first_packet->frameType;
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// Setting frame's playout delays to the same values
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// as of the first packet's.
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SetPlayoutDelay(first_packet->video_header.playout_delay);
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AllocateBitstreamBuffer(frame_size);
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bool bitstream_copied = packet_buffer_->GetBitstream(*this, _buffer);
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RTC_DCHECK(bitstream_copied);
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_encodedWidth = first_packet->width;
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_encodedHeight = first_packet->height;
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// EncodedFrame members
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SetTimestamp(first_packet->timestamp);
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VCMPacket* last_packet = packet_buffer_->GetPacket(last_seq_num);
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RTC_CHECK(last_packet);
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RTC_CHECK(last_packet->is_last_packet_in_frame);
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// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
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// ts_126114v120700p.pdf Section 7.4.5.
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// The MTSI client shall add the payload bytes as defined in this clause
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// onto the last RTP packet in each group of packets which make up a key
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// frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
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// (HEVC)).
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rotation_ = last_packet->video_header.rotation;
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SetColorSpace(last_packet->video_header.color_space
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? &last_packet->video_header.color_space.value()
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: nullptr);
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_rotation_set = true;
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content_type_ = last_packet->video_header.content_type;
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if (last_packet->video_header.video_timing.flags !=
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VideoSendTiming::kInvalid) {
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// ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
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// as this will be dealt with at the time of reporting.
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timing_.encode_start_ms =
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ntp_time_ms_ +
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last_packet->video_header.video_timing.encode_start_delta_ms;
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timing_.encode_finish_ms =
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ntp_time_ms_ +
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last_packet->video_header.video_timing.encode_finish_delta_ms;
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timing_.packetization_finish_ms =
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ntp_time_ms_ +
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last_packet->video_header.video_timing.packetization_finish_delta_ms;
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timing_.pacer_exit_ms =
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ntp_time_ms_ +
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last_packet->video_header.video_timing.pacer_exit_delta_ms;
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timing_.network_timestamp_ms =
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ntp_time_ms_ +
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last_packet->video_header.video_timing.network_timestamp_delta_ms;
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timing_.network2_timestamp_ms =
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ntp_time_ms_ +
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last_packet->video_header.video_timing.network2_timestamp_delta_ms;
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}
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timing_.flags = last_packet->video_header.video_timing.flags;
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timing_.receive_start_ms = first_packet->receive_time_ms;
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timing_.receive_finish_ms = last_packet->receive_time_ms;
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is_last_spatial_layer = last_packet->markerBit;
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}
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RtpFrameObject::~RtpFrameObject() {
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packet_buffer_->ReturnFrame(this);
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}
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uint16_t RtpFrameObject::first_seq_num() const {
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return first_seq_num_;
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}
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uint16_t RtpFrameObject::last_seq_num() const {
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return last_seq_num_;
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}
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int RtpFrameObject::times_nacked() const {
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return times_nacked_;
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}
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FrameType RtpFrameObject::frame_type() const {
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return frame_type_;
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}
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VideoCodecType RtpFrameObject::codec_type() const {
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return codec_type_;
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}
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int64_t RtpFrameObject::ReceivedTime() const {
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return received_time_;
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}
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int64_t RtpFrameObject::RenderTime() const {
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return _renderTimeMs;
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}
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void RtpFrameObject::SetSize(size_t size) {
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RTC_DCHECK_LE(size, _size);
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_length = size;
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}
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bool RtpFrameObject::delayed_by_retransmission() const {
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return times_nacked() > 0;
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}
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absl::optional<RTPVideoHeader> RtpFrameObject::GetRtpVideoHeader() const {
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rtc::CritScope lock(&packet_buffer_->crit_);
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VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
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if (!packet)
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return absl::nullopt;
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return packet->video_header;
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}
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absl::optional<RtpGenericFrameDescriptor>
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RtpFrameObject::GetGenericFrameDescriptor() const {
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rtc::CritScope lock(&packet_buffer_->crit_);
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VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
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if (!packet)
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return absl::nullopt;
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return packet->generic_descriptor;
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}
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absl::optional<FrameMarking> RtpFrameObject::GetFrameMarking() const {
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rtc::CritScope lock(&packet_buffer_->crit_);
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VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
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if (!packet)
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return absl::nullopt;
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return packet->video_header.frame_marking;
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}
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void RtpFrameObject::AllocateBitstreamBuffer(size_t frame_size) {
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// Since FFmpeg use an optimized bitstream reader that reads in chunks of
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// 32/64 bits we have to add at least that much padding to the buffer
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// to make sure the decoder doesn't read out of bounds.
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// NOTE! EncodedImage::_size is the size of the buffer (think capacity of
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// an std::vector) and EncodedImage::_length is the actual size of
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// the bitstream (think size of an std::vector).
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size_t new_size = frame_size + (codec_type_ == kVideoCodecH264
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? EncodedImage::kBufferPaddingBytesH264
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: 0);
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if (_size < new_size) {
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delete[] _buffer;
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_buffer = new uint8_t[new_size];
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_size = new_size;
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}
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_length = frame_size;
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}
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} // namespace video_coding
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} // namespace webrtc
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