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Bug: webrtc:9378 Change-Id: I6a66b9301cbadf1d6517bf7a96028099970a20a3 Reviewed-on: https://webrtc-review.googlesource.com/c/117964 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26585}
184 lines
6.2 KiB
C++
184 lines
6.2 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/critical_section.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 first_packet_received_time,
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int64_t last_packet_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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last_packet_received_time_(last_packet_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, data());
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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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_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_.receive_start_ms = first_packet_received_time;
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timing_.receive_finish_ms = last_packet_received_time;
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timing_.flags = last_packet->video_header.video_timing.flags;
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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 last_packet_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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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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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 (capacity() < new_size) {
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Allocate(new_size);
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}
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set_size(frame_size);
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}
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} // namespace video_coding
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} // namespace webrtc
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