webrtc/modules/video_coding/frame_object.cc
Niels Möller 87e2d785a0 Prepare for splitting FrameType into AudioFrameType and VideoFrameType
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.

After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.

Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
2019-03-07 10:12:57 +00:00

184 lines
6.3 KiB
C++

/*
* 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/critical_section.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 first_packet_received_time,
int64_t last_packet_received_time)
: packet_buffer_(packet_buffer),
first_seq_num_(first_seq_num),
last_seq_num_(last_seq_num),
last_packet_received_time_(last_packet_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, data());
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);
_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_received_time;
timing_.receive_finish_ms = last_packet_received_time;
timing_.flags = last_packet->video_header.video_timing.flags;
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_;
}
VideoFrameType RtpFrameObject::frame_type() const {
return frame_type_;
}
VideoCodecType RtpFrameObject::codec_type() const {
return codec_type_;
}
int64_t RtpFrameObject::ReceivedTime() const {
return last_packet_received_time_;
}
int64_t RtpFrameObject::RenderTime() const {
return _renderTimeMs;
}
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.
size_t new_size = frame_size + (codec_type_ == kVideoCodecH264
? EncodedImage::kBufferPaddingBytesH264
: 0);
if (capacity() < new_size) {
Allocate(new_size);
}
set_size(frame_size);
}
} // namespace video_coding
} // namespace webrtc