webrtc/modules/video_coding/deprecated/frame_buffer.cc
Danil Chapovalov c146b5f77b Represent unset VideoPlayoutDelay with nullopt rather than special value
Remove support for setting one limit without another limit
because related rtp header extension doesn't support such values.

Start morphing VideoPlayouDelay into a class and stricter type: add accessors returning TimeDelta

Bug: webrtc:13756
Change-Id: If0dd02620528dc870b015beeff3a8103e04022ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40570}
2023-08-18 13:17:50 +00:00

264 lines
8.2 KiB
C++

/*
* Copyright (c) 2012 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/deprecated/frame_buffer.h"
#include <string.h>
#include "api/video/encoded_image.h"
#include "api/video/video_timing.h"
#include "modules/video_coding/deprecated/packet.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
VCMFrameBuffer::VCMFrameBuffer()
: _state(kStateEmpty), _nackCount(0), _latestPacketTimeMs(-1) {}
VCMFrameBuffer::~VCMFrameBuffer() {}
webrtc::VideoFrameType VCMFrameBuffer::FrameType() const {
return _sessionInfo.FrameType();
}
int32_t VCMFrameBuffer::GetLowSeqNum() const {
return _sessionInfo.LowSequenceNumber();
}
int32_t VCMFrameBuffer::GetHighSeqNum() const {
return _sessionInfo.HighSequenceNumber();
}
int VCMFrameBuffer::PictureId() const {
return _sessionInfo.PictureId();
}
int VCMFrameBuffer::TemporalId() const {
return _sessionInfo.TemporalId();
}
bool VCMFrameBuffer::LayerSync() const {
return _sessionInfo.LayerSync();
}
int VCMFrameBuffer::Tl0PicId() const {
return _sessionInfo.Tl0PicId();
}
std::vector<NaluInfo> VCMFrameBuffer::GetNaluInfos() const {
return _sessionInfo.GetNaluInfos();
}
void VCMFrameBuffer::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::SetGofInfo");
_sessionInfo.SetGofInfo(gof_info, idx);
// TODO(asapersson): Consider adding hdr->VP9.ref_picture_id for testing.
_codecSpecificInfo.codecSpecific.VP9.temporal_idx =
gof_info.temporal_idx[idx];
_codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
gof_info.temporal_up_switch[idx];
}
// Insert packet
VCMFrameBufferEnum VCMFrameBuffer::InsertPacket(const VCMPacket& packet,
int64_t timeInMs,
const FrameData& frame_data) {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::InsertPacket");
RTC_DCHECK(!(NULL == packet.dataPtr && packet.sizeBytes > 0));
if (packet.dataPtr != NULL) {
_payloadType = packet.payloadType;
}
if (kStateEmpty == _state) {
// First packet (empty and/or media) inserted into this frame.
// store some info and set some initial values.
SetTimestamp(packet.timestamp);
// We only take the ntp timestamp of the first packet of a frame.
ntp_time_ms_ = packet.ntp_time_ms_;
_codec = packet.codec();
if (packet.video_header.frame_type != VideoFrameType::kEmptyFrame) {
// first media packet
SetState(kStateIncomplete);
}
}
size_t oldSize = encoded_image_buffer_ ? encoded_image_buffer_->size() : 0;
uint32_t requiredSizeBytes =
size() + packet.sizeBytes +
(packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
if (requiredSizeBytes > oldSize) {
const uint8_t* prevBuffer = data();
const uint32_t increments =
requiredSizeBytes / kBufferIncStepSizeBytes +
(requiredSizeBytes % kBufferIncStepSizeBytes > 0);
const uint32_t newSize = oldSize + increments * kBufferIncStepSizeBytes;
if (newSize > kMaxJBFrameSizeBytes) {
RTC_LOG(LS_ERROR) << "Failed to insert packet due to frame being too "
"big.";
return kSizeError;
}
if (data() == nullptr) {
encoded_image_buffer_ = EncodedImageBuffer::Create(newSize);
SetEncodedData(encoded_image_buffer_);
set_size(0);
} else {
RTC_CHECK(encoded_image_buffer_ != nullptr);
RTC_DCHECK_EQ(encoded_image_buffer_->data(), data());
encoded_image_buffer_->Realloc(newSize);
}
_sessionInfo.UpdateDataPointers(prevBuffer, data());
}
if (packet.width() > 0 && packet.height() > 0) {
_encodedWidth = packet.width();
_encodedHeight = packet.height();
}
// Don't copy payload specific data for empty packets (e.g padding packets).
if (packet.sizeBytes > 0)
CopyCodecSpecific(&packet.video_header);
int retVal = _sessionInfo.InsertPacket(
packet, encoded_image_buffer_ ? encoded_image_buffer_->data() : nullptr,
frame_data);
if (retVal == -1) {
return kSizeError;
} else if (retVal == -2) {
return kDuplicatePacket;
} else if (retVal == -3) {
return kOutOfBoundsPacket;
}
// update size
set_size(size() + static_cast<uint32_t>(retVal));
_latestPacketTimeMs = timeInMs;
// 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)).
if (packet.markerBit) {
rotation_ = packet.video_header.rotation;
content_type_ = packet.video_header.content_type;
if (packet.video_header.video_timing.flags != VideoSendTiming::kInvalid) {
timing_.encode_start_ms =
ntp_time_ms_ + packet.video_header.video_timing.encode_start_delta_ms;
timing_.encode_finish_ms =
ntp_time_ms_ +
packet.video_header.video_timing.encode_finish_delta_ms;
timing_.packetization_finish_ms =
ntp_time_ms_ +
packet.video_header.video_timing.packetization_finish_delta_ms;
timing_.pacer_exit_ms =
ntp_time_ms_ + packet.video_header.video_timing.pacer_exit_delta_ms;
timing_.network_timestamp_ms =
ntp_time_ms_ +
packet.video_header.video_timing.network_timestamp_delta_ms;
timing_.network2_timestamp_ms =
ntp_time_ms_ +
packet.video_header.video_timing.network2_timestamp_delta_ms;
}
timing_.flags = packet.video_header.video_timing.flags;
}
if (packet.is_first_packet_in_frame()) {
SetPlayoutDelay(packet.video_header.playout_delay);
}
if (_sessionInfo.complete()) {
SetState(kStateComplete);
return kCompleteSession;
}
return kIncomplete;
}
int64_t VCMFrameBuffer::LatestPacketTimeMs() const {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::LatestPacketTimeMs");
return _latestPacketTimeMs;
}
void VCMFrameBuffer::IncrementNackCount() {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::IncrementNackCount");
_nackCount++;
}
int16_t VCMFrameBuffer::GetNackCount() const {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::GetNackCount");
return _nackCount;
}
bool VCMFrameBuffer::HaveFirstPacket() const {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::HaveFirstPacket");
return _sessionInfo.HaveFirstPacket();
}
int VCMFrameBuffer::NumPackets() const {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::NumPackets");
return _sessionInfo.NumPackets();
}
void VCMFrameBuffer::Reset() {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::Reset");
set_size(0);
_sessionInfo.Reset();
_payloadType = 0;
_nackCount = 0;
_latestPacketTimeMs = -1;
_state = kStateEmpty;
VCMEncodedFrame::Reset();
}
// Set state of frame
void VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state) {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::SetState");
if (_state == state) {
return;
}
switch (state) {
case kStateIncomplete:
// we can go to this state from state kStateEmpty
RTC_DCHECK_EQ(_state, kStateEmpty);
// Do nothing, we received a packet
break;
case kStateComplete:
RTC_DCHECK(_state == kStateEmpty || _state == kStateIncomplete);
break;
case kStateEmpty:
// Should only be set to empty through Reset().
RTC_DCHECK_NOTREACHED();
break;
}
_state = state;
}
// Get current state of frame
VCMFrameBufferStateEnum VCMFrameBuffer::GetState() const {
return _state;
}
void VCMFrameBuffer::PrepareForDecode(bool continuous) {
TRACE_EVENT0("webrtc", "VCMFrameBuffer::PrepareForDecode");
size_t bytes_removed = _sessionInfo.MakeDecodable();
set_size(size() - bytes_removed);
// Transfer frame information to EncodedFrame and create any codec
// specific information.
_frameType = _sessionInfo.FrameType();
_missingFrame = !continuous;
}
} // namespace webrtc