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/*
* 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.
SetRtpTimestamp(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;
// 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