webrtc_m130/modules/video_coding/h264_sps_pps_tracker.cc

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/*
* 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/h264_sps_pps_tracker.h"
#include <string>
#include <utility>
#include "common_video/h264/h264_common.h"
#include "common_video/h264/pps_parser.h"
#include "common_video/h264/sps_parser.h"
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
#include "modules/video_coding/frame_object.h"
#include "modules/video_coding/packet_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace video_coding {
namespace {
const uint8_t start_code_h264[] = {0, 0, 0, 1};
} // namespace
H264SpsPpsTracker::H264SpsPpsTracker() = default;
H264SpsPpsTracker::~H264SpsPpsTracker() = default;
H264SpsPpsTracker::PpsInfo::PpsInfo() = default;
H264SpsPpsTracker::PpsInfo::PpsInfo(PpsInfo&& rhs) = default;
H264SpsPpsTracker::PpsInfo& H264SpsPpsTracker::PpsInfo::operator=(
PpsInfo&& rhs) = default;
H264SpsPpsTracker::PpsInfo::~PpsInfo() = default;
H264SpsPpsTracker::SpsInfo::SpsInfo() = default;
H264SpsPpsTracker::SpsInfo::SpsInfo(SpsInfo&& rhs) = default;
H264SpsPpsTracker::SpsInfo& H264SpsPpsTracker::SpsInfo::operator=(
SpsInfo&& rhs) = default;
H264SpsPpsTracker::SpsInfo::~SpsInfo() = default;
H264SpsPpsTracker::PacketAction H264SpsPpsTracker::CopyAndFixBitstream(
VCMPacket* packet) {
RTC_DCHECK(packet->codec() == kVideoCodecH264);
const uint8_t* data = packet->dataPtr;
const size_t data_size = packet->sizeBytes;
const RTPVideoHeader& video_header = packet->video_header;
auto& h264_header =
absl::get<RTPVideoHeaderH264>(packet->video_header.video_type_header);
bool append_sps_pps = false;
auto sps = sps_data_.end();
auto pps = pps_data_.end();
for (size_t i = 0; i < h264_header.nalus_length; ++i) {
const NaluInfo& nalu = h264_header.nalus[i];
switch (nalu.type) {
case H264::NaluType::kSps: {
sps_data_[nalu.sps_id].width = packet->width();
sps_data_[nalu.sps_id].height = packet->height();
break;
}
case H264::NaluType::kPps: {
pps_data_[nalu.pps_id].sps_id = nalu.sps_id;
break;
}
case H264::NaluType::kIdr: {
// If this is the first packet of an IDR, make sure we have the required
// SPS/PPS and also calculate how much extra space we need in the buffer
// to prepend the SPS/PPS to the bitstream with start codes.
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. Add RTC_DEPRACATed anonymous unions to not break downstream projects. Orignal issue's description: > commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff > Author: danilchap <danilchap@webrtc.org> > Date: Mon Dec 19 09:36:33 2016 -0800 > > Revert of Rename RTPVideoHeader.isFirstPacket to > .is_first_packet_in_frame. (patchset #1 id:1 of > https://codereview.webrtc.org/2574943003/ ) > > Reason for revert: > breaks downstream project. > > Can you make this change in a compatible way using anonymous > union: > union { > bool is_first_packet_in_frame; > RTC_DEPRECATED bool isFirstPacket; > }; > (unfortunetly this this treak breaks braced initialization in > rtp_rtcp_impl_unittest.cc, > so that should be rewritting in a more classic way) > > Original issue's description: > > Rename RTPVideoHeader.isFirstPacket to > > .is_first_packet_in_frame. > > > > Name should represent the actual meaning. > > > > BUG=None > > > > Review-Url: https://codereview.webrtc.org/2574943003 > > Cr-Commit-Position: refs/heads/master@{#15684} > > Committed: > > https://chromium.googlesource.com/external/webrtc/+/efde90838055f44ca05863ba020ca02c88b6d14c > > TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days > ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=None > > Review-Url: https://codereview.webrtc.org/2589783003 > Cr-Commit-Position: refs/heads/master@{#15686} > BUG=None Review-Url: https://codereview.webrtc.org/2614503002 Cr-Commit-Position: refs/heads/master@{#15987}
2017-01-10 04:21:35 -08:00
if (video_header.is_first_packet_in_frame) {
if (nalu.pps_id == -1) {
RTC_LOG(LS_WARNING) << "No PPS id in IDR nalu.";
return kRequestKeyframe;
}
pps = pps_data_.find(nalu.pps_id);
if (pps == pps_data_.end()) {
RTC_LOG(LS_WARNING)
<< "No PPS with id << " << nalu.pps_id << " received";
return kRequestKeyframe;
}
sps = sps_data_.find(pps->second.sps_id);
if (sps == sps_data_.end()) {
RTC_LOG(LS_WARNING)
<< "No SPS with id << " << pps->second.sps_id << " received";
return kRequestKeyframe;
}
// Since the first packet of every keyframe should have its width and
// height set we set it here in the case of it being supplied out of
// band.
packet->video_header.width = sps->second.width;
packet->video_header.height = sps->second.height;
// If the SPS/PPS was supplied out of band then we will have saved
// the actual bitstream in |data|.
if (sps->second.data && pps->second.data) {
RTC_DCHECK_GT(sps->second.size, 0);
RTC_DCHECK_GT(pps->second.size, 0);
append_sps_pps = true;
}
}
break;
}
default:
break;
}
}
RTC_CHECK(!append_sps_pps ||
(sps != sps_data_.end() && pps != pps_data_.end()));
// Calculate how much space we need for the rest of the bitstream.
size_t required_size = 0;
if (append_sps_pps) {
required_size += sps->second.size + sizeof(start_code_h264);
required_size += pps->second.size + sizeof(start_code_h264);
}
if (h264_header.packetization_type == kH264StapA) {
const uint8_t* nalu_ptr = data + 1;
while (nalu_ptr < data + data_size) {
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. Add RTC_DEPRACATed anonymous unions to not break downstream projects. Orignal issue's description: > commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff > Author: danilchap <danilchap@webrtc.org> > Date: Mon Dec 19 09:36:33 2016 -0800 > > Revert of Rename RTPVideoHeader.isFirstPacket to > .is_first_packet_in_frame. (patchset #1 id:1 of > https://codereview.webrtc.org/2574943003/ ) > > Reason for revert: > breaks downstream project. > > Can you make this change in a compatible way using anonymous > union: > union { > bool is_first_packet_in_frame; > RTC_DEPRECATED bool isFirstPacket; > }; > (unfortunetly this this treak breaks braced initialization in > rtp_rtcp_impl_unittest.cc, > so that should be rewritting in a more classic way) > > Original issue's description: > > Rename RTPVideoHeader.isFirstPacket to > > .is_first_packet_in_frame. > > > > Name should represent the actual meaning. > > > > BUG=None > > > > Review-Url: https://codereview.webrtc.org/2574943003 > > Cr-Commit-Position: refs/heads/master@{#15684} > > Committed: > > https://chromium.googlesource.com/external/webrtc/+/efde90838055f44ca05863ba020ca02c88b6d14c > > TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days > ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=None > > Review-Url: https://codereview.webrtc.org/2589783003 > Cr-Commit-Position: refs/heads/master@{#15686} > BUG=None Review-Url: https://codereview.webrtc.org/2614503002 Cr-Commit-Position: refs/heads/master@{#15987}
2017-01-10 04:21:35 -08:00
RTC_DCHECK(video_header.is_first_packet_in_frame);
required_size += sizeof(start_code_h264);
// The first two bytes describe the length of a segment.
uint16_t segment_length = nalu_ptr[0] << 8 | nalu_ptr[1];
nalu_ptr += 2;
required_size += segment_length;
nalu_ptr += segment_length;
}
} else {
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. Add RTC_DEPRACATed anonymous unions to not break downstream projects. Orignal issue's description: > commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff > Author: danilchap <danilchap@webrtc.org> > Date: Mon Dec 19 09:36:33 2016 -0800 > > Revert of Rename RTPVideoHeader.isFirstPacket to > .is_first_packet_in_frame. (patchset #1 id:1 of > https://codereview.webrtc.org/2574943003/ ) > > Reason for revert: > breaks downstream project. > > Can you make this change in a compatible way using anonymous > union: > union { > bool is_first_packet_in_frame; > RTC_DEPRECATED bool isFirstPacket; > }; > (unfortunetly this this treak breaks braced initialization in > rtp_rtcp_impl_unittest.cc, > so that should be rewritting in a more classic way) > > Original issue's description: > > Rename RTPVideoHeader.isFirstPacket to > > .is_first_packet_in_frame. > > > > Name should represent the actual meaning. > > > > BUG=None > > > > Review-Url: https://codereview.webrtc.org/2574943003 > > Cr-Commit-Position: refs/heads/master@{#15684} > > Committed: > > https://chromium.googlesource.com/external/webrtc/+/efde90838055f44ca05863ba020ca02c88b6d14c > > TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days > ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=None > > Review-Url: https://codereview.webrtc.org/2589783003 > Cr-Commit-Position: refs/heads/master@{#15686} > BUG=None Review-Url: https://codereview.webrtc.org/2614503002 Cr-Commit-Position: refs/heads/master@{#15987}
2017-01-10 04:21:35 -08:00
if (video_header.is_first_packet_in_frame)
required_size += sizeof(start_code_h264);
required_size += data_size;
}
// Then we copy to the new buffer.
uint8_t* buffer = new uint8_t[required_size];
uint8_t* insert_at = buffer;
if (append_sps_pps) {
// Insert SPS.
memcpy(insert_at, start_code_h264, sizeof(start_code_h264));
insert_at += sizeof(start_code_h264);
memcpy(insert_at, sps->second.data.get(), sps->second.size);
insert_at += sps->second.size;
// Insert PPS.
memcpy(insert_at, start_code_h264, sizeof(start_code_h264));
insert_at += sizeof(start_code_h264);
memcpy(insert_at, pps->second.data.get(), pps->second.size);
insert_at += pps->second.size;
// Update codec header to reflect the newly added SPS and PPS.
NaluInfo sps_info;
sps_info.type = H264::NaluType::kSps;
sps_info.sps_id = sps->first;
sps_info.pps_id = -1;
NaluInfo pps_info;
pps_info.type = H264::NaluType::kPps;
pps_info.sps_id = sps->first;
pps_info.pps_id = pps->first;
if (h264_header.nalus_length + 2 <= kMaxNalusPerPacket) {
h264_header.nalus[h264_header.nalus_length++] = sps_info;
h264_header.nalus[h264_header.nalus_length++] = pps_info;
} else {
RTC_LOG(LS_WARNING) << "Not enough space in H.264 codec header to insert "
"SPS/PPS provided out-of-band.";
}
}
// Copy the rest of the bitstream and insert start codes.
if (h264_header.packetization_type == kH264StapA) {
const uint8_t* nalu_ptr = data + 1;
while (nalu_ptr < data + data_size) {
memcpy(insert_at, start_code_h264, sizeof(start_code_h264));
insert_at += sizeof(start_code_h264);
// The first two bytes describe the length of a segment.
uint16_t segment_length = nalu_ptr[0] << 8 | nalu_ptr[1];
nalu_ptr += 2;
size_t copy_end = nalu_ptr - data + segment_length;
if (copy_end > data_size) {
delete[] buffer;
return kDrop;
}
memcpy(insert_at, nalu_ptr, segment_length);
insert_at += segment_length;
nalu_ptr += segment_length;
}
} else {
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. Add RTC_DEPRACATed anonymous unions to not break downstream projects. Orignal issue's description: > commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff > Author: danilchap <danilchap@webrtc.org> > Date: Mon Dec 19 09:36:33 2016 -0800 > > Revert of Rename RTPVideoHeader.isFirstPacket to > .is_first_packet_in_frame. (patchset #1 id:1 of > https://codereview.webrtc.org/2574943003/ ) > > Reason for revert: > breaks downstream project. > > Can you make this change in a compatible way using anonymous > union: > union { > bool is_first_packet_in_frame; > RTC_DEPRECATED bool isFirstPacket; > }; > (unfortunetly this this treak breaks braced initialization in > rtp_rtcp_impl_unittest.cc, > so that should be rewritting in a more classic way) > > Original issue's description: > > Rename RTPVideoHeader.isFirstPacket to > > .is_first_packet_in_frame. > > > > Name should represent the actual meaning. > > > > BUG=None > > > > Review-Url: https://codereview.webrtc.org/2574943003 > > Cr-Commit-Position: refs/heads/master@{#15684} > > Committed: > > https://chromium.googlesource.com/external/webrtc/+/efde90838055f44ca05863ba020ca02c88b6d14c > > TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days > ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=None > > Review-Url: https://codereview.webrtc.org/2589783003 > Cr-Commit-Position: refs/heads/master@{#15686} > BUG=None Review-Url: https://codereview.webrtc.org/2614503002 Cr-Commit-Position: refs/heads/master@{#15987}
2017-01-10 04:21:35 -08:00
if (video_header.is_first_packet_in_frame) {
memcpy(insert_at, start_code_h264, sizeof(start_code_h264));
insert_at += sizeof(start_code_h264);
}
memcpy(insert_at, data, data_size);
}
packet->dataPtr = buffer;
packet->sizeBytes = required_size;
return kInsert;
}
void H264SpsPpsTracker::InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
const std::vector<uint8_t>& pps) {
constexpr size_t kNaluHeaderOffset = 1;
if (sps.size() < kNaluHeaderOffset) {
RTC_LOG(LS_WARNING) << "SPS size " << sps.size() << " is smaller than "
<< kNaluHeaderOffset;
return;
}
if ((sps[0] & 0x1f) != H264::NaluType::kSps) {
RTC_LOG(LS_WARNING) << "SPS Nalu header missing";
return;
}
if (pps.size() < kNaluHeaderOffset) {
RTC_LOG(LS_WARNING) << "PPS size " << pps.size() << " is smaller than "
<< kNaluHeaderOffset;
return;
}
if ((pps[0] & 0x1f) != H264::NaluType::kPps) {
RTC_LOG(LS_WARNING) << "SPS Nalu header missing";
return;
}
absl::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
sps.data() + kNaluHeaderOffset, sps.size() - kNaluHeaderOffset);
absl::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
pps.data() + kNaluHeaderOffset, pps.size() - kNaluHeaderOffset);
if (!parsed_sps) {
RTC_LOG(LS_WARNING) << "Failed to parse SPS.";
}
if (!parsed_pps) {
RTC_LOG(LS_WARNING) << "Failed to parse PPS.";
}
if (!parsed_pps || !parsed_sps) {
return;
}
SpsInfo sps_info;
sps_info.size = sps.size();
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sps_info.width = parsed_sps->width;
sps_info.height = parsed_sps->height;
uint8_t* sps_data = new uint8_t[sps_info.size];
memcpy(sps_data, sps.data(), sps_info.size);
sps_info.data.reset(sps_data);
sps_data_[parsed_sps->id] = std::move(sps_info);
PpsInfo pps_info;
pps_info.size = pps.size();
pps_info.sps_id = parsed_pps->sps_id;
uint8_t* pps_data = new uint8_t[pps_info.size];
memcpy(pps_data, pps.data(), pps_info.size);
pps_info.data.reset(pps_data);
pps_data_[parsed_pps->id] = std::move(pps_info);
RTC_LOG(LS_INFO) << "Inserted SPS id " << parsed_sps->id << " and PPS id "
<< parsed_pps->id << " (referencing SPS "
<< parsed_pps->sps_id << ")";
}
} // namespace video_coding
} // namespace webrtc