webrtc_m130/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc

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/*
* Copyright (c) 2013 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 "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
namespace webrtc {
namespace {
bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const CodecInst& audio_codec) {
if (!payload.audio)
return false;
if (_stricmp(payload.name, audio_codec.plname) != 0)
return false;
const AudioPayload& audio_payload = payload.typeSpecific.Audio;
const uint32_t rate = std::max(0, audio_codec.rate);
return audio_payload.frequency == static_cast<uint32_t>(audio_codec.plfreq) &&
audio_payload.channels == audio_codec.channels &&
(audio_payload.rate == rate || audio_payload.rate == 0 || rate == 0);
}
bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const VideoCodec& video_codec) {
return !payload.audio && _stricmp(payload.name, video_codec.plName) == 0;
}
RtpUtility::Payload CreatePayloadType(const CodecInst& audio_codec) {
RtpUtility::Payload payload;
payload.name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload.name, audio_codec.plname, RTP_PAYLOAD_NAME_SIZE - 1);
RTC_DCHECK_GE(audio_codec.plfreq, 1000);
payload.typeSpecific.Audio.frequency = audio_codec.plfreq;
payload.typeSpecific.Audio.channels = audio_codec.channels;
payload.typeSpecific.Audio.rate = std::max(0, audio_codec.rate);
payload.audio = true;
return payload;
}
RtpVideoCodecTypes ConvertToRtpVideoCodecType(VideoCodecType type) {
switch (type) {
case kVideoCodecVP8:
return kRtpVideoVp8;
case kVideoCodecVP9:
return kRtpVideoVp9;
case kVideoCodecH264:
return kRtpVideoH264;
case kVideoCodecRED:
case kVideoCodecULPFEC:
return kRtpVideoNone;
default:
return kRtpVideoGeneric;
}
}
RtpUtility::Payload CreatePayloadType(const VideoCodec& video_codec) {
RtpUtility::Payload payload;
payload.name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload.name, video_codec.plName, RTP_PAYLOAD_NAME_SIZE - 1);
payload.typeSpecific.Video.videoCodecType =
ConvertToRtpVideoCodecType(video_codec.codecType);
payload.audio = false;
return payload;
}
bool IsPayloadTypeValid(int8_t payload_type) {
assert(payload_type >= 0);
// Sanity check.
switch (payload_type) {
// Reserved payload types to avoid RTCP conflicts when marker bit is set.
case 64: // 192 Full INTRA-frame request.
case 72: // 200 Sender report.
case 73: // 201 Receiver report.
case 74: // 202 Source description.
case 75: // 203 Goodbye.
case 76: // 204 Application-defined.
case 77: // 205 Transport layer FB message.
case 78: // 206 Payload-specific FB message.
case 79: // 207 Extended report.
LOG(LS_ERROR) << "Can't register invalid receiver payload type: "
<< payload_type;
return false;
default:
return true;
}
}
} // namespace
RTPPayloadRegistry::RTPPayloadRegistry()
: incoming_payload_type_(-1),
last_received_payload_type_(-1),
last_received_media_payload_type_(-1),
rtx_(false),
ssrc_rtx_(0) {}
RTPPayloadRegistry::~RTPPayloadRegistry() = default;
int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec,
bool* created_new_payload) {
*created_new_payload = false;
if (!IsPayloadTypeValid(audio_codec.pltype))
return -1;
rtc::CritScope cs(&crit_sect_);
auto it = payload_type_map_.find(audio_codec.pltype);
if (it != payload_type_map_.end()) {
// We already use this payload type. Check if it's the same as we already
// have. If same, ignore sending an error.
if (PayloadIsCompatible(it->second, audio_codec)) {
it->second.typeSpecific.Audio.rate = std::max(0, audio_codec.rate);
return 0;
}
LOG(LS_ERROR) << "Payload type already registered: " << audio_codec.pltype;
return -1;
}
// Audio codecs must be unique.
DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(audio_codec);
payload_type_map_[audio_codec.pltype] = CreatePayloadType(audio_codec);
*created_new_payload = true;
// Successful set of payload type, clear the value of last received payload
// type since it might mean something else.
last_received_payload_type_ = -1;
last_received_media_payload_type_ = -1;
return 0;
}
int32_t RTPPayloadRegistry::RegisterReceivePayload(
const VideoCodec& video_codec) {
if (!IsPayloadTypeValid(video_codec.plType))
return -1;
rtc::CritScope cs(&crit_sect_);
auto it = payload_type_map_.find(video_codec.plType);
if (it != payload_type_map_.end()) {
// We already use this payload type. Check if it's the same as we already
// have. If same, ignore sending an error.
if (PayloadIsCompatible(it->second, video_codec))
return 0;
LOG(LS_ERROR) << "Payload type already registered: "
<< static_cast<int>(video_codec.plType);
return -1;
}
payload_type_map_[video_codec.plType] = CreatePayloadType(video_codec);
// Successful set of payload type, clear the value of last received payload
// type since it might mean something else.
last_received_payload_type_ = -1;
last_received_media_payload_type_ = -1;
return 0;
}
int32_t RTPPayloadRegistry::DeRegisterReceivePayload(
const int8_t payload_type) {
rtc::CritScope cs(&crit_sect_);
payload_type_map_.erase(payload_type);
return 0;
}
// There can't be several codecs with the same rate, frequency and channels
// for audio codecs, but there can for video.
// Always called from within a critical section.
void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
const CodecInst& audio_codec) {
for (auto iterator = payload_type_map_.begin();
iterator != payload_type_map_.end(); ++iterator) {
if (PayloadIsCompatible(iterator->second, audio_codec)) {
// Remove old setting.
payload_type_map_.erase(iterator);
break;
}
}
}
int32_t RTPPayloadRegistry::ReceivePayloadType(const CodecInst& audio_codec,
int8_t* payload_type) const {
assert(payload_type);
rtc::CritScope cs(&crit_sect_);
for (const auto& it : payload_type_map_) {
if (PayloadIsCompatible(it.second, audio_codec)) {
*payload_type = it.first;
return 0;
}
}
return -1;
}
int32_t RTPPayloadRegistry::ReceivePayloadType(const VideoCodec& video_codec,
int8_t* payload_type) const {
assert(payload_type);
rtc::CritScope cs(&crit_sect_);
for (const auto& it : payload_type_map_) {
if (PayloadIsCompatible(it.second, video_codec)) {
*payload_type = it.first;
return 0;
}
}
return -1;
}
bool RTPPayloadRegistry::RtxEnabled() const {
rtc::CritScope cs(&crit_sect_);
return rtx_;
}
bool RTPPayloadRegistry::IsRtx(const RTPHeader& header) const {
rtc::CritScope cs(&crit_sect_);
return IsRtxInternal(header);
}
bool RTPPayloadRegistry::IsRtxInternal(const RTPHeader& header) const {
return rtx_ && ssrc_rtx_ == header.ssrc;
}
bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t* restored_packet,
const uint8_t* packet,
size_t* packet_length,
uint32_t original_ssrc,
Remove RED/RTX workaround from sender/receiver and VideoEngine2. In older Chrome versions, the associated payload type in the RTX header of retransmitted packets was always set to be the original media payload type, regardless of the actual payload type of the packet. This meant that packets encapsulated with RED headers had incorrect payload type information in the RTX header. Due to an assumption in the receiver, this incorrect payload type information would effectively be undone, leading to a working system. Albeit working, this behaviour was undesired, and thus removed. In the interim, several workarounds were introduced to not destroy interop between old and new Chrome versions: (1) https://codereview.webrtc.org/1649493004 - If no payload type mapping existed for RED over RTX, the payload type of the underlying media would be used. - If RED had been negotiated, received RTX packets would always be assumed to contain RED. (2) https://codereview.webrtc.org/1964473002 - If RED was removed from the remote description answer, it would be disabled in the local receiver as well. (3) https://codereview.webrtc.org/2033763002 - If RED was negotiated in the SDP, it would always be used, regardless if ULPFEC was negotiated and used, or not. Since the Chrome versions that exhibited the original bug now are very old, this CL removes the workarounds from (1) and (2). In particular, after this change, we will have the following behaviour: - We assume that a payload type mapping for RED over RTX always is set. If this is not the case, the RTX packet is not sent. - The associated payload type of received RTX packets will always be obeyed. - The (non)-existence of RED in the remote description does not affect the local receiver. The workaround in (3) still needs to exist, in order to interop with receivers that did not have the workarounds in (1) and (2) removed. The change in (3) can be removed in a couple of Chrome versions. TESTED=Using AppRTC between patched Chrome (connected to ethernet) and standard Chrome M54 (connected to lossy internal Google WiFi), with and without FEC turned off using AppRTC flag. Also using "Munge SDP" sample on patched Chrome over loopback interface, with 100ms delay and 5% packet loss simulated using tc. BUG=webrtc:6650 Review-Url: https://codereview.webrtc.org/2469093003 Cr-Commit-Position: refs/heads/master@{#15038}
2016-11-11 03:28:30 -08:00
const RTPHeader& header) {
if (kRtxHeaderSize + header.headerLength + header.paddingLength >
*packet_length) {
return false;
}
const uint8_t* rtx_header = packet + header.headerLength;
uint16_t original_sequence_number = (rtx_header[0] << 8) + rtx_header[1];
// Copy the packet into the restored packet, except for the RTX header.
memcpy(restored_packet, packet, header.headerLength);
memcpy(restored_packet + header.headerLength,
packet + header.headerLength + kRtxHeaderSize,
*packet_length - header.headerLength - kRtxHeaderSize);
*packet_length -= kRtxHeaderSize;
// Replace the SSRC and the sequence number with the originals.
ByteWriter<uint16_t>::WriteBigEndian(restored_packet + 2,
original_sequence_number);
ByteWriter<uint32_t>::WriteBigEndian(restored_packet + 8, original_ssrc);
rtc::CritScope cs(&crit_sect_);
if (!rtx_)
return true;
auto apt_mapping = rtx_payload_type_map_.find(header.payloadType);
Remove RED/RTX workaround from sender/receiver and VideoEngine2. In older Chrome versions, the associated payload type in the RTX header of retransmitted packets was always set to be the original media payload type, regardless of the actual payload type of the packet. This meant that packets encapsulated with RED headers had incorrect payload type information in the RTX header. Due to an assumption in the receiver, this incorrect payload type information would effectively be undone, leading to a working system. Albeit working, this behaviour was undesired, and thus removed. In the interim, several workarounds were introduced to not destroy interop between old and new Chrome versions: (1) https://codereview.webrtc.org/1649493004 - If no payload type mapping existed for RED over RTX, the payload type of the underlying media would be used. - If RED had been negotiated, received RTX packets would always be assumed to contain RED. (2) https://codereview.webrtc.org/1964473002 - If RED was removed from the remote description answer, it would be disabled in the local receiver as well. (3) https://codereview.webrtc.org/2033763002 - If RED was negotiated in the SDP, it would always be used, regardless if ULPFEC was negotiated and used, or not. Since the Chrome versions that exhibited the original bug now are very old, this CL removes the workarounds from (1) and (2). In particular, after this change, we will have the following behaviour: - We assume that a payload type mapping for RED over RTX always is set. If this is not the case, the RTX packet is not sent. - The associated payload type of received RTX packets will always be obeyed. - The (non)-existence of RED in the remote description does not affect the local receiver. The workaround in (3) still needs to exist, in order to interop with receivers that did not have the workarounds in (1) and (2) removed. The change in (3) can be removed in a couple of Chrome versions. TESTED=Using AppRTC between patched Chrome (connected to ethernet) and standard Chrome M54 (connected to lossy internal Google WiFi), with and without FEC turned off using AppRTC flag. Also using "Munge SDP" sample on patched Chrome over loopback interface, with 100ms delay and 5% packet loss simulated using tc. BUG=webrtc:6650 Review-Url: https://codereview.webrtc.org/2469093003 Cr-Commit-Position: refs/heads/master@{#15038}
2016-11-11 03:28:30 -08:00
if (apt_mapping == rtx_payload_type_map_.end()) {
// No associated payload type found. Warn, unless we have already done so.
if (payload_types_with_suppressed_warnings_.find(header.payloadType) ==
payload_types_with_suppressed_warnings_.end()) {
LOG(LS_WARNING)
<< "No RTX associated payload type mapping was available; "
"not able to restore original packet from RTX packet "
"with payload type: "
<< static_cast<int>(header.payloadType) << ". "
<< "Suppressing further warnings for this payload type.";
payload_types_with_suppressed_warnings_.insert(header.payloadType);
}
return false;
}
Remove RED/RTX workaround from sender/receiver and VideoEngine2. In older Chrome versions, the associated payload type in the RTX header of retransmitted packets was always set to be the original media payload type, regardless of the actual payload type of the packet. This meant that packets encapsulated with RED headers had incorrect payload type information in the RTX header. Due to an assumption in the receiver, this incorrect payload type information would effectively be undone, leading to a working system. Albeit working, this behaviour was undesired, and thus removed. In the interim, several workarounds were introduced to not destroy interop between old and new Chrome versions: (1) https://codereview.webrtc.org/1649493004 - If no payload type mapping existed for RED over RTX, the payload type of the underlying media would be used. - If RED had been negotiated, received RTX packets would always be assumed to contain RED. (2) https://codereview.webrtc.org/1964473002 - If RED was removed from the remote description answer, it would be disabled in the local receiver as well. (3) https://codereview.webrtc.org/2033763002 - If RED was negotiated in the SDP, it would always be used, regardless if ULPFEC was negotiated and used, or not. Since the Chrome versions that exhibited the original bug now are very old, this CL removes the workarounds from (1) and (2). In particular, after this change, we will have the following behaviour: - We assume that a payload type mapping for RED over RTX always is set. If this is not the case, the RTX packet is not sent. - The associated payload type of received RTX packets will always be obeyed. - The (non)-existence of RED in the remote description does not affect the local receiver. The workaround in (3) still needs to exist, in order to interop with receivers that did not have the workarounds in (1) and (2) removed. The change in (3) can be removed in a couple of Chrome versions. TESTED=Using AppRTC between patched Chrome (connected to ethernet) and standard Chrome M54 (connected to lossy internal Google WiFi), with and without FEC turned off using AppRTC flag. Also using "Munge SDP" sample on patched Chrome over loopback interface, with 100ms delay and 5% packet loss simulated using tc. BUG=webrtc:6650 Review-Url: https://codereview.webrtc.org/2469093003 Cr-Commit-Position: refs/heads/master@{#15038}
2016-11-11 03:28:30 -08:00
restored_packet[1] = static_cast<uint8_t>(apt_mapping->second);
if (header.markerBit) {
restored_packet[1] |= kRtpMarkerBitMask; // Marker bit is set.
}
return true;
}
void RTPPayloadRegistry::SetRtxSsrc(uint32_t ssrc) {
rtc::CritScope cs(&crit_sect_);
ssrc_rtx_ = ssrc;
rtx_ = true;
}
bool RTPPayloadRegistry::GetRtxSsrc(uint32_t* ssrc) const {
rtc::CritScope cs(&crit_sect_);
*ssrc = ssrc_rtx_;
return rtx_;
}
void RTPPayloadRegistry::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
rtc::CritScope cs(&crit_sect_);
if (payload_type < 0) {
LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
return;
}
rtx_payload_type_map_[payload_type] = associated_payload_type;
rtx_ = true;
}
bool RTPPayloadRegistry::IsRed(const RTPHeader& header) const {
rtc::CritScope cs(&crit_sect_);
auto it = payload_type_map_.find(header.payloadType);
return it != payload_type_map_.end() && _stricmp(it->second.name, "red") == 0;
}
bool RTPPayloadRegistry::IsEncapsulated(const RTPHeader& header) const {
return IsRed(header) || IsRtx(header);
}
bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type,
PayloadUnion* payload) const {
rtc::CritScope cs(&crit_sect_);
auto it = payload_type_map_.find(payload_type);
// Check that this is a registered payload type.
if (it == payload_type_map_.end()) {
return false;
}
*payload = it->second.typeSpecific;
return true;
}
int RTPPayloadRegistry::GetPayloadTypeFrequency(
uint8_t payload_type) const {
const RtpUtility::Payload* payload = PayloadTypeToPayload(payload_type);
if (!payload) {
return -1;
}
rtc::CritScope cs(&crit_sect_);
return payload->audio ? payload->typeSpecific.Audio.frequency
: kVideoPayloadTypeFrequency;
}
const RtpUtility::Payload* RTPPayloadRegistry::PayloadTypeToPayload(
uint8_t payload_type) const {
rtc::CritScope cs(&crit_sect_);
auto it = payload_type_map_.find(payload_type);
// Check that this is a registered payload type.
if (it == payload_type_map_.end()) {
return nullptr;
}
return &it->second;
}
void RTPPayloadRegistry::SetIncomingPayloadType(const RTPHeader& header) {
rtc::CritScope cs(&crit_sect_);
if (!IsRtxInternal(header))
incoming_payload_type_ = header.payloadType;
}
bool RTPPayloadRegistry::ReportMediaPayloadType(uint8_t media_payload_type) {
rtc::CritScope cs(&crit_sect_);
if (last_received_media_payload_type_ == media_payload_type) {
// Media type unchanged.
return true;
}
last_received_media_payload_type_ = media_payload_type;
return false;
}
// Returns -1 if a payload with name |payload_name| is not registered.
int8_t RTPPayloadRegistry::GetPayloadTypeWithName(
const char* payload_name) const {
rtc::CritScope cs(&crit_sect_);
for (const auto& it : payload_type_map_) {
if (_stricmp(it.second.name, payload_name) == 0)
return it.first;
}
return -1;
}
} // namespace webrtc