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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 "media/base/rtpdataengine.h"
#include <map>
#include "absl/strings/match.h"
#include "media/base/codec.h"
#include "media/base/mediaconstants.h"
#include "media/base/rtputils.h"
#include "media/base/streamparams.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/data_rate_limiter.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/sanitizer.h"
namespace cricket {
// We want to avoid IP fragmentation.
static const size_t kDataMaxRtpPacketLen = 1200U;
// We reserve space after the RTP header for future wiggle room.
static const unsigned char kReservedSpace[] = {0x00, 0x00, 0x00, 0x00};
// Amount of overhead SRTP may take. We need to leave room in the
// buffer for it, otherwise SRTP will fail later. If SRTP ever uses
// more than this, we need to increase this number.
static const size_t kMaxSrtpHmacOverhead = 16;
RtpDataEngine::RtpDataEngine() {
data_codecs_.push_back(
DataCodec(kGoogleRtpDataCodecPlType, kGoogleRtpDataCodecName));
}
DataMediaChannel* RtpDataEngine::CreateChannel(const MediaConfig& config) {
return new RtpDataMediaChannel(config);
}
static const DataCodec* FindCodecByName(const std::vector<DataCodec>& codecs,
const std::string& name) {
for (const DataCodec& codec : codecs) {
if (absl::EqualsIgnoreCase(name, codec.name))
return &codec;
}
return nullptr;
}
RtpDataMediaChannel::RtpDataMediaChannel(const MediaConfig& config)
: DataMediaChannel(config) {
Construct();
}
void RtpDataMediaChannel::Construct() {
sending_ = false;
receiving_ = false;
send_limiter_.reset(new rtc::DataRateLimiter(kDataMaxBandwidth / 8, 1.0));
}
RtpDataMediaChannel::~RtpDataMediaChannel() {
std::map<uint32_t, RtpClock*>::const_iterator iter;
for (iter = rtp_clock_by_send_ssrc_.begin();
iter != rtp_clock_by_send_ssrc_.end(); ++iter) {
delete iter->second;
}
}
void RTC_NO_SANITIZE("float-cast-overflow") // bugs.webrtc.org/8204
RtpClock::Tick(double now, int* seq_num, uint32_t* timestamp) {
*seq_num = ++last_seq_num_;
*timestamp = timestamp_offset_ + static_cast<uint32_t>(now * clockrate_);
// UBSan: 5.92374e+10 is outside the range of representable values of type
// 'unsigned int'
}
const DataCodec* FindUnknownCodec(const std::vector<DataCodec>& codecs) {
DataCodec data_codec(kGoogleRtpDataCodecPlType, kGoogleRtpDataCodecName);
std::vector<DataCodec>::const_iterator iter;
for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
if (!iter->Matches(data_codec)) {
return &(*iter);
}
}
return NULL;
}
const DataCodec* FindKnownCodec(const std::vector<DataCodec>& codecs) {
DataCodec data_codec(kGoogleRtpDataCodecPlType, kGoogleRtpDataCodecName);
std::vector<DataCodec>::const_iterator iter;
for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
if (iter->Matches(data_codec)) {
return &(*iter);
}
}
return NULL;
}
bool RtpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
const DataCodec* unknown_codec = FindUnknownCodec(codecs);
if (unknown_codec) {
RTC_LOG(LS_WARNING) << "Failed to SetRecvCodecs because of unknown codec: "
<< unknown_codec->ToString();
return false;
}
recv_codecs_ = codecs;
return true;
}
bool RtpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
const DataCodec* known_codec = FindKnownCodec(codecs);
if (!known_codec) {
RTC_LOG(LS_WARNING)
<< "Failed to SetSendCodecs because there is no known codec.";
return false;
}
send_codecs_ = codecs;
return true;
}
bool RtpDataMediaChannel::SetSendParameters(const DataSendParameters& params) {
return (SetSendCodecs(params.codecs) &&
SetMaxSendBandwidth(params.max_bandwidth_bps));
}
bool RtpDataMediaChannel::SetRecvParameters(const DataRecvParameters& params) {
return SetRecvCodecs(params.codecs);
}
bool RtpDataMediaChannel::AddSendStream(const StreamParams& stream) {
if (!stream.has_ssrcs()) {
return false;
}
if (GetStreamBySsrc(send_streams_, stream.first_ssrc())) {
RTC_LOG(LS_WARNING) << "Not adding data send stream '" << stream.id
<< "' with ssrc=" << stream.first_ssrc()
<< " because stream already exists.";
return false;
}
send_streams_.push_back(stream);
// TODO(pthatcher): This should be per-stream, not per-ssrc.
// And we should probably allow more than one per stream.
rtp_clock_by_send_ssrc_[stream.first_ssrc()] =
new RtpClock(kDataCodecClockrate, rtc::CreateRandomNonZeroId(),
rtc::CreateRandomNonZeroId());
RTC_LOG(LS_INFO) << "Added data send stream '" << stream.id
<< "' with ssrc=" << stream.first_ssrc();
return true;
}
bool RtpDataMediaChannel::RemoveSendStream(uint32_t ssrc) {
if (!GetStreamBySsrc(send_streams_, ssrc)) {
return false;
}
RemoveStreamBySsrc(&send_streams_, ssrc);
delete rtp_clock_by_send_ssrc_[ssrc];
rtp_clock_by_send_ssrc_.erase(ssrc);
return true;
}
bool RtpDataMediaChannel::AddRecvStream(const StreamParams& stream) {
if (!stream.has_ssrcs()) {
return false;
}
if (GetStreamBySsrc(recv_streams_, stream.first_ssrc())) {
RTC_LOG(LS_WARNING) << "Not adding data recv stream '" << stream.id
<< "' with ssrc=" << stream.first_ssrc()
<< " because stream already exists.";
return false;
}
recv_streams_.push_back(stream);
RTC_LOG(LS_INFO) << "Added data recv stream '" << stream.id
<< "' with ssrc=" << stream.first_ssrc();
return true;
}
bool RtpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
RemoveStreamBySsrc(&recv_streams_, ssrc);
return true;
}
void RtpDataMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
int64_t /* packet_time_us */) {
RtpHeader header;
if (!GetRtpHeader(packet->cdata(), packet->size(), &header)) {
return;
}
size_t header_length;
if (!GetRtpHeaderLen(packet->cdata(), packet->size(), &header_length)) {
return;
}
const char* data =
packet->cdata<char>() + header_length + sizeof(kReservedSpace);
size_t data_len = packet->size() - header_length - sizeof(kReservedSpace);
if (!receiving_) {
RTC_LOG(LS_WARNING) << "Not receiving packet " << header.ssrc << ":"
<< header.seq_num << " before SetReceive(true) called.";
return;
}
if (!FindCodecById(recv_codecs_, header.payload_type)) {
return;
}
if (!GetStreamBySsrc(recv_streams_, header.ssrc)) {
RTC_LOG(LS_WARNING) << "Received packet for unknown ssrc: " << header.ssrc;
return;
}
// Uncomment this for easy debugging.
// const auto* found_stream = GetStreamBySsrc(recv_streams_, header.ssrc);
// RTC_LOG(LS_INFO) << "Received packet"
// << " groupid=" << found_stream.groupid
// << ", ssrc=" << header.ssrc
// << ", seqnum=" << header.seq_num
// << ", timestamp=" << header.timestamp
// << ", len=" << data_len;
ReceiveDataParams params;
params.ssrc = header.ssrc;
params.seq_num = header.seq_num;
params.timestamp = header.timestamp;
SignalDataReceived(params, data, data_len);
}
bool RtpDataMediaChannel::SetMaxSendBandwidth(int bps) {
if (bps <= 0) {
bps = kDataMaxBandwidth;
}
send_limiter_.reset(new rtc::DataRateLimiter(bps / 8, 1.0));
RTC_LOG(LS_INFO) << "RtpDataMediaChannel::SetSendBandwidth to " << bps
<< "bps.";
return true;
}
bool RtpDataMediaChannel::SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result) {
if (result) {
// If we return true, we'll set this to SDR_SUCCESS.
*result = SDR_ERROR;
}
if (!sending_) {
RTC_LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssrc
<< " len=" << payload.size()
<< " before SetSend(true).";
return false;
}
if (params.type != cricket::DMT_TEXT) {
RTC_LOG(LS_WARNING)
<< "Not sending data because binary type is unsupported.";
return false;
}
const StreamParams* found_stream =
GetStreamBySsrc(send_streams_, params.ssrc);
if (!found_stream) {
RTC_LOG(LS_WARNING) << "Not sending data because ssrc is unknown: "
<< params.ssrc;
return false;
}
const DataCodec* found_codec =
FindCodecByName(send_codecs_, kGoogleRtpDataCodecName);
if (!found_codec) {
RTC_LOG(LS_WARNING) << "Not sending data because codec is unknown: "
<< kGoogleRtpDataCodecName;
return false;
}
size_t packet_len = (kMinRtpPacketLen + sizeof(kReservedSpace) +
payload.size() + kMaxSrtpHmacOverhead);
if (packet_len > kDataMaxRtpPacketLen) {
return false;
}
double now =
rtc::TimeMicros() / static_cast<double>(rtc::kNumMicrosecsPerSec);
if (!send_limiter_->CanUse(packet_len, now)) {
RTC_LOG(LS_VERBOSE) << "Dropped data packet of len=" << packet_len
<< "; already sent " << send_limiter_->used_in_period()
<< "/" << send_limiter_->max_per_period();
return false;
}
RtpHeader header;
header.payload_type = found_codec->id;
header.ssrc = params.ssrc;
rtp_clock_by_send_ssrc_[header.ssrc]->Tick(now, &header.seq_num,
&header.timestamp);
rtc::CopyOnWriteBuffer packet(kMinRtpPacketLen, packet_len);
if (!SetRtpHeader(packet.data(), packet.size(), header)) {
return false;
}
packet.AppendData(kReservedSpace);
packet.AppendData(payload);
RTC_LOG(LS_VERBOSE) << "Sent RTP data packet: "
<< " stream=" << found_stream->id
<< " ssrc=" << header.ssrc
<< ", seqnum=" << header.seq_num
<< ", timestamp=" << header.timestamp
<< ", len=" << payload.size();
rtc::PacketOptions options;
options.info_signaled_after_sent.packet_type = rtc::PacketType::kData;
MediaChannel::SendPacket(&packet, options);
send_limiter_->Use(packet_len, now);
if (result) {
*result = SDR_SUCCESS;
}
return true;
}
rtc::DiffServCodePoint RtpDataMediaChannel::PreferredDscp() const {
return rtc::DSCP_AF41;
}
} // namespace cricket