Remove RTP data implementation

Bug: webrtc:6625
Change-Id: Ie68d7a938d8b7be95a01cca74a176104e4e44e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215321
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33759}

media/BUILD.gn

@@ -142,8 +142,6 @@ rtc_library("rtc_media_base") {
     "base/media_engine.h",
     "base/rid_description.cc",
     "base/rid_description.h",
-    "base/rtp_data_engine.cc",
-    "base/rtp_data_engine.h",
     "base/rtp_utils.cc",
     "base/rtp_utils.h",
     "base/stream_params.cc",
@@ -628,7 +626,6 @@ if (rtc_include_tests) {
       sources = [
         "base/codec_unittest.cc",
         "base/media_engine_unittest.cc",
-        "base/rtp_data_engine_unittest.cc",
         "base/rtp_utils_unittest.cc",
         "base/sdp_fmtp_utils_unittest.cc",
         "base/stream_params_unittest.cc",

media/base/fake_media_engine.cc

@@ -422,93 +422,6 @@ void FakeVideoMediaChannel::ClearRecordableEncodedFrameCallback(uint32_t ssrc) {
 
 void FakeVideoMediaChannel::GenerateKeyFrame(uint32_t ssrc) {}
 
-FakeDataMediaChannel::FakeDataMediaChannel(void* unused,
-                                           const DataOptions& options)
-    : send_blocked_(false), max_bps_(-1) {}
-FakeDataMediaChannel::~FakeDataMediaChannel() {}
-const std::vector<DataCodec>& FakeDataMediaChannel::recv_codecs() const {
-  return recv_codecs_;
-}
-const std::vector<DataCodec>& FakeDataMediaChannel::send_codecs() const {
-  return send_codecs_;
-}
-const std::vector<DataCodec>& FakeDataMediaChannel::codecs() const {
-  return send_codecs();
-}
-int FakeDataMediaChannel::max_bps() const {
-  return max_bps_;
-}
-bool FakeDataMediaChannel::SetSendParameters(const DataSendParameters& params) {
-  set_send_rtcp_parameters(params.rtcp);
-  return (SetSendCodecs(params.codecs) &&
-          SetMaxSendBandwidth(params.max_bandwidth_bps));
-}
-bool FakeDataMediaChannel::SetRecvParameters(const DataRecvParameters& params) {
-  set_recv_rtcp_parameters(params.rtcp);
-  return SetRecvCodecs(params.codecs);
-}
-bool FakeDataMediaChannel::SetSend(bool send) {
-  return set_sending(send);
-}
-bool FakeDataMediaChannel::SetReceive(bool receive) {
-  set_playout(receive);
-  return true;
-}
-bool FakeDataMediaChannel::AddRecvStream(const StreamParams& sp) {
-  if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
-    return false;
-  return true;
-}
-bool FakeDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
-  if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
-    return false;
-  return true;
-}
-bool FakeDataMediaChannel::SendData(const SendDataParams& params,
-                                    const rtc::CopyOnWriteBuffer& payload,
-                                    SendDataResult* result) {
-  if (send_blocked_) {
-    *result = SDR_BLOCK;
-    return false;
-  } else {
-    last_sent_data_params_ = params;
-    last_sent_data_ = std::string(payload.data<char>(), payload.size());
-    return true;
-  }
-}
-SendDataParams FakeDataMediaChannel::last_sent_data_params() {
-  return last_sent_data_params_;
-}
-std::string FakeDataMediaChannel::last_sent_data() {
-  return last_sent_data_;
-}
-bool FakeDataMediaChannel::is_send_blocked() {
-  return send_blocked_;
-}
-void FakeDataMediaChannel::set_send_blocked(bool blocked) {
-  send_blocked_ = blocked;
-}
-bool FakeDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
-  if (fail_set_recv_codecs()) {
-    // Fake the failure in SetRecvCodecs.
-    return false;
-  }
-  recv_codecs_ = codecs;
-  return true;
-}
-bool FakeDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
-  if (fail_set_send_codecs()) {
-    // Fake the failure in SetSendCodecs.
-    return false;
-  }
-  send_codecs_ = codecs;
-  return true;
-}
-bool FakeDataMediaChannel::SetMaxSendBandwidth(int bps) {
-  max_bps_ = bps;
-  return true;
-}
-
 FakeVoiceEngine::FakeVoiceEngine() : fail_create_channel_(false) {
   // Add a fake audio codec. Note that the name must not be "" as there are
   // sanity checks against that.
@@ -668,22 +581,4 @@ void FakeMediaEngine::set_fail_create_channel(bool fail) {
   video_->fail_create_channel_ = fail;
 }
 
-DataMediaChannel* FakeDataEngine::CreateChannel(const MediaConfig& config) {
-  FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
-  channels_.push_back(ch);
-  return ch;
-}
-FakeDataMediaChannel* FakeDataEngine::GetChannel(size_t index) {
-  return (channels_.size() > index) ? channels_[index] : NULL;
-}
-void FakeDataEngine::UnregisterChannel(DataMediaChannel* channel) {
-  channels_.erase(absl::c_find(channels_, channel));
-}
-void FakeDataEngine::SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
-  data_codecs_ = data_codecs;
-}
-const std::vector<DataCodec>& FakeDataEngine::data_codecs() {
-  return data_codecs_;
-}
-
 }  // namespace cricket

media/base/fake_media_engine.h

@@ -472,48 +472,6 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
   int max_bps_;
 };
 
-// Dummy option class, needed for the DataTraits abstraction in
-// channel_unittest.c.
-class DataOptions {};
-
-class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
- public:
-  explicit FakeDataMediaChannel(void* unused, const DataOptions& options);
-  ~FakeDataMediaChannel();
-  const std::vector<DataCodec>& recv_codecs() const;
-  const std::vector<DataCodec>& send_codecs() const;
-  const std::vector<DataCodec>& codecs() const;
-  int max_bps() const;
-
-  bool SetSendParameters(const DataSendParameters& params) override;
-  bool SetRecvParameters(const DataRecvParameters& params) override;
-  bool SetSend(bool send) override;
-  bool SetReceive(bool receive) override;
-  bool AddRecvStream(const StreamParams& sp) override;
-  bool RemoveRecvStream(uint32_t ssrc) override;
-
-  bool SendData(const SendDataParams& params,
-                const rtc::CopyOnWriteBuffer& payload,
-                SendDataResult* result) override;
-
-  SendDataParams last_sent_data_params();
-  std::string last_sent_data();
-  bool is_send_blocked();
-  void set_send_blocked(bool blocked);
-
- private:
-  bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
-  bool SetSendCodecs(const std::vector<DataCodec>& codecs);
-  bool SetMaxSendBandwidth(int bps);
-
-  std::vector<DataCodec> recv_codecs_;
-  std::vector<DataCodec> send_codecs_;
-  SendDataParams last_sent_data_params_;
-  std::string last_sent_data_;
-  bool send_blocked_;
-  int max_bps_;
-};
-
 class FakeVoiceEngine : public VoiceEngineInterface {
  public:
   FakeVoiceEngine();
@@ -609,25 +567,6 @@ class FakeMediaEngine : public CompositeMediaEngine {
   FakeVideoEngine* const video_;
 };
 
-// Have to come afterwards due to declaration order
-
-class FakeDataEngine : public DataEngineInterface {
- public:
-  DataMediaChannel* CreateChannel(const MediaConfig& config) override;
-
-  FakeDataMediaChannel* GetChannel(size_t index);
-
-  void UnregisterChannel(DataMediaChannel* channel);
-
-  void SetDataCodecs(const std::vector<DataCodec>& data_codecs);
-
-  const std::vector<DataCodec>& data_codecs() override;
-
- private:
-  std::vector<FakeDataMediaChannel*> channels_;
-  std::vector<DataCodec> data_codecs_;
-};
-
 }  // namespace cricket
 
 #endif  // MEDIA_BASE_FAKE_MEDIA_ENGINE_H_

media/base/media_channel.h

@@ -955,11 +955,8 @@ enum DataMessageType {
 // signal fires, on up the chain.
 struct ReceiveDataParams {
   // The in-packet stream indentifier.
-  // RTP data channels use SSRCs, SCTP data channels use SIDs.
+  // SCTP data channels use SIDs.
-  union {
+  int sid = 0;
-    uint32_t ssrc;
-    int sid = 0;
-  };
   // The type of message (binary, text, or control).
   DataMessageType type = DMT_TEXT;
   // A per-stream value incremented per packet in the stream.
@@ -970,11 +967,7 @@ struct ReceiveDataParams {
 
 struct SendDataParams {
   // The in-packet stream indentifier.
-  // RTP data channels use SSRCs, SCTP data channels use SIDs.
+  int sid = 0;
-  union {
-    uint32_t ssrc;
-    int sid = 0;
-  };
   // The type of message (binary, text, or control).
   DataMessageType type = DMT_TEXT;
 

media/base/media_engine.h

@@ -158,17 +158,9 @@ class CompositeMediaEngine : public MediaEngineInterface {
 
 enum DataChannelType {
   DCT_NONE = 0,
-  DCT_RTP = 1,
   DCT_SCTP = 2,
 };
 
-class DataEngineInterface {
- public:
-  virtual ~DataEngineInterface() {}
-  virtual DataMediaChannel* CreateChannel(const MediaConfig& config) = 0;
-  virtual const std::vector<DataCodec>& data_codecs() = 0;
-};
-
 webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
 webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
 

media/base/rtp_data_engine.cc

@@ -1,340 +0,0 @@
-/*
- *  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/rtp_data_engine.h"
-
-#include <map>
-
-#include "absl/strings/match.h"
-#include "media/base/codec.h"
-#include "media/base/media_constants.h"
-#include "media/base/rtp_utils.h"
-#include "media/base/stream_params.h"
-#include "rtc_base/copy_on_write_buffer.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();
-  SetPreferredDscp(rtc::DSCP_AF41);
-}
-
-void RtpDataMediaChannel::Construct() {
-  sending_ = false;
-  receiving_ = false;
-  send_limiter_.reset(new rtc::DataRateLimiter(kRtpDataMaxBandwidth / 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;
-}
-
-// Not implemented.
-void RtpDataMediaChannel::ResetUnsignaledRecvStream() {}
-void RtpDataMediaChannel::OnDemuxerCriteriaUpdatePending() {}
-void RtpDataMediaChannel::OnDemuxerCriteriaUpdateComplete() {}
-
-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 = kRtpDataMaxBandwidth;
-  }
-  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.MutableData(), 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;
-}
-
-}  // namespace cricket

media/base/rtp_data_engine.h

@@ -1,111 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef MEDIA_BASE_RTP_DATA_ENGINE_H_
-#define MEDIA_BASE_RTP_DATA_ENGINE_H_
-
-#include <map>
-#include <memory>
-#include <string>
-#include <vector>
-
-#include "media/base/codec.h"
-#include "media/base/media_channel.h"
-#include "media/base/media_constants.h"
-#include "media/base/media_engine.h"
-
-namespace rtc {
-class DataRateLimiter;
-}
-
-namespace cricket {
-
-class RtpDataEngine : public DataEngineInterface {
- public:
-  RtpDataEngine();
-
-  virtual DataMediaChannel* CreateChannel(const MediaConfig& config);
-
-  virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
-
- private:
-  std::vector<DataCodec> data_codecs_;
-};
-
-// Keep track of sequence number and timestamp of an RTP stream.  The
-// sequence number starts with a "random" value and increments.  The
-// timestamp starts with a "random" value and increases monotonically
-// according to the clockrate.
-class RtpClock {
- public:
-  RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset)
-      : clockrate_(clockrate),
-        last_seq_num_(first_seq_num),
-        timestamp_offset_(timestamp_offset) {}
-
-  // Given the current time (in number of seconds which must be
-  // monotonically increasing), Return the next sequence number and
-  // timestamp.
-  void Tick(double now, int* seq_num, uint32_t* timestamp);
-
- private:
-  int clockrate_;
-  uint16_t last_seq_num_;
-  uint32_t timestamp_offset_;
-};
-
-class RtpDataMediaChannel : public DataMediaChannel {
- public:
-  explicit RtpDataMediaChannel(const MediaConfig& config);
-  virtual ~RtpDataMediaChannel();
-
-  virtual bool SetSendParameters(const DataSendParameters& params);
-  virtual bool SetRecvParameters(const DataRecvParameters& params);
-  virtual bool AddSendStream(const StreamParams& sp);
-  virtual bool RemoveSendStream(uint32_t ssrc);
-  virtual bool AddRecvStream(const StreamParams& sp);
-  virtual bool RemoveRecvStream(uint32_t ssrc);
-  virtual void ResetUnsignaledRecvStream();
-  virtual void OnDemuxerCriteriaUpdatePending();
-  virtual void OnDemuxerCriteriaUpdateComplete();
-  virtual bool SetSend(bool send) {
-    sending_ = send;
-    return true;
-  }
-  virtual bool SetReceive(bool receive) {
-    receiving_ = receive;
-    return true;
-  }
-  virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
-                                int64_t packet_time_us);
-  virtual void OnReadyToSend(bool ready) {}
-  virtual bool SendData(const SendDataParams& params,
-                        const rtc::CopyOnWriteBuffer& payload,
-                        SendDataResult* result);
-
- private:
-  void Construct();
-  bool SetMaxSendBandwidth(int bps);
-  bool SetSendCodecs(const std::vector<DataCodec>& codecs);
-  bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
-
-  bool sending_;
-  bool receiving_;
-  std::vector<DataCodec> send_codecs_;
-  std::vector<DataCodec> recv_codecs_;
-  std::vector<StreamParams> send_streams_;
-  std::vector<StreamParams> recv_streams_;
-  std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
-  std::unique_ptr<rtc::DataRateLimiter> send_limiter_;
-};
-
-}  // namespace cricket
-
-#endif  // MEDIA_BASE_RTP_DATA_ENGINE_H_

media/base/rtp_data_engine_unittest.cc

@@ -1,362 +0,0 @@
-/*
- *  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/rtp_data_engine.h"
-
-#include <string.h>
-
-#include <memory>
-#include <string>
-
-#include "media/base/fake_network_interface.h"
-#include "media/base/media_constants.h"
-#include "media/base/rtp_utils.h"
-#include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/fake_clock.h"
-#include "rtc_base/third_party/sigslot/sigslot.h"
-#include "rtc_base/time_utils.h"
-#include "test/gtest.h"
-
-class FakeDataReceiver : public sigslot::has_slots<> {
- public:
-  FakeDataReceiver() : has_received_data_(false) {}
-
-  void OnDataReceived(const cricket::ReceiveDataParams& params,
-                      const char* data,
-                      size_t len) {
-    has_received_data_ = true;
-    last_received_data_ = std::string(data, len);
-    last_received_data_len_ = len;
-    last_received_data_params_ = params;
-  }
-
-  bool has_received_data() const { return has_received_data_; }
-  std::string last_received_data() const { return last_received_data_; }
-  size_t last_received_data_len() const { return last_received_data_len_; }
-  cricket::ReceiveDataParams last_received_data_params() const {
-    return last_received_data_params_;
-  }
-
- private:
-  bool has_received_data_;
-  std::string last_received_data_;
-  size_t last_received_data_len_;
-  cricket::ReceiveDataParams last_received_data_params_;
-};
-
-class RtpDataMediaChannelTest : public ::testing::Test {
- protected:
-  virtual void SetUp() {
-    // Seed needed for each test to satisfy expectations.
-    iface_.reset(new cricket::FakeNetworkInterface());
-    dme_.reset(CreateEngine());
-    receiver_.reset(new FakeDataReceiver());
-  }
-
-  void SetNow(double now) { clock_.SetTime(webrtc::Timestamp::Seconds(now)); }
-
-  cricket::RtpDataEngine* CreateEngine() {
-    cricket::RtpDataEngine* dme = new cricket::RtpDataEngine();
-    return dme;
-  }
-
-  cricket::RtpDataMediaChannel* CreateChannel() {
-    return CreateChannel(dme_.get());
-  }
-
-  cricket::RtpDataMediaChannel* CreateChannel(cricket::RtpDataEngine* dme) {
-    cricket::MediaConfig config;
-    cricket::RtpDataMediaChannel* channel =
-        static_cast<cricket::RtpDataMediaChannel*>(dme->CreateChannel(config));
-    channel->SetInterface(iface_.get());
-    channel->SignalDataReceived.connect(receiver_.get(),
-                                        &FakeDataReceiver::OnDataReceived);
-    return channel;
-  }
-
-  FakeDataReceiver* receiver() { return receiver_.get(); }
-
-  bool HasReceivedData() { return receiver_->has_received_data(); }
-
-  std::string GetReceivedData() { return receiver_->last_received_data(); }
-
-  size_t GetReceivedDataLen() { return receiver_->last_received_data_len(); }
-
-  cricket::ReceiveDataParams GetReceivedDataParams() {
-    return receiver_->last_received_data_params();
-  }
-
-  bool HasSentData(int count) { return (iface_->NumRtpPackets() > count); }
-
-  std::string GetSentData(int index) {
-    // Assume RTP header of length 12
-    std::unique_ptr<const rtc::CopyOnWriteBuffer> packet(
-        iface_->GetRtpPacket(index));
-    if (packet->size() > 12) {
-      return std::string(packet->data<char>() + 12, packet->size() - 12);
-    } else {
-      return "";
-    }
-  }
-
-  cricket::RtpHeader GetSentDataHeader(int index) {
-    std::unique_ptr<const rtc::CopyOnWriteBuffer> packet(
-        iface_->GetRtpPacket(index));
-    cricket::RtpHeader header;
-    GetRtpHeader(packet->data(), packet->size(), &header);
-    return header;
-  }
-
- private:
-  std::unique_ptr<cricket::RtpDataEngine> dme_;
-  rtc::ScopedFakeClock clock_;
-  std::unique_ptr<cricket::FakeNetworkInterface> iface_;
-  std::unique_ptr<FakeDataReceiver> receiver_;
-};
-
-TEST_F(RtpDataMediaChannelTest, SetUnknownCodecs) {
-  std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
-
-  cricket::DataCodec known_codec;
-  known_codec.id = 103;
-  known_codec.name = "google-data";
-  cricket::DataCodec unknown_codec;
-  unknown_codec.id = 104;
-  unknown_codec.name = "unknown-data";
-
-  cricket::DataSendParameters send_parameters_known;
-  send_parameters_known.codecs.push_back(known_codec);
-  cricket::DataRecvParameters recv_parameters_known;
-  recv_parameters_known.codecs.push_back(known_codec);
-
-  cricket::DataSendParameters send_parameters_unknown;
-  send_parameters_unknown.codecs.push_back(unknown_codec);
-  cricket::DataRecvParameters recv_parameters_unknown;
-  recv_parameters_unknown.codecs.push_back(unknown_codec);
-
-  cricket::DataSendParameters send_parameters_mixed;
-  send_parameters_mixed.codecs.push_back(known_codec);
-  send_parameters_mixed.codecs.push_back(unknown_codec);
-  cricket::DataRecvParameters recv_parameters_mixed;
-  recv_parameters_mixed.codecs.push_back(known_codec);
-  recv_parameters_mixed.codecs.push_back(unknown_codec);
-
-  EXPECT_TRUE(dmc->SetSendParameters(send_parameters_known));
-  EXPECT_FALSE(dmc->SetSendParameters(send_parameters_unknown));
-  EXPECT_TRUE(dmc->SetSendParameters(send_parameters_mixed));
-  EXPECT_TRUE(dmc->SetRecvParameters(recv_parameters_known));
-  EXPECT_FALSE(dmc->SetRecvParameters(recv_parameters_unknown));
-  EXPECT_FALSE(dmc->SetRecvParameters(recv_parameters_mixed));
-}
-
-TEST_F(RtpDataMediaChannelTest, AddRemoveSendStream) {
-  std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
-
-  cricket::StreamParams stream1;
-  stream1.add_ssrc(41);
-  EXPECT_TRUE(dmc->AddSendStream(stream1));
-  cricket::StreamParams stream2;
-  stream2.add_ssrc(42);
-  EXPECT_TRUE(dmc->AddSendStream(stream2));
-
-  EXPECT_TRUE(dmc->RemoveSendStream(41));
-  EXPECT_TRUE(dmc->RemoveSendStream(42));
-  EXPECT_FALSE(dmc->RemoveSendStream(43));
-}
-
-TEST_F(RtpDataMediaChannelTest, AddRemoveRecvStream) {
-  std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
-
-  cricket::StreamParams stream1;
-  stream1.add_ssrc(41);
-  EXPECT_TRUE(dmc->AddRecvStream(stream1));
-  cricket::StreamParams stream2;
-  stream2.add_ssrc(42);
-  EXPECT_TRUE(dmc->AddRecvStream(stream2));
-  EXPECT_FALSE(dmc->AddRecvStream(stream2));
-
-  EXPECT_TRUE(dmc->RemoveRecvStream(41));
-  EXPECT_TRUE(dmc->RemoveRecvStream(42));
-}
-
-TEST_F(RtpDataMediaChannelTest, SendData) {
-  std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
-
-  cricket::SendDataParams params;
-  params.ssrc = 42;
-  unsigned char data[] = "food";
-  rtc::CopyOnWriteBuffer payload(data, 4);
-  unsigned char padded_data[] = {
-      0x00, 0x00, 0x00, 0x00, 'f', 'o', 'o', 'd',
-  };
-  cricket::SendDataResult result;
-
-  // Not sending
-  EXPECT_FALSE(dmc->SendData(params, payload, &result));
-  EXPECT_EQ(cricket::SDR_ERROR, result);
-  EXPECT_FALSE(HasSentData(0));
-  ASSERT_TRUE(dmc->SetSend(true));
-
-  // Unknown stream name.
-  EXPECT_FALSE(dmc->SendData(params, payload, &result));
-  EXPECT_EQ(cricket::SDR_ERROR, result);
-  EXPECT_FALSE(HasSentData(0));
-
-  cricket::StreamParams stream;
-  stream.add_ssrc(42);
-  ASSERT_TRUE(dmc->AddSendStream(stream));
-
-  // Unknown codec;
-  EXPECT_FALSE(dmc->SendData(params, payload, &result));
-  EXPECT_EQ(cricket::SDR_ERROR, result);
-  EXPECT_FALSE(HasSentData(0));
-
-  cricket::DataCodec codec;
-  codec.id = 103;
-  codec.name = cricket::kGoogleRtpDataCodecName;
-  cricket::DataSendParameters parameters;
-  parameters.codecs.push_back(codec);
-  ASSERT_TRUE(dmc->SetSendParameters(parameters));
-
-  // Length too large;
-  std::string x10000(10000, 'x');
-  EXPECT_FALSE(dmc->SendData(
-      params, rtc::CopyOnWriteBuffer(x10000.data(), x10000.length()), &result));
-  EXPECT_EQ(cricket::SDR_ERROR, result);
-  EXPECT_FALSE(HasSentData(0));
-
-  // Finally works!
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
-  ASSERT_TRUE(HasSentData(0));
-  EXPECT_EQ(sizeof(padded_data), GetSentData(0).length());
-  EXPECT_EQ(0, memcmp(padded_data, GetSentData(0).data(), sizeof(padded_data)));
-  cricket::RtpHeader header0 = GetSentDataHeader(0);
-  EXPECT_NE(0, header0.seq_num);
-  EXPECT_NE(0U, header0.timestamp);
-  EXPECT_EQ(header0.ssrc, 42U);
-  EXPECT_EQ(header0.payload_type, 103);
-
-  // Should bump timestamp by 180000 because the clock rate is 90khz.
-  SetNow(2);
-
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  ASSERT_TRUE(HasSentData(1));
-  EXPECT_EQ(sizeof(padded_data), GetSentData(1).length());
-  EXPECT_EQ(0, memcmp(padded_data, GetSentData(1).data(), sizeof(padded_data)));
-  cricket::RtpHeader header1 = GetSentDataHeader(1);
-  EXPECT_EQ(header1.ssrc, 42U);
-  EXPECT_EQ(header1.payload_type, 103);
-  EXPECT_EQ(static_cast<uint16_t>(header0.seq_num + 1),
-            static_cast<uint16_t>(header1.seq_num));
-  EXPECT_EQ(header0.timestamp + 180000, header1.timestamp);
-}
-
-TEST_F(RtpDataMediaChannelTest, SendDataRate) {
-  std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
-
-  ASSERT_TRUE(dmc->SetSend(true));
-
-  cricket::DataCodec codec;
-  codec.id = 103;
-  codec.name = cricket::kGoogleRtpDataCodecName;
-  cricket::DataSendParameters parameters;
-  parameters.codecs.push_back(codec);
-  ASSERT_TRUE(dmc->SetSendParameters(parameters));
-
-  cricket::StreamParams stream;
-  stream.add_ssrc(42);
-  ASSERT_TRUE(dmc->AddSendStream(stream));
-
-  cricket::SendDataParams params;
-  params.ssrc = 42;
-  unsigned char data[] = "food";
-  rtc::CopyOnWriteBuffer payload(data, 4);
-  cricket::SendDataResult result;
-
-  // With rtp overhead of 32 bytes, each one of our packets is 36
-  // bytes, or 288 bits.  So, a limit of 872bps will allow 3 packets,
-  // but not four.
-  parameters.max_bandwidth_bps = 872;
-  ASSERT_TRUE(dmc->SetSendParameters(parameters));
-
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  EXPECT_FALSE(dmc->SendData(params, payload, &result));
-  EXPECT_FALSE(dmc->SendData(params, payload, &result));
-
-  SetNow(0.9);
-  EXPECT_FALSE(dmc->SendData(params, payload, &result));
-
-  SetNow(1.1);
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  SetNow(1.9);
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-
-  SetNow(2.2);
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  EXPECT_TRUE(dmc->SendData(params, payload, &result));
-  EXPECT_FALSE(dmc->SendData(params, payload, &result));
-}
-
-TEST_F(RtpDataMediaChannelTest, ReceiveData) {
-  // PT= 103, SN=2, TS=3, SSRC = 4, data = "abcde"
-  unsigned char data[] = {0x80, 0x67, 0x00, 0x02, 0x00, 0x00, 0x00,
-                          0x03, 0x00, 0x00, 0x00, 0x2A, 0x00, 0x00,
-                          0x00, 0x00, 'a',  'b',  'c',  'd',  'e'};
-  rtc::CopyOnWriteBuffer packet(data, sizeof(data));
-
-  std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
-
-  // SetReceived not called.
-  dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
-  EXPECT_FALSE(HasReceivedData());
-
-  dmc->SetReceive(true);
-
-  // Unknown payload id
-  dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
-  EXPECT_FALSE(HasReceivedData());
-
-  cricket::DataCodec codec;
-  codec.id = 103;
-  codec.name = cricket::kGoogleRtpDataCodecName;
-  cricket::DataRecvParameters parameters;
-  parameters.codecs.push_back(codec);
-  ASSERT_TRUE(dmc->SetRecvParameters(parameters));
-
-  // Unknown stream
-  dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
-  EXPECT_FALSE(HasReceivedData());
-
-  cricket::StreamParams stream;
-  stream.add_ssrc(42);
-  ASSERT_TRUE(dmc->AddRecvStream(stream));
-
-  // Finally works!
-  dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
-  EXPECT_TRUE(HasReceivedData());
-  EXPECT_EQ("abcde", GetReceivedData());
-  EXPECT_EQ(5U, GetReceivedDataLen());
-}
-
-TEST_F(RtpDataMediaChannelTest, InvalidRtpPackets) {
-  unsigned char data[] = {0x80, 0x65, 0x00, 0x02};
-  rtc::CopyOnWriteBuffer packet(data, sizeof(data));
-
-  std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
-
-  // Too short
-  dmc->OnPacketReceived(packet, /* packet_time_us */ -1);
-  EXPECT_FALSE(HasReceivedData());
-}

pc/BUILD.gn

@@ -190,8 +190,6 @@ rtc_library("peerconnection") {
     "rtc_stats_collector.h",
     "rtc_stats_traversal.cc",
     "rtc_stats_traversal.h",
-    "rtp_data_channel.cc",
-    "rtp_data_channel.h",
     "sctp_data_channel.cc",
     "sctp_data_channel.h",
     "sdp_offer_answer.cc",  # TODO: Make separate target when not circular

pc/channel.cc

@@ -1281,243 +1281,4 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   return true;
 }
 
-RtpDataChannel::RtpDataChannel(rtc::Thread* worker_thread,
-                               rtc::Thread* network_thread,
-                               rtc::Thread* signaling_thread,
-                               std::unique_ptr<DataMediaChannel> media_channel,
-                               const std::string& content_name,
-                               bool srtp_required,
-                               webrtc::CryptoOptions crypto_options,
-                               UniqueRandomIdGenerator* ssrc_generator)
-    : BaseChannel(worker_thread,
-                  network_thread,
-                  signaling_thread,
-                  std::move(media_channel),
-                  content_name,
-                  srtp_required,
-                  crypto_options,
-                  ssrc_generator) {}
-
-RtpDataChannel::~RtpDataChannel() {
-  TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel");
-  // this can't be done in the base class, since it calls a virtual
-  DisableMedia_w();
-  Deinit();
-}
-
-void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
-  BaseChannel::Init_w(rtp_transport);
-  media_channel()->SignalDataReceived.connect(this,
-                                              &RtpDataChannel::OnDataReceived);
-  media_channel()->SignalReadyToSend.connect(
-      this, &RtpDataChannel::OnDataChannelReadyToSend);
-}
-
-bool RtpDataChannel::SendData(const SendDataParams& params,
-                              const rtc::CopyOnWriteBuffer& payload,
-                              SendDataResult* result) {
-  DataMediaChannel* mc = media_channel();
-  return InvokeOnWorker<bool>(RTC_FROM_HERE, [mc, &params, &payload, result] {
-    return mc->SendData(params, payload, result);
-  });
-}
-
-bool RtpDataChannel::CheckDataChannelTypeFromContent(
-    const MediaContentDescription* content,
-    std::string* error_desc) {
-  if (!content->as_rtp_data()) {
-    if (content->as_sctp()) {
-      SafeSetError("Data channel type mismatch. Expected RTP, got SCTP.",
-                   error_desc);
-    } else {
-      SafeSetError("Data channel is not RTP or SCTP.", error_desc);
-    }
-    return false;
-  }
-  return true;
-}
-
-bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content,
-                                       SdpType type,
-                                       std::string* error_desc) {
-  TRACE_EVENT0("webrtc", "RtpDataChannel::SetLocalContent_w");
-  RTC_DCHECK_RUN_ON(worker_thread());
-  RTC_LOG(LS_INFO) << "Setting local data description for " << ToString();
-
-  RTC_DCHECK(content);
-  if (!content) {
-    SafeSetError("Can't find data content in local description.", error_desc);
-    return false;
-  }
-
-  if (!CheckDataChannelTypeFromContent(content, error_desc)) {
-    return false;
-  }
-  const RtpDataContentDescription* data = content->as_rtp_data();
-
-  RtpHeaderExtensions rtp_header_extensions =
-      GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
-
-  DataRecvParameters recv_params = last_recv_params_;
-  RtpParametersFromMediaDescription(
-      data, rtp_header_extensions,
-      webrtc::RtpTransceiverDirectionHasRecv(data->direction()), &recv_params);
-  if (!media_channel()->SetRecvParameters(recv_params)) {
-    SafeSetError(
-        "Failed to set remote data description recv parameters for m-section "
-        "with mid='" +
-            content_name() + "'.",
-        error_desc);
-    return false;
-  }
-  for (const DataCodec& codec : data->codecs()) {
-    MaybeAddHandledPayloadType(codec.id);
-  }
-  // Need to re-register the sink to update the handled payload.
-  if (!RegisterRtpDemuxerSink_w()) {
-    RTC_LOG(LS_ERROR) << "Failed to set up data demuxing for " << ToString();
-    return false;
-  }
-
-  last_recv_params_ = recv_params;
-
-  // TODO(pthatcher): Move local streams into DataSendParameters, and
-  // only give it to the media channel once we have a remote
-  // description too (without a remote description, we won't be able
-  // to send them anyway).
-  if (!UpdateLocalStreams_w(data->streams(), type, error_desc)) {
-    SafeSetError(
-        "Failed to set local data description streams for m-section with "
-        "mid='" +
-            content_name() + "'.",
-        error_desc);
-    return false;
-  }
-
-  set_local_content_direction(content->direction());
-  UpdateMediaSendRecvState_w();
-  return true;
-}
-
-bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content,
-                                        SdpType type,
-                                        std::string* error_desc) {
-  TRACE_EVENT0("webrtc", "RtpDataChannel::SetRemoteContent_w");
-  RTC_DCHECK_RUN_ON(worker_thread());
-  RTC_LOG(LS_INFO) << "Setting remote data description for " << ToString();
-
-  RTC_DCHECK(content);
-  if (!content) {
-    SafeSetError("Can't find data content in remote description.", error_desc);
-    return false;
-  }
-
-  if (!CheckDataChannelTypeFromContent(content, error_desc)) {
-    return false;
-  }
-
-  const RtpDataContentDescription* data = content->as_rtp_data();
-
-  // If the remote data doesn't have codecs, it must be empty, so ignore it.
-  if (!data->has_codecs()) {
-    return true;
-  }
-
-  RtpHeaderExtensions rtp_header_extensions =
-      GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
-
-  RTC_LOG(LS_INFO) << "Setting remote data description for " << ToString();
-  DataSendParameters send_params = last_send_params_;
-  RtpSendParametersFromMediaDescription<DataCodec>(
-      data, rtp_header_extensions,
-      webrtc::RtpTransceiverDirectionHasRecv(data->direction()), &send_params);
-  if (!media_channel()->SetSendParameters(send_params)) {
-    SafeSetError(
-        "Failed to set remote data description send parameters for m-section "
-        "with mid='" +
-            content_name() + "'.",
-        error_desc);
-    return false;
-  }
-  last_send_params_ = send_params;
-
-  // TODO(pthatcher): Move remote streams into DataRecvParameters,
-  // and only give it to the media channel once we have a local
-  // description too (without a local description, we won't be able to
-  // recv them anyway).
-  if (!UpdateRemoteStreams_w(data->streams(), type, error_desc)) {
-    SafeSetError(
-        "Failed to set remote data description streams for m-section with "
-        "mid='" +
-            content_name() + "'.",
-        error_desc);
-    return false;
-  }
-
-  set_remote_content_direction(content->direction());
-  UpdateMediaSendRecvState_w();
-  return true;
-}
-
-void RtpDataChannel::UpdateMediaSendRecvState_w() {
-  // Render incoming data if we're the active call, and we have the local
-  // content. We receive data on the default channel and multiplexed streams.
-  RTC_DCHECK_RUN_ON(worker_thread());
-  bool recv = IsReadyToReceiveMedia_w();
-  if (!media_channel()->SetReceive(recv)) {
-    RTC_LOG(LS_ERROR) << "Failed to SetReceive on data channel: " << ToString();
-  }
-
-  // Send outgoing data if we're the active call, we have the remote content,
-  // and we have had some form of connectivity.
-  bool send = IsReadyToSendMedia_w();
-  if (!media_channel()->SetSend(send)) {
-    RTC_LOG(LS_ERROR) << "Failed to SetSend on data channel: " << ToString();
-  }
-
-  // Trigger SignalReadyToSendData asynchronously.
-  OnDataChannelReadyToSend(send);
-
-  RTC_LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send
-                   << " for " << ToString();
-}
-
-void RtpDataChannel::OnMessage(rtc::Message* pmsg) {
-  switch (pmsg->message_id) {
-    case MSG_READYTOSENDDATA: {
-      DataChannelReadyToSendMessageData* data =
-          static_cast<DataChannelReadyToSendMessageData*>(pmsg->pdata);
-      ready_to_send_data_ = data->data();
-      SignalReadyToSendData(ready_to_send_data_);
-      delete data;
-      break;
-    }
-    case MSG_DATARECEIVED: {
-      DataReceivedMessageData* data =
-          static_cast<DataReceivedMessageData*>(pmsg->pdata);
-      SignalDataReceived(data->params, data->payload);
-      delete data;
-      break;
-    }
-    default:
-      BaseChannel::OnMessage(pmsg);
-      break;
-  }
-}
-
-void RtpDataChannel::OnDataReceived(const ReceiveDataParams& params,
-                                    const char* data,
-                                    size_t len) {
-  DataReceivedMessageData* msg = new DataReceivedMessageData(params, data, len);
-  signaling_thread()->Post(RTC_FROM_HERE, this, MSG_DATARECEIVED, msg);
-}
-
-void RtpDataChannel::OnDataChannelReadyToSend(bool writable) {
-  // This is usded for congestion control to indicate that the stream is ready
-  // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
-  // that the transport channel is ready.
-  signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
-                           new DataChannelReadyToSendMessageData(writable));
-}
-
 }  // namespace cricket

pc/channel.h

@@ -471,104 +471,6 @@ class VideoChannel : public BaseChannel {
   VideoRecvParameters last_recv_params_;
 };
 
-// RtpDataChannel is a specialization for data.
-class RtpDataChannel : public BaseChannel {
- public:
-  RtpDataChannel(rtc::Thread* worker_thread,
-                 rtc::Thread* network_thread,
-                 rtc::Thread* signaling_thread,
-                 std::unique_ptr<DataMediaChannel> channel,
-                 const std::string& content_name,
-                 bool srtp_required,
-                 webrtc::CryptoOptions crypto_options,
-                 rtc::UniqueRandomIdGenerator* ssrc_generator);
-  ~RtpDataChannel();
-  // TODO(zhihuang): Remove this once the RtpTransport can be shared between
-  // BaseChannels.
-  void Init_w(DtlsTransportInternal* rtp_dtls_transport,
-              DtlsTransportInternal* rtcp_dtls_transport,
-              rtc::PacketTransportInternal* rtp_packet_transport,
-              rtc::PacketTransportInternal* rtcp_packet_transport);
-  void Init_w(webrtc::RtpTransportInternal* rtp_transport) override;
-
-  virtual bool SendData(const SendDataParams& params,
-                        const rtc::CopyOnWriteBuffer& payload,
-                        SendDataResult* result);
-
-  // Should be called on the signaling thread only.
-  bool ready_to_send_data() const { return ready_to_send_data_; }
-
-  sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&>
-      SignalDataReceived;
-  // Signal for notifying when the channel becomes ready to send data.
-  // That occurs when the channel is enabled, the transport is writable,
-  // both local and remote descriptions are set, and the channel is unblocked.
-  sigslot::signal1<bool> SignalReadyToSendData;
-  cricket::MediaType media_type() const override {
-    return cricket::MEDIA_TYPE_DATA;
-  }
-
- protected:
-  // downcasts a MediaChannel.
-  DataMediaChannel* media_channel() const override {
-    return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
-  }
-
- private:
-  struct SendDataMessageData : public rtc::MessageData {
-    SendDataMessageData(const SendDataParams& params,
-                        const rtc::CopyOnWriteBuffer* payload,
-                        SendDataResult* result)
-        : params(params), payload(payload), result(result), succeeded(false) {}
-
-    const SendDataParams& params;
-    const rtc::CopyOnWriteBuffer* payload;
-    SendDataResult* result;
-    bool succeeded;
-  };
-
-  struct DataReceivedMessageData : public rtc::MessageData {
-    // We copy the data because the data will become invalid after we
-    // handle DataMediaChannel::SignalDataReceived but before we fire
-    // SignalDataReceived.
-    DataReceivedMessageData(const ReceiveDataParams& params,
-                            const char* data,
-                            size_t len)
-        : params(params), payload(data, len) {}
-    const ReceiveDataParams params;
-    const rtc::CopyOnWriteBuffer payload;
-  };
-
-  typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData;
-
-  // overrides from BaseChannel
-  // Checks that data channel type is RTP.
-  bool CheckDataChannelTypeFromContent(const MediaContentDescription* content,
-                                       std::string* error_desc);
-  bool SetLocalContent_w(const MediaContentDescription* content,
-                         webrtc::SdpType type,
-                         std::string* error_desc) override;
-  bool SetRemoteContent_w(const MediaContentDescription* content,
-                          webrtc::SdpType type,
-                          std::string* error_desc) override;
-  void UpdateMediaSendRecvState_w() override;
-
-  void OnMessage(rtc::Message* pmsg) override;
-  void OnDataReceived(const ReceiveDataParams& params,
-                      const char* data,
-                      size_t len);
-  void OnDataChannelReadyToSend(bool writable);
-
-  bool ready_to_send_data_ = false;
-
-  // Last DataSendParameters sent down to the media_channel() via
-  // SetSendParameters.
-  DataSendParameters last_send_params_;
-  // Last DataRecvParameters sent down to the media_channel() via
-  // SetRecvParameters.
-  DataRecvParameters last_recv_params_;
-};
-
 }  // namespace cricket
 
 #endif  // PC_CHANNEL_H_

pc/channel_manager.cc

@@ -28,35 +28,29 @@ namespace cricket {
 // static
 std::unique_ptr<ChannelManager> ChannelManager::Create(
     std::unique_ptr<MediaEngineInterface> media_engine,
-    std::unique_ptr<DataEngineInterface> data_engine,
     bool enable_rtx,
     rtc::Thread* worker_thread,
     rtc::Thread* network_thread) {
   RTC_DCHECK_RUN_ON(worker_thread);
   RTC_DCHECK(network_thread);
   RTC_DCHECK(worker_thread);
-  RTC_DCHECK(data_engine);
 
   if (media_engine)
     media_engine->Init();
 
-  return absl::WrapUnique(new ChannelManager(std::move(media_engine),
+  return absl::WrapUnique(new ChannelManager(
-                                             std::move(data_engine), enable_rtx,
+      std::move(media_engine), enable_rtx, worker_thread, network_thread));
-                                             worker_thread, network_thread));
 }
 
 ChannelManager::ChannelManager(
     std::unique_ptr<MediaEngineInterface> media_engine,
-    std::unique_ptr<DataEngineInterface> data_engine,
     bool enable_rtx,
     rtc::Thread* worker_thread,
     rtc::Thread* network_thread)
     : media_engine_(std::move(media_engine)),
-      data_engine_(std::move(data_engine)),
       worker_thread_(worker_thread),
       network_thread_(network_thread),
       enable_rtx_(enable_rtx) {
-  RTC_DCHECK(data_engine_);
   RTC_DCHECK(worker_thread_);
   RTC_DCHECK(network_thread_);
   RTC_DCHECK_RUN_ON(worker_thread_);
@@ -116,11 +110,6 @@ void ChannelManager::GetSupportedVideoReceiveCodecs(
   }
 }
 
-void ChannelManager::GetSupportedDataCodecs(
-    std::vector<DataCodec>* codecs) const {
-  *codecs = data_engine_->data_codecs();
-}
-
 RtpHeaderExtensions ChannelManager::GetDefaultEnabledAudioRtpHeaderExtensions()
     const {
   if (!media_engine_)
@@ -273,61 +262,6 @@ void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
       }));
 }
 
-RtpDataChannel* ChannelManager::CreateRtpDataChannel(
-    const MediaConfig& media_config,
-    webrtc::RtpTransportInternal* rtp_transport,
-    rtc::Thread* signaling_thread,
-    const std::string& content_name,
-    bool srtp_required,
-    const webrtc::CryptoOptions& crypto_options,
-    rtc::UniqueRandomIdGenerator* ssrc_generator) {
-  if (!worker_thread_->IsCurrent()) {
-    return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
-      return CreateRtpDataChannel(media_config, rtp_transport, signaling_thread,
-                                  content_name, srtp_required, crypto_options,
-                                  ssrc_generator);
-    });
-  }
-
-  RTC_DCHECK_RUN_ON(worker_thread_);
-
-  // This is ok to alloc from a thread other than the worker thread.
-  DataMediaChannel* media_channel = data_engine_->CreateChannel(media_config);
-  if (!media_channel) {
-    RTC_LOG(LS_WARNING) << "Failed to create RTP data channel.";
-    return nullptr;
-  }
-
-  auto data_channel = std::make_unique<RtpDataChannel>(
-      worker_thread_, network_thread_, signaling_thread,
-      absl::WrapUnique(media_channel), content_name, srtp_required,
-      crypto_options, ssrc_generator);
-
-  // Media Transports are not supported with Rtp Data Channel.
-  data_channel->Init_w(rtp_transport);
-
-  RtpDataChannel* data_channel_ptr = data_channel.get();
-  data_channels_.push_back(std::move(data_channel));
-  return data_channel_ptr;
-}
-
-void ChannelManager::DestroyRtpDataChannel(RtpDataChannel* data_channel) {
-  TRACE_EVENT0("webrtc", "ChannelManager::DestroyRtpDataChannel");
-  RTC_DCHECK(data_channel);
-
-  if (!worker_thread_->IsCurrent()) {
-    worker_thread_->Invoke<void>(
-        RTC_FROM_HERE, [&] { return DestroyRtpDataChannel(data_channel); });
-    return;
-  }
-  RTC_DCHECK_RUN_ON(worker_thread_);
-
-  data_channels_.erase(absl::c_find_if(
-      data_channels_, [&](const std::unique_ptr<RtpDataChannel>& p) {
-        return p.get() == data_channel;
-      }));
-}
-
 bool ChannelManager::StartAecDump(webrtc::FileWrapper file,
                                   int64_t max_size_bytes) {
   RTC_DCHECK_RUN_ON(worker_thread_);

pc/channel_manager.h

@@ -50,7 +50,6 @@ class ChannelManager final {
   // will own that reference and media engine initialization
   static std::unique_ptr<ChannelManager> Create(
       std::unique_ptr<MediaEngineInterface> media_engine,
-      std::unique_ptr<DataEngineInterface> data_engine,
       bool enable_rtx,
       rtc::Thread* worker_thread,
       rtc::Thread* network_thread);
@@ -110,17 +109,6 @@ class ChannelManager final {
   // Destroys a video channel created by CreateVideoChannel.
   void DestroyVideoChannel(VideoChannel* video_channel);
 
-  RtpDataChannel* CreateRtpDataChannel(
-      const MediaConfig& media_config,
-      webrtc::RtpTransportInternal* rtp_transport,
-      rtc::Thread* signaling_thread,
-      const std::string& content_name,
-      bool srtp_required,
-      const webrtc::CryptoOptions& crypto_options,
-      rtc::UniqueRandomIdGenerator* ssrc_generator);
-  // Destroys a data channel created by CreateRtpDataChannel.
-  void DestroyRtpDataChannel(RtpDataChannel* data_channel);
-
   // Starts AEC dump using existing file, with a specified maximum file size in
   // bytes. When the limit is reached, logging will stop and the file will be
   // closed. If max_size_bytes is set to <= 0, no limit will be used.
@@ -131,13 +119,11 @@ class ChannelManager final {
 
  private:
   ChannelManager(std::unique_ptr<MediaEngineInterface> media_engine,
-                 std::unique_ptr<DataEngineInterface> data_engine,
                  bool enable_rtx,
                  rtc::Thread* worker_thread,
                  rtc::Thread* network_thread);
 
   const std::unique_ptr<MediaEngineInterface> media_engine_;  // Nullable.
-  const std::unique_ptr<DataEngineInterface> data_engine_;    // Non-null.
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
 
@@ -146,8 +132,6 @@ class ChannelManager final {
       RTC_GUARDED_BY(worker_thread_);
   std::vector<std::unique_ptr<VideoChannel>> video_channels_
       RTC_GUARDED_BY(worker_thread_);
-  std::vector<std::unique_ptr<RtpDataChannel>> data_channels_
-      RTC_GUARDED_BY(worker_thread_);
 
   const bool enable_rtx_;
 };

pc/channel_manager_unittest.cc

@@ -58,7 +58,6 @@ class ChannelManagerTest : public ::testing::Test {
         video_bitrate_allocator_factory_(
             webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
         cm_(cricket::ChannelManager::Create(CreateFakeMediaEngine(),
-                                            std::make_unique<FakeDataEngine>(),
                                             false,
                                             worker_,
                                             network_.get())),
@@ -80,14 +79,8 @@ class ChannelManagerTest : public ::testing::Test {
         webrtc::CryptoOptions(), &ssrc_generator_, VideoOptions(),
         video_bitrate_allocator_factory_.get());
     EXPECT_TRUE(video_channel != nullptr);
-    cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel(
-        cricket::MediaConfig(), rtp_transport, rtc::Thread::Current(),
-        cricket::CN_DATA, kDefaultSrtpRequired, webrtc::CryptoOptions(),
-        &ssrc_generator_);
-    EXPECT_TRUE(rtp_data_channel != nullptr);
     cm_->DestroyVideoChannel(video_channel);
     cm_->DestroyVoiceChannel(voice_channel);
-    cm_->DestroyRtpDataChannel(rtp_data_channel);
   }
 
   std::unique_ptr<rtc::Thread> network_;
@@ -112,7 +105,6 @@ TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
 
   // Enable and check.
   cm_ = cricket::ChannelManager::Create(CreateFakeMediaEngine(),
-                                        std::make_unique<FakeDataEngine>(),
                                         true, worker_, network_.get());
   cm_->GetSupportedVideoSendCodecs(&send_codecs);
   EXPECT_TRUE(ContainsMatchingCodec(send_codecs, rtx_codec));
@@ -121,7 +113,6 @@ TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
 
   // Disable and check.
   cm_ = cricket::ChannelManager::Create(CreateFakeMediaEngine(),
-                                        std::make_unique<FakeDataEngine>(),
                                         false, worker_, network_.get());
   cm_->GetSupportedVideoSendCodecs(&send_codecs);
   EXPECT_FALSE(ContainsMatchingCodec(send_codecs, rtx_codec));

pc/channel_unittest.cc

@@ -93,13 +93,6 @@ class VideoTraits : public Traits<cricket::VideoChannel,
                                   cricket::VideoMediaInfo,
                                   cricket::VideoOptions> {};
 
-class DataTraits : public Traits<cricket::RtpDataChannel,
-                                 cricket::FakeDataMediaChannel,
-                                 cricket::RtpDataContentDescription,
-                                 cricket::RtpDataCodec,
-                                 cricket::DataMediaInfo,
-                                 cricket::DataOptions> {};
-
 // Base class for Voice/Video/RtpDataChannel tests
 template <class T>
 class ChannelTest : public ::testing::Test, public sigslot::has_slots<> {
@@ -2274,220 +2267,5 @@ TEST_F(VideoChannelDoubleThreadTest, SocketOptionsMergedOnSetTransport) {
   Base::SocketOptionsMergedOnSetTransport();
 }
 
-// RtpDataChannelSingleThreadTest
-class RtpDataChannelSingleThreadTest : public ChannelTest<DataTraits> {
- public:
-  typedef ChannelTest<DataTraits> Base;
-  RtpDataChannelSingleThreadTest()
-      : Base(true, kDataPacket, kRtcpReport, NetworkIsWorker::Yes) {}
-};
-
-// RtpDataChannelDoubleThreadTest
-class RtpDataChannelDoubleThreadTest : public ChannelTest<DataTraits> {
- public:
-  typedef ChannelTest<DataTraits> Base;
-  RtpDataChannelDoubleThreadTest()
-      : Base(true, kDataPacket, kRtcpReport, NetworkIsWorker::No) {}
-};
-
-// Override to avoid engine channel parameter.
-template <>
-std::unique_ptr<cricket::RtpDataChannel> ChannelTest<DataTraits>::CreateChannel(
-    rtc::Thread* worker_thread,
-    rtc::Thread* network_thread,
-    std::unique_ptr<cricket::FakeDataMediaChannel> ch,
-    webrtc::RtpTransportInternal* rtp_transport,
-    int flags) {
-  rtc::Thread* signaling_thread = rtc::Thread::Current();
-  auto channel = std::make_unique<cricket::RtpDataChannel>(
-      worker_thread, network_thread, signaling_thread, std::move(ch),
-      cricket::CN_DATA, (flags & DTLS) != 0, webrtc::CryptoOptions(),
-      &ssrc_generator_);
-  channel->Init_w(rtp_transport);
-  return channel;
-}
-
-template <>
-void ChannelTest<DataTraits>::CreateContent(
-    int flags,
-    const cricket::AudioCodec& audio_codec,
-    const cricket::VideoCodec& video_codec,
-    cricket::RtpDataContentDescription* data) {
-  data->AddCodec(kGoogleDataCodec);
-  data->set_rtcp_mux((flags & RTCP_MUX) != 0);
-}
-
-template <>
-void ChannelTest<DataTraits>::CopyContent(
-    const cricket::RtpDataContentDescription& source,
-    cricket::RtpDataContentDescription* data) {
-  *data = source;
-}
-
-template <>
-bool ChannelTest<DataTraits>::CodecMatches(const cricket::DataCodec& c1,
-                                           const cricket::DataCodec& c2) {
-  return c1.name == c2.name;
-}
-
-template <>
-void ChannelTest<DataTraits>::AddLegacyStreamInContent(
-    uint32_t ssrc,
-    int flags,
-    cricket::RtpDataContentDescription* data) {
-  data->AddLegacyStream(ssrc);
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestInit) {
-  Base::TestInit();
-  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestDeinit) {
-  Base::TestDeinit();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestSetContents) {
-  Base::TestSetContents();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestSetContentsNullOffer) {
-  Base::TestSetContentsNullOffer();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestSetContentsRtcpMux) {
-  Base::TestSetContentsRtcpMux();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestChangeStreamParamsInContent) {
-  Base::TestChangeStreamParamsInContent();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestPlayoutAndSendingStates) {
-  Base::TestPlayoutAndSendingStates();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestMediaContentDirection) {
-  Base::TestMediaContentDirection();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestCallSetup) {
-  Base::TestCallSetup();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestCallTeardownRtcpMux) {
-  Base::TestCallTeardownRtcpMux();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestOnTransportReadyToSend) {
-  Base::TestOnTransportReadyToSend();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, SendRtpToRtp) {
-  Base::SendRtpToRtp();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, SendRtpToRtpOnThread) {
-  Base::SendRtpToRtpOnThread();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, SendWithWritabilityLoss) {
-  Base::SendWithWritabilityLoss();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, SocketOptionsMergedOnSetTransport) {
-  Base::SocketOptionsMergedOnSetTransport();
-}
-
-TEST_F(RtpDataChannelSingleThreadTest, TestSendData) {
-  CreateChannels(0, 0);
-  EXPECT_TRUE(SendInitiate());
-  EXPECT_TRUE(SendAccept());
-
-  cricket::SendDataParams params;
-  params.ssrc = 42;
-  unsigned char data[] = {'f', 'o', 'o'};
-  rtc::CopyOnWriteBuffer payload(data, 3);
-  cricket::SendDataResult result;
-  ASSERT_TRUE(media_channel1_->SendData(params, payload, &result));
-  EXPECT_EQ(params.ssrc, media_channel1_->last_sent_data_params().ssrc);
-  EXPECT_EQ("foo", media_channel1_->last_sent_data());
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestInit) {
-  Base::TestInit();
-  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestDeinit) {
-  Base::TestDeinit();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestSetContents) {
-  Base::TestSetContents();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestSetContentsNullOffer) {
-  Base::TestSetContentsNullOffer();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestSetContentsRtcpMux) {
-  Base::TestSetContentsRtcpMux();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestChangeStreamParamsInContent) {
-  Base::TestChangeStreamParamsInContent();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
-  Base::TestPlayoutAndSendingStates();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestMediaContentDirection) {
-  Base::TestMediaContentDirection();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestCallSetup) {
-  Base::TestCallSetup();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestCallTeardownRtcpMux) {
-  Base::TestCallTeardownRtcpMux();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestOnTransportReadyToSend) {
-  Base::TestOnTransportReadyToSend();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, SendRtpToRtp) {
-  Base::SendRtpToRtp();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, SendRtpToRtpOnThread) {
-  Base::SendRtpToRtpOnThread();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, SendWithWritabilityLoss) {
-  Base::SendWithWritabilityLoss();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, SocketOptionsMergedOnSetTransport) {
-  Base::SocketOptionsMergedOnSetTransport();
-}
-
-TEST_F(RtpDataChannelDoubleThreadTest, TestSendData) {
-  CreateChannels(0, 0);
-  EXPECT_TRUE(SendInitiate());
-  EXPECT_TRUE(SendAccept());
-
-  cricket::SendDataParams params;
-  params.ssrc = 42;
-  unsigned char data[] = {'f', 'o', 'o'};
-  rtc::CopyOnWriteBuffer payload(data, 3);
-  cricket::SendDataResult result;
-  ASSERT_TRUE(media_channel1_->SendData(params, payload, &result));
-  EXPECT_EQ(params.ssrc, media_channel1_->last_sent_data_params().ssrc);
-  EXPECT_EQ("foo", media_channel1_->last_sent_data());
-}
 
 // TODO(pthatcher): TestSetReceiver?

pc/connection_context.cc

@@ -15,7 +15,6 @@
 #include <utility>
 
 #include "api/transport/field_trial_based_config.h"
-#include "media/base/rtp_data_engine.h"
 #include "media/sctp/sctp_transport_factory.h"
 #include "rtc_base/helpers.h"
 #include "rtc_base/ref_counted_object.h"
@@ -127,8 +126,7 @@ ConnectionContext::ConnectionContext(
   worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
     channel_manager_ = cricket::ChannelManager::Create(
         std::move(dependencies->media_engine),
-        std::make_unique<cricket::RtpDataEngine>(), /*enable_rtx=*/true,
+        /*enable_rtx=*/true, worker_thread(), network_thread());
-        worker_thread(), network_thread());
   });
 
   // Set warning levels on the threads, to give warnings when response

pc/data_channel_controller.cc

@@ -28,7 +28,7 @@ namespace webrtc {
 
 bool DataChannelController::HasDataChannels() const {
   RTC_DCHECK_RUN_ON(signaling_thread());
-  return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
+  return !sctp_data_channels_.empty();
 }
 
 bool DataChannelController::SendData(const cricket::SendDataParams& params,
@@ -36,40 +36,10 @@ bool DataChannelController::SendData(const cricket::SendDataParams& params,
                                      cricket::SendDataResult* result) {
   if (data_channel_transport())
     return DataChannelSendData(params, payload, result);
-  if (rtp_data_channel())
-    return rtp_data_channel()->SendData(params, payload, result);
   RTC_LOG(LS_ERROR) << "SendData called before transport is ready";
   return false;
 }
 
-bool DataChannelController::ConnectDataChannel(
-    RtpDataChannel* webrtc_data_channel) {
-  RTC_DCHECK_RUN_ON(signaling_thread());
-  if (!rtp_data_channel()) {
-    // Don't log an error here, because DataChannels are expected to call
-    // ConnectDataChannel in this state. It's the only way to initially tell
-    // whether or not the underlying transport is ready.
-    return false;
-  }
-  rtp_data_channel()->SignalReadyToSendData.connect(
-      webrtc_data_channel, &RtpDataChannel::OnChannelReady);
-  rtp_data_channel()->SignalDataReceived.connect(
-      webrtc_data_channel, &RtpDataChannel::OnDataReceived);
-  return true;
-}
-
-void DataChannelController::DisconnectDataChannel(
-    RtpDataChannel* webrtc_data_channel) {
-  RTC_DCHECK_RUN_ON(signaling_thread());
-  if (!rtp_data_channel()) {
-    RTC_LOG(LS_ERROR)
-        << "DisconnectDataChannel called when rtp_data_channel_ is NULL.";
-    return;
-  }
-  rtp_data_channel()->SignalReadyToSendData.disconnect(webrtc_data_channel);
-  rtp_data_channel()->SignalDataReceived.disconnect(webrtc_data_channel);
-}
-
 bool DataChannelController::ConnectDataChannel(
     SctpDataChannel* webrtc_data_channel) {
   RTC_DCHECK_RUN_ON(signaling_thread());
@@ -126,8 +96,7 @@ void DataChannelController::RemoveSctpDataStream(int sid) {
 
 bool DataChannelController::ReadyToSendData() const {
   RTC_DCHECK_RUN_ON(signaling_thread());
-  return (rtp_data_channel() && rtp_data_channel()->ready_to_send_data()) ||
+  return (data_channel_transport() && data_channel_transport_ready_to_send_);
-         (data_channel_transport() && data_channel_transport_ready_to_send_);
 }
 
 void DataChannelController::OnDataReceived(
@@ -258,10 +227,10 @@ bool DataChannelController::HandleOpenMessage_s(
     // be created.
     std::string label;
     InternalDataChannelInit config;
-    config.id = params.ssrc;
+    config.id = params.sid;
     if (!ParseDataChannelOpenMessage(buffer, &label, &config)) {
-      RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for ssrc "
+      RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
-                          << params.ssrc;
+                          << params.sid;
       return true;
     }
     config.open_handshake_role = InternalDataChannelInit::kAcker;
@@ -304,38 +273,11 @@ DataChannelController::InternalCreateDataChannelWithProxy(
     if (channel) {
       return SctpDataChannel::CreateProxy(channel);
     }
-  } else if (data_channel_type() == cricket::DCT_RTP) {
-    rtc::scoped_refptr<RtpDataChannel> channel =
-        InternalCreateRtpDataChannel(label, config);
-    if (channel) {
-      return RtpDataChannel::CreateProxy(channel);
-    }
   }
 
   return nullptr;
 }
 
-rtc::scoped_refptr<RtpDataChannel>
-DataChannelController::InternalCreateRtpDataChannel(
-    const std::string& label,
-    const DataChannelInit* config) {
-  RTC_DCHECK_RUN_ON(signaling_thread());
-  DataChannelInit new_config = config ? (*config) : DataChannelInit();
-  rtc::scoped_refptr<RtpDataChannel> channel(
-      RtpDataChannel::Create(this, label, new_config, signaling_thread()));
-  if (!channel) {
-    return nullptr;
-  }
-  if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
-    RTC_LOG(LS_ERROR) << "DataChannel with label " << channel->label()
-                      << " already exists.";
-    return nullptr;
-  }
-  rtp_data_channels_[channel->label()] = channel;
-  SignalRtpDataChannelCreated_(channel.get());
-  return channel;
-}
-
 rtc::scoped_refptr<SctpDataChannel>
 DataChannelController::InternalCreateSctpDataChannel(
     const std::string& label,
@@ -416,14 +358,8 @@ void DataChannelController::OnSctpDataChannelClosed(SctpDataChannel* channel) {
 
 void DataChannelController::OnTransportChannelClosed() {
   RTC_DCHECK_RUN_ON(signaling_thread());
-  // Use a temporary copy of the RTP/SCTP DataChannel list because the
+  // Use a temporary copy of the SCTP DataChannel list because the
   // DataChannel may callback to us and try to modify the list.
-  std::map<std::string, rtc::scoped_refptr<RtpDataChannel>> temp_rtp_dcs;
-  temp_rtp_dcs.swap(rtp_data_channels_);
-  for (const auto& kv : temp_rtp_dcs) {
-    kv.second->OnTransportChannelClosed();
-  }
-
   std::vector<rtc::scoped_refptr<SctpDataChannel>> temp_sctp_dcs;
   temp_sctp_dcs.swap(sctp_data_channels_);
   for (const auto& channel : temp_sctp_dcs) {
@@ -441,58 +377,6 @@ SctpDataChannel* DataChannelController::FindDataChannelBySid(int sid) const {
   return nullptr;
 }
 
-void DataChannelController::UpdateLocalRtpDataChannels(
-    const cricket::StreamParamsVec& streams) {
-  std::vector<std::string> existing_channels;
-
-  RTC_DCHECK_RUN_ON(signaling_thread());
-  // Find new and active data channels.
-  for (const cricket::StreamParams& params : streams) {
-    // |it->sync_label| is actually the data channel label. The reason is that
-    // we use the same naming of data channels as we do for
-    // MediaStreams and Tracks.
-    // For MediaStreams, the sync_label is the MediaStream label and the
-    // track label is the same as |streamid|.
-    const std::string& channel_label = params.first_stream_id();
-    auto data_channel_it = rtp_data_channels()->find(channel_label);
-    if (data_channel_it == rtp_data_channels()->end()) {
-      RTC_LOG(LS_ERROR) << "channel label not found";
-      continue;
-    }
-    // Set the SSRC the data channel should use for sending.
-    data_channel_it->second->SetSendSsrc(params.first_ssrc());
-    existing_channels.push_back(data_channel_it->first);
-  }
-
-  UpdateClosingRtpDataChannels(existing_channels, true);
-}
-
-void DataChannelController::UpdateRemoteRtpDataChannels(
-    const cricket::StreamParamsVec& streams) {
-  RTC_DCHECK_RUN_ON(signaling_thread());
-
-  std::vector<std::string> existing_channels;
-
-  // Find new and active data channels.
-  for (const cricket::StreamParams& params : streams) {
-    // The data channel label is either the mslabel or the SSRC if the mslabel
-    // does not exist. Ex a=ssrc:444330170 mslabel:test1.
-    std::string label = params.first_stream_id().empty()
-                            ? rtc::ToString(params.first_ssrc())
-                            : params.first_stream_id();
-    auto data_channel_it = rtp_data_channels()->find(label);
-    if (data_channel_it == rtp_data_channels()->end()) {
-      // This is a new data channel.
-      CreateRemoteRtpDataChannel(label, params.first_ssrc());
-    } else {
-      data_channel_it->second->SetReceiveSsrc(params.first_ssrc());
-    }
-    existing_channels.push_back(label);
-  }
-
-  UpdateClosingRtpDataChannels(existing_channels, false);
-}
-
 cricket::DataChannelType DataChannelController::data_channel_type() const {
   // TODO(bugs.webrtc.org/9987): Should be restricted to the signaling thread.
   // RTC_DCHECK_RUN_ON(signaling_thread());
@@ -505,19 +389,6 @@ void DataChannelController::set_data_channel_type(
   data_channel_type_ = type;
 }
 
-cricket::RtpDataChannel* DataChannelController::rtp_data_channel() const {
-  // TODO(bugs.webrtc.org/9987): Only allow this accessor to be called on the
-  // network thread.
-  // RTC_DCHECK_RUN_ON(network_thread());
-  return rtp_data_channel_;
-}
-
-void DataChannelController::set_rtp_data_channel(
-    cricket::RtpDataChannel* channel) {
-  RTC_DCHECK_RUN_ON(network_thread());
-  rtp_data_channel_ = channel;
-}
-
 DataChannelTransportInterface* DataChannelController::data_channel_transport()
     const {
   // TODO(bugs.webrtc.org/11547): Only allow this accessor to be called on the
@@ -532,56 +403,6 @@ void DataChannelController::set_data_channel_transport(
   data_channel_transport_ = transport;
 }
 
-const std::map<std::string, rtc::scoped_refptr<RtpDataChannel>>*
-DataChannelController::rtp_data_channels() const {
-  RTC_DCHECK_RUN_ON(signaling_thread());
-  return &rtp_data_channels_;
-}
-
-void DataChannelController::UpdateClosingRtpDataChannels(
-    const std::vector<std::string>& active_channels,
-    bool is_local_update) {
-  auto it = rtp_data_channels_.begin();
-  while (it != rtp_data_channels_.end()) {
-    RtpDataChannel* data_channel = it->second;
-    if (absl::c_linear_search(active_channels, data_channel->label())) {
-      ++it;
-      continue;
-    }
-
-    if (is_local_update) {
-      data_channel->SetSendSsrc(0);
-    } else {
-      data_channel->RemotePeerRequestClose();
-    }
-
-    if (data_channel->state() == RtpDataChannel::kClosed) {
-      rtp_data_channels_.erase(it);
-      it = rtp_data_channels_.begin();
-    } else {
-      ++it;
-    }
-  }
-}
-
-void DataChannelController::CreateRemoteRtpDataChannel(const std::string& label,
-                                                       uint32_t remote_ssrc) {
-  if (data_channel_type() != cricket::DCT_RTP) {
-    return;
-  }
-  rtc::scoped_refptr<RtpDataChannel> channel(
-      InternalCreateRtpDataChannel(label, nullptr));
-  if (!channel.get()) {
-    RTC_LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
-                           "CreateDataChannel failed.";
-    return;
-  }
-  channel->SetReceiveSsrc(remote_ssrc);
-  rtc::scoped_refptr<DataChannelInterface> proxy_channel =
-      RtpDataChannel::CreateProxy(std::move(channel));
-  pc_->Observer()->OnDataChannel(std::move(proxy_channel));
-}
-
 bool DataChannelController::DataChannelSendData(
     const cricket::SendDataParams& params,
     const rtc::CopyOnWriteBuffer& payload,

pc/data_channel_controller.h

@@ -27,7 +27,6 @@
 #include "media/base/stream_params.h"
 #include "pc/channel.h"
 #include "pc/data_channel_utils.h"
-#include "pc/rtp_data_channel.h"
 #include "pc/sctp_data_channel.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
@@ -41,8 +40,7 @@ namespace webrtc {
 
 class PeerConnection;
 
-class DataChannelController : public RtpDataChannelProviderInterface,
+class DataChannelController : public SctpDataChannelProviderInterface,
-                              public SctpDataChannelProviderInterface,
                               public DataChannelSink {
  public:
   explicit DataChannelController(PeerConnection* pc) : pc_(pc) {}
@@ -53,13 +51,11 @@ class DataChannelController : public RtpDataChannelProviderInterface,
   DataChannelController(DataChannelController&&) = delete;
   DataChannelController& operator=(DataChannelController&& other) = delete;
 
-  // Implements RtpDataChannelProviderInterface/
+  // Implements
   // SctpDataChannelProviderInterface.
   bool SendData(const cricket::SendDataParams& params,
                 const rtc::CopyOnWriteBuffer& payload,
                 cricket::SendDataResult* result) override;
-  bool ConnectDataChannel(RtpDataChannel* webrtc_data_channel) override;
-  void DisconnectDataChannel(RtpDataChannel* webrtc_data_channel) override;
   bool ConnectDataChannel(SctpDataChannel* webrtc_data_channel) override;
   void DisconnectDataChannel(SctpDataChannel* webrtc_data_channel) override;
   void AddSctpDataStream(int sid) override;
@@ -104,28 +100,13 @@ class DataChannelController : public RtpDataChannelProviderInterface,
     RTC_DCHECK_RUN_ON(signaling_thread());
     return !sctp_data_channels_.empty();
   }
-  bool HasRtpDataChannels() const {
-    RTC_DCHECK_RUN_ON(signaling_thread());
-    return !rtp_data_channels_.empty();
-  }
-
-  void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams);
-  void UpdateRemoteRtpDataChannels(const cricket::StreamParamsVec& streams);
 
   // Accessors
   cricket::DataChannelType data_channel_type() const;
   void set_data_channel_type(cricket::DataChannelType type);
-  cricket::RtpDataChannel* rtp_data_channel() const;
-  void set_rtp_data_channel(cricket::RtpDataChannel* channel);
   DataChannelTransportInterface* data_channel_transport() const;
   void set_data_channel_transport(DataChannelTransportInterface* transport);
-  const std::map<std::string, rtc::scoped_refptr<RtpDataChannel>>*
-  rtp_data_channels() const;
 
-  sigslot::signal1<RtpDataChannel*>& SignalRtpDataChannelCreated() {
-    RTC_DCHECK_RUN_ON(signaling_thread());
-    return SignalRtpDataChannelCreated_;
-  }
   sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() {
     RTC_DCHECK_RUN_ON(signaling_thread());
     return SignalSctpDataChannelCreated_;
@@ -136,10 +117,6 @@ class DataChannelController : public RtpDataChannelProviderInterface,
   void OnSctpDataChannelClosed(SctpDataChannel* channel);
 
  private:
-  rtc::scoped_refptr<RtpDataChannel> InternalCreateRtpDataChannel(
-      const std::string& label,
-      const DataChannelInit* config) /* RTC_RUN_ON(signaling_thread()) */;
-
   rtc::scoped_refptr<SctpDataChannel> InternalCreateSctpDataChannel(
       const std::string& label,
       const InternalDataChannelInit*
@@ -155,14 +132,6 @@ class DataChannelController : public RtpDataChannelProviderInterface,
                                 const InternalDataChannelInit& config)
       RTC_RUN_ON(signaling_thread());
 
-  void CreateRemoteRtpDataChannel(const std::string& label,
-                                  uint32_t remote_ssrc)
-      RTC_RUN_ON(signaling_thread());
-
-  void UpdateClosingRtpDataChannels(
-      const std::vector<std::string>& active_channels,
-      bool is_local_update) RTC_RUN_ON(signaling_thread());
-
   // Called from SendData when data_channel_transport() is true.
   bool DataChannelSendData(const cricket::SendDataParams& params,
                            const rtc::CopyOnWriteBuffer& payload,
@@ -175,13 +144,7 @@ class DataChannelController : public RtpDataChannelProviderInterface,
   rtc::Thread* network_thread() const;
   rtc::Thread* signaling_thread() const;
 
-  // Specifies which kind of data channel is allowed. This is controlled
+  // Specifies whether or not SCTP data channels are allowed.
-  // by the chrome command-line flag and constraints:
-  // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
-  // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
-  // not set or false, SCTP is allowed (DCT_SCTP);
-  // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
-  // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
   cricket::DataChannelType data_channel_type_ =
       cricket::DCT_NONE;  // TODO(bugs.webrtc.org/9987): Accessed on both
                           // signaling and network thread.
@@ -197,22 +160,12 @@ class DataChannelController : public RtpDataChannelProviderInterface,
   bool data_channel_transport_ready_to_send_
       RTC_GUARDED_BY(signaling_thread()) = false;
 
-  // |rtp_data_channel_| is used if in RTP data channel mode,
-  // |data_channel_transport_| when using SCTP.
-  // TODO(bugs.webrtc.org/9987): Accessed on both signaling and network
-  // thread.
-  cricket::RtpDataChannel* rtp_data_channel_ = nullptr;
-
   SctpSidAllocator sid_allocator_ /* RTC_GUARDED_BY(signaling_thread()) */;
   std::vector<rtc::scoped_refptr<SctpDataChannel>> sctp_data_channels_
       RTC_GUARDED_BY(signaling_thread());
   std::vector<rtc::scoped_refptr<SctpDataChannel>> sctp_data_channels_to_free_
       RTC_GUARDED_BY(signaling_thread());
 
-  // Map of label -> DataChannel
-  std::map<std::string, rtc::scoped_refptr<RtpDataChannel>> rtp_data_channels_
-      RTC_GUARDED_BY(signaling_thread());
-
   // Signals from |data_channel_transport_|.  These are invoked on the
   // signaling thread.
   // TODO(bugs.webrtc.org/11547): These '_s' signals likely all belong on the
@@ -228,8 +181,6 @@ class DataChannelController : public RtpDataChannelProviderInterface,
   sigslot::signal1<int> SignalDataChannelTransportChannelClosed_s
       RTC_GUARDED_BY(signaling_thread());
 
-  sigslot::signal1<RtpDataChannel*> SignalRtpDataChannelCreated_
-      RTC_GUARDED_BY(signaling_thread());
   sigslot::signal1<SctpDataChannel*> SignalSctpDataChannelCreated_
       RTC_GUARDED_BY(signaling_thread());
 

pc/data_channel_integrationtest.cc

@@ -553,11 +553,10 @@ TEST_F(DataChannelIntegrationTestUnifiedPlan,
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
+  ASSERT_TRUE_WAIT(caller()->pc()->GetSctpTransport(), kDefaultTimeout);
-    ASSERT_EQ_WAIT(SctpTransportState::kConnected,
+  ASSERT_EQ_WAIT(SctpTransportState::kConnected,
-                   caller()->pc()->GetSctpTransport()->Information().state(),
+                 caller()->pc()->GetSctpTransport()->Information().state(),
-                   kDefaultTimeout);
+                 kDefaultTimeout);
-  });
   ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
 }

pc/data_channel_unittest.cc

@@ -287,8 +287,7 @@ TEST_F(SctpDataChannelTest, OpenMessageSent) {
   SetChannelReady();
   EXPECT_GE(webrtc_data_channel_->id(), 0);
   EXPECT_EQ(cricket::DMT_CONTROL, provider_->last_send_data_params().type);
-  EXPECT_EQ(provider_->last_send_data_params().ssrc,
+  EXPECT_EQ(provider_->last_send_data_params().sid, webrtc_data_channel_->id());
-            static_cast<uint32_t>(webrtc_data_channel_->id()));
 }
 
 TEST_F(SctpDataChannelTest, QueuedOpenMessageSent) {
@@ -297,8 +296,7 @@ TEST_F(SctpDataChannelTest, QueuedOpenMessageSent) {
   provider_->set_send_blocked(false);
 
   EXPECT_EQ(cricket::DMT_CONTROL, provider_->last_send_data_params().type);
-  EXPECT_EQ(provider_->last_send_data_params().ssrc,
+  EXPECT_EQ(provider_->last_send_data_params().sid, webrtc_data_channel_->id());
-            static_cast<uint32_t>(webrtc_data_channel_->id()));
 }
 
 // Tests that the DataChannel created after transport gets ready can enter OPEN
@@ -334,7 +332,7 @@ TEST_F(SctpDataChannelTest, SendUnorderedAfterReceivesOpenAck) {
 
   // Emulates receiving an OPEN_ACK message.
   cricket::ReceiveDataParams params;
-  params.ssrc = init.id;
+  params.sid = init.id;
   params.type = cricket::DMT_CONTROL;
   rtc::CopyOnWriteBuffer payload;
   webrtc::WriteDataChannelOpenAckMessage(&payload);
@@ -360,7 +358,7 @@ TEST_F(SctpDataChannelTest, SendUnorderedAfterReceiveData) {
 
   // Emulates receiving a DATA message.
   cricket::ReceiveDataParams params;
-  params.ssrc = init.id;
+  params.sid = init.id;
   params.type = cricket::DMT_TEXT;
   webrtc::DataBuffer buffer("data");
   dc->OnDataReceived(params, buffer.data);
@@ -406,39 +404,39 @@ TEST_F(SctpDataChannelTest, QueuedCloseFlushes) {
   EXPECT_EQ(cricket::DMT_TEXT, provider_->last_send_data_params().type);
 }
 
-// Tests that messages are sent with the right ssrc.
+// Tests that messages are sent with the right id.
-TEST_F(SctpDataChannelTest, SendDataSsrc) {
+TEST_F(SctpDataChannelTest, SendDataId) {
   webrtc_data_channel_->SetSctpSid(1);
   SetChannelReady();
   webrtc::DataBuffer buffer("data");
   EXPECT_TRUE(webrtc_data_channel_->Send(buffer));
-  EXPECT_EQ(1U, provider_->last_send_data_params().ssrc);
+  EXPECT_EQ(1, provider_->last_send_data_params().sid);
 }
 
-// Tests that the incoming messages with wrong ssrcs are rejected.
+// Tests that the incoming messages with wrong ids are rejected.
-TEST_F(SctpDataChannelTest, ReceiveDataWithInvalidSsrc) {
+TEST_F(SctpDataChannelTest, ReceiveDataWithInvalidId) {
   webrtc_data_channel_->SetSctpSid(1);
   SetChannelReady();
 
   AddObserver();
 
   cricket::ReceiveDataParams params;
-  params.ssrc = 0;
+  params.sid = 0;
   webrtc::DataBuffer buffer("abcd");
   webrtc_data_channel_->OnDataReceived(params, buffer.data);
 
   EXPECT_EQ(0U, observer_->messages_received());
 }
 
-// Tests that the incoming messages with right ssrcs are acceted.
+// Tests that the incoming messages with right ids are accepted.
-TEST_F(SctpDataChannelTest, ReceiveDataWithValidSsrc) {
+TEST_F(SctpDataChannelTest, ReceiveDataWithValidId) {
   webrtc_data_channel_->SetSctpSid(1);
   SetChannelReady();
 
   AddObserver();
 
   cricket::ReceiveDataParams params;
-  params.ssrc = 1;
+  params.sid = 1;
   webrtc::DataBuffer buffer("abcd");
 
   webrtc_data_channel_->OnDataReceived(params, buffer.data);
@@ -459,7 +457,7 @@ TEST_F(SctpDataChannelTest, NoMsgSentIfNegotiatedAndNotFromOpenMsg) {
                               rtc::Thread::Current(), rtc::Thread::Current());
 
   EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000);
-  EXPECT_EQ(0U, provider_->last_send_data_params().ssrc);
+  EXPECT_EQ(0, provider_->last_send_data_params().sid);
 }
 
 // Tests that DataChannel::messages_received() and DataChannel::bytes_received()
@@ -477,7 +475,7 @@ TEST_F(SctpDataChannelTest, VerifyMessagesAndBytesReceived) {
 
   webrtc_data_channel_->SetSctpSid(1);
   cricket::ReceiveDataParams params;
-  params.ssrc = 1;
+  params.sid = 1;
 
   // Default values.
   EXPECT_EQ(0U, webrtc_data_channel_->messages_received());
@@ -524,8 +522,7 @@ TEST_F(SctpDataChannelTest, OpenAckSentIfCreatedFromOpenMessage) {
 
   EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000);
 
-  EXPECT_EQ(static_cast<unsigned int>(config.id),
+  EXPECT_EQ(config.id, provider_->last_send_data_params().sid);
-            provider_->last_send_data_params().ssrc);
   EXPECT_EQ(cricket::DMT_CONTROL, provider_->last_send_data_params().type);
 }
 
@@ -584,7 +581,7 @@ TEST_F(SctpDataChannelTest, ClosedWhenReceivedBufferFull) {
   memset(buffer.MutableData(), 0, buffer.size());
 
   cricket::ReceiveDataParams params;
-  params.ssrc = 0;
+  params.sid = 0;
 
   // Receiving data without having an observer will overflow the buffer.
   for (size_t i = 0; i < 16 * 1024 + 1; ++i) {

pc/media_session.cc

@@ -1380,14 +1380,6 @@ void MediaDescriptionOptions::AddVideoSender(
                     num_sim_layers);
 }
 
-void MediaDescriptionOptions::AddRtpDataChannel(const std::string& track_id,
-                                                const std::string& stream_id) {
-  RTC_DCHECK(type == MEDIA_TYPE_DATA);
-  // TODO(steveanton): Is it the case that RtpDataChannel will never have more
-  // than one stream?
-  AddSenderInternal(track_id, {stream_id}, {}, SimulcastLayerList(), 1);
-}
-
 void MediaDescriptionOptions::AddSenderInternal(
     const std::string& track_id,
     const std::vector<std::string>& stream_ids,
@@ -1428,7 +1420,6 @@ MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
   channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_);
   channel_manager->GetSupportedVideoSendCodecs(&video_send_codecs_);
   channel_manager->GetSupportedVideoReceiveCodecs(&video_recv_codecs_);
-  channel_manager->GetSupportedDataCodecs(&rtp_data_codecs_);
   ComputeAudioCodecsIntersectionAndUnion();
   ComputeVideoCodecsIntersectionAndUnion();
 }
@@ -1521,12 +1512,8 @@ std::unique_ptr<SessionDescription> MediaSessionDescriptionFactory::CreateOffer(
 
   AudioCodecs offer_audio_codecs;
   VideoCodecs offer_video_codecs;
-  RtpDataCodecs offer_rtp_data_codecs;
+  GetCodecsForOffer(current_active_contents, &offer_audio_codecs,
-  GetCodecsForOffer(
+                    &offer_video_codecs);
-      current_active_contents, &offer_audio_codecs, &offer_video_codecs,
-      session_options.data_channel_type == DataChannelType::DCT_SCTP
-          ? nullptr
-          : &offer_rtp_data_codecs);
   if (!session_options.vad_enabled) {
     // If application doesn't want CN codecs in offer.
     StripCNCodecs(&offer_audio_codecs);
@@ -1574,8 +1561,8 @@ std::unique_ptr<SessionDescription> MediaSessionDescriptionFactory::CreateOffer(
       case MEDIA_TYPE_DATA:
         if (!AddDataContentForOffer(media_description_options, session_options,
                                     current_content, current_description,
-                                    offer_rtp_data_codecs, &current_streams,
+                                    &current_streams, offer.get(),
-                                    offer.get(), &ice_credentials)) {
+                                    &ice_credentials)) {
           return nullptr;
         }
         break;
@@ -1673,9 +1660,8 @@ MediaSessionDescriptionFactory::CreateAnswer(
   // sections.
   AudioCodecs answer_audio_codecs;
   VideoCodecs answer_video_codecs;
-  RtpDataCodecs answer_rtp_data_codecs;
   GetCodecsForAnswer(current_active_contents, *offer, &answer_audio_codecs,
-                     &answer_video_codecs, &answer_rtp_data_codecs);
+                     &answer_video_codecs);
 
   if (!session_options.vad_enabled) {
     // If application doesn't want CN codecs in answer.
@@ -1732,11 +1718,11 @@ MediaSessionDescriptionFactory::CreateAnswer(
         }
         break;
       case MEDIA_TYPE_DATA:
-        if (!AddDataContentForAnswer(
+        if (!AddDataContentForAnswer(media_description_options, session_options,
-                media_description_options, session_options, offer_content,
+                                     offer_content, offer, current_content,
-                offer, current_content, current_description,
+                                     current_description,
-                bundle_transport.get(), answer_rtp_data_codecs,
+                                     bundle_transport.get(), &current_streams,
-                &current_streams, answer.get(), &ice_credentials)) {
+                                     answer.get(), &ice_credentials)) {
           return nullptr;
         }
         break;
@@ -1899,7 +1885,6 @@ void MergeCodecsFromDescription(
     const std::vector<const ContentInfo*>& current_active_contents,
     AudioCodecs* audio_codecs,
     VideoCodecs* video_codecs,
-    RtpDataCodecs* rtp_data_codecs,
     UsedPayloadTypes* used_pltypes) {
   for (const ContentInfo* content : current_active_contents) {
     if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
@@ -1910,14 +1895,6 @@ void MergeCodecsFromDescription(
       const VideoContentDescription* video =
           content->media_description()->as_video();
       MergeCodecs<VideoCodec>(video->codecs(), video_codecs, used_pltypes);
-    } else if (IsMediaContentOfType(content, MEDIA_TYPE_DATA)) {
-      const RtpDataContentDescription* data =
-          content->media_description()->as_rtp_data();
-      if (data) {
-        // Only relevant for RTP datachannels
-        MergeCodecs<RtpDataCodec>(data->codecs(), rtp_data_codecs,
-                                  used_pltypes);
-      }
     }
   }
 }
@@ -1931,22 +1908,17 @@ void MergeCodecsFromDescription(
 void MediaSessionDescriptionFactory::GetCodecsForOffer(
     const std::vector<const ContentInfo*>& current_active_contents,
     AudioCodecs* audio_codecs,
-    VideoCodecs* video_codecs,
+    VideoCodecs* video_codecs) const {
-    RtpDataCodecs* rtp_data_codecs) const {
   // First - get all codecs from the current description if the media type
   // is used. Add them to |used_pltypes| so the payload type is not reused if a
   // new media type is added.
   UsedPayloadTypes used_pltypes;
   MergeCodecsFromDescription(current_active_contents, audio_codecs,
-                             video_codecs, rtp_data_codecs, &used_pltypes);
+                             video_codecs, &used_pltypes);
 
   // Add our codecs that are not in the current description.
   MergeCodecs<AudioCodec>(all_audio_codecs_, audio_codecs, &used_pltypes);
   MergeCodecs<VideoCodec>(all_video_codecs_, video_codecs, &used_pltypes);
-  // Only allocate a payload type for rtp datachannels when using rtp data
-  // channels.
-  if (rtp_data_codecs)
-    MergeCodecs<DataCodec>(rtp_data_codecs_, rtp_data_codecs, &used_pltypes);
 }
 
 // Getting codecs for an answer involves these steps:
@@ -1960,19 +1932,17 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
     const std::vector<const ContentInfo*>& current_active_contents,
     const SessionDescription& remote_offer,
     AudioCodecs* audio_codecs,
-    VideoCodecs* video_codecs,
+    VideoCodecs* video_codecs) const {
-    RtpDataCodecs* rtp_data_codecs) const {
   // First - get all codecs from the current description if the media type
   // is used. Add them to |used_pltypes| so the payload type is not reused if a
   // new media type is added.
   UsedPayloadTypes used_pltypes;
   MergeCodecsFromDescription(current_active_contents, audio_codecs,
-                             video_codecs, rtp_data_codecs, &used_pltypes);
+                             video_codecs, &used_pltypes);
 
   // Second - filter out codecs that we don't support at all and should ignore.
   AudioCodecs filtered_offered_audio_codecs;
   VideoCodecs filtered_offered_video_codecs;
-  RtpDataCodecs filtered_offered_rtp_data_codecs;
   for (const ContentInfo& content : remote_offer.contents()) {
     if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) {
       const AudioContentDescription* audio =
@@ -1998,22 +1968,6 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
           filtered_offered_video_codecs.push_back(offered_video_codec);
         }
       }
-    } else if (IsMediaContentOfType(&content, MEDIA_TYPE_DATA)) {
-      const RtpDataContentDescription* data =
-          content.media_description()->as_rtp_data();
-      if (data) {
-        // RTP data. This part is inactive for SCTP data.
-        for (const RtpDataCodec& offered_rtp_data_codec : data->codecs()) {
-          if (!FindMatchingCodec<RtpDataCodec>(
-                  data->codecs(), filtered_offered_rtp_data_codecs,
-                  offered_rtp_data_codec, nullptr) &&
-              FindMatchingCodec<RtpDataCodec>(data->codecs(), rtp_data_codecs_,
-                                              offered_rtp_data_codec,
-                                              nullptr)) {
-            filtered_offered_rtp_data_codecs.push_back(offered_rtp_data_codec);
-          }
-        }
-      }
     }
   }
 
@@ -2023,8 +1977,6 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
                           &used_pltypes);
   MergeCodecs<VideoCodec>(filtered_offered_video_codecs, video_codecs,
                           &used_pltypes);
-  MergeCodecs<DataCodec>(filtered_offered_rtp_data_codecs, rtp_data_codecs,
-                         &used_pltypes);
 }
 
 MediaSessionDescriptionFactory::AudioVideoRtpHeaderExtensions
@@ -2333,7 +2285,7 @@ bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
   return true;
 }
 
-bool MediaSessionDescriptionFactory::AddSctpDataContentForOffer(
+bool MediaSessionDescriptionFactory::AddDataContentForOffer(
     const MediaDescriptionOptions& media_description_options,
     const MediaSessionOptions& session_options,
     const ContentInfo* current_content,
@@ -2378,73 +2330,6 @@ bool MediaSessionDescriptionFactory::AddSctpDataContentForOffer(
   return true;
 }
 
-bool MediaSessionDescriptionFactory::AddRtpDataContentForOffer(
-    const MediaDescriptionOptions& media_description_options,
-    const MediaSessionOptions& session_options,
-    const ContentInfo* current_content,
-    const SessionDescription* current_description,
-    const RtpDataCodecs& rtp_data_codecs,
-    StreamParamsVec* current_streams,
-    SessionDescription* desc,
-    IceCredentialsIterator* ice_credentials) const {
-  auto data = std::make_unique<RtpDataContentDescription>();
-  bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
-
-  cricket::SecurePolicy sdes_policy =
-      IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
-                                                         : secure();
-  std::vector<std::string> crypto_suites;
-  GetSupportedDataSdesCryptoSuiteNames(session_options.crypto_options,
-                                       &crypto_suites);
-  if (!CreateMediaContentOffer(media_description_options, session_options,
-                               rtp_data_codecs, sdes_policy,
-                               GetCryptos(current_content), crypto_suites,
-                               RtpHeaderExtensions(), ssrc_generator_,
-                               current_streams, data.get())) {
-    return false;
-  }
-
-  data->set_bandwidth(kRtpDataMaxBandwidth);
-  SetMediaProtocol(secure_transport, data.get());
-  desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
-                   media_description_options.stopped, std::move(data));
-  if (!AddTransportOffer(media_description_options.mid,
-                         media_description_options.transport_options,
-                         current_description, desc, ice_credentials)) {
-    return false;
-  }
-  return true;
-}
-
-bool MediaSessionDescriptionFactory::AddDataContentForOffer(
-    const MediaDescriptionOptions& media_description_options,
-    const MediaSessionOptions& session_options,
-    const ContentInfo* current_content,
-    const SessionDescription* current_description,
-    const RtpDataCodecs& rtp_data_codecs,
-    StreamParamsVec* current_streams,
-    SessionDescription* desc,
-    IceCredentialsIterator* ice_credentials) const {
-  bool is_sctp = (session_options.data_channel_type == DCT_SCTP);
-  // If the DataChannel type is not specified, use the DataChannel type in
-  // the current description.
-  if (session_options.data_channel_type == DCT_NONE && current_content) {
-    RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_DATA));
-    is_sctp = (current_content->media_description()->protocol() ==
-               kMediaProtocolSctp);
-  }
-  if (is_sctp) {
-    return AddSctpDataContentForOffer(
-        media_description_options, session_options, current_content,
-        current_description, current_streams, desc, ice_credentials);
-  } else {
-    return AddRtpDataContentForOffer(media_description_options, session_options,
-                                     current_content, current_description,
-                                     rtp_data_codecs, current_streams, desc,
-                                     ice_credentials);
-  }
-}
-
 bool MediaSessionDescriptionFactory::AddUnsupportedContentForOffer(
     const MediaDescriptionOptions& media_description_options,
     const MediaSessionOptions& session_options,
@@ -2718,7 +2603,6 @@ bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
     const ContentInfo* current_content,
     const SessionDescription* current_description,
     const TransportInfo* bundle_transport,
-    const RtpDataCodecs& rtp_data_codecs,
     StreamParamsVec* current_streams,
     SessionDescription* answer,
     IceCredentialsIterator* ice_credentials) const {
@@ -2766,25 +2650,7 @@ bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
     bool offer_uses_sctpmap = offer_data_description->use_sctpmap();
     data_answer->as_sctp()->set_use_sctpmap(offer_uses_sctpmap);
   } else {
-    // RTP offer
+    RTC_NOTREACHED() << "Non-SCTP data content found";
-    data_answer = std::make_unique<RtpDataContentDescription>();
-
-    const RtpDataContentDescription* offer_data_description =
-        offer_content->media_description()->as_rtp_data();
-    RTC_CHECK(offer_data_description);
-    if (!SetCodecsInAnswer(offer_data_description, rtp_data_codecs,
-                           media_description_options, session_options,
-                           ssrc_generator_, current_streams,
-                           data_answer->as_rtp_data())) {
-      return false;
-    }
-    if (!CreateMediaContentAnswer(
-            offer_data_description, media_description_options, session_options,
-            sdes_policy, GetCryptos(current_content), RtpHeaderExtensions(),
-            ssrc_generator_, enable_encrypted_rtp_header_extensions_,
-            current_streams, bundle_enabled, data_answer.get())) {
-      return false;  // Fails the session setup.
-    }
   }
 
   bool secure = bundle_transport ? bundle_transport->description.secure()
@@ -2800,13 +2666,6 @@ bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
     return false;
   }
 
-  if (!rejected && session_options.data_channel_type == DCT_RTP) {
-    data_answer->set_bandwidth(kRtpDataMaxBandwidth);
-  } else {
-    // RFC 3264
-    // The answer MUST contain the same number of m-lines as the offer.
-    RTC_LOG(LS_INFO) << "Data is not supported in the answer.";
-  }
   answer->AddContent(media_description_options.mid, offer_content->type,
                      rejected, std::move(data_answer));
   return true;
@@ -2991,12 +2850,6 @@ const VideoContentDescription* GetFirstVideoContentDescription(
   return desc ? desc->as_video() : nullptr;
 }
 
-const RtpDataContentDescription* GetFirstRtpDataContentDescription(
-    const SessionDescription* sdesc) {
-  auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);
-  return desc ? desc->as_rtp_data() : nullptr;
-}
-
 const SctpDataContentDescription* GetFirstSctpDataContentDescription(
     const SessionDescription* sdesc) {
   auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);
@@ -3069,12 +2922,6 @@ VideoContentDescription* GetFirstVideoContentDescription(
   return desc ? desc->as_video() : nullptr;
 }
 
-RtpDataContentDescription* GetFirstRtpDataContentDescription(
-    SessionDescription* sdesc) {
-  auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);
-  return desc ? desc->as_rtp_data() : nullptr;
-}
-
 SctpDataContentDescription* GetFirstSctpDataContentDescription(
     SessionDescription* sdesc) {
   auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);

pc/media_session.h

@@ -73,10 +73,6 @@ struct MediaDescriptionOptions {
                       const SimulcastLayerList& simulcast_layers,
                       int num_sim_layers);
 
-  // Internally just uses sender_options.
-  void AddRtpDataChannel(const std::string& track_id,
-                         const std::string& stream_id);
-
   MediaType type;
   std::string mid;
   webrtc::RtpTransceiverDirection direction;
@@ -162,10 +158,6 @@ class MediaSessionDescriptionFactory {
                         const VideoCodecs& recv_codecs);
   RtpHeaderExtensions filtered_rtp_header_extensions(
       RtpHeaderExtensions extensions) const;
-  const RtpDataCodecs& rtp_data_codecs() const { return rtp_data_codecs_; }
-  void set_rtp_data_codecs(const RtpDataCodecs& codecs) {
-    rtp_data_codecs_ = codecs;
-  }
   SecurePolicy secure() const { return secure_; }
   void set_secure(SecurePolicy s) { secure_ = s; }
 
@@ -204,14 +196,12 @@ class MediaSessionDescriptionFactory {
   void GetCodecsForOffer(
       const std::vector<const ContentInfo*>& current_active_contents,
       AudioCodecs* audio_codecs,
-      VideoCodecs* video_codecs,
+      VideoCodecs* video_codecs) const;
-      RtpDataCodecs* rtp_data_codecs) const;
   void GetCodecsForAnswer(
       const std::vector<const ContentInfo*>& current_active_contents,
       const SessionDescription& remote_offer,
       AudioCodecs* audio_codecs,
-      VideoCodecs* video_codecs,
+      VideoCodecs* video_codecs) const;
-      RtpDataCodecs* rtp_data_codecs) const;
   AudioVideoRtpHeaderExtensions GetOfferedRtpHeaderExtensionsWithIds(
       const std::vector<const ContentInfo*>& current_active_contents,
       bool extmap_allow_mixed,
@@ -261,32 +251,11 @@ class MediaSessionDescriptionFactory {
       SessionDescription* desc,
       IceCredentialsIterator* ice_credentials) const;
 
-  bool AddSctpDataContentForOffer(
-      const MediaDescriptionOptions& media_description_options,
-      const MediaSessionOptions& session_options,
-      const ContentInfo* current_content,
-      const SessionDescription* current_description,
-      StreamParamsVec* current_streams,
-      SessionDescription* desc,
-      IceCredentialsIterator* ice_credentials) const;
-  bool AddRtpDataContentForOffer(
-      const MediaDescriptionOptions& media_description_options,
-      const MediaSessionOptions& session_options,
-      const ContentInfo* current_content,
-      const SessionDescription* current_description,
-      const RtpDataCodecs& rtp_data_codecs,
-      StreamParamsVec* current_streams,
-      SessionDescription* desc,
-      IceCredentialsIterator* ice_credentials) const;
-  // This function calls either AddRtpDataContentForOffer or
-  // AddSctpDataContentForOffer depending on protocol.
-  // The codecs argument is ignored for SCTP.
   bool AddDataContentForOffer(
       const MediaDescriptionOptions& media_description_options,
       const MediaSessionOptions& session_options,
       const ContentInfo* current_content,
       const SessionDescription* current_description,
-      const RtpDataCodecs& rtp_data_codecs,
       StreamParamsVec* current_streams,
       SessionDescription* desc,
       IceCredentialsIterator* ice_credentials) const;
@@ -335,7 +304,6 @@ class MediaSessionDescriptionFactory {
       const ContentInfo* current_content,
       const SessionDescription* current_description,
       const TransportInfo* bundle_transport,
-      const RtpDataCodecs& rtp_data_codecs,
       StreamParamsVec* current_streams,
       SessionDescription* answer,
       IceCredentialsIterator* ice_credentials) const;
@@ -368,7 +336,6 @@ class MediaSessionDescriptionFactory {
   VideoCodecs video_sendrecv_codecs_;
   // Union of send and recv.
   VideoCodecs all_video_codecs_;
-  RtpDataCodecs rtp_data_codecs_;
   // This object is not owned by the channel so it must outlive it.
   rtc::UniqueRandomIdGenerator* const ssrc_generator_;
   bool enable_encrypted_rtp_header_extensions_ = false;
@@ -398,8 +365,6 @@ const AudioContentDescription* GetFirstAudioContentDescription(
     const SessionDescription* sdesc);
 const VideoContentDescription* GetFirstVideoContentDescription(
     const SessionDescription* sdesc);
-const RtpDataContentDescription* GetFirstRtpDataContentDescription(
-    const SessionDescription* sdesc);
 const SctpDataContentDescription* GetFirstSctpDataContentDescription(
     const SessionDescription* sdesc);
 // Non-const versions of the above functions.
@@ -417,8 +382,6 @@ AudioContentDescription* GetFirstAudioContentDescription(
     SessionDescription* sdesc);
 VideoContentDescription* GetFirstVideoContentDescription(
     SessionDescription* sdesc);
-RtpDataContentDescription* GetFirstRtpDataContentDescription(
-    SessionDescription* sdesc);
 SctpDataContentDescription* GetFirstSctpDataContentDescription(
     SessionDescription* sdesc);
 

pc/media_session_unittest.cc

@@ -50,7 +50,6 @@ using cricket::CryptoParamsVec;
 using cricket::GetFirstAudioContent;
 using cricket::GetFirstAudioContentDescription;
 using cricket::GetFirstDataContent;
-using cricket::GetFirstRtpDataContentDescription;
 using cricket::GetFirstVideoContent;
 using cricket::GetFirstVideoContentDescription;
 using cricket::kAutoBandwidth;
@@ -65,8 +64,6 @@ using cricket::MediaSessionOptions;
 using cricket::MediaType;
 using cricket::RidDescription;
 using cricket::RidDirection;
-using cricket::RtpDataCodec;
-using cricket::RtpDataContentDescription;
 using cricket::SctpDataContentDescription;
 using cricket::SEC_DISABLED;
 using cricket::SEC_ENABLED;
@@ -133,15 +130,6 @@ static const VideoCodec kVideoCodecs2[] = {VideoCodec(126, "H264"),
 
 static const VideoCodec kVideoCodecsAnswer[] = {VideoCodec(97, "H264")};
 
-static const RtpDataCodec kDataCodecs1[] = {RtpDataCodec(98, "binary-data"),
-                                            RtpDataCodec(99, "utf8-text")};
-
-static const RtpDataCodec kDataCodecs2[] = {RtpDataCodec(126, "binary-data"),
-                                            RtpDataCodec(127, "utf8-text")};
-
-static const RtpDataCodec kDataCodecsAnswer[] = {
-    RtpDataCodec(98, "binary-data"), RtpDataCodec(99, "utf8-text")};
-
 static const RtpExtension kAudioRtpExtension1[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
     RtpExtension("http://google.com/testing/audio_something", 10),
@@ -260,9 +248,6 @@ static const char kVideoTrack2[] = "video_2";
 static const char kAudioTrack1[] = "audio_1";
 static const char kAudioTrack2[] = "audio_2";
 static const char kAudioTrack3[] = "audio_3";
-static const char kDataTrack1[] = "data_1";
-static const char kDataTrack2[] = "data_2";
-static const char kDataTrack3[] = "data_3";
 
 static const char* kMediaProtocols[] = {"RTP/AVP", "RTP/SAVP", "RTP/AVPF",
                                         "RTP/SAVPF"};
@@ -369,10 +354,6 @@ static void AttachSenderToMediaDescriptionOptions(
       it->AddVideoSender(track_id, stream_ids, rids, simulcast_layers,
                          num_sim_layer);
       break;
-    case MEDIA_TYPE_DATA:
-      RTC_CHECK(stream_ids.size() == 1U);
-      it->AddRtpDataChannel(track_id, stream_ids[0]);
-      break;
     default:
       RTC_NOTREACHED();
   }
@@ -437,12 +418,10 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
                          MAKE_VECTOR(kAudioCodecs1));
     f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1),
                          MAKE_VECTOR(kVideoCodecs1));
-    f1_.set_rtp_data_codecs(MAKE_VECTOR(kDataCodecs1));
     f2_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs2),
                          MAKE_VECTOR(kAudioCodecs2));
     f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2),
                          MAKE_VECTOR(kVideoCodecs2));
-    f2_.set_rtp_data_codecs(MAKE_VECTOR(kDataCodecs2));
     tdf1_.set_certificate(rtc::RTCCertificate::Create(
         std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
     tdf2_.set_certificate(rtc::RTCCertificate::Create(
@@ -604,8 +583,6 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
     f1_.set_secure(SEC_ENABLED);
     MediaSessionOptions options;
     AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-    AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                   &options);
     std::unique_ptr<SessionDescription> ref_desc;
     std::unique_ptr<SessionDescription> desc;
     if (offer) {
@@ -862,30 +839,21 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoOffer) {
 TEST_F(MediaSessionDescriptionFactoryTest, TestBundleOfferWithSameCodecPlType) {
   const VideoCodec& offered_video_codec = f2_.video_sendrecv_codecs()[0];
   const AudioCodec& offered_audio_codec = f2_.audio_sendrecv_codecs()[0];
-  const RtpDataCodec& offered_data_codec = f2_.rtp_data_codecs()[0];
   ASSERT_EQ(offered_video_codec.id, offered_audio_codec.id);
-  ASSERT_EQ(offered_video_codec.id, offered_data_codec.id);
 
   MediaSessionOptions opts;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
   opts.bundle_enabled = true;
   std::unique_ptr<SessionDescription> offer = f2_.CreateOffer(opts, NULL);
   const VideoContentDescription* vcd =
       GetFirstVideoContentDescription(offer.get());
   const AudioContentDescription* acd =
       GetFirstAudioContentDescription(offer.get());
-  const RtpDataContentDescription* dcd =
-      GetFirstRtpDataContentDescription(offer.get());
   ASSERT_TRUE(NULL != vcd);
   ASSERT_TRUE(NULL != acd);
-  ASSERT_TRUE(NULL != dcd);
   EXPECT_NE(vcd->codecs()[0].id, acd->codecs()[0].id);
-  EXPECT_NE(vcd->codecs()[0].id, dcd->codecs()[0].id);
-  EXPECT_NE(acd->codecs()[0].id, dcd->codecs()[0].id);
   EXPECT_EQ(vcd->codecs()[0].name, offered_video_codec.name);
   EXPECT_EQ(acd->codecs()[0].name, offered_audio_codec.name);
-  EXPECT_EQ(dcd->codecs()[0].name, offered_data_codec.name);
 }
 
 // Test creating an updated offer with bundle, audio, video and data
@@ -909,8 +877,6 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 
   MediaSessionOptions updated_opts;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &updated_opts);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &updated_opts);
   updated_opts.bundle_enabled = true;
   std::unique_ptr<SessionDescription> updated_offer(
       f1_.CreateOffer(updated_opts, answer.get()));
@@ -919,51 +885,13 @@ TEST_F(MediaSessionDescriptionFactoryTest,
       GetFirstAudioContentDescription(updated_offer.get());
   const VideoContentDescription* vcd =
       GetFirstVideoContentDescription(updated_offer.get());
-  const RtpDataContentDescription* dcd =
-      GetFirstRtpDataContentDescription(updated_offer.get());
   EXPECT_TRUE(NULL != vcd);
   EXPECT_TRUE(NULL != acd);
-  EXPECT_TRUE(NULL != dcd);
 
   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
   EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
   ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
   EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
-  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
-}
-
-// Create a RTP data offer, and ensure it matches what we expect.
-TEST_F(MediaSessionDescriptionFactoryTest, TestCreateRtpDataOffer) {
-  MediaSessionOptions opts;
-  AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
-  f1_.set_secure(SEC_ENABLED);
-  std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
-  ASSERT_TRUE(offer.get() != NULL);
-  const ContentInfo* ac = offer->GetContentByName("audio");
-  const ContentInfo* dc = offer->GetContentByName("data");
-  ASSERT_TRUE(ac != NULL);
-  ASSERT_TRUE(dc != NULL);
-  EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  EXPECT_EQ(MediaProtocolType::kRtp, dc->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
-  EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
-  EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
-  EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attched.
-  EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
-  EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
-  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
-  EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
-  EXPECT_EQ(f1_.rtp_data_codecs(), dcd->codecs());
-  EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached.
-  EXPECT_EQ(cricket::kRtpDataMaxBandwidth,
-            dcd->bandwidth());   // default bandwidth (auto)
-  EXPECT_TRUE(dcd->rtcp_mux());  // rtcp-mux defaults on
-  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
 }
 
 // Create an SCTP data offer with bundle without error.
@@ -1350,74 +1278,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerGcmAnswer) {
   TestVideoGcmCipher(false, true);
 }
 
-TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswer) {
-  MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
-  f1_.set_secure(SEC_ENABLED);
-  f2_.set_secure(SEC_ENABLED);
-  std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
-  ASSERT_TRUE(offer.get() != NULL);
-  std::unique_ptr<SessionDescription> answer =
-      f2_.CreateAnswer(offer.get(), opts, NULL);
-  const ContentInfo* ac = answer->GetContentByName("audio");
-  const ContentInfo* dc = answer->GetContentByName("data");
-  ASSERT_TRUE(ac != NULL);
-  ASSERT_TRUE(dc != NULL);
-  EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  EXPECT_EQ(MediaProtocolType::kRtp, dc->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
-  EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
-  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
-  EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
-  EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
-  EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
-  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
-  EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
-  EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached
-  EXPECT_TRUE(dcd->rtcp_mux());      // negotiated rtcp-mux
-  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
-}
-
-TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerGcm) {
-  MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
-  opts.crypto_options.srtp.enable_gcm_crypto_suites = true;
-  f1_.set_secure(SEC_ENABLED);
-  f2_.set_secure(SEC_ENABLED);
-  std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
-  ASSERT_TRUE(offer.get() != NULL);
-  for (cricket::ContentInfo& content : offer->contents()) {
-    auto cryptos = content.media_description()->cryptos();
-    PreferGcmCryptoParameters(&cryptos);
-    content.media_description()->set_cryptos(cryptos);
-  }
-  std::unique_ptr<SessionDescription> answer =
-      f2_.CreateAnswer(offer.get(), opts, NULL);
-  const ContentInfo* ac = answer->GetContentByName("audio");
-  const ContentInfo* dc = answer->GetContentByName("data");
-  ASSERT_TRUE(ac != NULL);
-  ASSERT_TRUE(dc != NULL);
-  EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  EXPECT_EQ(MediaProtocolType::kRtp, dc->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
-  EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
-  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
-  EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
-  EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
-  EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
-  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
-  EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
-  EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
-  EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached
-  EXPECT_TRUE(dcd->rtcp_mux());      // negotiated rtcp-mux
-  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuiteGcm);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
-}
-
 // The use_sctpmap flag should be set in an Sctp DataContentDescription by
 // default. The answer's use_sctpmap flag should match the offer's.
 TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerUsesSctpmap) {
@@ -1621,35 +1481,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerToInactiveOffer) {
                              RtpTransceiverDirection::kInactive);
 }
 
-// Test that a data content with an unknown protocol is rejected in an answer.
-TEST_F(MediaSessionDescriptionFactoryTest,
-       CreateDataAnswerToOfferWithUnknownProtocol) {
-  MediaSessionOptions opts;
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
-  f1_.set_secure(SEC_ENABLED);
-  f2_.set_secure(SEC_ENABLED);
-  std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
-  ContentInfo* dc_offer = offer->GetContentByName("data");
-  ASSERT_TRUE(dc_offer != NULL);
-  RtpDataContentDescription* dcd_offer =
-      dc_offer->media_description()->as_rtp_data();
-  ASSERT_TRUE(dcd_offer != NULL);
-  // Offer must be acceptable as an RTP protocol in order to be set.
-  std::string protocol = "RTP/a weird unknown protocol";
-  dcd_offer->set_protocol(protocol);
-
-  std::unique_ptr<SessionDescription> answer =
-      f2_.CreateAnswer(offer.get(), opts, NULL);
-
-  const ContentInfo* dc_answer = answer->GetContentByName("data");
-  ASSERT_TRUE(dc_answer != NULL);
-  EXPECT_TRUE(dc_answer->rejected);
-  const RtpDataContentDescription* dcd_answer =
-      dc_answer->media_description()->as_rtp_data();
-  ASSERT_TRUE(dcd_answer != NULL);
-  EXPECT_EQ(protocol, dcd_answer->protocol());
-}
-
 // Test that the media protocol is RTP/AVPF if DTLS and SDES are disabled.
 TEST_F(MediaSessionDescriptionFactoryTest, AudioOfferAnswerWithCryptoDisabled) {
   MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
@@ -2169,36 +2000,28 @@ TEST_F(MediaSessionDescriptionFactoryTest,
        TestCreateAnswerWithoutLegacyStreams) {
   MediaSessionOptions opts;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
   std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
   ASSERT_TRUE(offer.get() != NULL);
   std::unique_ptr<SessionDescription> answer =
       f2_.CreateAnswer(offer.get(), opts, NULL);
   const ContentInfo* ac = answer->GetContentByName("audio");
   const ContentInfo* vc = answer->GetContentByName("video");
-  const ContentInfo* dc = answer->GetContentByName("data");
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc != NULL);
   const AudioContentDescription* acd = ac->media_description()->as_audio();
   const VideoContentDescription* vcd = vc->media_description()->as_video();
-  const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
 
   EXPECT_FALSE(acd->has_ssrcs());  // No StreamParams.
   EXPECT_FALSE(vcd->has_ssrcs());  // No StreamParams.
-  EXPECT_FALSE(dcd->has_ssrcs());  // No StreamParams.
 }
 
 // Create a typical video answer, and ensure it matches what we expect.
 TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerRtcpMux) {
   MediaSessionOptions offer_opts;
   AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &offer_opts);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kSendRecv,
-                 &offer_opts);
 
   MediaSessionOptions answer_opts;
   AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &answer_opts);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kSendRecv,
-                 &answer_opts);
 
   std::unique_ptr<SessionDescription> offer;
   std::unique_ptr<SessionDescription> answer;
@@ -2209,16 +2032,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerRtcpMux) {
   answer = f2_.CreateAnswer(offer.get(), answer_opts, NULL);
   ASSERT_TRUE(NULL != GetFirstAudioContentDescription(offer.get()));
   ASSERT_TRUE(NULL != GetFirstVideoContentDescription(offer.get()));
-  ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(offer.get()));
   ASSERT_TRUE(NULL != GetFirstAudioContentDescription(answer.get()));
   ASSERT_TRUE(NULL != GetFirstVideoContentDescription(answer.get()));
-  ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(answer.get()));
   EXPECT_TRUE(GetFirstAudioContentDescription(offer.get())->rtcp_mux());
   EXPECT_TRUE(GetFirstVideoContentDescription(offer.get())->rtcp_mux());
-  EXPECT_TRUE(GetFirstRtpDataContentDescription(offer.get())->rtcp_mux());
   EXPECT_TRUE(GetFirstAudioContentDescription(answer.get())->rtcp_mux());
   EXPECT_TRUE(GetFirstVideoContentDescription(answer.get())->rtcp_mux());
-  EXPECT_TRUE(GetFirstRtpDataContentDescription(answer.get())->rtcp_mux());
 
   offer_opts.rtcp_mux_enabled = true;
   answer_opts.rtcp_mux_enabled = false;
@@ -2226,16 +2045,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerRtcpMux) {
   answer = f2_.CreateAnswer(offer.get(), answer_opts, NULL);
   ASSERT_TRUE(NULL != GetFirstAudioContentDescription(offer.get()));
   ASSERT_TRUE(NULL != GetFirstVideoContentDescription(offer.get()));
-  ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(offer.get()));
   ASSERT_TRUE(NULL != GetFirstAudioContentDescription(answer.get()));
   ASSERT_TRUE(NULL != GetFirstVideoContentDescription(answer.get()));
-  ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(answer.get()));
   EXPECT_TRUE(GetFirstAudioContentDescription(offer.get())->rtcp_mux());
   EXPECT_TRUE(GetFirstVideoContentDescription(offer.get())->rtcp_mux());
-  EXPECT_TRUE(GetFirstRtpDataContentDescription(offer.get())->rtcp_mux());
   EXPECT_FALSE(GetFirstAudioContentDescription(answer.get())->rtcp_mux());
   EXPECT_FALSE(GetFirstVideoContentDescription(answer.get())->rtcp_mux());
-  EXPECT_FALSE(GetFirstRtpDataContentDescription(answer.get())->rtcp_mux());
 
   offer_opts.rtcp_mux_enabled = false;
   answer_opts.rtcp_mux_enabled = true;
@@ -2243,16 +2058,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerRtcpMux) {
   answer = f2_.CreateAnswer(offer.get(), answer_opts, NULL);
   ASSERT_TRUE(NULL != GetFirstAudioContentDescription(offer.get()));
   ASSERT_TRUE(NULL != GetFirstVideoContentDescription(offer.get()));
-  ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(offer.get()));
   ASSERT_TRUE(NULL != GetFirstAudioContentDescription(answer.get()));
   ASSERT_TRUE(NULL != GetFirstVideoContentDescription(answer.get()));
-  ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(answer.get()));
   EXPECT_FALSE(GetFirstAudioContentDescription(offer.get())->rtcp_mux());
   EXPECT_FALSE(GetFirstVideoContentDescription(offer.get())->rtcp_mux());
-  EXPECT_FALSE(GetFirstRtpDataContentDescription(offer.get())->rtcp_mux());
   EXPECT_FALSE(GetFirstAudioContentDescription(answer.get())->rtcp_mux());
   EXPECT_FALSE(GetFirstVideoContentDescription(answer.get())->rtcp_mux());
-  EXPECT_FALSE(GetFirstRtpDataContentDescription(answer.get())->rtcp_mux());
 
   offer_opts.rtcp_mux_enabled = false;
   answer_opts.rtcp_mux_enabled = false;
@@ -2260,16 +2071,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerRtcpMux) {
   answer = f2_.CreateAnswer(offer.get(), answer_opts, NULL);
   ASSERT_TRUE(NULL != GetFirstAudioContentDescription(offer.get()));
   ASSERT_TRUE(NULL != GetFirstVideoContentDescription(offer.get()));
-  ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(offer.get()));
   ASSERT_TRUE(NULL != GetFirstAudioContentDescription(answer.get()));
   ASSERT_TRUE(NULL != GetFirstVideoContentDescription(answer.get()));
-  ASSERT_TRUE(NULL != GetFirstRtpDataContentDescription(answer.get()));
   EXPECT_FALSE(GetFirstAudioContentDescription(offer.get())->rtcp_mux());
   EXPECT_FALSE(GetFirstVideoContentDescription(offer.get())->rtcp_mux());
-  EXPECT_FALSE(GetFirstRtpDataContentDescription(offer.get())->rtcp_mux());
   EXPECT_FALSE(GetFirstAudioContentDescription(answer.get())->rtcp_mux());
   EXPECT_FALSE(GetFirstVideoContentDescription(answer.get())->rtcp_mux());
-  EXPECT_FALSE(GetFirstRtpDataContentDescription(answer.get())->rtcp_mux());
 }
 
 // Create an audio-only answer to a video offer.
@@ -2295,55 +2102,27 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerToVideo) {
   EXPECT_TRUE(vc->rejected);
 }
 
-// Create an audio-only answer to an offer with data.
-TEST_F(MediaSessionDescriptionFactoryTest, TestCreateNoDataAnswerToDataOffer) {
-  MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
-  opts.data_channel_type = cricket::DCT_RTP;
-  AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data",
-                             RtpTransceiverDirection::kRecvOnly, kActive,
-                             &opts);
-  std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
-  ASSERT_TRUE(offer.get() != NULL);
-
-  opts.media_description_options[1].stopped = true;
-  std::unique_ptr<SessionDescription> answer =
-      f2_.CreateAnswer(offer.get(), opts, NULL);
-  const ContentInfo* ac = answer->GetContentByName("audio");
-  const ContentInfo* dc = answer->GetContentByName("data");
-  ASSERT_TRUE(ac != NULL);
-  ASSERT_TRUE(dc != NULL);
-  ASSERT_TRUE(dc->media_description() != NULL);
-  EXPECT_TRUE(dc->rejected);
-}
-
 // Create an answer that rejects the contents which are rejected in the offer.
 TEST_F(MediaSessionDescriptionFactoryTest,
        CreateAnswerToOfferWithRejectedMedia) {
   MediaSessionOptions opts;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
   std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
   ASSERT_TRUE(offer.get() != NULL);
   ContentInfo* ac = offer->GetContentByName("audio");
   ContentInfo* vc = offer->GetContentByName("video");
-  ContentInfo* dc = offer->GetContentByName("data");
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc != NULL);
-  ASSERT_TRUE(dc != NULL);
   ac->rejected = true;
   vc->rejected = true;
-  dc->rejected = true;
   std::unique_ptr<SessionDescription> answer =
       f2_.CreateAnswer(offer.get(), opts, NULL);
   ac = answer->GetContentByName("audio");
   vc = answer->GetContentByName("video");
-  dc = answer->GetContentByName("data");
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc != NULL);
-  ASSERT_TRUE(dc != NULL);
   EXPECT_TRUE(ac->rejected);
   EXPECT_TRUE(vc->rejected);
-  EXPECT_TRUE(dc->rejected);
 }
 
 TEST_F(MediaSessionDescriptionFactoryTest,
@@ -2458,7 +2237,6 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 // Create an audio and video offer with:
 // - one video track
 // - two audio tracks
-// - two data tracks
 // and ensure it matches what we expect. Also updates the initial offer by
 // adding a new video track and replaces one of the audio tracks.
 TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
@@ -2471,25 +2249,16 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
   AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack2,
                                         {kMediaStream1}, 1, &opts);
 
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kSendRecv, &opts);
-  AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack1,
-                                        {kMediaStream1}, 1, &opts);
-  AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack2,
-                                        {kMediaStream1}, 1, &opts);
-
   f1_.set_secure(SEC_ENABLED);
   std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(opts, NULL);
 
   ASSERT_TRUE(offer.get() != NULL);
   const ContentInfo* ac = offer->GetContentByName("audio");
   const ContentInfo* vc = offer->GetContentByName("video");
-  const ContentInfo* dc = offer->GetContentByName("data");
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc != NULL);
-  ASSERT_TRUE(dc != NULL);
   const AudioContentDescription* acd = ac->media_description()->as_audio();
   const VideoContentDescription* vcd = vc->media_description()->as_video();
-  const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
 
@@ -2518,25 +2287,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
   EXPECT_EQ(kAutoBandwidth, vcd->bandwidth());  // default bandwidth (auto)
   EXPECT_TRUE(vcd->rtcp_mux());                 // rtcp-mux defaults on
 
-  EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
-  EXPECT_EQ(f1_.rtp_data_codecs(), dcd->codecs());
-  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
-
-  const StreamParamsVec& data_streams = dcd->streams();
-  ASSERT_EQ(2U, data_streams.size());
-  EXPECT_EQ(data_streams[0].cname, data_streams[1].cname);
-  EXPECT_EQ(kDataTrack1, data_streams[0].id);
-  ASSERT_EQ(1U, data_streams[0].ssrcs.size());
-  EXPECT_NE(0U, data_streams[0].ssrcs[0]);
-  EXPECT_EQ(kDataTrack2, data_streams[1].id);
-  ASSERT_EQ(1U, data_streams[1].ssrcs.size());
-  EXPECT_NE(0U, data_streams[1].ssrcs[0]);
-
-  EXPECT_EQ(cricket::kRtpDataMaxBandwidth,
-            dcd->bandwidth());   // default bandwidth (auto)
-  EXPECT_TRUE(dcd->rtcp_mux());  // rtcp-mux defaults on
-  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
-
   // Update the offer. Add a new video track that is not synched to the
   // other tracks and replace audio track 2 with audio track 3.
   AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack2,
@@ -2544,38 +2294,27 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
   DetachSenderFromMediaSection("audio", kAudioTrack2, &opts);
   AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack3,
                                         {kMediaStream1}, 1, &opts);
-  DetachSenderFromMediaSection("data", kDataTrack2, &opts);
-  AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack3,
-                                        {kMediaStream1}, 1, &opts);
   std::unique_ptr<SessionDescription> updated_offer(
       f1_.CreateOffer(opts, offer.get()));
 
   ASSERT_TRUE(updated_offer.get() != NULL);
   ac = updated_offer->GetContentByName("audio");
   vc = updated_offer->GetContentByName("video");
-  dc = updated_offer->GetContentByName("data");
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc != NULL);
-  ASSERT_TRUE(dc != NULL);
   const AudioContentDescription* updated_acd =
       ac->media_description()->as_audio();
   const VideoContentDescription* updated_vcd =
       vc->media_description()->as_video();
-  const RtpDataContentDescription* updated_dcd =
-      dc->media_description()->as_rtp_data();
 
   EXPECT_EQ(acd->type(), updated_acd->type());
   EXPECT_EQ(acd->codecs(), updated_acd->codecs());
   EXPECT_EQ(vcd->type(), updated_vcd->type());
   EXPECT_EQ(vcd->codecs(), updated_vcd->codecs());
-  EXPECT_EQ(dcd->type(), updated_dcd->type());
-  EXPECT_EQ(dcd->codecs(), updated_dcd->codecs());
   ASSERT_CRYPTO(updated_acd, 1U, kDefaultSrtpCryptoSuite);
   EXPECT_TRUE(CompareCryptoParams(acd->cryptos(), updated_acd->cryptos()));
   ASSERT_CRYPTO(updated_vcd, 1U, kDefaultSrtpCryptoSuite);
   EXPECT_TRUE(CompareCryptoParams(vcd->cryptos(), updated_vcd->cryptos()));
-  ASSERT_CRYPTO(updated_dcd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_TRUE(CompareCryptoParams(dcd->cryptos(), updated_dcd->cryptos()));
 
   const StreamParamsVec& updated_audio_streams = updated_acd->streams();
   ASSERT_EQ(2U, updated_audio_streams.size());
@@ -2591,18 +2330,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
   EXPECT_EQ(kVideoTrack2, updated_video_streams[1].id);
   // All the media streams in one PeerConnection share one RTCP CNAME.
   EXPECT_EQ(updated_video_streams[1].cname, updated_video_streams[0].cname);
-
-  const StreamParamsVec& updated_data_streams = updated_dcd->streams();
-  ASSERT_EQ(2U, updated_data_streams.size());
-  EXPECT_EQ(data_streams[0], updated_data_streams[0]);
-  EXPECT_EQ(kDataTrack3, updated_data_streams[1].id);  // New data track.
-  ASSERT_EQ(1U, updated_data_streams[1].ssrcs.size());
-  EXPECT_NE(0U, updated_data_streams[1].ssrcs[0]);
-  EXPECT_EQ(updated_data_streams[0].cname, updated_data_streams[1].cname);
-  // The stream correctly got the CNAME from the MediaSessionOptions.
-  // The Expected RTCP CNAME is the default one as we are using the default
-  // MediaSessionOptions.
-  EXPECT_EQ(updated_data_streams[0].cname, cricket::kDefaultRtcpCname);
 }
 
 // Create an offer with simulcast video stream.
@@ -2805,10 +2532,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
   AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
                              RtpTransceiverDirection::kRecvOnly, kActive,
                              &offer_opts);
-  offer_opts.data_channel_type = cricket::DCT_RTP;
-  AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data",
-                             RtpTransceiverDirection::kRecvOnly, kActive,
-                             &offer_opts);
   f1_.set_secure(SEC_ENABLED);
   f2_.set_secure(SEC_ENABLED);
   std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(offer_opts, NULL);
@@ -2827,31 +2550,18 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
   AttachSenderToMediaDescriptionOptions("audio", MEDIA_TYPE_AUDIO, kAudioTrack2,
                                         {kMediaStream1}, 1, &answer_opts);
 
-  AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data",
-                             RtpTransceiverDirection::kSendRecv, kActive,
-                             &answer_opts);
-  AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack1,
-                                        {kMediaStream1}, 1, &answer_opts);
-  AttachSenderToMediaDescriptionOptions("data", MEDIA_TYPE_DATA, kDataTrack2,
-                                        {kMediaStream1}, 1, &answer_opts);
-  answer_opts.data_channel_type = cricket::DCT_RTP;
-
   std::unique_ptr<SessionDescription> answer =
       f2_.CreateAnswer(offer.get(), answer_opts, NULL);
 
   ASSERT_TRUE(answer.get() != NULL);
   const ContentInfo* ac = answer->GetContentByName("audio");
   const ContentInfo* vc = answer->GetContentByName("video");
-  const ContentInfo* dc = answer->GetContentByName("data");
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc != NULL);
-  ASSERT_TRUE(dc != NULL);
   const AudioContentDescription* acd = ac->media_description()->as_audio();
   const VideoContentDescription* vcd = vc->media_description()->as_video();
-  const RtpDataContentDescription* dcd = dc->media_description()->as_rtp_data();
   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
   ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
-  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
 
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
@@ -2879,59 +2589,33 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
   EXPECT_EQ(kAutoBandwidth, vcd->bandwidth());  // default bandwidth (auto)
   EXPECT_TRUE(vcd->rtcp_mux());                 // rtcp-mux defaults on
 
-  EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
-  EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
-
-  const StreamParamsVec& data_streams = dcd->streams();
-  ASSERT_EQ(2U, data_streams.size());
-  EXPECT_TRUE(data_streams[0].cname == data_streams[1].cname);
-  EXPECT_EQ(kDataTrack1, data_streams[0].id);
-  ASSERT_EQ(1U, data_streams[0].ssrcs.size());
-  EXPECT_NE(0U, data_streams[0].ssrcs[0]);
-  EXPECT_EQ(kDataTrack2, data_streams[1].id);
-  ASSERT_EQ(1U, data_streams[1].ssrcs.size());
-  EXPECT_NE(0U, data_streams[1].ssrcs[0]);
-
-  EXPECT_EQ(cricket::kRtpDataMaxBandwidth,
-            dcd->bandwidth());   // default bandwidth (auto)
-  EXPECT_TRUE(dcd->rtcp_mux());  // rtcp-mux defaults on
-
   // Update the answer. Add a new video track that is not synched to the
   // other tracks and remove 1 audio track.
   AttachSenderToMediaDescriptionOptions("video", MEDIA_TYPE_VIDEO, kVideoTrack2,
                                         {kMediaStream2}, 1, &answer_opts);
   DetachSenderFromMediaSection("audio", kAudioTrack2, &answer_opts);
-  DetachSenderFromMediaSection("data", kDataTrack2, &answer_opts);
   std::unique_ptr<SessionDescription> updated_answer(
       f2_.CreateAnswer(offer.get(), answer_opts, answer.get()));
 
   ASSERT_TRUE(updated_answer.get() != NULL);
   ac = updated_answer->GetContentByName("audio");
   vc = updated_answer->GetContentByName("video");
-  dc = updated_answer->GetContentByName("data");
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc != NULL);
-  ASSERT_TRUE(dc != NULL);
   const AudioContentDescription* updated_acd =
       ac->media_description()->as_audio();
   const VideoContentDescription* updated_vcd =
       vc->media_description()->as_video();
-  const RtpDataContentDescription* updated_dcd =
-      dc->media_description()->as_rtp_data();
 
   ASSERT_CRYPTO(updated_acd, 1U, kDefaultSrtpCryptoSuite);
   EXPECT_TRUE(CompareCryptoParams(acd->cryptos(), updated_acd->cryptos()));
   ASSERT_CRYPTO(updated_vcd, 1U, kDefaultSrtpCryptoSuite);
   EXPECT_TRUE(CompareCryptoParams(vcd->cryptos(), updated_vcd->cryptos()));
-  ASSERT_CRYPTO(updated_dcd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_TRUE(CompareCryptoParams(dcd->cryptos(), updated_dcd->cryptos()));
 
   EXPECT_EQ(acd->type(), updated_acd->type());
   EXPECT_EQ(acd->codecs(), updated_acd->codecs());
   EXPECT_EQ(vcd->type(), updated_vcd->type());
   EXPECT_EQ(vcd->codecs(), updated_vcd->codecs());
-  EXPECT_EQ(dcd->type(), updated_dcd->type());
-  EXPECT_EQ(dcd->codecs(), updated_dcd->codecs());
 
   const StreamParamsVec& updated_audio_streams = updated_acd->streams();
   ASSERT_EQ(1U, updated_audio_streams.size());
@@ -2943,10 +2627,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
   EXPECT_EQ(kVideoTrack2, updated_video_streams[1].id);
   // All media streams in one PeerConnection share one CNAME.
   EXPECT_EQ(updated_video_streams[1].cname, updated_video_streams[0].cname);
-
-  const StreamParamsVec& updated_data_streams = updated_dcd->streams();
-  ASSERT_EQ(1U, updated_data_streams.size());
-  EXPECT_TRUE(data_streams[0] == updated_data_streams[0]);
 }
 
 // Create an updated offer after creating an answer to the original offer and
@@ -3790,8 +3470,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferAudioCurrent) {
 TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferMultimedia) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
   TestTransportInfo(true, options, false);
 }
 
@@ -3799,16 +3477,12 @@ TEST_F(MediaSessionDescriptionFactoryTest,
        TestTransportInfoOfferMultimediaCurrent) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
   TestTransportInfo(true, options, true);
 }
 
 TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferBundle) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
   options.bundle_enabled = true;
   TestTransportInfo(true, options, false);
 }
@@ -3817,8 +3491,6 @@ TEST_F(MediaSessionDescriptionFactoryTest,
        TestTransportInfoOfferBundleCurrent) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
   options.bundle_enabled = true;
   TestTransportInfo(true, options, true);
 }
@@ -3854,8 +3526,6 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoAnswerMultimedia) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
   TestTransportInfo(false, options, false);
 }
 
@@ -3863,16 +3533,12 @@ TEST_F(MediaSessionDescriptionFactoryTest,
        TestTransportInfoAnswerMultimediaCurrent) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
   TestTransportInfo(false, options, true);
 }
 
 TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoAnswerBundle) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
   options.bundle_enabled = true;
   TestTransportInfo(false, options, false);
 }
@@ -3881,8 +3547,6 @@ TEST_F(MediaSessionDescriptionFactoryTest,
        TestTransportInfoAnswerBundleCurrent) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
   options.bundle_enabled = true;
   TestTransportInfo(false, options, true);
 }
@@ -4072,8 +3736,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoOfferDtlsButNotSdes) {
   tdf2_.set_secure(SEC_ENABLED);
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
-  AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
-                 &options);
 
   // Generate an offer with DTLS but without SDES.
   std::unique_ptr<SessionDescription> offer = f1_.CreateOffer(options, NULL);
@@ -4085,9 +3747,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoOfferDtlsButNotSdes) {
   const VideoContentDescription* video_offer =
       GetFirstVideoContentDescription(offer.get());
   ASSERT_TRUE(video_offer->cryptos().empty());
-  const RtpDataContentDescription* data_offer =
-      GetFirstRtpDataContentDescription(offer.get());
-  ASSERT_TRUE(data_offer->cryptos().empty());
 
   const cricket::TransportDescription* audio_offer_trans_desc =
       offer->GetTransportDescriptionByName("audio");
@@ -4095,9 +3754,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoOfferDtlsButNotSdes) {
   const cricket::TransportDescription* video_offer_trans_desc =
       offer->GetTransportDescriptionByName("video");
   ASSERT_TRUE(video_offer_trans_desc->identity_fingerprint.get() != NULL);
-  const cricket::TransportDescription* data_offer_trans_desc =
-      offer->GetTransportDescriptionByName("data");
-  ASSERT_TRUE(data_offer_trans_desc->identity_fingerprint.get() != NULL);
 
   // Generate an answer with DTLS.
   std::unique_ptr<SessionDescription> answer =
@@ -4110,9 +3766,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoOfferDtlsButNotSdes) {
   const cricket::TransportDescription* video_answer_trans_desc =
       answer->GetTransportDescriptionByName("video");
   EXPECT_TRUE(video_answer_trans_desc->identity_fingerprint.get() != NULL);
-  const cricket::TransportDescription* data_answer_trans_desc =
-      answer->GetTransportDescriptionByName("data");
-  EXPECT_TRUE(data_answer_trans_desc->identity_fingerprint.get() != NULL);
 }
 
 // Verifies if vad_enabled option is set to false, CN codecs are not present in
@@ -4618,12 +4271,10 @@ class MediaProtocolTest : public ::testing::TestWithParam<const char*> {
                          MAKE_VECTOR(kAudioCodecs1));
     f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1),
                          MAKE_VECTOR(kVideoCodecs1));
-    f1_.set_rtp_data_codecs(MAKE_VECTOR(kDataCodecs1));
     f2_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs2),
                          MAKE_VECTOR(kAudioCodecs2));
     f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2),
                          MAKE_VECTOR(kVideoCodecs2));
-    f2_.set_rtp_data_codecs(MAKE_VECTOR(kDataCodecs2));
     f1_.set_secure(SEC_ENABLED);
     f2_.set_secure(SEC_ENABLED);
     tdf1_.set_certificate(rtc::RTCCertificate::Create(

pc/peer_connection.cc

@@ -1302,7 +1302,7 @@ rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
 
   // Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
   // the first SCTP DataChannel.
-  if (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
+  if (first_datachannel) {
     sdp_handler_->UpdateNegotiationNeeded();
   }
   NoteUsageEvent(UsageEvent::DATA_ADDED);
@@ -1768,7 +1768,6 @@ void PeerConnection::Close() {
     // TODO(tommi): ^^ That's not exactly optimal since this is yet another
     // blocking hop to the network thread during Close(). Further still, the
     // voice/video/data channels will be cleared on the worker thread.
-    RTC_DCHECK(!data_channel_controller_.rtp_data_channel());
     transport_controller_.reset();
     port_allocator_->DiscardCandidatePool();
     if (network_thread_safety_) {
@@ -1945,11 +1944,6 @@ void PeerConnection::OnSelectedCandidatePairChanged(
 absl::optional<std::string> PeerConnection::GetDataMid() const {
   RTC_DCHECK_RUN_ON(signaling_thread());
   switch (data_channel_type()) {
-    case cricket::DCT_RTP:
-      if (!data_channel_controller_.rtp_data_channel()) {
-        return absl::nullopt;
-      }
-      return data_channel_controller_.rtp_data_channel()->content_name();
     case cricket::DCT_SCTP:
       return sctp_mid_s_;
     default:
@@ -2106,10 +2100,6 @@ cricket::ChannelInterface* PeerConnection::GetChannel(
       return channel;
     }
   }
-  if (rtp_data_channel() &&
-      rtp_data_channel()->content_name() == content_name) {
-    return rtp_data_channel();
-  }
   return nullptr;
 }
 
@@ -2211,11 +2201,6 @@ std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
           channel->transport_name();
     }
   }
-  if (data_channel_controller_.rtp_data_channel()) {
-    transport_names_by_mid[data_channel_controller_.rtp_data_channel()
-                               ->content_name()] =
-        data_channel_controller_.rtp_data_channel()->transport_name();
-  }
   if (sctp_mid_n_) {
     cricket::DtlsTransportInternal* dtls_transport =
         transport_controller_->GetDtlsTransport(*sctp_mid_n_);
@@ -2448,22 +2433,7 @@ bool PeerConnection::SetupDataChannelTransport_n(const std::string& mid) {
   return true;
 }
 
-void PeerConnection::SetupRtpDataChannelTransport_n(
-    cricket::RtpDataChannel* data_channel) {
-  data_channel_controller_.set_rtp_data_channel(data_channel);
-  if (!data_channel)
-    return;
-
-  // TODO(bugs.webrtc.org/9987): OnSentPacket_w needs to be changed to
-  // OnSentPacket_n (and be called on the network thread).
-  data_channel->SignalSentPacket().connect(this,
-                                           &PeerConnection::OnSentPacket_w);
-}
-
 void PeerConnection::TeardownDataChannelTransport_n() {
-  // Clear the RTP data channel if any.
-  data_channel_controller_.set_rtp_data_channel(nullptr);
-
   if (sctp_mid_n_) {
     // |sctp_mid_| may still be active through an SCTP transport.  If not, unset
     // it.
@@ -2707,11 +2677,6 @@ void PeerConnection::ReportTransportStats() {
     }
   }
 
-  if (rtp_data_channel()) {
-    media_types_by_transport_name[rtp_data_channel()->transport_name()].insert(
-        cricket::MEDIA_TYPE_DATA);
-  }
-
   if (sctp_mid_n_) {
     cricket::DtlsTransportInternal* dtls_transport =
         transport_controller_->GetDtlsTransport(*sctp_mid_n_);

pc/peer_connection.h

@@ -72,7 +72,6 @@
 #include "pc/peer_connection_internal.h"
 #include "pc/peer_connection_message_handler.h"
 #include "pc/rtc_stats_collector.h"
-#include "pc/rtp_data_channel.h"
 #include "pc/rtp_receiver.h"
 #include "pc/rtp_sender.h"
 #include "pc/rtp_transceiver.h"
@@ -287,18 +286,10 @@ class PeerConnection : public PeerConnectionInternal,
     return rtp_manager()->transceivers()->List();
   }
 
-  sigslot::signal1<RtpDataChannel*>& SignalRtpDataChannelCreated() override {
-    return data_channel_controller_.SignalRtpDataChannelCreated();
-  }
-
   sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() override {
     return data_channel_controller_.SignalSctpDataChannelCreated();
   }
 
-  cricket::RtpDataChannel* rtp_data_channel() const override {
-    return data_channel_controller_.rtp_data_channel();
-  }
-
   std::vector<DataChannelStats> GetDataChannelStats() const override;
 
   absl::optional<std::string> sctp_transport_name() const override;
@@ -404,8 +395,8 @@ class PeerConnection : public PeerConnectionInternal,
   }
   bool ValidateBundleSettings(const cricket::SessionDescription* desc);
 
-  // Returns the MID for the data section associated with either the
+  // Returns the MID for the data section associated with the
-  // RtpDataChannel or SCTP data channel, if it has been set. If no data
+  // SCTP data channel, if it has been set. If no data
   // channels are configured this will return nullopt.
   absl::optional<std::string> GetDataMid() const;
 
@@ -443,8 +434,6 @@ class PeerConnection : public PeerConnectionInternal,
 
   bool SetupDataChannelTransport_n(const std::string& mid)
       RTC_RUN_ON(network_thread());
-  void SetupRtpDataChannelTransport_n(cricket::RtpDataChannel* data_channel)
-      RTC_RUN_ON(network_thread());
   void TeardownDataChannelTransport_n() RTC_RUN_ON(network_thread());
   cricket::ChannelInterface* GetChannel(const std::string& content_name);
 

pc/peer_connection_data_channel_unittest.cc

@@ -330,8 +330,9 @@ TEST_P(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
 
   auto answer = callee->CreateAnswer();
   ChangeSctpPortOnDescription(answer->description(), kNewRecvPort);
+  std::string sdp;
+  answer->ToString(&sdp);
   ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
-
   auto* callee_transport =
       callee->sctp_transport_factory()->last_fake_sctp_transport();
   ASSERT_TRUE(callee_transport);

pc/peer_connection_internal.h

@@ -19,7 +19,6 @@
 
 #include "api/peer_connection_interface.h"
 #include "call/call.h"
-#include "pc/rtp_data_channel.h"
 #include "pc/rtp_transceiver.h"
 #include "pc/sctp_data_channel.h"
 
@@ -41,13 +40,9 @@ class PeerConnectionInternal : public PeerConnectionInterface {
       rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
   GetTransceiversInternal() const = 0;
 
-  virtual sigslot::signal1<RtpDataChannel*>& SignalRtpDataChannelCreated() = 0;
   virtual sigslot::signal1<SctpDataChannel*>&
   SignalSctpDataChannelCreated() = 0;
 
-  // Only valid when using deprecated RTP data channels.
-  virtual cricket::RtpDataChannel* rtp_data_channel() const = 0;
-
   // Call on the network thread to fetch stats for all the data channels.
   // TODO(tommi): Make pure virtual after downstream updates.
   virtual std::vector<DataChannelStats> GetDataChannelStats() const {

pc/rtc_stats_collector.cc

@@ -1118,8 +1118,6 @@ RTCStatsCollector::RTCStatsCollector(PeerConnectionInternal* pc,
   RTC_DCHECK(worker_thread_);
   RTC_DCHECK(network_thread_);
   RTC_DCHECK_GE(cache_lifetime_us_, 0);
-  pc_->SignalRtpDataChannelCreated().connect(
-      this, &RTCStatsCollector::OnRtpDataChannelCreated);
   pc_->SignalSctpDataChannelCreated().connect(
       this, &RTCStatsCollector::OnSctpDataChannelCreated);
 }
@@ -2150,20 +2148,12 @@ std::set<std::string> RTCStatsCollector::PrepareTransportNames_s() const {
           transceiver->internal()->channel()->transport_name());
     }
   }
-  if (pc_->rtp_data_channel()) {
-    transport_names.insert(pc_->rtp_data_channel()->transport_name());
-  }
   if (pc_->sctp_transport_name()) {
     transport_names.insert(*pc_->sctp_transport_name());
   }
   return transport_names;
 }
 
-void RTCStatsCollector::OnRtpDataChannelCreated(RtpDataChannel* channel) {
-  channel->SignalOpened.connect(this, &RTCStatsCollector::OnDataChannelOpened);
-  channel->SignalClosed.connect(this, &RTCStatsCollector::OnDataChannelClosed);
-}
-
 void RTCStatsCollector::OnSctpDataChannelCreated(SctpDataChannel* channel) {
   channel->SignalOpened.connect(this, &RTCStatsCollector::OnDataChannelOpened);
   channel->SignalClosed.connect(this, &RTCStatsCollector::OnDataChannelClosed);

pc/rtc_stats_collector.h

@@ -29,7 +29,6 @@
 #include "media/base/media_channel.h"
 #include "pc/data_channel_utils.h"
 #include "pc/peer_connection_internal.h"
-#include "pc/rtp_data_channel.h"
 #include "pc/rtp_receiver.h"
 #include "pc/rtp_sender.h"
 #include "pc/rtp_transceiver.h"
@@ -239,7 +238,6 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
   void MergeNetworkReport_s();
 
   // Slots for signals (sigslot) that are wired up to |pc_|.
-  void OnRtpDataChannelCreated(RtpDataChannel* channel);
   void OnSctpDataChannelCreated(SctpDataChannel* channel);
   // Slots for signals (sigslot) that are wired up to |channel|.
   void OnDataChannelOpened(DataChannelInterface* channel);

pc/rtp_data_channel.cc

@@ -1,394 +0,0 @@
-/*
- *  Copyright 2020 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 "pc/rtp_data_channel.h"
-
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "api/proxy.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/location.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/ref_counted_object.h"
-#include "rtc_base/thread.h"
-
-namespace webrtc {
-
-namespace {
-
-static size_t kMaxQueuedReceivedDataBytes = 16 * 1024 * 1024;
-
-static std::atomic<int> g_unique_id{0};
-
-int GenerateUniqueId() {
-  return ++g_unique_id;
-}
-
-// Define proxy for DataChannelInterface.
-BEGIN_PRIMARY_PROXY_MAP(DataChannel)
-PROXY_PRIMARY_THREAD_DESTRUCTOR()
-PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*)
-PROXY_METHOD0(void, UnregisterObserver)
-BYPASS_PROXY_CONSTMETHOD0(std::string, label)
-BYPASS_PROXY_CONSTMETHOD0(bool, reliable)
-BYPASS_PROXY_CONSTMETHOD0(bool, ordered)
-BYPASS_PROXY_CONSTMETHOD0(uint16_t, maxRetransmitTime)
-BYPASS_PROXY_CONSTMETHOD0(uint16_t, maxRetransmits)
-BYPASS_PROXY_CONSTMETHOD0(absl::optional<int>, maxRetransmitsOpt)
-BYPASS_PROXY_CONSTMETHOD0(absl::optional<int>, maxPacketLifeTime)
-BYPASS_PROXY_CONSTMETHOD0(std::string, protocol)
-BYPASS_PROXY_CONSTMETHOD0(bool, negotiated)
-// Can't bypass the proxy since the id may change.
-PROXY_CONSTMETHOD0(int, id)
-BYPASS_PROXY_CONSTMETHOD0(Priority, priority)
-PROXY_CONSTMETHOD0(DataState, state)
-PROXY_CONSTMETHOD0(RTCError, error)
-PROXY_CONSTMETHOD0(uint32_t, messages_sent)
-PROXY_CONSTMETHOD0(uint64_t, bytes_sent)
-PROXY_CONSTMETHOD0(uint32_t, messages_received)
-PROXY_CONSTMETHOD0(uint64_t, bytes_received)
-PROXY_CONSTMETHOD0(uint64_t, buffered_amount)
-PROXY_METHOD0(void, Close)
-// TODO(bugs.webrtc.org/11547): Change to run on the network thread.
-PROXY_METHOD1(bool, Send, const DataBuffer&)
-END_PROXY_MAP()
-
-}  // namespace
-
-rtc::scoped_refptr<RtpDataChannel> RtpDataChannel::Create(
-    RtpDataChannelProviderInterface* provider,
-    const std::string& label,
-    const DataChannelInit& config,
-    rtc::Thread* signaling_thread) {
-  rtc::scoped_refptr<RtpDataChannel> channel(
-      new rtc::RefCountedObject<RtpDataChannel>(config, provider, label,
-                                                signaling_thread));
-  if (!channel->Init()) {
-    return nullptr;
-  }
-  return channel;
-}
-
-// static
-rtc::scoped_refptr<DataChannelInterface> RtpDataChannel::CreateProxy(
-    rtc::scoped_refptr<RtpDataChannel> channel) {
-  return DataChannelProxy::Create(channel->signaling_thread_, channel.get());
-}
-
-RtpDataChannel::RtpDataChannel(const DataChannelInit& config,
-                               RtpDataChannelProviderInterface* provider,
-                               const std::string& label,
-                               rtc::Thread* signaling_thread)
-    : signaling_thread_(signaling_thread),
-      internal_id_(GenerateUniqueId()),
-      label_(label),
-      config_(config),
-      provider_(provider) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-}
-
-bool RtpDataChannel::Init() {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  if (config_.reliable || config_.id != -1 || config_.maxRetransmits ||
-      config_.maxRetransmitTime) {
-    RTC_LOG(LS_ERROR) << "Failed to initialize the RTP data channel due to "
-                         "invalid DataChannelInit.";
-    return false;
-  }
-
-  return true;
-}
-
-RtpDataChannel::~RtpDataChannel() {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-}
-
-void RtpDataChannel::RegisterObserver(DataChannelObserver* observer) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  observer_ = observer;
-  DeliverQueuedReceivedData();
-}
-
-void RtpDataChannel::UnregisterObserver() {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  observer_ = nullptr;
-}
-
-void RtpDataChannel::Close() {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  if (state_ == kClosed)
-    return;
-  send_ssrc_ = 0;
-  send_ssrc_set_ = false;
-  SetState(kClosing);
-  UpdateState();
-}
-
-RtpDataChannel::DataState RtpDataChannel::state() const {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  return state_;
-}
-
-RTCError RtpDataChannel::error() const {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  return error_;
-}
-
-uint32_t RtpDataChannel::messages_sent() const {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  return messages_sent_;
-}
-
-uint64_t RtpDataChannel::bytes_sent() const {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  return bytes_sent_;
-}
-
-uint32_t RtpDataChannel::messages_received() const {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  return messages_received_;
-}
-
-uint64_t RtpDataChannel::bytes_received() const {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  return bytes_received_;
-}
-
-bool RtpDataChannel::Send(const DataBuffer& buffer) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-
-  if (state_ != kOpen) {
-    return false;
-  }
-
-  // TODO(jiayl): the spec is unclear about if the remote side should get the
-  // onmessage event. We need to figure out the expected behavior and change the
-  // code accordingly.
-  if (buffer.size() == 0) {
-    return true;
-  }
-
-  return SendDataMessage(buffer);
-}
-
-void RtpDataChannel::SetReceiveSsrc(uint32_t receive_ssrc) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-
-  if (receive_ssrc_set_) {
-    return;
-  }
-  receive_ssrc_ = receive_ssrc;
-  receive_ssrc_set_ = true;
-  UpdateState();
-}
-
-void RtpDataChannel::OnTransportChannelClosed() {
-  RTCError error = RTCError(RTCErrorType::OPERATION_ERROR_WITH_DATA,
-                            "Transport channel closed");
-  CloseAbruptlyWithError(std::move(error));
-}
-
-DataChannelStats RtpDataChannel::GetStats() const {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  DataChannelStats stats{internal_id_,        id(),         label(),
-                         protocol(),          state(),      messages_sent(),
-                         messages_received(), bytes_sent(), bytes_received()};
-  return stats;
-}
-
-// The remote peer request that this channel shall be closed.
-void RtpDataChannel::RemotePeerRequestClose() {
-  // Close with error code explicitly set to OK.
-  CloseAbruptlyWithError(RTCError());
-}
-
-void RtpDataChannel::SetSendSsrc(uint32_t send_ssrc) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  if (send_ssrc_set_) {
-    return;
-  }
-  send_ssrc_ = send_ssrc;
-  send_ssrc_set_ = true;
-  UpdateState();
-}
-
-void RtpDataChannel::OnDataReceived(const cricket::ReceiveDataParams& params,
-                                    const rtc::CopyOnWriteBuffer& payload) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  if (params.ssrc != receive_ssrc_) {
-    return;
-  }
-
-  RTC_DCHECK(params.type == cricket::DMT_BINARY ||
-             params.type == cricket::DMT_TEXT);
-
-  RTC_LOG(LS_VERBOSE) << "DataChannel received DATA message, sid = "
-                      << params.sid;
-
-  bool binary = (params.type == cricket::DMT_BINARY);
-  auto buffer = std::make_unique<DataBuffer>(payload, binary);
-  if (state_ == kOpen && observer_) {
-    ++messages_received_;
-    bytes_received_ += buffer->size();
-    observer_->OnMessage(*buffer.get());
-  } else {
-    if (queued_received_data_.byte_count() + payload.size() >
-        kMaxQueuedReceivedDataBytes) {
-      RTC_LOG(LS_ERROR) << "Queued received data exceeds the max buffer size.";
-
-      queued_received_data_.Clear();
-      CloseAbruptlyWithError(
-          RTCError(RTCErrorType::RESOURCE_EXHAUSTED,
-                   "Queued received data exceeds the max buffer size."));
-
-      return;
-    }
-    queued_received_data_.PushBack(std::move(buffer));
-  }
-}
-
-void RtpDataChannel::OnChannelReady(bool writable) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-
-  writable_ = writable;
-  if (!writable) {
-    return;
-  }
-
-  UpdateState();
-}
-
-void RtpDataChannel::CloseAbruptlyWithError(RTCError error) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-
-  if (state_ == kClosed) {
-    return;
-  }
-
-  if (connected_to_provider_) {
-    DisconnectFromProvider();
-  }
-
-  // Still go to "kClosing" before "kClosed", since observers may be expecting
-  // that.
-  SetState(kClosing);
-  error_ = std::move(error);
-  SetState(kClosed);
-}
-
-void RtpDataChannel::UpdateState() {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  // UpdateState determines what to do from a few state variables.  Include
-  // all conditions required for each state transition here for
-  // clarity.
-  switch (state_) {
-    case kConnecting: {
-      if (send_ssrc_set_ == receive_ssrc_set_) {
-        if (!connected_to_provider_) {
-          connected_to_provider_ = provider_->ConnectDataChannel(this);
-        }
-        if (connected_to_provider_ && writable_) {
-          SetState(kOpen);
-          // If we have received buffers before the channel got writable.
-          // Deliver them now.
-          DeliverQueuedReceivedData();
-        }
-      }
-      break;
-    }
-    case kOpen: {
-      break;
-    }
-    case kClosing: {
-      // For RTP data channels, we can go to "closed" after we finish
-      // sending data and the send/recv SSRCs are unset.
-      if (connected_to_provider_) {
-        DisconnectFromProvider();
-      }
-      if (!send_ssrc_set_ && !receive_ssrc_set_) {
-        SetState(kClosed);
-      }
-      break;
-    }
-    case kClosed:
-      break;
-  }
-}
-
-void RtpDataChannel::SetState(DataState state) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  if (state_ == state) {
-    return;
-  }
-
-  state_ = state;
-  if (observer_) {
-    observer_->OnStateChange();
-  }
-  if (state_ == kOpen) {
-    SignalOpened(this);
-  } else if (state_ == kClosed) {
-    SignalClosed(this);
-  }
-}
-
-void RtpDataChannel::DisconnectFromProvider() {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  if (!connected_to_provider_)
-    return;
-
-  provider_->DisconnectDataChannel(this);
-  connected_to_provider_ = false;
-}
-
-void RtpDataChannel::DeliverQueuedReceivedData() {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  if (!observer_) {
-    return;
-  }
-
-  while (!queued_received_data_.Empty()) {
-    std::unique_ptr<DataBuffer> buffer = queued_received_data_.PopFront();
-    ++messages_received_;
-    bytes_received_ += buffer->size();
-    observer_->OnMessage(*buffer);
-  }
-}
-
-bool RtpDataChannel::SendDataMessage(const DataBuffer& buffer) {
-  RTC_DCHECK_RUN_ON(signaling_thread_);
-  cricket::SendDataParams send_params;
-
-  send_params.ssrc = send_ssrc_;
-  send_params.type = buffer.binary ? cricket::DMT_BINARY : cricket::DMT_TEXT;
-
-  cricket::SendDataResult send_result = cricket::SDR_SUCCESS;
-  bool success = provider_->SendData(send_params, buffer.data, &send_result);
-
-  if (success) {
-    ++messages_sent_;
-    bytes_sent_ += buffer.size();
-    if (observer_ && buffer.size() > 0) {
-      observer_->OnBufferedAmountChange(buffer.size());
-    }
-    return true;
-  }
-
-  return false;
-}
-
-// static
-void RtpDataChannel::ResetInternalIdAllocatorForTesting(int new_value) {
-  g_unique_id = new_value;
-}
-
-}  // namespace webrtc

pc/rtp_data_channel.h

@@ -1,204 +0,0 @@
-/*
- *  Copyright 2020 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.
- */
-
-#ifndef PC_RTP_DATA_CHANNEL_H_
-#define PC_RTP_DATA_CHANNEL_H_
-
-#include <stdint.h>
-
-#include <memory>
-#include <string>
-
-#include "absl/types/optional.h"
-#include "api/data_channel_interface.h"
-#include "api/priority.h"
-#include "api/rtc_error.h"
-#include "api/scoped_refptr.h"
-#include "api/transport/data_channel_transport_interface.h"
-#include "media/base/media_channel.h"
-#include "pc/channel.h"
-#include "pc/data_channel_utils.h"
-#include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/third_party/sigslot/sigslot.h"
-#include "rtc_base/thread.h"
-#include "rtc_base/thread_annotations.h"
-
-namespace webrtc {
-
-class RtpDataChannel;
-
-// TODO(deadbeef): Once RTP data channels go away, get rid of this and have
-// DataChannel depend on SctpTransportInternal (pure virtual SctpTransport
-// interface) instead.
-class RtpDataChannelProviderInterface {
- public:
-  // Sends the data to the transport.
-  virtual bool SendData(const cricket::SendDataParams& params,
-                        const rtc::CopyOnWriteBuffer& payload,
-                        cricket::SendDataResult* result) = 0;
-  // Connects to the transport signals.
-  virtual bool ConnectDataChannel(RtpDataChannel* data_channel) = 0;
-  // Disconnects from the transport signals.
-  virtual void DisconnectDataChannel(RtpDataChannel* data_channel) = 0;
-  // Returns true if the transport channel is ready to send data.
-  virtual bool ReadyToSendData() const = 0;
-
- protected:
-  virtual ~RtpDataChannelProviderInterface() {}
-};
-
-// RtpDataChannel is an implementation of the DataChannelInterface based on
-// libjingle's data engine. It provides an implementation of unreliable data
-// channels.
-
-// DataChannel states:
-// kConnecting: The channel has been created the transport might not yet be
-//              ready.
-// kOpen: The channel have a local SSRC set by a call to UpdateSendSsrc
-//        and a remote SSRC set by call to UpdateReceiveSsrc and the transport
-//        has been writable once.
-// kClosing: DataChannelInterface::Close has been called or UpdateReceiveSsrc
-//           has been called with SSRC==0
-// kClosed: Both UpdateReceiveSsrc and UpdateSendSsrc has been called with
-//          SSRC==0.
-class RtpDataChannel : public DataChannelInterface,
-                       public sigslot::has_slots<> {
- public:
-  static rtc::scoped_refptr<RtpDataChannel> Create(
-      RtpDataChannelProviderInterface* provider,
-      const std::string& label,
-      const DataChannelInit& config,
-      rtc::Thread* signaling_thread);
-
-  // Instantiates an API proxy for a DataChannel instance that will be handed
-  // out to external callers.
-  static rtc::scoped_refptr<DataChannelInterface> CreateProxy(
-      rtc::scoped_refptr<RtpDataChannel> channel);
-
-  void RegisterObserver(DataChannelObserver* observer) override;
-  void UnregisterObserver() override;
-
-  std::string label() const override { return label_; }
-  bool reliable() const override { return false; }
-  bool ordered() const override { return config_.ordered; }
-  // Backwards compatible accessors
-  uint16_t maxRetransmitTime() const override {
-    return config_.maxRetransmitTime ? *config_.maxRetransmitTime
-                                     : static_cast<uint16_t>(-1);
-  }
-  uint16_t maxRetransmits() const override {
-    return config_.maxRetransmits ? *config_.maxRetransmits
-                                  : static_cast<uint16_t>(-1);
-  }
-  absl::optional<int> maxPacketLifeTime() const override {
-    return config_.maxRetransmitTime;
-  }
-  absl::optional<int> maxRetransmitsOpt() const override {
-    return config_.maxRetransmits;
-  }
-  std::string protocol() const override { return config_.protocol; }
-  bool negotiated() const override { return config_.negotiated; }
-  int id() const override { return config_.id; }
-  Priority priority() const override {
-    return config_.priority ? *config_.priority : Priority::kLow;
-  }
-
-  virtual int internal_id() const { return internal_id_; }
-
-  uint64_t buffered_amount() const override { return 0; }
-  void Close() override;
-  DataState state() const override;
-  RTCError error() const override;
-  uint32_t messages_sent() const override;
-  uint64_t bytes_sent() const override;
-  uint32_t messages_received() const override;
-  uint64_t bytes_received() const override;
-  bool Send(const DataBuffer& buffer) override;
-
-  // Close immediately, ignoring any queued data or closing procedure.
-  // This is called when SDP indicates a channel should be removed.
-  void CloseAbruptlyWithError(RTCError error);
-
-  // Called when the channel's ready to use.  That can happen when the
-  // underlying DataMediaChannel becomes ready, or when this channel is a new
-  // stream on an existing DataMediaChannel, and we've finished negotiation.
-  void OnChannelReady(bool writable);
-
-  // Slots for provider to connect signals to.
-  void OnDataReceived(const cricket::ReceiveDataParams& params,
-                      const rtc::CopyOnWriteBuffer& payload);
-
-  // Called when the transport channel is unusable.
-  // This method makes sure the DataChannel is disconnected and changes state
-  // to kClosed.
-  void OnTransportChannelClosed();
-
-  DataChannelStats GetStats() const;
-
-  // The remote peer requested that this channel should be closed.
-  void RemotePeerRequestClose();
-  // Set the SSRC this channel should use to send data on the
-  // underlying data engine. |send_ssrc| == 0 means that the channel is no
-  // longer part of the session negotiation.
-  void SetSendSsrc(uint32_t send_ssrc);
-  // Set the SSRC this channel should use to receive data from the
-  // underlying data engine.
-  void SetReceiveSsrc(uint32_t receive_ssrc);
-
-  // Emitted when state transitions to kOpen.
-  sigslot::signal1<DataChannelInterface*> SignalOpened;
-  // Emitted when state transitions to kClosed.
-  sigslot::signal1<DataChannelInterface*> SignalClosed;
-
-  // Reset the allocator for internal ID values for testing, so that
-  // the internal IDs generated are predictable. Test only.
-  static void ResetInternalIdAllocatorForTesting(int new_value);
-
- protected:
-  RtpDataChannel(const DataChannelInit& config,
-                 RtpDataChannelProviderInterface* client,
-                 const std::string& label,
-                 rtc::Thread* signaling_thread);
-  ~RtpDataChannel() override;
-
- private:
-  bool Init();
-  void UpdateState();
-  void SetState(DataState state);
-  void DisconnectFromProvider();
-
-  void DeliverQueuedReceivedData();
-
-  bool SendDataMessage(const DataBuffer& buffer);
-
-  rtc::Thread* const signaling_thread_;
-  const int internal_id_;
-  const std::string label_;
-  const DataChannelInit config_;
-  DataChannelObserver* observer_ RTC_GUARDED_BY(signaling_thread_) = nullptr;
-  DataState state_ RTC_GUARDED_BY(signaling_thread_) = kConnecting;
-  RTCError error_ RTC_GUARDED_BY(signaling_thread_);
-  uint32_t messages_sent_ RTC_GUARDED_BY(signaling_thread_) = 0;
-  uint64_t bytes_sent_ RTC_GUARDED_BY(signaling_thread_) = 0;
-  uint32_t messages_received_ RTC_GUARDED_BY(signaling_thread_) = 0;
-  uint64_t bytes_received_ RTC_GUARDED_BY(signaling_thread_) = 0;
-  RtpDataChannelProviderInterface* const provider_;
-  bool connected_to_provider_ RTC_GUARDED_BY(signaling_thread_) = false;
-  bool send_ssrc_set_ RTC_GUARDED_BY(signaling_thread_) = false;
-  bool receive_ssrc_set_ RTC_GUARDED_BY(signaling_thread_) = false;
-  bool writable_ RTC_GUARDED_BY(signaling_thread_) = false;
-  uint32_t send_ssrc_ RTC_GUARDED_BY(signaling_thread_) = 0;
-  uint32_t receive_ssrc_ RTC_GUARDED_BY(signaling_thread_) = 0;
-  PacketQueue queued_received_data_ RTC_GUARDED_BY(signaling_thread_);
-};
-
-}  // namespace webrtc
-
-#endif  // PC_RTP_DATA_CHANNEL_H_

pc/rtp_sender_receiver_unittest.cc

@@ -37,7 +37,6 @@
 #include "media/base/media_channel.h"
 #include "media/base/media_config.h"
 #include "media/base/media_engine.h"
-#include "media/base/rtp_data_engine.h"
 #include "media/base/stream_params.h"
 #include "media/base/test_utils.h"
 #include "media/engine/fake_webrtc_call.h"
@@ -112,8 +111,7 @@ class RtpSenderReceiverTest
         local_stream_(MediaStream::Create(kStreamId1)) {
     worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
       channel_manager_ = cricket::ChannelManager::Create(
-          absl::WrapUnique(media_engine_),
+          absl::WrapUnique(media_engine_), false, worker_thread_,
-          std::make_unique<cricket::RtpDataEngine>(), false, worker_thread_,
           network_thread_);
     });
 

pc/rtp_transceiver_unittest.cc

@@ -84,7 +84,6 @@ class RtpTransceiverUnifiedPlanTest : public ::testing::Test {
   RtpTransceiverUnifiedPlanTest()
       : channel_manager_(cricket::ChannelManager::Create(
             std::make_unique<cricket::FakeMediaEngine>(),
-            std::make_unique<cricket::FakeDataEngine>(),
             false,
             rtc::Thread::Current(),
             rtc::Thread::Current())),
@@ -121,7 +120,6 @@ class RtpTransceiverTestForHeaderExtensions : public ::testing::Test {
   RtpTransceiverTestForHeaderExtensions()
       : channel_manager_(cricket::ChannelManager::Create(
             std::make_unique<cricket::FakeMediaEngine>(),
-            std::make_unique<cricket::FakeDataEngine>(),
             false,
             rtc::Thread::Current(),
             rtc::Thread::Current())),

pc/sdp_offer_answer.cc

@@ -43,7 +43,6 @@
 #include "pc/media_stream.h"
 #include "pc/peer_connection.h"
 #include "pc/peer_connection_message_handler.h"
-#include "pc/rtp_data_channel.h"
 #include "pc/rtp_media_utils.h"
 #include "pc/rtp_sender.h"
 #include "pc/rtp_transport_internal.h"
@@ -693,27 +692,6 @@ std::string GenerateRtcpCname() {
   return cname;
 }
 
-// Add options to |session_options| from |rtp_data_channels|.
-void AddRtpDataChannelOptions(
-    const std::map<std::string, rtc::scoped_refptr<RtpDataChannel>>&
-        rtp_data_channels,
-    cricket::MediaDescriptionOptions* data_media_description_options) {
-  if (!data_media_description_options) {
-    return;
-  }
-  // Check for data channels.
-  for (const auto& kv : rtp_data_channels) {
-    const RtpDataChannel* channel = kv.second;
-    if (channel->state() == RtpDataChannel::kConnecting ||
-        channel->state() == RtpDataChannel::kOpen) {
-      // Legacy RTP data channels are signaled with the track/stream ID set to
-      // the data channel's label.
-      data_media_description_options->AddRtpDataChannel(channel->label(),
-                                                        channel->label());
-    }
-  }
-}
-
 // Check if we can send |new_stream| on a PeerConnection.
 bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
                             webrtc::MediaStreamInterface* new_stream) {
@@ -1464,17 +1442,7 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
     }
   }
 
-  const cricket::ContentInfo* data_content =
+  // This function does nothing with data content.
-      GetFirstDataContent(local_description()->description());
-  if (data_content) {
-    const cricket::RtpDataContentDescription* rtp_data_desc =
-        data_content->media_description()->as_rtp_data();
-    // rtp_data_desc will be null if this is an SCTP description.
-    if (rtp_data_desc) {
-      data_channel_controller()->UpdateLocalRtpDataChannels(
-          rtp_data_desc->streams());
-    }
-  }
 
   if (type == SdpType::kAnswer &&
       local_ice_credentials_to_replace_->SatisfiesIceRestart(
@@ -1802,8 +1770,6 @@ RTCError SdpOfferAnswerHandler::ApplyRemoteDescription(
       GetFirstAudioContentDescription(remote_description()->description());
   const cricket::VideoContentDescription* video_desc =
       GetFirstVideoContentDescription(remote_description()->description());
-  const cricket::RtpDataContentDescription* rtp_data_desc =
-      GetFirstRtpDataContentDescription(remote_description()->description());
 
   // Check if the descriptions include streams, just in case the peer supports
   // MSID, but doesn't indicate so with "a=msid-semantic".
@@ -1856,13 +1822,6 @@ RTCError SdpOfferAnswerHandler::ApplyRemoteDescription(
       }
     }
 
-    // If this is an RTP data transport, update the DataChannels with the
-    // information from the remote peer.
-    if (rtp_data_desc) {
-      data_channel_controller()->UpdateRemoteRtpDataChannels(
-          GetActiveStreams(rtp_data_desc));
-    }
-
     // Iterate new_streams and notify the observer about new MediaStreams.
     auto observer = pc_->Observer();
     for (size_t i = 0; i < new_streams->count(); ++i) {
@@ -2515,7 +2474,6 @@ RTCError SdpOfferAnswerHandler::UpdateSessionState(
     RTC_DCHECK(type == SdpType::kAnswer);
     ChangeSignalingState(PeerConnectionInterface::kStable);
     transceivers()->DiscardStableStates();
-    have_pending_rtp_data_channel_ = false;
   }
 
   // Update internal objects according to the session description's media
@@ -2739,10 +2697,6 @@ RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
     transceiver->internal()->set_mline_index(state.mline_index());
   }
   transport_controller()->RollbackTransports();
-  if (have_pending_rtp_data_channel_) {
-    DestroyDataChannelTransport();
-    have_pending_rtp_data_channel_ = false;
-  }
   transceivers()->DiscardStableStates();
   pending_local_description_.reset();
   pending_remote_description_.reset();
@@ -3407,21 +3361,13 @@ RTCError SdpOfferAnswerHandler::UpdateDataChannel(
     RTC_LOG(LS_INFO) << "Rejected data channel, mid=" << content.mid();
     DestroyDataChannelTransport();
   } else {
-    if (!data_channel_controller()->rtp_data_channel() &&
+    if (!data_channel_controller()->data_channel_transport()) {
-        !data_channel_controller()->data_channel_transport()) {
       RTC_LOG(LS_INFO) << "Creating data channel, mid=" << content.mid();
       if (!CreateDataChannel(content.name)) {
         LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
                              "Failed to create data channel.");
       }
     }
-    if (source == cricket::CS_REMOTE) {
-      const MediaContentDescription* data_desc = content.media_description();
-      if (data_desc && cricket::IsRtpProtocol(data_desc->protocol())) {
-        data_channel_controller()->UpdateRemoteRtpDataChannels(
-            GetActiveStreams(data_desc));
-      }
-    }
   }
   return RTCError::OK();
 }
@@ -3548,15 +3494,7 @@ void SdpOfferAnswerHandler::GetOptionsForOffer(
     GetOptionsForPlanBOffer(offer_answer_options, session_options);
   }
 
-  // Intentionally unset the data channel type for RTP data channel with the
+  session_options->data_channel_type = pc_->data_channel_type();
-  // second condition. Otherwise the RTP data channels would be successfully
-  // negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
-  // when building with chromium. We want to leave RTP data channels broken, so
-  // people won't try to use them.
-  if (data_channel_controller()->HasRtpDataChannels() ||
-      pc_->data_channel_type() != cricket::DCT_RTP) {
-    session_options->data_channel_type = pc_->data_channel_type();
-  }
 
   // Apply ICE restart flag and renomination flag.
   bool ice_restart = offer_answer_options.ice_restart || HasNewIceCredentials();
@@ -3815,14 +3753,7 @@ void SdpOfferAnswerHandler::GetOptionsForAnswer(
     GetOptionsForPlanBAnswer(offer_answer_options, session_options);
   }
 
-  // Intentionally unset the data channel type for RTP data channel. Otherwise
+  session_options->data_channel_type = pc_->data_channel_type();
-  // the RTP data channels would be successfully negotiated by default and the
-  // unit tests in WebRtcDataBrowserTest will fail when building with chromium.
-  // We want to leave RTP data channels broken, so people won't try to use them.
-  if (data_channel_controller()->HasRtpDataChannels() ||
-      pc_->data_channel_type() != cricket::DCT_RTP) {
-    session_options->data_channel_type = pc_->data_channel_type();
-  }
 
   // Apply ICE renomination flag.
   for (auto& options : session_options->media_description_options) {
@@ -4234,11 +4165,6 @@ void SdpOfferAnswerHandler::EnableSending() {
       channel->Enable(true);
     }
   }
-
-  if (data_channel_controller()->rtp_data_channel() &&
-      !data_channel_controller()->rtp_data_channel()->enabled()) {
-    data_channel_controller()->rtp_data_channel()->Enable(true);
-  }
 }
 
 RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
@@ -4280,29 +4206,6 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
     }
   }
 
-  // If using the RtpDataChannel, push down the new SDP section for it too.
-  if (data_channel_controller()->rtp_data_channel()) {
-    const ContentInfo* data_content =
-        cricket::GetFirstDataContent(sdesc->description());
-    if (data_content && !data_content->rejected) {
-      const MediaContentDescription* data_desc =
-          data_content->media_description();
-      if (data_desc) {
-        std::string error;
-        bool success = (source == cricket::CS_LOCAL)
-                           ? data_channel_controller()
-                                 ->rtp_data_channel()
-                                 ->SetLocalContent(data_desc, type, &error)
-                           : data_channel_controller()
-                                 ->rtp_data_channel()
-                                 ->SetRemoteContent(data_desc, type, &error);
-        if (!success) {
-          LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
-        }
-      }
-    }
-  }
-
   // Need complete offer/answer with an SCTP m= section before starting SCTP,
   // according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
   if (pc_->sctp_mid() && local_description() && remote_description()) {
@@ -4599,8 +4502,7 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
 
   const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
   if (pc_->data_channel_type() != cricket::DCT_NONE && data &&
-      !data->rejected && !data_channel_controller()->rtp_data_channel() &&
+      !data->rejected && !data_channel_controller()->data_channel_transport()) {
-      !data_channel_controller()->data_channel_transport()) {
     if (!CreateDataChannel(data->name)) {
       LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
                            "Failed to create data channel.");
@@ -4678,23 +4580,10 @@ bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
       // out of sync (transport name not set while the mid is set).
       pc_->SetSctpDataMid(mid);
       break;
-    case cricket::DCT_RTP:
+    case cricket::DCT_NONE:
-    default:
+      // User error.
-      RtpTransportInternal* rtp_transport = pc_->GetRtpTransport(mid);
+      RTC_NOTREACHED();
-      cricket::RtpDataChannel* data_channel =
+      return false;
-          channel_manager()->CreateRtpDataChannel(
-              pc_->configuration()->media_config, rtp_transport,
-              signaling_thread(), mid, pc_->SrtpRequired(),
-              pc_->GetCryptoOptions(), &ssrc_generator_);
-      if (!data_channel)
-        return false;
-
-      pc_->network_thread()->Invoke<void>(RTC_FROM_HERE, [this, data_channel] {
-        RTC_DCHECK_RUN_ON(pc_->network_thread());
-        pc_->SetupRtpDataChannelTransport_n(data_channel);
-      });
-      have_pending_rtp_data_channel_ = true;
-      break;
   }
   return true;
 }
@@ -4716,9 +4605,9 @@ void SdpOfferAnswerHandler::DestroyTransceiverChannel(
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
   if (channel) {
     // TODO(tommi): VideoRtpReceiver::SetMediaChannel blocks and jumps to the
-    // worker thread. When being set to nullptr, there are additional blocking
+    // worker thread. When being set to nullptrpus, there are additional
-    // calls to e.g. ClearRecordableEncodedFrameCallback which triggers another
+    // blocking calls to e.g. ClearRecordableEncodedFrameCallback which triggers
-    // blocking call or Stop() for video channels.
+    // another blocking call or Stop() for video channels.
     transceiver->internal()->SetChannel(nullptr);
     RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
     // TODO(tommi): All channel objects end up getting deleted on the
@@ -4731,9 +4620,8 @@ void SdpOfferAnswerHandler::DestroyTransceiverChannel(
 void SdpOfferAnswerHandler::DestroyDataChannelTransport() {
   RTC_DCHECK_RUN_ON(signaling_thread());
   const bool has_sctp = pc_->sctp_mid().has_value();
-  auto* rtp_data_channel = data_channel_controller()->rtp_data_channel();
 
-  if (has_sctp || rtp_data_channel)
+  if (has_sctp)
     data_channel_controller()->OnTransportChannelClosed();
 
   pc_->network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
@@ -4743,9 +4631,6 @@ void SdpOfferAnswerHandler::DestroyDataChannelTransport() {
 
   if (has_sctp)
     pc_->ResetSctpDataMid();
-
-  if (rtp_data_channel)
-    DestroyChannelInterface(rtp_data_channel);
 }
 
 void SdpOfferAnswerHandler::DestroyChannelInterface(
@@ -4770,8 +4655,8 @@ void SdpOfferAnswerHandler::DestroyChannelInterface(
           static_cast<cricket::VideoChannel*>(channel));
       break;
     case cricket::MEDIA_TYPE_DATA:
-      channel_manager()->DestroyRtpDataChannel(
+      RTC_NOTREACHED()
-          static_cast<cricket::RtpDataChannel*>(channel));
+          << "Trying to destroy datachannel through DestroyChannelInterface";
       break;
     default:
       RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
@@ -4887,8 +4772,6 @@ SdpOfferAnswerHandler::GetMediaDescriptionOptionsForActiveData(
   cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
                                            RtpTransceiverDirection::kSendRecv,
                                            /*stopped=*/false);
-  AddRtpDataChannelOptions(*(data_channel_controller()->rtp_data_channels()),
-                           &options);
   return options;
 }
 
@@ -4899,8 +4782,6 @@ SdpOfferAnswerHandler::GetMediaDescriptionOptionsForRejectedData(
   cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
                                            RtpTransceiverDirection::kInactive,
                                            /*stopped=*/true);
-  AddRtpDataChannelOptions(*(data_channel_controller()->rtp_data_channels()),
-                           &options);
   return options;
 }
 

pc/sdp_offer_answer.h

@@ -632,10 +632,6 @@ class SdpOfferAnswerHandler : public SdpStateProvider,
   rtc::scoped_refptr<MediaStreamInterface> missing_msid_default_stream_
       RTC_GUARDED_BY(signaling_thread());
 
-  // Used when rolling back RTP data channels.
-  bool have_pending_rtp_data_channel_ RTC_GUARDED_BY(signaling_thread()) =
-      false;
-
   // Updates the error state, signaling if necessary.
   void SetSessionError(SessionError error, const std::string& error_desc);
 

pc/session_description.h

@@ -60,7 +60,6 @@ const int kAutoBandwidth = -1;
 
 class AudioContentDescription;
 class VideoContentDescription;
-class RtpDataContentDescription;
 class SctpDataContentDescription;
 class UnsupportedContentDescription;
 
@@ -83,11 +82,6 @@ class MediaContentDescription {
   virtual VideoContentDescription* as_video() { return nullptr; }
   virtual const VideoContentDescription* as_video() const { return nullptr; }
 
-  virtual RtpDataContentDescription* as_rtp_data() { return nullptr; }
-  virtual const RtpDataContentDescription* as_rtp_data() const {
-    return nullptr;
-  }
-
   virtual SctpDataContentDescription* as_sctp() { return nullptr; }
   virtual const SctpDataContentDescription* as_sctp() const { return nullptr; }
 
@@ -361,20 +355,6 @@ class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> {
   }
 };
 
-class RtpDataContentDescription
-    : public MediaContentDescriptionImpl<RtpDataCodec> {
- public:
-  RtpDataContentDescription() {}
-  MediaType type() const override { return MEDIA_TYPE_DATA; }
-  RtpDataContentDescription* as_rtp_data() override { return this; }
-  const RtpDataContentDescription* as_rtp_data() const override { return this; }
-
- private:
-  RtpDataContentDescription* CloneInternal() const override {
-    return new RtpDataContentDescription(*this);
-  }
-};
-
 class SctpDataContentDescription : public MediaContentDescription {
  public:
   SctpDataContentDescription() {}

pc/session_description_unittest.cc

@@ -130,16 +130,6 @@ TEST(SessionDescriptionTest, AddContentTransfersExtmapAllowMixedSetting) {
   EXPECT_EQ(MediaContentDescription::kSession,
             session_desc.GetContentDescriptionByName("video")
                 ->extmap_allow_mixed_enum());
-
-  // Session level setting overrides media level when new content is added.
-  std::unique_ptr<MediaContentDescription> data_desc =
-      std::make_unique<RtpDataContentDescription>();
-  data_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
-  session_desc.AddContent("data", MediaProtocolType::kRtp,
-                          std::move(data_desc));
-  EXPECT_EQ(MediaContentDescription::kSession,
-            session_desc.GetContentDescriptionByName("data")
-                ->extmap_allow_mixed_enum());
 }
 
 }  // namespace cricket

pc/test/fake_peer_connection_base.h

@@ -248,16 +248,10 @@ class FakePeerConnectionBase : public PeerConnectionInternal {
     return {};
   }
 
-  sigslot::signal1<RtpDataChannel*>& SignalRtpDataChannelCreated() override {
-    return SignalRtpDataChannelCreated_;
-  }
-
   sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() override {
     return SignalSctpDataChannelCreated_;
   }
 
-  cricket::RtpDataChannel* rtp_data_channel() const override { return nullptr; }
-
   absl::optional<std::string> sctp_transport_name() const override {
     return absl::nullopt;
   }
@@ -298,7 +292,6 @@ class FakePeerConnectionBase : public PeerConnectionInternal {
   }
 
  protected:
-  sigslot::signal1<RtpDataChannel*> SignalRtpDataChannelCreated_;
   sigslot::signal1<SctpDataChannel*> SignalSctpDataChannelCreated_;
 };
 

pc/webrtc_sdp.cc

@@ -82,7 +82,6 @@ using cricket::MediaContentDescription;
 using cricket::MediaProtocolType;
 using cricket::MediaType;
 using cricket::RidDescription;
-using cricket::RtpDataContentDescription;
 using cricket::RtpHeaderExtensions;
 using cricket::SctpDataContentDescription;
 using cricket::SimulcastDescription;
@@ -1415,12 +1414,7 @@ void BuildMediaDescription(const ContentInfo* content_info,
         fmt.append(kDefaultSctpmapProtocol);
       }
     } else {
-      const RtpDataContentDescription* rtp_data_desc =
+      RTC_NOTREACHED() << "Data description without SCTP";
-          media_desc->as_rtp_data();
-      for (const cricket::RtpDataCodec& codec : rtp_data_desc->codecs()) {
-        fmt.append(" ");
-        fmt.append(rtc::ToString(codec.id));
-      }
     }
   } else if (media_type == cricket::MEDIA_TYPE_UNSUPPORTED) {
     const UnsupportedContentDescription* unsupported_desc =
@@ -1972,19 +1966,6 @@ void BuildRtpMap(const MediaContentDescription* media_desc,
       ptime = std::max(ptime, max_minptime);
       AddAttributeLine(kCodecParamPTime, ptime, message);
     }
-  } else if (media_type == cricket::MEDIA_TYPE_DATA) {
-    if (media_desc->as_rtp_data()) {
-      for (const cricket::RtpDataCodec& codec :
-           media_desc->as_rtp_data()->codecs()) {
-        // RFC 4566
-        // a=rtpmap:<payload type> <encoding name>/<clock rate>
-        // [/<encodingparameters>]
-        InitAttrLine(kAttributeRtpmap, &os);
-        os << kSdpDelimiterColon << codec.id << " " << codec.name << "/"
-           << codec.clockrate;
-        AddLine(os.str(), message);
-      }
-    }
   }
 }
 
@@ -2739,14 +2720,6 @@ bool ParseMediaDescription(
         }
         data_desc->set_protocol(protocol);
         content = std::move(data_desc);
-      } else if (cricket::IsRtpProtocol(protocol)) {
-        // RTP
-        std::unique_ptr<RtpDataContentDescription> data_desc =
-            ParseContentDescription<RtpDataContentDescription>(
-                message, cricket::MEDIA_TYPE_DATA, mline_index, protocol,
-                payload_types, pos, &content_name, &bundle_only,
-                &section_msid_signaling, &transport, candidates, error);
-        content = std::move(data_desc);
       } else {
         return ParseFailed(line, "Unsupported protocol for media type", error);
       }
@@ -3671,11 +3644,6 @@ bool ParseRtpmapAttribute(const std::string& line,
     AudioContentDescription* audio_desc = media_desc->as_audio();
     UpdateCodec(payload_type, encoding_name, clock_rate, 0, channels,
                 audio_desc);
-  } else if (media_type == cricket::MEDIA_TYPE_DATA) {
-    RtpDataContentDescription* data_desc = media_desc->as_rtp_data();
-    if (data_desc) {
-      data_desc->AddCodec(cricket::RtpDataCodec(payload_type, encoding_name));
-    }
   }
   return true;
 }

pc/webrtc_sdp_unittest.cc

@@ -65,7 +65,6 @@ using cricket::MediaProtocolType;
 using cricket::RELAY_PORT_TYPE;
 using cricket::RidDescription;
 using cricket::RidDirection;
-using cricket::RtpDataContentDescription;
 using cricket::SctpDataContentDescription;
 using cricket::SessionDescription;
 using cricket::SimulcastDescription;
@@ -263,22 +262,6 @@ static const char kSdpString[] =
     "a=ssrc:3 mslabel:local_stream_1\r\n"
     "a=ssrc:3 label:video_track_id_1\r\n";
 
-static const char kSdpRtpDataChannelString[] =
-    "m=application 9 RTP/SAVPF 101\r\n"
-    "c=IN IP4 0.0.0.0\r\n"
-    "a=rtcp:9 IN IP4 0.0.0.0\r\n"
-    "a=ice-ufrag:ufrag_data\r\n"
-    "a=ice-pwd:pwd_data\r\n"
-    "a=mid:data_content_name\r\n"
-    "a=sendrecv\r\n"
-    "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
-    "inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5\r\n"
-    "a=rtpmap:101 google-data/90000\r\n"
-    "a=ssrc:10 cname:data_channel_cname\r\n"
-    "a=ssrc:10 msid:data_channel data_channeld0\r\n"
-    "a=ssrc:10 mslabel:data_channel\r\n"
-    "a=ssrc:10 label:data_channeld0\r\n";
-
 // draft-ietf-mmusic-sctp-sdp-03
 static const char kSdpSctpDataChannelString[] =
     "m=application 9 UDP/DTLS/SCTP 5000\r\n"
@@ -906,12 +889,6 @@ static const uint32_t kVideoTrack3Ssrc = 6;
 static const char kAudioTrackId3[] = "audio_track_id_3";
 static const uint32_t kAudioTrack3Ssrc = 7;
 
-// DataChannel
-static const char kDataChannelLabel[] = "data_channel";
-static const char kDataChannelMsid[] = "data_channeld0";
-static const char kDataChannelCname[] = "data_channel_cname";
-static const uint32_t kDataChannelSsrc = 10;
-
 // Candidate
 static const char kDummyMid[] = "dummy_mid";
 static const int kDummyIndex = 123;
@@ -1466,11 +1443,6 @@ class WebRtcSdpTest : public ::testing::Test {
               simulcast2.receive_layers().size());
   }
 
-  void CompareRtpDataContentDescription(const RtpDataContentDescription* dcd1,
-                                        const RtpDataContentDescription* dcd2) {
-    CompareMediaContentDescription<RtpDataContentDescription>(dcd1, dcd2);
-  }
-
   void CompareSctpDataContentDescription(
       const SctpDataContentDescription* dcd1,
       const SctpDataContentDescription* dcd2) {
@@ -1521,14 +1493,6 @@ class WebRtcSdpTest : public ::testing::Test {
         const SctpDataContentDescription* scd2 =
             c2.media_description()->as_sctp();
         CompareSctpDataContentDescription(scd1, scd2);
-      } else {
-        if (IsDataContent(&c1)) {
-          const RtpDataContentDescription* dcd1 =
-              c1.media_description()->as_rtp_data();
-          const RtpDataContentDescription* dcd2 =
-              c2.media_description()->as_rtp_data();
-          CompareRtpDataContentDescription(dcd1, dcd2);
-        }
       }
 
       CompareSimulcastDescription(
@@ -1816,28 +1780,6 @@ class WebRtcSdpTest : public ::testing::Test {
         kDataContentName, TransportDescription(kUfragData, kPwdData)));
   }
 
-  void AddRtpDataChannel() {
-    std::unique_ptr<RtpDataContentDescription> data(
-        new RtpDataContentDescription());
-    data_desc_ = data.get();
-
-    data_desc_->AddCodec(DataCodec(101, "google-data"));
-    StreamParams data_stream;
-    data_stream.id = kDataChannelMsid;
-    data_stream.cname = kDataChannelCname;
-    data_stream.set_stream_ids({kDataChannelLabel});
-    data_stream.ssrcs.push_back(kDataChannelSsrc);
-    data_desc_->AddStream(data_stream);
-    data_desc_->AddCrypto(
-        CryptoParams(1, "AES_CM_128_HMAC_SHA1_80",
-                     "inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5", ""));
-    data_desc_->set_protocol(cricket::kMediaProtocolSavpf);
-    desc_.AddContent(kDataContentName, MediaProtocolType::kRtp,
-                     std::move(data));
-    desc_.AddTransportInfo(TransportInfo(
-        kDataContentName, TransportDescription(kUfragData, kPwdData)));
-  }
-
   bool TestDeserializeDirection(RtpTransceiverDirection direction) {
     std::string new_sdp = kSdpFullString;
     ReplaceDirection(direction, &new_sdp);
@@ -2103,7 +2045,6 @@ class WebRtcSdpTest : public ::testing::Test {
   SessionDescription desc_;
   AudioContentDescription* audio_desc_;
   VideoContentDescription* video_desc_;
-  RtpDataContentDescription* data_desc_;
   SctpDataContentDescription* sctp_desc_;
   Candidates candidates_;
   std::unique_ptr<IceCandidateInterface> jcandidate_;
@@ -2269,18 +2210,6 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithAudioVideoRejected) {
   EXPECT_TRUE(TestSerializeRejected(true, true));
 }
 
-TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithRtpDataChannel) {
-  AddRtpDataChannel();
-  JsepSessionDescription jsep_desc(kDummyType);
-
-  MakeDescriptionWithoutCandidates(&jsep_desc);
-  std::string message = webrtc::SdpSerialize(jsep_desc);
-
-  std::string expected_sdp = kSdpString;
-  expected_sdp.append(kSdpRtpDataChannelString);
-  EXPECT_EQ(expected_sdp, message);
-}
-
 TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithSctpDataChannel) {
   bool use_sctpmap = true;
   AddSctpDataChannel(use_sctpmap);
@@ -2327,22 +2256,6 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
   EXPECT_EQ(expected_sdp, message);
 }
 
-TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
-  JsepSessionDescription jsep_desc(kDummyType);
-  AddRtpDataChannel();
-  data_desc_->set_bandwidth(100 * 1000);
-  data_desc_->set_bandwidth_type("AS");
-  MakeDescriptionWithoutCandidates(&jsep_desc);
-  std::string message = webrtc::SdpSerialize(jsep_desc);
-
-  std::string expected_sdp = kSdpString;
-  expected_sdp.append(kSdpRtpDataChannelString);
-  // Serializing data content shouldn't ignore bandwidth settings.
-  InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
-              "b=AS:100\r\n", &expected_sdp);
-  EXPECT_EQ(expected_sdp, message);
-}
-
 TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmapAllowMixed) {
   jdesc_.description()->set_extmap_allow_mixed(true);
   TestSerialize(jdesc_);
@@ -2913,21 +2826,6 @@ TEST_F(WebRtcSdpTest, DeserializeInvalidCandidiate) {
   EXPECT_FALSE(SdpDeserializeCandidate(kSdpTcpInvalidCandidate, &jcandidate));
 }
 
-TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpDataChannels) {
-  AddRtpDataChannel();
-  JsepSessionDescription jdesc(kDummyType);
-  ASSERT_TRUE(jdesc.Initialize(desc_.Clone(), kSessionId, kSessionVersion));
-
-  std::string sdp_with_data = kSdpString;
-  sdp_with_data.append(kSdpRtpDataChannelString);
-  JsepSessionDescription jdesc_output(kDummyType);
-
-  // Deserialize
-  EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
-  // Verify
-  EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
-}
-
 TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannels) {
   bool use_sctpmap = true;
   AddSctpDataChannel(use_sctpmap);
@@ -3088,8 +2986,9 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpmapAttribute) {
 }
 
 TEST_F(WebRtcSdpTest, DeserializeSdpWithStrangeApplicationProtocolNames) {
-  static const char* bad_strings[] = {"DTLS/SCTPRTP/", "obviously-bogus",
+  static const char* bad_strings[] = {
-                                      "UDP/TL/RTSP/SAVPF", "UDP/TL/RTSP/S"};
+      "DTLS/SCTPRTP/", "obviously-bogus",   "UDP/TL/RTSP/SAVPF",
+      "UDP/TL/RTSP/S", "DTLS/SCTP/RTP/FOO", "obviously-bogus/RTP/"};
   for (auto proto : bad_strings) {
     std::string sdp_with_data = kSdpString;
     sdp_with_data.append("m=application 9 ");
@@ -3099,18 +2998,6 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithStrangeApplicationProtocolNames) {
     EXPECT_FALSE(SdpDeserialize(sdp_with_data, &jdesc_output))
         << "Parsing should have failed on " << proto;
   }
-  // The following strings are strange, but acceptable as RTP.
-  static const char* weird_strings[] = {"DTLS/SCTP/RTP/FOO",
-                                        "obviously-bogus/RTP/"};
-  for (auto proto : weird_strings) {
-    std::string sdp_with_data = kSdpString;
-    sdp_with_data.append("m=application 9 ");
-    sdp_with_data.append(proto);
-    sdp_with_data.append(" 47\r\n");
-    JsepSessionDescription jdesc_output(kDummyType);
-    EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output))
-        << "Parsing should have succeeded on " << proto;
-  }
 }
 
 // For crbug/344475.
@@ -3168,21 +3055,6 @@ TEST_F(WebRtcSdpTest,
   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
 }
 
-TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpDataChannelsAndBandwidth) {
-  // We want to test that deserializing data content limits bandwidth
-  // settings (it should never be greater than the default).
-  // This should prevent someone from using unlimited data bandwidth through
-  // JS and "breaking the Internet".
-  // See: https://code.google.com/p/chromium/issues/detail?id=280726
-  std::string sdp_with_bandwidth = kSdpString;
-  sdp_with_bandwidth.append(kSdpRtpDataChannelString);
-  InjectAfter("a=mid:data_content_name\r\n", "b=AS:100\r\n",
-              &sdp_with_bandwidth);
-  JsepSessionDescription jdesc_with_bandwidth(kDummyType);
-
-  EXPECT_FALSE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
-}
-
 TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsAndBandwidth) {
   bool use_sctpmap = true;
   AddSctpDataChannel(use_sctpmap);

test/pc/sctp/fake_sctp_transport.h

@@ -40,8 +40,14 @@ class FakeSctpTransport : public cricket::SctpTransportInternal {
   int max_message_size() const { return max_message_size_; }
   absl::optional<int> max_outbound_streams() const { return absl::nullopt; }
   absl::optional<int> max_inbound_streams() const { return absl::nullopt; }
-  int local_port() const { return *local_port_; }
+  int local_port() const {
-  int remote_port() const { return *remote_port_; }
+    RTC_DCHECK(local_port_);
+    return *local_port_;
+  }
+  int remote_port() const {
+    RTC_DCHECK(remote_port_);
+    return *remote_port_;
+  }
 
  private:
   absl::optional<int> local_port_;