pc/media_session.cc

/*
 *  Copyright 2004 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/media_session.h"

#include <algorithm>
#include <functional>
#include <map>
#include <memory>
#include <set>
#include <unordered_map>
#include <utility>

#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/crypto_params.h"
#include "media/base/h264_profile_level_id.h"
#include "media/base/media_constants.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "pc/channel_manager.h"
#include "pc/media_protocol_names.h"
#include "pc/rtp_media_utils.h"
#include "pc/srtp_filter.h"
#include "pc/used_ids.h"
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/third_party/base64/base64.h"
#include "rtc_base/unique_id_generator.h"

namespace {

using rtc::UniqueRandomIdGenerator;
using webrtc::RtpTransceiverDirection;

const char kInline[] = "inline:";

void GetSupportedSdesCryptoSuiteNames(
    void (*func)(const webrtc::CryptoOptions&, std::vector<int>*),
    const webrtc::CryptoOptions& crypto_options,
    std::vector<std::string>* names) {
  std::vector<int> crypto_suites;
  func(crypto_options, &crypto_suites);
  for (const auto crypto : crypto_suites) {
    names->push_back(rtc::SrtpCryptoSuiteToName(crypto));
  }
}

}  // namespace

namespace cricket {

// RTP Profile names
// http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml
// RFC4585
const char kMediaProtocolAvpf[] = "RTP/AVPF";
// RFC5124
const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF";

// We always generate offers with "UDP/TLS/RTP/SAVPF" when using DTLS-SRTP,
// but we tolerate "RTP/SAVPF" in offers we receive, for compatibility.
const char kMediaProtocolSavpf[] = "RTP/SAVPF";

// Note that the below functions support some protocol strings purely for
// legacy compatibility, as required by JSEP in Section 5.1.2, Profile Names
// and Interoperability.

static bool IsDtlsRtp(const std::string& protocol) {
  // Most-likely values first.
  return protocol == "UDP/TLS/RTP/SAVPF" || protocol == "TCP/TLS/RTP/SAVPF" ||
         protocol == "UDP/TLS/RTP/SAVP" || protocol == "TCP/TLS/RTP/SAVP";
}

static bool IsPlainRtp(const std::string& protocol) {
  // Most-likely values first.
  return protocol == "RTP/SAVPF" || protocol == "RTP/AVPF" ||
         protocol == "RTP/SAVP" || protocol == "RTP/AVP";
}

static RtpTransceiverDirection NegotiateRtpTransceiverDirection(
    RtpTransceiverDirection offer,
    RtpTransceiverDirection wants) {
  bool offer_send = webrtc::RtpTransceiverDirectionHasSend(offer);
  bool offer_recv = webrtc::RtpTransceiverDirectionHasRecv(offer);
  bool wants_send = webrtc::RtpTransceiverDirectionHasSend(wants);
  bool wants_recv = webrtc::RtpTransceiverDirectionHasRecv(wants);
  return webrtc::RtpTransceiverDirectionFromSendRecv(offer_recv && wants_send,
                                                     offer_send && wants_recv);
}

static bool IsMediaContentOfType(const ContentInfo* content,
                                 MediaType media_type) {
  if (!content || !content->media_description()) {
    return false;
  }
  return content->media_description()->type() == media_type;
}

static bool CreateCryptoParams(int tag,
                               const std::string& cipher,
                               CryptoParams* crypto_out) {
  int key_len;
  int salt_len;
  if (!rtc::GetSrtpKeyAndSaltLengths(rtc::SrtpCryptoSuiteFromName(cipher),
                                     &key_len, &salt_len)) {
    return false;
  }

  int master_key_len = key_len + salt_len;
  std::string master_key;
  if (!rtc::CreateRandomData(master_key_len, &master_key)) {
    return false;
  }

  RTC_CHECK_EQ(master_key_len, master_key.size());
  std::string key = rtc::Base64::Encode(master_key);

  crypto_out->tag = tag;
  crypto_out->cipher_suite = cipher;
  crypto_out->key_params = kInline;
  crypto_out->key_params += key;
  return true;
}

static bool AddCryptoParams(const std::string& cipher_suite,
                            CryptoParamsVec* cryptos_out) {
  int size = static_cast<int>(cryptos_out->size());

  cryptos_out->resize(size + 1);
  return CreateCryptoParams(size, cipher_suite, &cryptos_out->at(size));
}

void AddMediaCryptos(const CryptoParamsVec& cryptos,
                     MediaContentDescription* media) {
  for (const CryptoParams& crypto : cryptos) {
    media->AddCrypto(crypto);
  }
}

bool CreateMediaCryptos(const std::vector<std::string>& crypto_suites,
                        MediaContentDescription* media) {
  CryptoParamsVec cryptos;
  for (const std::string& crypto_suite : crypto_suites) {
    if (!AddCryptoParams(crypto_suite, &cryptos)) {
      return false;
    }
  }
  AddMediaCryptos(cryptos, media);
  return true;
}

const CryptoParamsVec* GetCryptos(const ContentInfo* content) {
  if (!content || !content->media_description()) {
    return nullptr;
  }
  return &content->media_description()->cryptos();
}

bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
                        const CryptoParams& crypto,
                        CryptoParams* crypto_out) {
  auto it = absl::c_find_if(
      cryptos, [&crypto](const CryptoParams& c) { return crypto.Matches(c); });
  if (it == cryptos.end()) {
    return false;
  }
  *crypto_out = *it;
  return true;
}

// For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the
// low overhead.
void GetSupportedAudioSdesCryptoSuites(
    const webrtc::CryptoOptions& crypto_options,
    std::vector<int>* crypto_suites) {
  if (crypto_options.srtp.enable_gcm_crypto_suites) {
    crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
    crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
  }
  if (crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher) {
    crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
  }
  crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
}

void GetSupportedAudioSdesCryptoSuiteNames(
    const webrtc::CryptoOptions& crypto_options,
    std::vector<std::string>* crypto_suite_names) {
  GetSupportedSdesCryptoSuiteNames(GetSupportedAudioSdesCryptoSuites,
                                   crypto_options, crypto_suite_names);
}

void GetSupportedVideoSdesCryptoSuites(
    const webrtc::CryptoOptions& crypto_options,
    std::vector<int>* crypto_suites) {
  if (crypto_options.srtp.enable_gcm_crypto_suites) {
    crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
    crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
  }
  crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
}

void GetSupportedVideoSdesCryptoSuiteNames(
    const webrtc::CryptoOptions& crypto_options,
    std::vector<std::string>* crypto_suite_names) {
  GetSupportedSdesCryptoSuiteNames(GetSupportedVideoSdesCryptoSuites,
                                   crypto_options, crypto_suite_names);
}

void GetSupportedDataSdesCryptoSuites(
    const webrtc::CryptoOptions& crypto_options,
    std::vector<int>* crypto_suites) {
  if (crypto_options.srtp.enable_gcm_crypto_suites) {
    crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
    crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
  }
  crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
}

void GetSupportedDataSdesCryptoSuiteNames(
    const webrtc::CryptoOptions& crypto_options,
    std::vector<std::string>* crypto_suite_names) {
  GetSupportedSdesCryptoSuiteNames(GetSupportedDataSdesCryptoSuites,
                                   crypto_options, crypto_suite_names);
}

// Support any GCM cipher (if enabled through options). For video support only
// 80-bit SHA1 HMAC. For audio 32-bit HMAC is tolerated (if enabled) unless
// bundle is enabled because it is low overhead.
// Pick the crypto in the list that is supported.
static bool SelectCrypto(const MediaContentDescription* offer,
                         bool bundle,
                         const webrtc::CryptoOptions& crypto_options,
                         CryptoParams* crypto_out) {
  bool audio = offer->type() == MEDIA_TYPE_AUDIO;
  const CryptoParamsVec& cryptos = offer->cryptos();

  for (const CryptoParams& crypto : cryptos) {
    if ((crypto_options.srtp.enable_gcm_crypto_suites &&
         rtc::IsGcmCryptoSuiteName(crypto.cipher_suite)) ||
        rtc::CS_AES_CM_128_HMAC_SHA1_80 == crypto.cipher_suite ||
        (rtc::CS_AES_CM_128_HMAC_SHA1_32 == crypto.cipher_suite && audio &&
         !bundle && crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher)) {
      return CreateCryptoParams(crypto.tag, crypto.cipher_suite, crypto_out);
    }
  }
  return false;
}

// Finds all StreamParams of all media types and attach them to stream_params.
static StreamParamsVec GetCurrentStreamParams(
    const std::vector<const ContentInfo*>& active_local_contents) {
  StreamParamsVec stream_params;
  for (const ContentInfo* content : active_local_contents) {
    for (const StreamParams& params : content->media_description()->streams()) {
      stream_params.push_back(params);
    }
  }
  return stream_params;
}

// Filters the data codecs for the data channel type.
void FilterDataCodecs(std::vector<DataCodec>* codecs, bool sctp) {
  // Filter RTP codec for SCTP and vice versa.
  const char* codec_name =
      sctp ? kGoogleRtpDataCodecName : kGoogleSctpDataCodecName;
  codecs->erase(std::remove_if(codecs->begin(), codecs->end(),
                               [&codec_name](const DataCodec& codec) {
                                 return absl::EqualsIgnoreCase(codec.name,
                                                               codec_name);
                               }),
                codecs->end());
}

static StreamParams CreateStreamParamsForNewSenderWithSsrcs(
    const SenderOptions& sender,
    const std::string& rtcp_cname,
    bool include_rtx_streams,
    bool include_flexfec_stream,
    UniqueRandomIdGenerator* ssrc_generator) {
  StreamParams result;
  result.id = sender.track_id;

  // TODO(brandtr): Update when we support multistream protection.
  if (include_flexfec_stream && sender.num_sim_layers > 1) {
    include_flexfec_stream = false;
    RTC_LOG(LS_WARNING)
        << "Our FlexFEC implementation only supports protecting "
           "a single media streams. This session has multiple "
           "media streams however, so no FlexFEC SSRC will be generated.";
  }

  result.GenerateSsrcs(sender.num_sim_layers, include_rtx_streams,
                       include_flexfec_stream, ssrc_generator);

  result.cname = rtcp_cname;
  result.set_stream_ids(sender.stream_ids);

  return result;
}

static bool ValidateSimulcastLayers(
    const std::vector<RidDescription>& rids,
    const SimulcastLayerList& simulcast_layers) {
  return absl::c_all_of(
      simulcast_layers.GetAllLayers(), [&rids](const SimulcastLayer& layer) {
        return absl::c_any_of(rids, [&layer](const RidDescription& rid) {
          return rid.rid == layer.rid;
        });
      });
}

static StreamParams CreateStreamParamsForNewSenderWithRids(
    const SenderOptions& sender,
    const std::string& rtcp_cname) {
  RTC_DCHECK(!sender.rids.empty());
  RTC_DCHECK_EQ(sender.num_sim_layers, 0)
      << "RIDs are the compliant way to indicate simulcast.";
  RTC_DCHECK(ValidateSimulcastLayers(sender.rids, sender.simulcast_layers));
  StreamParams result;
  result.id = sender.track_id;
  result.cname = rtcp_cname;
  result.set_stream_ids(sender.stream_ids);

  // More than one rid should be signaled.
  if (sender.rids.size() > 1) {
    result.set_rids(sender.rids);
  }

  return result;
}

// Adds SimulcastDescription if indicated by the media description options.
// MediaContentDescription should already be set up with the send rids.
static void AddSimulcastToMediaDescription(
    const MediaDescriptionOptions& media_description_options,
    MediaContentDescription* description) {
  RTC_DCHECK(description);

  // Check if we are using RIDs in this scenario.
  if (absl::c_all_of(description->streams(), [](const StreamParams& params) {
        return !params.has_rids();
      })) {
    return;
  }

  RTC_DCHECK_EQ(1, description->streams().size())
      << "RIDs are only supported in Unified Plan semantics.";
  RTC_DCHECK_EQ(1, media_description_options.sender_options.size());
  RTC_DCHECK(description->type() == MediaType::MEDIA_TYPE_AUDIO ||
             description->type() == MediaType::MEDIA_TYPE_VIDEO);

  // One RID or less indicates that simulcast is not needed.
  if (description->streams()[0].rids().size() <= 1) {
    return;
  }

  // Only negotiate the send layers.
  SimulcastDescription simulcast;
  simulcast.send_layers() =
      media_description_options.sender_options[0].simulcast_layers;
  description->set_simulcast_description(simulcast);
}

// Adds a StreamParams for each SenderOptions in |sender_options| to
// content_description.
// |current_params| - All currently known StreamParams of any media type.
template <class C>
static bool AddStreamParams(
    const std::vector<SenderOptions>& sender_options,
    const std::string& rtcp_cname,
    UniqueRandomIdGenerator* ssrc_generator,
    StreamParamsVec* current_streams,
    MediaContentDescriptionImpl<C>* content_description) {
  // SCTP streams are not negotiated using SDP/ContentDescriptions.
  if (IsSctpProtocol(content_description->protocol())) {
    return true;
  }

  const bool include_rtx_streams =
      ContainsRtxCodec(content_description->codecs());

  const bool include_flexfec_stream =
      ContainsFlexfecCodec(content_description->codecs());

  for (const SenderOptions& sender : sender_options) {
    // groupid is empty for StreamParams generated using
    // MediaSessionDescriptionFactory.
    StreamParams* param =
        GetStreamByIds(*current_streams, "" /*group_id*/, sender.track_id);
    if (!param) {
      // This is a new sender.
      StreamParams stream_param =
          sender.rids.empty()
              ?
              // Signal SSRCs and legacy simulcast (if requested).
              CreateStreamParamsForNewSenderWithSsrcs(
                  sender, rtcp_cname, include_rtx_streams,
                  include_flexfec_stream, ssrc_generator)
              :
              // Signal RIDs and spec-compliant simulcast (if requested).
              CreateStreamParamsForNewSenderWithRids(sender, rtcp_cname);

      content_description->AddStream(stream_param);

      // Store the new StreamParams in current_streams.
      // This is necessary so that we can use the CNAME for other media types.
      current_streams->push_back(stream_param);
    } else {
      // Use existing generated SSRCs/groups, but update the sync_label if
      // necessary. This may be needed if a MediaStreamTrack was moved from one
      // MediaStream to another.
      param->set_stream_ids(sender.stream_ids);
      content_description->AddStream(*param);
    }
  }
  return true;
}

// Updates the transport infos of the |sdesc| according to the given
// |bundle_group|. The transport infos of the content names within the
// |bundle_group| should be updated to use the ufrag, pwd and DTLS role of the
// first content within the |bundle_group|.
static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
                                         SessionDescription* sdesc) {
  // The bundle should not be empty.
  if (!sdesc || !bundle_group.FirstContentName()) {
    return false;
  }

  // We should definitely have a transport for the first content.
  const std::string& selected_content_name = *bundle_group.FirstContentName();
  const TransportInfo* selected_transport_info =
      sdesc->GetTransportInfoByName(selected_content_name);
  if (!selected_transport_info) {
    return false;
  }

  // Set the other contents to use the same ICE credentials.
  const std::string& selected_ufrag =
      selected_transport_info->description.ice_ufrag;
  const std::string& selected_pwd =
      selected_transport_info->description.ice_pwd;
  ConnectionRole selected_connection_role =
      selected_transport_info->description.connection_role;
  const absl::optional<OpaqueTransportParameters>& selected_opaque_parameters =
      selected_transport_info->description.opaque_parameters;
  for (TransportInfo& transport_info : sdesc->transport_infos()) {
    if (bundle_group.HasContentName(transport_info.content_name) &&
        transport_info.content_name != selected_content_name) {
      transport_info.description.ice_ufrag = selected_ufrag;
      transport_info.description.ice_pwd = selected_pwd;
      transport_info.description.connection_role = selected_connection_role;
      transport_info.description.opaque_parameters = selected_opaque_parameters;
    }
  }
  return true;
}

// Gets the CryptoParamsVec of the given |content_name| from |sdesc|, and
// sets it to |cryptos|.
static bool GetCryptosByName(const SessionDescription* sdesc,
                             const std::string& content_name,
                             CryptoParamsVec* cryptos) {
  if (!sdesc || !cryptos) {
    return false;
  }
  const ContentInfo* content = sdesc->GetContentByName(content_name);
  if (!content || !content->media_description()) {
    return false;
  }
  *cryptos = content->media_description()->cryptos();
  return true;
}

// Prunes the |target_cryptos| by removing the crypto params (cipher_suite)
// which are not available in |filter|.
static void PruneCryptos(const CryptoParamsVec& filter,
                         CryptoParamsVec* target_cryptos) {
  if (!target_cryptos) {
    return;
  }

  target_cryptos->erase(
      std::remove_if(target_cryptos->begin(), target_cryptos->end(),
                     // Returns true if the |crypto|'s cipher_suite is not
                     // found in |filter|.
                     [&filter](const CryptoParams& crypto) {
                       for (const CryptoParams& entry : filter) {
                         if (entry.cipher_suite == crypto.cipher_suite)
                           return false;
                       }
                       return true;
                     }),
      target_cryptos->end());
}

static bool IsRtpContent(SessionDescription* sdesc,
                         const std::string& content_name) {
  bool is_rtp = false;
  ContentInfo* content = sdesc->GetContentByName(content_name);
  if (content && content->media_description()) {
    is_rtp = IsRtpProtocol(content->media_description()->protocol());
  }
  return is_rtp;
}

// Updates the crypto parameters of the |sdesc| according to the given
// |bundle_group|. The crypto parameters of all the contents within the
// |bundle_group| should be updated to use the common subset of the
// available cryptos.
static bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group,
                                        SessionDescription* sdesc) {
  // The bundle should not be empty.
  if (!sdesc || !bundle_group.FirstContentName()) {
    return false;
  }

  bool common_cryptos_needed = false;
  // Get the common cryptos.
  const ContentNames& content_names = bundle_group.content_names();
  CryptoParamsVec common_cryptos;
  bool first = true;
  for (const std::string& content_name : content_names) {
    if (!IsRtpContent(sdesc, content_name)) {
      continue;
    }
    // The common cryptos are needed if any of the content does not have DTLS
    // enabled.
    if (!sdesc->GetTransportInfoByName(content_name)->description.secure()) {
      common_cryptos_needed = true;
    }
    if (first) {
      first = false;
      // Initial the common_cryptos with the first content in the bundle group.
      if (!GetCryptosByName(sdesc, content_name, &common_cryptos)) {
        return false;
      }
      if (common_cryptos.empty()) {
        // If there's no crypto params, we should just return.
        return true;
      }
    } else {
      CryptoParamsVec cryptos;
      if (!GetCryptosByName(sdesc, content_name, &cryptos)) {
        return false;
      }
      PruneCryptos(cryptos, &common_cryptos);
    }
  }

  if (common_cryptos.empty() && common_cryptos_needed) {
    return false;
  }

  // Update to use the common cryptos.
  for (const std::string& content_name : content_names) {
    if (!IsRtpContent(sdesc, content_name)) {
      continue;
    }
    ContentInfo* content = sdesc->GetContentByName(content_name);
    if (IsMediaContent(content)) {
      MediaContentDescription* media_desc = content->media_description();
      if (!media_desc) {
        return false;
      }
      media_desc->set_cryptos(common_cryptos);
    }
  }
  return true;
}

static std::vector<const ContentInfo*> GetActiveContents(
    const SessionDescription& description,
    const MediaSessionOptions& session_options) {
  std::vector<const ContentInfo*> active_contents;
  for (size_t i = 0; i < description.contents().size(); ++i) {
    RTC_DCHECK_LT(i, session_options.media_description_options.size());
    const ContentInfo& content = description.contents()[i];
    const MediaDescriptionOptions& media_options =
        session_options.media_description_options[i];
    if (!content.rejected && !media_options.stopped &&
        content.name == media_options.mid) {
      active_contents.push_back(&content);
    }
  }
  return active_contents;
}

template <class C>
static bool ContainsRtxCodec(const std::vector<C>& codecs) {
  for (const auto& codec : codecs) {
    if (IsRtxCodec(codec)) {
      return true;
    }
  }
  return false;
}

template <class C>
static bool IsRtxCodec(const C& codec) {
  return absl::EqualsIgnoreCase(codec.name, kRtxCodecName);
}

template <class C>
static bool ContainsFlexfecCodec(const std::vector<C>& codecs) {
  for (const auto& codec : codecs) {
    if (IsFlexfecCodec(codec)) {
      return true;
    }
  }
  return false;
}

template <class C>
static bool IsFlexfecCodec(const C& codec) {
  return absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName);
}

// Create a media content to be offered for the given |sender_options|,
// according to the given options.rtcp_mux, session_options.is_muc, codecs,
// secure_transport, crypto, and current_streams. If we don't currently have
// crypto (in current_cryptos) and it is enabled (in secure_policy), crypto is
// created (according to crypto_suites). The created content is added to the
// offer.
static bool CreateContentOffer(
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const SecurePolicy& secure_policy,
    const CryptoParamsVec* current_cryptos,
    const std::vector<std::string>& crypto_suites,
    const RtpHeaderExtensions& rtp_extensions,
    UniqueRandomIdGenerator* ssrc_generator,
    StreamParamsVec* current_streams,
    MediaContentDescription* offer) {
  offer->set_rtcp_mux(session_options.rtcp_mux_enabled);
  if (offer->type() == cricket::MEDIA_TYPE_VIDEO) {
    offer->set_rtcp_reduced_size(true);
  }
  offer->set_rtp_header_extensions(rtp_extensions);

  AddSimulcastToMediaDescription(media_description_options, offer);

  if (secure_policy != SEC_DISABLED) {
    if (current_cryptos) {
      AddMediaCryptos(*current_cryptos, offer);
    }
    if (offer->cryptos().empty()) {
      if (!CreateMediaCryptos(crypto_suites, offer)) {
        return false;
      }
    }
  }

  offer->set_alt_protocol(media_description_options.alt_protocol);

  if (secure_policy == SEC_REQUIRED && offer->cryptos().empty()) {
    return false;
  }
  return true;
}
template <class C>
static bool CreateMediaContentOffer(
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const std::vector<C>& codecs,
    const SecurePolicy& secure_policy,
    const CryptoParamsVec* current_cryptos,
    const std::vector<std::string>& crypto_suites,
    const RtpHeaderExtensions& rtp_extensions,
    UniqueRandomIdGenerator* ssrc_generator,
    StreamParamsVec* current_streams,
    MediaContentDescriptionImpl<C>* offer) {
  offer->AddCodecs(codecs);
  if (!AddStreamParams(media_description_options.sender_options,
                       session_options.rtcp_cname, ssrc_generator,
                       current_streams, offer)) {
    return false;
  }

  return CreateContentOffer(media_description_options, session_options,
                            secure_policy, current_cryptos, crypto_suites,
                            rtp_extensions, ssrc_generator, current_streams,
                            offer);
}

template <class C>
static bool ReferencedCodecsMatch(const std::vector<C>& codecs1,
                                  const int codec1_id,
                                  const std::vector<C>& codecs2,
                                  const int codec2_id) {
  const C* codec1 = FindCodecById(codecs1, codec1_id);
  const C* codec2 = FindCodecById(codecs2, codec2_id);
  return codec1 != nullptr && codec2 != nullptr && codec1->Matches(*codec2);
}

template <class C>
static void NegotiatePacketization(const C& local_codec,
                                   const C& remote_codec,
                                   C* negotiated_codec) {}

template <>
void NegotiatePacketization(const VideoCodec& local_codec,
                            const VideoCodec& remote_codec,
                            VideoCodec* negotiated_codec) {
  negotiated_codec->packetization =
      VideoCodec::IntersectPacketization(local_codec, remote_codec);
}

template <class C>
static void NegotiateCodecs(const std::vector<C>& local_codecs,
                            const std::vector<C>& offered_codecs,
                            std::vector<C>* negotiated_codecs,
                            bool keep_offer_order) {
  for (const C& ours : local_codecs) {
    C theirs;
    // Note that we intentionally only find one matching codec for each of our
    // local codecs, in case the remote offer contains duplicate codecs.
    if (FindMatchingCodec(local_codecs, offered_codecs, ours, &theirs)) {
      C negotiated = ours;
      NegotiatePacketization(ours, theirs, &negotiated);
      negotiated.IntersectFeedbackParams(theirs);
      if (IsRtxCodec(negotiated)) {
        const auto apt_it =
            theirs.params.find(kCodecParamAssociatedPayloadType);
        // FindMatchingCodec shouldn't return something with no apt value.
        RTC_DCHECK(apt_it != theirs.params.end());
        negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_it->second);
      }
      if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) {
        webrtc::H264::GenerateProfileLevelIdForAnswer(
            ours.params, theirs.params, &negotiated.params);
      }
      negotiated.id = theirs.id;
      negotiated.name = theirs.name;
      negotiated_codecs->push_back(std::move(negotiated));
    }
  }
  if (keep_offer_order) {
    // RFC3264: Although the answerer MAY list the formats in their desired
    // order of preference, it is RECOMMENDED that unless there is a
    // specific reason, the answerer list formats in the same relative order
    // they were present in the offer.
    // This can be skipped when the transceiver has any codec preferences.
    std::unordered_map<int, int> payload_type_preferences;
    int preference = static_cast<int>(offered_codecs.size() + 1);
    for (const C& codec : offered_codecs) {
      payload_type_preferences[codec.id] = preference--;
    }
    absl::c_sort(*negotiated_codecs, [&payload_type_preferences](const C& a,
                                                                 const C& b) {
      return payload_type_preferences[a.id] > payload_type_preferences[b.id];
    });
  }
}

// Finds a codec in |codecs2| that matches |codec_to_match|, which is
// a member of |codecs1|. If |codec_to_match| is an RTX codec, both
// the codecs themselves and their associated codecs must match.
template <class C>
static bool FindMatchingCodec(const std::vector<C>& codecs1,
                              const std::vector<C>& codecs2,
                              const C& codec_to_match,
                              C* found_codec) {
  // |codec_to_match| should be a member of |codecs1|, in order to look up RTX
  // codecs' associated codecs correctly. If not, that's a programming error.
  RTC_DCHECK(absl::c_any_of(codecs1, [&codec_to_match](const C& codec) {
    return &codec == &codec_to_match;
  }));
  for (const C& potential_match : codecs2) {
    if (potential_match.Matches(codec_to_match)) {
      if (IsRtxCodec(codec_to_match)) {
        int apt_value_1 = 0;
        int apt_value_2 = 0;
        if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType,
                                     &apt_value_1) ||
            !potential_match.GetParam(kCodecParamAssociatedPayloadType,
                                      &apt_value_2)) {
          RTC_LOG(LS_WARNING) << "RTX missing associated payload type.";
          continue;
        }
        if (!ReferencedCodecsMatch(codecs1, apt_value_1, codecs2,
                                   apt_value_2)) {
          continue;
        }
      }
      if (found_codec) {
        *found_codec = potential_match;
      }
      return true;
    }
  }
  return false;
}

// Find the codec in |codec_list| that |rtx_codec| is associated with.
template <class C>
static const C* GetAssociatedCodec(const std::vector<C>& codec_list,
                                   const C& rtx_codec) {
  std::string associated_pt_str;
  if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
                          &associated_pt_str)) {
    RTC_LOG(LS_WARNING) << "RTX codec " << rtx_codec.name
                        << " is missing an associated payload type.";
    return nullptr;
  }

  int associated_pt;
  if (!rtc::FromString(associated_pt_str, &associated_pt)) {
    RTC_LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str
                        << " of RTX codec " << rtx_codec.name
                        << " to an integer.";
    return nullptr;
  }

  // Find the associated reference codec for the reference RTX codec.
  const C* associated_codec = FindCodecById(codec_list, associated_pt);
  if (!associated_codec) {
    RTC_LOG(LS_WARNING) << "Couldn't find associated codec with payload type "
                        << associated_pt << " for RTX codec " << rtx_codec.name
                        << ".";
  }
  return associated_codec;
}

// Adds all codecs from |reference_codecs| to |offered_codecs| that don't
// already exist in |offered_codecs| and ensure the payload types don't
// collide.
template <class C>
static void MergeCodecs(const std::vector<C>& reference_codecs,
                        std::vector<C>* offered_codecs,
                        UsedPayloadTypes* used_pltypes) {
  // Add all new codecs that are not RTX codecs.
  for (const C& reference_codec : reference_codecs) {
    if (!IsRtxCodec(reference_codec) &&
        !FindMatchingCodec<C>(reference_codecs, *offered_codecs,
                              reference_codec, nullptr)) {
      C codec = reference_codec;
      used_pltypes->FindAndSetIdUsed(&codec);
      offered_codecs->push_back(codec);
    }
  }

  // Add all new RTX codecs.
  for (const C& reference_codec : reference_codecs) {
    if (IsRtxCodec(reference_codec) &&
        !FindMatchingCodec<C>(reference_codecs, *offered_codecs,
                              reference_codec, nullptr)) {
      C rtx_codec = reference_codec;
      const C* associated_codec =
          GetAssociatedCodec(reference_codecs, rtx_codec);
      if (!associated_codec) {
        continue;
      }
      // Find a codec in the offered list that matches the reference codec.
      // Its payload type may be different than the reference codec.
      C matching_codec;
      if (!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
                                *associated_codec, &matching_codec)) {
        RTC_LOG(LS_WARNING)
            << "Couldn't find matching " << associated_codec->name << " codec.";
        continue;
      }

      rtx_codec.params[kCodecParamAssociatedPayloadType] =
          rtc::ToString(matching_codec.id);
      used_pltypes->FindAndSetIdUsed(&rtx_codec);
      offered_codecs->push_back(rtx_codec);
    }
  }
}

template <typename Codecs>
static Codecs MatchCodecPreference(
    const std::vector<webrtc::RtpCodecCapability>& codec_preferences,
    const Codecs& codecs) {
  Codecs filtered_codecs;
  std::set<std::string> kept_codecs_ids;
  bool want_rtx = false;

  for (const auto& codec_preference : codec_preferences) {
    auto found_codec = absl::c_find_if(
        codecs, [&codec_preference](const typename Codecs::value_type& codec) {
          webrtc::RtpCodecParameters codec_parameters =
              codec.ToCodecParameters();
          return codec_parameters.name == codec_preference.name &&
                 codec_parameters.kind == codec_preference.kind &&
                 codec_parameters.num_channels ==
                     codec_preference.num_channels &&
                 codec_parameters.clock_rate == codec_preference.clock_rate &&
                 codec_parameters.parameters == codec_preference.parameters;
        });

    if (found_codec != codecs.end()) {
      filtered_codecs.push_back(*found_codec);
      kept_codecs_ids.insert(std::to_string(found_codec->id));
    } else if (IsRtxCodec(codec_preference)) {
      want_rtx = true;
    }
  }

  if (want_rtx) {
    for (const auto& codec : codecs) {
      if (IsRtxCodec(codec)) {
        const auto apt =
            codec.params.find(cricket::kCodecParamAssociatedPayloadType);
        if (apt != codec.params.end() &&
            kept_codecs_ids.count(apt->second) > 0) {
          filtered_codecs.push_back(codec);
        }
      }
    }
  }

  return filtered_codecs;
}

static bool FindByUriAndEncryption(const RtpHeaderExtensions& extensions,
                                   const webrtc::RtpExtension& ext_to_match,
                                   webrtc::RtpExtension* found_extension) {
  auto it = absl::c_find_if(
      extensions, [&ext_to_match](const webrtc::RtpExtension& extension) {
        // We assume that all URIs are given in a canonical
        // format.
        return extension.uri == ext_to_match.uri &&
               extension.encrypt == ext_to_match.encrypt;
      });
  if (it == extensions.end()) {
    return false;
  }
  if (found_extension) {
    *found_extension = *it;
  }
  return true;
}

static bool FindByUri(const RtpHeaderExtensions& extensions,
                      const webrtc::RtpExtension& ext_to_match,
                      webrtc::RtpExtension* found_extension) {
  // We assume that all URIs are given in a canonical format.
  const webrtc::RtpExtension* found =
      webrtc::RtpExtension::FindHeaderExtensionByUri(extensions,
                                                     ext_to_match.uri);
  if (!found) {
    return false;
  }
  if (found_extension) {
    *found_extension = *found;
  }
  return true;
}

static bool FindByUriWithEncryptionPreference(
    const RtpHeaderExtensions& extensions,
    absl::string_view uri_to_match,
    bool encryption_preference,
    webrtc::RtpExtension* found_extension) {
  const webrtc::RtpExtension* unencrypted_extension = nullptr;
  for (const webrtc::RtpExtension& extension : extensions) {
    // We assume that all URIs are given in a canonical format.
    if (extension.uri == uri_to_match) {
      if (!encryption_preference || extension.encrypt) {
        if (found_extension) {
          *found_extension = extension;
        }
        return true;
      }
      unencrypted_extension = &extension;
    }
  }
  if (unencrypted_extension) {
    if (found_extension) {
      *found_extension = *unencrypted_extension;
    }
    return true;
  }
  return false;
}

// Adds all extensions from |reference_extensions| to |offered_extensions| that
// don't already exist in |offered_extensions| and ensure the IDs don't
// collide. If an extension is added, it's also added to |regular_extensions| or
// |encrypted_extensions|, and if the extension is in |regular_extensions| or
// |encrypted_extensions|, its ID is marked as used in |used_ids|.
// |offered_extensions| is for either audio or video while |regular_extensions|
// and |encrypted_extensions| are used for both audio and video. There could be
// overlap between audio extensions and video extensions.
static void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions,
                            RtpHeaderExtensions* offered_extensions,
                            RtpHeaderExtensions* regular_extensions,
                            RtpHeaderExtensions* encrypted_extensions,
                            UsedRtpHeaderExtensionIds* used_ids) {
  for (auto reference_extension : reference_extensions) {
    if (!FindByUriAndEncryption(*offered_extensions, reference_extension,
                                nullptr)) {
      webrtc::RtpExtension existing;
      if (reference_extension.encrypt) {
        if (FindByUriAndEncryption(*encrypted_extensions, reference_extension,
                                   &existing)) {
          offered_extensions->push_back(existing);
        } else {
          used_ids->FindAndSetIdUsed(&reference_extension);
          encrypted_extensions->push_back(reference_extension);
          offered_extensions->push_back(reference_extension);
        }
      } else {
        if (FindByUriAndEncryption(*regular_extensions, reference_extension,
                                   &existing)) {
          offered_extensions->push_back(existing);
        } else {
          used_ids->FindAndSetIdUsed(&reference_extension);
          regular_extensions->push_back(reference_extension);
          offered_extensions->push_back(reference_extension);
        }
      }
    }
  }
}

static void AddEncryptedVersionsOfHdrExts(RtpHeaderExtensions* extensions,
                                          RtpHeaderExtensions* all_extensions,
                                          UsedRtpHeaderExtensionIds* used_ids) {
  RtpHeaderExtensions encrypted_extensions;
  for (const webrtc::RtpExtension& extension : *extensions) {
    webrtc::RtpExtension existing;
    // Don't add encrypted extensions again that were already included in a
    // previous offer or regular extensions that are also included as encrypted
    // extensions.
    if (extension.encrypt ||
        !webrtc::RtpExtension::IsEncryptionSupported(extension.uri) ||
        (FindByUriWithEncryptionPreference(*extensions, extension.uri, true,
                                           &existing) &&
         existing.encrypt)) {
      continue;
    }

    if (FindByUri(*all_extensions, extension, &existing)) {
      encrypted_extensions.push_back(existing);
    } else {
      webrtc::RtpExtension encrypted(extension);
      encrypted.encrypt = true;
      used_ids->FindAndSetIdUsed(&encrypted);
      all_extensions->push_back(encrypted);
      encrypted_extensions.push_back(encrypted);
    }
  }
  extensions->insert(extensions->end(), encrypted_extensions.begin(),
                     encrypted_extensions.end());
}

static void NegotiateRtpHeaderExtensions(
    const RtpHeaderExtensions& local_extensions,
    const RtpHeaderExtensions& offered_extensions,
    bool enable_encrypted_rtp_header_extensions,
    RtpHeaderExtensions* negotiated_extensions) {
  // TransportSequenceNumberV2 is not offered by default. The special logic for
  // the TransportSequenceNumber extensions works as follows:
  // Offer       Answer
  // V1          V1 if in local_extensions.
  // V1 and V2   V2 regardless of local_extensions.
  // V2          V2 regardless of local_extensions.
  const webrtc::RtpExtension* transport_sequence_number_v2_offer =
      webrtc::RtpExtension::FindHeaderExtensionByUri(
          offered_extensions,
          webrtc::RtpExtension::kTransportSequenceNumberV2Uri);

  bool frame_descriptor_in_local = false;
  bool dependency_descriptor_in_local = false;
  bool abs_capture_time_in_local = false;

  for (const webrtc::RtpExtension& ours : local_extensions) {
    if (ours.uri == webrtc::RtpExtension::kGenericFrameDescriptorUri00)
      frame_descriptor_in_local = true;
    else if (ours.uri == webrtc::RtpExtension::kDependencyDescriptorUri)
      dependency_descriptor_in_local = true;
    else if (ours.uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri)
      abs_capture_time_in_local = true;
    webrtc::RtpExtension theirs;
    if (FindByUriWithEncryptionPreference(
            offered_extensions, ours.uri,
            enable_encrypted_rtp_header_extensions, &theirs)) {
      if (transport_sequence_number_v2_offer &&
          ours.uri == webrtc::RtpExtension::kTransportSequenceNumberUri) {
        // Don't respond to
        // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
        // if we get an offer including
        // http://www.webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02
        continue;
      } else {
        // We respond with their RTP header extension id.
        negotiated_extensions->push_back(theirs);
      }
    }
  }

  if (transport_sequence_number_v2_offer) {
    // Respond that we support kTransportSequenceNumberV2Uri.
    negotiated_extensions->push_back(*transport_sequence_number_v2_offer);
  }

  // Frame descriptors support. If the extension is not present locally, but is
  // in the offer, we add it to the list.
  webrtc::RtpExtension theirs;
  if (!dependency_descriptor_in_local &&
      FindByUriWithEncryptionPreference(
          offered_extensions, webrtc::RtpExtension::kDependencyDescriptorUri,
          enable_encrypted_rtp_header_extensions, &theirs)) {
    negotiated_extensions->push_back(theirs);
  }
  if (!frame_descriptor_in_local &&
      FindByUriWithEncryptionPreference(
          offered_extensions,
          webrtc::RtpExtension::kGenericFrameDescriptorUri00,
          enable_encrypted_rtp_header_extensions, &theirs)) {
    negotiated_extensions->push_back(theirs);
  }

  // Absolute capture time support. If the extension is not present locally, but
  // is in the offer, we add it to the list.
  if (!abs_capture_time_in_local &&
      FindByUriWithEncryptionPreference(
          offered_extensions, webrtc::RtpExtension::kAbsoluteCaptureTimeUri,
          enable_encrypted_rtp_header_extensions, &theirs)) {
    negotiated_extensions->push_back(theirs);
  }
}

static void StripCNCodecs(AudioCodecs* audio_codecs) {
  audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(),
                                     [](const AudioCodec& codec) {
                                       return absl::EqualsIgnoreCase(
                                           codec.name, kComfortNoiseCodecName);
                                     }),
                      audio_codecs->end());
}

template <class C>
static bool SetCodecsInAnswer(
    const MediaContentDescriptionImpl<C>* offer,
    const std::vector<C>& local_codecs,
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    UniqueRandomIdGenerator* ssrc_generator,
    StreamParamsVec* current_streams,
    MediaContentDescriptionImpl<C>* answer) {
  std::vector<C> negotiated_codecs;
  NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs,
                  media_description_options.codec_preferences.empty());
  answer->AddCodecs(negotiated_codecs);
  answer->set_protocol(offer->protocol());
  if (!AddStreamParams(media_description_options.sender_options,
                       session_options.rtcp_cname, ssrc_generator,
                       current_streams, answer)) {
    return false;  // Something went seriously wrong.
  }
  return true;
}

// Create a media content to be answered for the given |sender_options|
// according to the given session_options.rtcp_mux, session_options.streams,
// codecs, crypto, and current_streams.  If we don't currently have crypto (in
// current_cryptos) and it is enabled (in secure_policy), crypto is created
// (according to crypto_suites). The codecs, rtcp_mux, and crypto are all
// negotiated with the offer. If the negotiation fails, this method returns
// false.  The created content is added to the offer.
static bool CreateMediaContentAnswer(
    const MediaContentDescription* offer,
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const SecurePolicy& sdes_policy,
    const CryptoParamsVec* current_cryptos,
    const RtpHeaderExtensions& local_rtp_extenstions,
    UniqueRandomIdGenerator* ssrc_generator,
    bool enable_encrypted_rtp_header_extensions,
    StreamParamsVec* current_streams,
    bool bundle_enabled,
    MediaContentDescription* answer) {
  answer->set_extmap_allow_mixed_enum(offer->extmap_allow_mixed_enum());
  RtpHeaderExtensions negotiated_rtp_extensions;
  NegotiateRtpHeaderExtensions(
      local_rtp_extenstions, offer->rtp_header_extensions(),
      enable_encrypted_rtp_header_extensions, &negotiated_rtp_extensions);
  answer->set_rtp_header_extensions(negotiated_rtp_extensions);

  answer->set_rtcp_mux(session_options.rtcp_mux_enabled && offer->rtcp_mux());
  if (answer->type() == cricket::MEDIA_TYPE_VIDEO) {
    answer->set_rtcp_reduced_size(offer->rtcp_reduced_size());
  }

  answer->set_remote_estimate(offer->remote_estimate());

  if (sdes_policy != SEC_DISABLED) {
    CryptoParams crypto;
    if (SelectCrypto(offer, bundle_enabled, session_options.crypto_options,
                     &crypto)) {
      if (current_cryptos) {
        FindMatchingCrypto(*current_cryptos, crypto, &crypto);
      }
      answer->AddCrypto(crypto);
    }
  }

  if (answer->cryptos().empty() && sdes_policy == SEC_REQUIRED) {
    return false;
  }

  AddSimulcastToMediaDescription(media_description_options, answer);

  answer->set_direction(NegotiateRtpTransceiverDirection(
      offer->direction(), media_description_options.direction));

  if (offer->alt_protocol() == media_description_options.alt_protocol) {
    answer->set_alt_protocol(media_description_options.alt_protocol);
  }
  return true;
}

static bool IsMediaProtocolSupported(MediaType type,
                                     const std::string& protocol,
                                     bool secure_transport) {
  // Since not all applications serialize and deserialize the media protocol,
  // we will have to accept |protocol| to be empty.
  if (protocol.empty()) {
    return true;
  }

  if (type == MEDIA_TYPE_DATA) {
    // Check for SCTP, but also for RTP for RTP-based data channels.
    // TODO(pthatcher): Remove RTP once RTP-based data channels are gone.
    if (secure_transport) {
      // Most likely scenarios first.
      return IsDtlsSctp(protocol) || IsDtlsRtp(protocol) ||
             IsPlainRtp(protocol);
    } else {
      return IsPlainSctp(protocol) || IsPlainRtp(protocol);
    }
  }

  // Allow for non-DTLS RTP protocol even when using DTLS because that's what
  // JSEP specifies.
  if (secure_transport) {
    // Most likely scenarios first.
    return IsDtlsRtp(protocol) || IsPlainRtp(protocol);
  } else {
    return IsPlainRtp(protocol);
  }
}

static void SetMediaProtocol(bool secure_transport,
                             MediaContentDescription* desc) {
  if (!desc->cryptos().empty())
    desc->set_protocol(kMediaProtocolSavpf);
  else if (secure_transport)
    desc->set_protocol(kMediaProtocolDtlsSavpf);
  else
    desc->set_protocol(kMediaProtocolAvpf);
}

// Gets the TransportInfo of the given |content_name| from the
// |current_description|. If doesn't exist, returns a new one.
static const TransportDescription* GetTransportDescription(
    const std::string& content_name,
    const SessionDescription* current_description) {
  const TransportDescription* desc = NULL;
  if (current_description) {
    const TransportInfo* info =
        current_description->GetTransportInfoByName(content_name);
    if (info) {
      desc = &info->description;
    }
  }
  return desc;
}

// Gets the current DTLS state from the transport description.
static bool IsDtlsActive(const ContentInfo* content,
                         const SessionDescription* current_description) {
  if (!content) {
    return false;
  }

  size_t msection_index = content - &current_description->contents()[0];

  if (current_description->transport_infos().size() <= msection_index) {
    return false;
  }

  return current_description->transport_infos()[msection_index]
      .description.secure();
}

void MediaDescriptionOptions::AddAudioSender(
    const std::string& track_id,
    const std::vector<std::string>& stream_ids) {
  RTC_DCHECK(type == MEDIA_TYPE_AUDIO);
  AddSenderInternal(track_id, stream_ids, {}, SimulcastLayerList(), 1);
}

void MediaDescriptionOptions::AddVideoSender(
    const std::string& track_id,
    const std::vector<std::string>& stream_ids,
    const std::vector<RidDescription>& rids,
    const SimulcastLayerList& simulcast_layers,
    int num_sim_layers) {
  RTC_DCHECK(type == MEDIA_TYPE_VIDEO);
  RTC_DCHECK(rids.empty() || num_sim_layers == 0)
      << "RIDs are the compliant way to indicate simulcast.";
  RTC_DCHECK(ValidateSimulcastLayers(rids, simulcast_layers));
  AddSenderInternal(track_id, stream_ids, rids, simulcast_layers,
                    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,
    const std::vector<RidDescription>& rids,
    const SimulcastLayerList& simulcast_layers,
    int num_sim_layers) {
  // TODO(steveanton): Support any number of stream ids.
  RTC_CHECK(stream_ids.size() == 1U);
  SenderOptions options;
  options.track_id = track_id;
  options.stream_ids = stream_ids;
  options.simulcast_layers = simulcast_layers;
  options.rids = rids;
  options.num_sim_layers = num_sim_layers;
  sender_options.push_back(options);
}

bool MediaSessionOptions::HasMediaDescription(MediaType type) const {
  return absl::c_any_of(
      media_description_options,
      [type](const MediaDescriptionOptions& t) { return t.type == type; });
}

MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
    const TransportDescriptionFactory* transport_desc_factory,
    rtc::UniqueRandomIdGenerator* ssrc_generator)
    : ssrc_generator_(ssrc_generator),
      transport_desc_factory_(transport_desc_factory) {
  RTC_DCHECK(ssrc_generator_);
}

MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
    ChannelManager* channel_manager,
    const TransportDescriptionFactory* transport_desc_factory,
    rtc::UniqueRandomIdGenerator* ssrc_generator)
    : MediaSessionDescriptionFactory(transport_desc_factory, ssrc_generator) {
  channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
  channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_);
  audio_rtp_extensions_ =
      channel_manager->GetDefaultEnabledAudioRtpHeaderExtensions();
  channel_manager->GetSupportedVideoSendCodecs(&video_send_codecs_);
  channel_manager->GetSupportedVideoReceiveCodecs(&video_recv_codecs_);
  video_rtp_extensions_ =
      channel_manager->GetDefaultEnabledVideoRtpHeaderExtensions();
  channel_manager->GetSupportedDataCodecs(&rtp_data_codecs_);
  ComputeAudioCodecsIntersectionAndUnion();
  ComputeVideoCodecsIntersectionAndUnion();
}

const AudioCodecs& MediaSessionDescriptionFactory::audio_sendrecv_codecs()
    const {
  return audio_sendrecv_codecs_;
}

const AudioCodecs& MediaSessionDescriptionFactory::audio_send_codecs() const {
  return audio_send_codecs_;
}

const AudioCodecs& MediaSessionDescriptionFactory::audio_recv_codecs() const {
  return audio_recv_codecs_;
}

void MediaSessionDescriptionFactory::set_audio_codecs(
    const AudioCodecs& send_codecs,
    const AudioCodecs& recv_codecs) {
  audio_send_codecs_ = send_codecs;
  audio_recv_codecs_ = recv_codecs;
  ComputeAudioCodecsIntersectionAndUnion();
}

const VideoCodecs& MediaSessionDescriptionFactory::video_sendrecv_codecs()
    const {
  return video_sendrecv_codecs_;
}

const VideoCodecs& MediaSessionDescriptionFactory::video_send_codecs() const {
  return video_send_codecs_;
}

const VideoCodecs& MediaSessionDescriptionFactory::video_recv_codecs() const {
  return video_recv_codecs_;
}

void MediaSessionDescriptionFactory::set_video_codecs(
    const VideoCodecs& send_codecs,
    const VideoCodecs& recv_codecs) {
  video_send_codecs_ = send_codecs;
  video_recv_codecs_ = recv_codecs;
  ComputeVideoCodecsIntersectionAndUnion();
}

static void RemoveUnifiedPlanExtensions(RtpHeaderExtensions* extensions) {
  RTC_DCHECK(extensions);

  extensions->erase(
      std::remove_if(extensions->begin(), extensions->end(),
                     [](auto extension) {
                       return extension.uri == webrtc::RtpExtension::kMidUri ||
                              extension.uri == webrtc::RtpExtension::kRidUri ||
                              extension.uri ==
                                  webrtc::RtpExtension::kRepairedRidUri;
                     }),
      extensions->end());
}

RtpHeaderExtensions
MediaSessionDescriptionFactory::audio_rtp_header_extensions() const {
  RtpHeaderExtensions extensions = audio_rtp_extensions_;
  if (!is_unified_plan_) {
    RemoveUnifiedPlanExtensions(&extensions);
  }

  return extensions;
}

RtpHeaderExtensions
MediaSessionDescriptionFactory::video_rtp_header_extensions() const {
  RtpHeaderExtensions extensions = video_rtp_extensions_;
  if (!is_unified_plan_) {
    RemoveUnifiedPlanExtensions(&extensions);
  }

  return extensions;
}

std::unique_ptr<SessionDescription> MediaSessionDescriptionFactory::CreateOffer(
    const MediaSessionOptions& session_options,
    const SessionDescription* current_description) const {
  // Must have options for each existing section.
  if (current_description) {
    RTC_DCHECK_LE(current_description->contents().size(),
                  session_options.media_description_options.size());
  }

  IceCredentialsIterator ice_credentials(
      session_options.pooled_ice_credentials);

  std::vector<const ContentInfo*> current_active_contents;
  if (current_description) {
    current_active_contents =
        GetActiveContents(*current_description, session_options);
  }

  StreamParamsVec current_streams =
      GetCurrentStreamParams(current_active_contents);

  AudioCodecs offer_audio_codecs;
  VideoCodecs offer_video_codecs;
  RtpDataCodecs offer_rtp_data_codecs;
  GetCodecsForOffer(current_active_contents, &offer_audio_codecs,
                    &offer_video_codecs, &offer_rtp_data_codecs);

  if (!session_options.vad_enabled) {
    // If application doesn't want CN codecs in offer.
    StripCNCodecs(&offer_audio_codecs);
  }
  FilterDataCodecs(&offer_rtp_data_codecs,
                   session_options.data_channel_type == DCT_SCTP);

  RtpHeaderExtensions audio_rtp_extensions;
  RtpHeaderExtensions video_rtp_extensions;
  GetRtpHdrExtsToOffer(current_active_contents,
                       session_options.offer_extmap_allow_mixed,
                       &audio_rtp_extensions, &video_rtp_extensions);

  auto offer = std::make_unique<SessionDescription>();

  // Iterate through the media description options, matching with existing media
  // descriptions in |current_description|.
  size_t msection_index = 0;
  for (const MediaDescriptionOptions& media_description_options :
       session_options.media_description_options) {
    const ContentInfo* current_content = nullptr;
    if (current_description &&
        msection_index < current_description->contents().size()) {
      current_content = &current_description->contents()[msection_index];
      // Media type must match unless this media section is being recycled.
      RTC_DCHECK(current_content->name != media_description_options.mid ||
                 IsMediaContentOfType(current_content,
                                      media_description_options.type));
    }
    switch (media_description_options.type) {
      case MEDIA_TYPE_AUDIO:
        if (!AddAudioContentForOffer(
                media_description_options, session_options, current_content,
                current_description, audio_rtp_extensions, offer_audio_codecs,
                &current_streams, offer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      case MEDIA_TYPE_VIDEO:
        if (!AddVideoContentForOffer(
                media_description_options, session_options, current_content,
                current_description, video_rtp_extensions, offer_video_codecs,
                &current_streams, offer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      case MEDIA_TYPE_DATA:
        if (!AddDataContentForOffer(media_description_options, session_options,
                                    current_content, current_description,
                                    offer_rtp_data_codecs, &current_streams,
                                    offer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      default:
        RTC_NOTREACHED();
    }
    ++msection_index;
  }

  // Bundle the contents together, if we've been asked to do so, and update any
  // parameters that need to be tweaked for BUNDLE.
  if (session_options.bundle_enabled) {
    ContentGroup offer_bundle(GROUP_TYPE_BUNDLE);
    for (const ContentInfo& content : offer->contents()) {
      if (content.rejected) {
        continue;
      }
      // TODO(deadbeef): There are conditions that make bundling two media
      // descriptions together illegal. For example, they use the same payload
      // type to represent different codecs, or same IDs for different header
      // extensions. We need to detect this and not try to bundle those media
      // descriptions together.
      offer_bundle.AddContentName(content.name);
    }
    if (!offer_bundle.content_names().empty()) {
      offer->AddGroup(offer_bundle);
      if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) {
        RTC_LOG(LS_ERROR)
            << "CreateOffer failed to UpdateTransportInfoForBundle.";
        return nullptr;
      }
      if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) {
        RTC_LOG(LS_ERROR)
            << "CreateOffer failed to UpdateCryptoParamsForBundle.";
        return nullptr;
      }
    }
  }

  // The following determines how to signal MSIDs to ensure compatibility with
  // older endpoints (in particular, older Plan B endpoints).
  if (is_unified_plan_) {
    // Be conservative and signal using both a=msid and a=ssrc lines. Unified
    // Plan answerers will look at a=msid and Plan B answerers will look at the
    // a=ssrc MSID line.
    offer->set_msid_signaling(cricket::kMsidSignalingMediaSection |
                              cricket::kMsidSignalingSsrcAttribute);
  } else {
    // Plan B always signals MSID using a=ssrc lines.
    offer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
  }

  offer->set_extmap_allow_mixed(session_options.offer_extmap_allow_mixed);

  return offer;
}

std::unique_ptr<SessionDescription>
MediaSessionDescriptionFactory::CreateAnswer(
    const SessionDescription* offer,
    const MediaSessionOptions& session_options,
    const SessionDescription* current_description) const {
  if (!offer) {
    return nullptr;
  }

  // Must have options for exactly as many sections as in the offer.
  RTC_DCHECK_EQ(offer->contents().size(),
                session_options.media_description_options.size());

  IceCredentialsIterator ice_credentials(
      session_options.pooled_ice_credentials);

  std::vector<const ContentInfo*> current_active_contents;
  if (current_description) {
    current_active_contents =
        GetActiveContents(*current_description, session_options);
  }

  StreamParamsVec current_streams =
      GetCurrentStreamParams(current_active_contents);

  // Get list of all possible codecs that respects existing payload type
  // mappings and uses a single payload type space.
  //
  // Note that these lists may be further filtered for each m= section; this
  // step is done just to establish the payload type mappings shared by all
  // 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);

  if (!session_options.vad_enabled) {
    // If application doesn't want CN codecs in answer.
    StripCNCodecs(&answer_audio_codecs);
  }
  FilterDataCodecs(&answer_rtp_data_codecs,
                   session_options.data_channel_type == DCT_SCTP);

  auto answer = std::make_unique<SessionDescription>();

  // If the offer supports BUNDLE, and we want to use it too, create a BUNDLE
  // group in the answer with the appropriate content names.
  const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE);
  ContentGroup answer_bundle(GROUP_TYPE_BUNDLE);
  // Transport info shared by the bundle group.
  std::unique_ptr<TransportInfo> bundle_transport;

  answer->set_extmap_allow_mixed(offer->extmap_allow_mixed());

  // Iterate through the media description options, matching with existing
  // media descriptions in |current_description|.
  size_t msection_index = 0;
  for (const MediaDescriptionOptions& media_description_options :
       session_options.media_description_options) {
    const ContentInfo* offer_content = &offer->contents()[msection_index];
    // Media types and MIDs must match between the remote offer and the
    // MediaDescriptionOptions.
    RTC_DCHECK(
        IsMediaContentOfType(offer_content, media_description_options.type));
    RTC_DCHECK(media_description_options.mid == offer_content->name);
    const ContentInfo* current_content = nullptr;
    if (current_description &&
        msection_index < current_description->contents().size()) {
      current_content = &current_description->contents()[msection_index];
    }
    switch (media_description_options.type) {
      case MEDIA_TYPE_AUDIO:
        if (!AddAudioContentForAnswer(
                media_description_options, session_options, offer_content,
                offer, current_content, current_description,
                bundle_transport.get(), answer_audio_codecs, &current_streams,
                answer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      case MEDIA_TYPE_VIDEO:
        if (!AddVideoContentForAnswer(
                media_description_options, session_options, offer_content,
                offer, current_content, current_description,
                bundle_transport.get(), answer_video_codecs, &current_streams,
                answer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      case MEDIA_TYPE_DATA:
        if (!AddDataContentForAnswer(
                media_description_options, session_options, offer_content,
                offer, current_content, current_description,
                bundle_transport.get(), answer_rtp_data_codecs,
                &current_streams, answer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      default:
        RTC_NOTREACHED();
    }
    ++msection_index;
    // See if we can add the newly generated m= section to the BUNDLE group in
    // the answer.
    ContentInfo& added = answer->contents().back();
    if (!added.rejected && session_options.bundle_enabled && offer_bundle &&
        offer_bundle->HasContentName(added.name)) {
      answer_bundle.AddContentName(added.name);
      bundle_transport.reset(
          new TransportInfo(*answer->GetTransportInfoByName(added.name)));
    }
  }

  // If a BUNDLE group was offered, put a BUNDLE group in the answer even if
  // it's empty. RFC5888 says:
  //
  //   A SIP entity that receives an offer that contains an "a=group" line
  //   with semantics that are understood MUST return an answer that
  //   contains an "a=group" line with the same semantics.
  if (offer_bundle) {
    answer->AddGroup(answer_bundle);
  }

  if (answer_bundle.FirstContentName()) {
    // Share the same ICE credentials and crypto params across all contents,
    // as BUNDLE requires.
    if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) {
      RTC_LOG(LS_ERROR)
          << "CreateAnswer failed to UpdateTransportInfoForBundle.";
      return NULL;
    }

    if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) {
      RTC_LOG(LS_ERROR)
          << "CreateAnswer failed to UpdateCryptoParamsForBundle.";
      return NULL;
    }
  }

  // The following determines how to signal MSIDs to ensure compatibility with
  // older endpoints (in particular, older Plan B endpoints).
  if (is_unified_plan_) {
    // Unified Plan needs to look at what the offer included to find the most
    // compatible answer.
    if (offer->msid_signaling() == 0) {
      // We end up here in one of three cases:
      // 1. An empty offer. We'll reply with an empty answer so it doesn't
      //    matter what we pick here.
      // 2. A data channel only offer. We won't add any MSIDs to the answer so
      //    it also doesn't matter what we pick here.
      // 3. Media that's either sendonly or inactive from the remote endpoint.
      //    We don't have any information to say whether the endpoint is Plan B
      //    or Unified Plan, so be conservative and send both.
      answer->set_msid_signaling(cricket::kMsidSignalingMediaSection |
                                 cricket::kMsidSignalingSsrcAttribute);
    } else if (offer->msid_signaling() ==
               (cricket::kMsidSignalingMediaSection |
                cricket::kMsidSignalingSsrcAttribute)) {
      // If both a=msid and a=ssrc MSID signaling methods were used, we're
      // probably talking to a Unified Plan endpoint so respond with just
      // a=msid.
      answer->set_msid_signaling(cricket::kMsidSignalingMediaSection);
    } else {
      // Otherwise, it's clear which method the offerer is using so repeat that
      // back to them.
      answer->set_msid_signaling(offer->msid_signaling());
    }
  } else {
    // Plan B always signals MSID using a=ssrc lines.
    answer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
  }

  return answer;
}

const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer(
    const RtpTransceiverDirection& direction) const {
  switch (direction) {
    // If stream is inactive - generate list as if sendrecv.
    case RtpTransceiverDirection::kSendRecv:
    case RtpTransceiverDirection::kInactive:
      return audio_sendrecv_codecs_;
    case RtpTransceiverDirection::kSendOnly:
      return audio_send_codecs_;
    case RtpTransceiverDirection::kRecvOnly:
      return audio_recv_codecs_;
    case RtpTransceiverDirection::kStopped:
      RTC_NOTREACHED();
      return audio_sendrecv_codecs_;
  }
}

const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer(
    const RtpTransceiverDirection& offer,
    const RtpTransceiverDirection& answer) const {
  switch (answer) {
    // For inactive and sendrecv answers, generate lists as if we were to accept
    // the offer's direction. See RFC 3264 Section 6.1.
    case RtpTransceiverDirection::kSendRecv:
    case RtpTransceiverDirection::kInactive:
      return GetAudioCodecsForOffer(
          webrtc::RtpTransceiverDirectionReversed(offer));
    case RtpTransceiverDirection::kSendOnly:
      return audio_send_codecs_;
    case RtpTransceiverDirection::kRecvOnly:
      return audio_recv_codecs_;
    case RtpTransceiverDirection::kStopped:
      RTC_NOTREACHED();
      return audio_sendrecv_codecs_;
  }
}

const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForOffer(
    const RtpTransceiverDirection& direction) const {
  switch (direction) {
    // If stream is inactive - generate list as if sendrecv.
    case RtpTransceiverDirection::kSendRecv:
    case RtpTransceiverDirection::kInactive:
      return video_sendrecv_codecs_;
    case RtpTransceiverDirection::kSendOnly:
      return video_send_codecs_;
    case RtpTransceiverDirection::kRecvOnly:
      return video_recv_codecs_;
    case RtpTransceiverDirection::kStopped:
      RTC_NOTREACHED();
      return video_sendrecv_codecs_;
  }
}

const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForAnswer(
    const RtpTransceiverDirection& offer,
    const RtpTransceiverDirection& answer) const {
  switch (answer) {
    // For inactive and sendrecv answers, generate lists as if we were to accept
    // the offer's direction. See RFC 3264 Section 6.1.
    case RtpTransceiverDirection::kSendRecv:
    case RtpTransceiverDirection::kInactive:
      return GetVideoCodecsForOffer(
          webrtc::RtpTransceiverDirectionReversed(offer));
    case RtpTransceiverDirection::kSendOnly:
      return video_send_codecs_;
    case RtpTransceiverDirection::kRecvOnly:
      return video_recv_codecs_;
    case RtpTransceiverDirection::kStopped:
      RTC_NOTREACHED();
      return video_sendrecv_codecs_;
  }
}

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)) {
      const AudioContentDescription* audio =
          content->media_description()->as_audio();
      MergeCodecs<AudioCodec>(audio->codecs(), audio_codecs, used_pltypes);
    } else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
      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);
      }
    }
  }
}

// Getting codecs for an offer involves these steps:
//
// 1. Construct payload type -> codec mappings for current description.
// 2. Add any reference codecs that weren't already present
// 3. For each individual media description (m= section), filter codecs based
//    on the directional attribute (happens in another method).
void MediaSessionDescriptionFactory::GetCodecsForOffer(
    const std::vector<const ContentInfo*>& current_active_contents,
    AudioCodecs* audio_codecs,
    VideoCodecs* video_codecs,
    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);

  // 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);
  MergeCodecs<DataCodec>(rtp_data_codecs_, rtp_data_codecs, &used_pltypes);
}

// Getting codecs for an answer involves these steps:
//
// 1. Construct payload type -> codec mappings for current description.
// 2. Add any codecs from the offer that weren't already present.
// 3. Add any remaining codecs that weren't already present.
// 4. For each individual media description (m= section), filter codecs based
//    on the directional attribute (happens in another method).
void MediaSessionDescriptionFactory::GetCodecsForAnswer(
    const std::vector<const ContentInfo*>& current_active_contents,
    const SessionDescription& remote_offer,
    AudioCodecs* audio_codecs,
    VideoCodecs* video_codecs,
    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);

  // 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 =
          content.media_description()->as_audio();
      for (const AudioCodec& offered_audio_codec : audio->codecs()) {
        if (!FindMatchingCodec<AudioCodec>(audio->codecs(),
                                           filtered_offered_audio_codecs,
                                           offered_audio_codec, nullptr) &&
            FindMatchingCodec<AudioCodec>(audio->codecs(), all_audio_codecs_,
                                          offered_audio_codec, nullptr)) {
          filtered_offered_audio_codecs.push_back(offered_audio_codec);
        }
      }
    } else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) {
      const VideoContentDescription* video =
          content.media_description()->as_video();
      for (const VideoCodec& offered_video_codec : video->codecs()) {
        if (!FindMatchingCodec<VideoCodec>(video->codecs(),
                                           filtered_offered_video_codecs,
                                           offered_video_codec, nullptr) &&
            FindMatchingCodec<VideoCodec>(video->codecs(), all_video_codecs_,
                                          offered_video_codec, nullptr)) {
          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);
          }
        }
      }
    }
  }

  // Add codecs that are not in the current description but were in
  // |remote_offer|.
  MergeCodecs<AudioCodec>(filtered_offered_audio_codecs, audio_codecs,
                          &used_pltypes);
  MergeCodecs<VideoCodec>(filtered_offered_video_codecs, video_codecs,
                          &used_pltypes);
  MergeCodecs<DataCodec>(filtered_offered_rtp_data_codecs, rtp_data_codecs,
                         &used_pltypes);
}

void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
    const std::vector<const ContentInfo*>& current_active_contents,
    bool extmap_allow_mixed,
    RtpHeaderExtensions* offer_audio_extensions,
    RtpHeaderExtensions* offer_video_extensions) const {
  // All header extensions allocated from the same range to avoid potential
  // issues when using BUNDLE.

  // Strictly speaking the SDP attribute extmap_allow_mixed signals that the
  // receiver supports an RTP stream where one- and two-byte RTP header
  // extensions are mixed. For backwards compatibility reasons it's used in
  // WebRTC to signal that two-byte RTP header extensions are supported.
  UsedRtpHeaderExtensionIds used_ids(
      extmap_allow_mixed ? UsedRtpHeaderExtensionIds::IdDomain::kTwoByteAllowed
                         : UsedRtpHeaderExtensionIds::IdDomain::kOneByteOnly);
  RtpHeaderExtensions all_regular_extensions;
  RtpHeaderExtensions all_encrypted_extensions;

  // First - get all extensions from the current description if the media type
  // is used.
  // Add them to |used_ids| so the local ids are not reused if a new media
  // type is added.
  for (const ContentInfo* content : current_active_contents) {
    if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
      const AudioContentDescription* audio =
          content->media_description()->as_audio();
      MergeRtpHdrExts(audio->rtp_header_extensions(), offer_audio_extensions,
                      &all_regular_extensions, &all_encrypted_extensions,
                      &used_ids);
    } else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
      const VideoContentDescription* video =
          content->media_description()->as_video();
      MergeRtpHdrExts(video->rtp_header_extensions(), offer_video_extensions,
                      &all_regular_extensions, &all_encrypted_extensions,
                      &used_ids);
    }
  }

  // Add our default RTP header extensions that are not in the current
  // description.
  MergeRtpHdrExts(audio_rtp_header_extensions(), offer_audio_extensions,
                  &all_regular_extensions, &all_encrypted_extensions,
                  &used_ids);
  MergeRtpHdrExts(video_rtp_header_extensions(), offer_video_extensions,
                  &all_regular_extensions, &all_encrypted_extensions,
                  &used_ids);

  // TODO(jbauch): Support adding encrypted header extensions to existing
  // sessions.
  if (enable_encrypted_rtp_header_extensions_ &&
      current_active_contents.empty()) {
    AddEncryptedVersionsOfHdrExts(offer_audio_extensions,
                                  &all_encrypted_extensions, &used_ids);
    AddEncryptedVersionsOfHdrExts(offer_video_extensions,
                                  &all_encrypted_extensions, &used_ids);
  }
}

bool MediaSessionDescriptionFactory::AddTransportOffer(
    const std::string& content_name,
    const TransportOptions& transport_options,
    const SessionDescription* current_desc,
    SessionDescription* offer_desc,
    IceCredentialsIterator* ice_credentials) const {
  if (!transport_desc_factory_)
    return false;
  const TransportDescription* current_tdesc =
      GetTransportDescription(content_name, current_desc);
  std::unique_ptr<TransportDescription> new_tdesc(
      transport_desc_factory_->CreateOffer(transport_options, current_tdesc,
                                           ice_credentials));
  if (!new_tdesc) {
    RTC_LOG(LS_ERROR) << "Failed to AddTransportOffer, content name="
                      << content_name;
  }
  offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc));
  return true;
}

std::unique_ptr<TransportDescription>
MediaSessionDescriptionFactory::CreateTransportAnswer(
    const std::string& content_name,
    const SessionDescription* offer_desc,
    const TransportOptions& transport_options,
    const SessionDescription* current_desc,
    bool require_transport_attributes,
    IceCredentialsIterator* ice_credentials) const {
  if (!transport_desc_factory_)
    return NULL;
  const TransportDescription* offer_tdesc =
      GetTransportDescription(content_name, offer_desc);
  const TransportDescription* current_tdesc =
      GetTransportDescription(content_name, current_desc);
  return transport_desc_factory_->CreateAnswer(offer_tdesc, transport_options,
                                               require_transport_attributes,
                                               current_tdesc, ice_credentials);
}

bool MediaSessionDescriptionFactory::AddTransportAnswer(
    const std::string& content_name,
    const TransportDescription& transport_desc,
    SessionDescription* answer_desc) const {
  answer_desc->AddTransportInfo(TransportInfo(content_name, transport_desc));
  return true;
}

// |audio_codecs| = set of all possible codecs that can be used, with correct
// payload type mappings
//
// |supported_audio_codecs| = set of codecs that are supported for the direction
// of this m= section
//
// acd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from acd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const ContentInfo* current_content,
    const SessionDescription* current_description,
    const RtpHeaderExtensions& audio_rtp_extensions,
    const AudioCodecs& audio_codecs,
    StreamParamsVec* current_streams,
    SessionDescription* desc,
    IceCredentialsIterator* ice_credentials) const {
  // Filter audio_codecs (which includes all codecs, with correctly remapped
  // payload types) based on transceiver direction.
  const AudioCodecs& supported_audio_codecs =
      GetAudioCodecsForOffer(media_description_options.direction);

  AudioCodecs filtered_codecs;

  if (!media_description_options.codec_preferences.empty()) {
    // Add the codecs from the current transceiver's codec preferences.
    // They override any existing codecs from previous negotiations.
    filtered_codecs = MatchCodecPreference(
        media_description_options.codec_preferences, supported_audio_codecs);
  } else {
    // Add the codecs from current content if it exists and is not rejected nor
    // recycled.
    if (current_content && !current_content->rejected &&
        current_content->name == media_description_options.mid) {
      RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO));
      const AudioContentDescription* acd =
          current_content->media_description()->as_audio();
      for (const AudioCodec& codec : acd->codecs()) {
        if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec,
                                          nullptr)) {
          filtered_codecs.push_back(codec);
        }
      }
    }
    // Add other supported audio codecs.
    AudioCodec found_codec;
    for (const AudioCodec& codec : supported_audio_codecs) {
      if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs,
                                        codec, &found_codec) &&
          !FindMatchingCodec<AudioCodec>(supported_audio_codecs,
                                         filtered_codecs, codec, nullptr)) {
        // Use the |found_codec| from |audio_codecs| because it has the
        // correctly mapped payload type.
        filtered_codecs.push_back(found_codec);
      }
    }
  }

  cricket::SecurePolicy sdes_policy =
      IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
                                                         : secure();

  auto audio = std::make_unique<AudioContentDescription>();
  std::vector<std::string> crypto_suites;
  GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options,
                                        &crypto_suites);
  if (!CreateMediaContentOffer(media_description_options, session_options,
                               filtered_codecs, sdes_policy,
                               GetCryptos(current_content), crypto_suites,
                               audio_rtp_extensions, ssrc_generator_,
                               current_streams, audio.get())) {
    return false;
  }

  bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
  SetMediaProtocol(secure_transport, audio.get());

  audio->set_direction(media_description_options.direction);

  desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
                   media_description_options.stopped, std::move(audio));
  if (!AddTransportOffer(media_description_options.mid,
                         media_description_options.transport_options,
                         current_description, desc, ice_credentials)) {
    return false;
  }

  return true;
}

// TODO(kron): This function is very similar to AddAudioContentForOffer.
// Refactor to reuse shared code.
bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const ContentInfo* current_content,
    const SessionDescription* current_description,
    const RtpHeaderExtensions& video_rtp_extensions,
    const VideoCodecs& video_codecs,
    StreamParamsVec* current_streams,
    SessionDescription* desc,
    IceCredentialsIterator* ice_credentials) const {
  // Filter video_codecs (which includes all codecs, with correctly remapped
  // payload types) based on transceiver direction.
  const VideoCodecs& supported_video_codecs =
      GetVideoCodecsForOffer(media_description_options.direction);

  VideoCodecs filtered_codecs;

  if (!media_description_options.codec_preferences.empty()) {
    // Add the codecs from the current transceiver's codec preferences.
    // They override any existing codecs from previous negotiations.
    filtered_codecs = MatchCodecPreference(
        media_description_options.codec_preferences, supported_video_codecs);
  } else {
    // Add the codecs from current content if it exists and is not rejected nor
    // recycled.
    if (current_content && !current_content->rejected &&
        current_content->name == media_description_options.mid) {
      RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO));
      const VideoContentDescription* vcd =
          current_content->media_description()->as_video();
      for (const VideoCodec& codec : vcd->codecs()) {
        if (FindMatchingCodec<VideoCodec>(vcd->codecs(), video_codecs, codec,
                                          nullptr)) {
          filtered_codecs.push_back(codec);
        }
      }
    }
    // Add other supported video codecs.
    VideoCodec found_codec;
    for (const VideoCodec& codec : supported_video_codecs) {
      if (FindMatchingCodec<VideoCodec>(supported_video_codecs, video_codecs,
                                        codec, &found_codec) &&
          !FindMatchingCodec<VideoCodec>(supported_video_codecs,
                                         filtered_codecs, codec, nullptr)) {
        // Use the |found_codec| from |video_codecs| because it has the
        // correctly mapped payload type.
        filtered_codecs.push_back(found_codec);
      }
    }
  }

  if (session_options.raw_packetization_for_video) {
    for (VideoCodec& codec : filtered_codecs) {
      if (codec.GetCodecType() == VideoCodec::CODEC_VIDEO) {
        codec.packetization = kPacketizationParamRaw;
      }
    }
  }

  cricket::SecurePolicy sdes_policy =
      IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
                                                         : secure();
  auto video = std::make_unique<VideoContentDescription>();
  std::vector<std::string> crypto_suites;
  GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options,
                                        &crypto_suites);
  if (!CreateMediaContentOffer(media_description_options, session_options,
                               filtered_codecs, sdes_policy,
                               GetCryptos(current_content), crypto_suites,
                               video_rtp_extensions, ssrc_generator_,
                               current_streams, video.get())) {
    return false;
  }

  video->set_bandwidth(kAutoBandwidth);

  bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
  SetMediaProtocol(secure_transport, video.get());

  video->set_direction(media_description_options.direction);

  desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
                   media_description_options.stopped, std::move(video));
  if (!AddTransportOffer(media_description_options.mid,
                         media_description_options.transport_options,
                         current_description, desc, ice_credentials)) {
    return false;
  }

  return true;
}

bool MediaSessionDescriptionFactory::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 {
  auto data = std::make_unique<SctpDataContentDescription>();

  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;
  // SDES doesn't make sense for SCTP, so we disable it, and we only
  // get SDES crypto suites for RTP-based data channels.
  sdes_policy = cricket::SEC_DISABLED;
  // Unlike SetMediaProtocol below, we need to set the protocol
  // before we call CreateMediaContentOffer.  Otherwise,
  // CreateMediaContentOffer won't know this is SCTP and will
  // generate SSRCs rather than SIDs.
  data->set_protocol(secure_transport ? kMediaProtocolUdpDtlsSctp
                                      : kMediaProtocolSctp);
  data->set_use_sctpmap(session_options.use_obsolete_sctp_sdp);
  data->set_max_message_size(kSctpSendBufferSize);

  if (!CreateContentOffer(media_description_options, session_options,
                          sdes_policy, GetCryptos(current_content),
                          crypto_suites, RtpHeaderExtensions(), ssrc_generator_,
                          current_streams, data.get())) {
    return false;
  }

  desc->AddContent(media_description_options.mid, MediaProtocolType::kSctp,
                   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::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(kDataMaxBandwidth);
  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 ||
       session_options.data_channel_type == DCT_DATA_CHANNEL_TRANSPORT_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);
  }
}

// |audio_codecs| = set of all possible codecs that can be used, with correct
// payload type mappings
//
// |supported_audio_codecs| = set of codecs that are supported for the direction
// of this m= section
//
// acd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from acd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const ContentInfo* offer_content,
    const SessionDescription* offer_description,
    const ContentInfo* current_content,
    const SessionDescription* current_description,
    const TransportInfo* bundle_transport,
    const AudioCodecs& audio_codecs,
    StreamParamsVec* current_streams,
    SessionDescription* answer,
    IceCredentialsIterator* ice_credentials) const {
  RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO));
  const AudioContentDescription* offer_audio_description =
      offer_content->media_description()->as_audio();

  std::unique_ptr<TransportDescription> audio_transport = CreateTransportAnswer(
      media_description_options.mid, offer_description,
      media_description_options.transport_options, current_description,
      bundle_transport != nullptr, ice_credentials);
  if (!audio_transport) {
    return false;
  }

  // Pick codecs based on the requested communications direction in the offer
  // and the selected direction in the answer.
  // Note these will be filtered one final time in CreateMediaContentAnswer.
  auto wants_rtd = media_description_options.direction;
  auto offer_rtd = offer_audio_description->direction();
  auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
  AudioCodecs supported_audio_codecs =
      GetAudioCodecsForAnswer(offer_rtd, answer_rtd);

  AudioCodecs filtered_codecs;

  if (!media_description_options.codec_preferences.empty()) {
    filtered_codecs = MatchCodecPreference(
        media_description_options.codec_preferences, supported_audio_codecs);
  } else {
    // Add the codecs from current content if it exists and is not rejected nor
    // recycled.
    if (current_content && !current_content->rejected &&
        current_content->name == media_description_options.mid) {
      RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO));
      const AudioContentDescription* acd =
          current_content->media_description()->as_audio();
      for (const AudioCodec& codec : acd->codecs()) {
        if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec,
                                          nullptr)) {
          filtered_codecs.push_back(codec);
        }
      }
    }
    // Add other supported audio codecs.
    for (const AudioCodec& codec : supported_audio_codecs) {
      if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs,
                                        codec, nullptr) &&
          !FindMatchingCodec<AudioCodec>(supported_audio_codecs,
                                         filtered_codecs, codec, nullptr)) {
        // We should use the local codec with local parameters and the codec id
        // would be correctly mapped in |NegotiateCodecs|.
        filtered_codecs.push_back(codec);
      }
    }
  }

  bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
                        session_options.bundle_enabled;
  auto audio_answer = std::make_unique<AudioContentDescription>();
  // Do not require or create SDES cryptos if DTLS is used.
  cricket::SecurePolicy sdes_policy =
      audio_transport->secure() ? cricket::SEC_DISABLED : secure();
  if (!SetCodecsInAnswer(offer_audio_description, filtered_codecs,
                         media_description_options, session_options,
                         ssrc_generator_, current_streams,
                         audio_answer.get())) {
    return false;
  }
  if (!CreateMediaContentAnswer(
          offer_audio_description, media_description_options, session_options,
          sdes_policy, GetCryptos(current_content),
          audio_rtp_header_extensions(), ssrc_generator_,
          enable_encrypted_rtp_header_extensions_, current_streams,
          bundle_enabled, audio_answer.get())) {
    return false;  // Fails the session setup.
  }

  bool secure = bundle_transport ? bundle_transport->description.secure()
                                 : audio_transport->secure();
  bool rejected = media_description_options.stopped ||
                  offer_content->rejected ||
                  !IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
                                            audio_answer->protocol(), secure);
  if (!AddTransportAnswer(media_description_options.mid,
                          *(audio_transport.get()), answer)) {
    return false;
  }

  if (rejected) {
    RTC_LOG(LS_INFO) << "Audio m= section '" << media_description_options.mid
                     << "' being rejected in answer.";
  }

  answer->AddContent(media_description_options.mid, offer_content->type,
                     rejected, std::move(audio_answer));
  return true;
}

// TODO(kron): This function is very similar to AddAudioContentForAnswer.
// Refactor to reuse shared code.
bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const ContentInfo* offer_content,
    const SessionDescription* offer_description,
    const ContentInfo* current_content,
    const SessionDescription* current_description,
    const TransportInfo* bundle_transport,
    const VideoCodecs& video_codecs,
    StreamParamsVec* current_streams,
    SessionDescription* answer,
    IceCredentialsIterator* ice_credentials) const {
  RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_VIDEO));
  const VideoContentDescription* offer_video_description =
      offer_content->media_description()->as_video();

  std::unique_ptr<TransportDescription> video_transport = CreateTransportAnswer(
      media_description_options.mid, offer_description,
      media_description_options.transport_options, current_description,
      bundle_transport != nullptr, ice_credentials);
  if (!video_transport) {
    return false;
  }

  // Pick codecs based on the requested communications direction in the offer
  // and the selected direction in the answer.
  // Note these will be filtered one final time in CreateMediaContentAnswer.
  auto wants_rtd = media_description_options.direction;
  auto offer_rtd = offer_video_description->direction();
  auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
  VideoCodecs supported_video_codecs =
      GetVideoCodecsForAnswer(offer_rtd, answer_rtd);

  VideoCodecs filtered_codecs;

  if (!media_description_options.codec_preferences.empty()) {
    filtered_codecs = MatchCodecPreference(
        media_description_options.codec_preferences, supported_video_codecs);
  } else {
    // Add the codecs from current content if it exists and is not rejected nor
    // recycled.
    if (current_content && !current_content->rejected &&
        current_content->name == media_description_options.mid) {
      RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO));
      const VideoContentDescription* vcd =
          current_content->media_description()->as_video();
      for (const VideoCodec& codec : vcd->codecs()) {
        if (FindMatchingCodec<VideoCodec>(vcd->codecs(), video_codecs, codec,
                                          nullptr)) {
          filtered_codecs.push_back(codec);
        }
      }
    }
    // Add other supported video codecs.
    for (const VideoCodec& codec : supported_video_codecs) {
      if (FindMatchingCodec<VideoCodec>(supported_video_codecs, video_codecs,
                                        codec, nullptr) &&
          !FindMatchingCodec<VideoCodec>(supported_video_codecs,
                                         filtered_codecs, codec, nullptr)) {
        // We should use the local codec with local parameters and the codec id
        // would be correctly mapped in |NegotiateCodecs|.
        filtered_codecs.push_back(codec);
      }
    }
  }

  if (session_options.raw_packetization_for_video) {
    for (VideoCodec& codec : filtered_codecs) {
      if (codec.GetCodecType() == VideoCodec::CODEC_VIDEO) {
        codec.packetization = kPacketizationParamRaw;
      }
    }
  }

  bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
                        session_options.bundle_enabled;
  auto video_answer = std::make_unique<VideoContentDescription>();
  // Do not require or create SDES cryptos if DTLS is used.
  cricket::SecurePolicy sdes_policy =
      video_transport->secure() ? cricket::SEC_DISABLED : secure();
  if (!SetCodecsInAnswer(offer_video_description, filtered_codecs,
                         media_description_options, session_options,
                         ssrc_generator_, current_streams,
                         video_answer.get())) {
    return false;
  }
  if (!CreateMediaContentAnswer(
          offer_video_description, media_description_options, session_options,
          sdes_policy, GetCryptos(current_content),
          video_rtp_header_extensions(), ssrc_generator_,
          enable_encrypted_rtp_header_extensions_, current_streams,
          bundle_enabled, video_answer.get())) {
    return false;  // Failed the sessin setup.
  }
  bool secure = bundle_transport ? bundle_transport->description.secure()
                                 : video_transport->secure();
  bool rejected = media_description_options.stopped ||
                  offer_content->rejected ||
                  !IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
                                            video_answer->protocol(), secure);
  if (!AddTransportAnswer(media_description_options.mid,
                          *(video_transport.get()), answer)) {
    return false;
  }

  if (!rejected) {
    video_answer->set_bandwidth(kAutoBandwidth);
  } else {
    RTC_LOG(LS_INFO) << "Video m= section '" << media_description_options.mid
                     << "' being rejected in answer.";
  }
  answer->AddContent(media_description_options.mid, offer_content->type,
                     rejected, std::move(video_answer));
  return true;
}

bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const ContentInfo* offer_content,
    const SessionDescription* offer_description,
    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 {
  std::unique_ptr<TransportDescription> data_transport = CreateTransportAnswer(
      media_description_options.mid, offer_description,
      media_description_options.transport_options, current_description,
      bundle_transport != nullptr, ice_credentials);
  if (!data_transport) {
    return false;
  }

  // Do not require or create SDES cryptos if DTLS is used.
  cricket::SecurePolicy sdes_policy =
      data_transport->secure() ? cricket::SEC_DISABLED : secure();
  bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
                        session_options.bundle_enabled;
  RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_DATA));
  std::unique_ptr<MediaContentDescription> data_answer;
  if (offer_content->media_description()->as_sctp()) {
    // SCTP data content
    data_answer = std::make_unique<SctpDataContentDescription>();
    const SctpDataContentDescription* offer_data_description =
        offer_content->media_description()->as_sctp();
    // Respond with the offerer's proto, whatever it is.
    data_answer->as_sctp()->set_protocol(offer_data_description->protocol());
    // Respond with our max message size or the remote max messsage size,
    // whichever is smaller.
    // 0 is treated specially - it means "I can accept any size". Since
    // we do not implement infinite size messages, reply with
    // kSctpSendBufferSize.
    if (offer_data_description->max_message_size() == 0) {
      data_answer->as_sctp()->set_max_message_size(kSctpSendBufferSize);
    } else {
      data_answer->as_sctp()->set_max_message_size(std::min(
          offer_data_description->max_message_size(), kSctpSendBufferSize));
    }
    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.
    }
    // Respond with sctpmap if the offer uses sctpmap.
    bool offer_uses_sctpmap = offer_data_description->use_sctpmap();
    data_answer->as_sctp()->set_use_sctpmap(offer_uses_sctpmap);
  } else {
    // RTP offer
    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()
                                 : data_transport->secure();

  bool rejected = session_options.data_channel_type == DCT_NONE ||
                  media_description_options.stopped ||
                  offer_content->rejected ||
                  !IsMediaProtocolSupported(MEDIA_TYPE_DATA,
                                            data_answer->protocol(), secure);
  if (!AddTransportAnswer(media_description_options.mid,
                          *(data_transport.get()), answer)) {
    return false;
  }

  if (!rejected) {
    data_answer->set_bandwidth(kDataMaxBandwidth);
  } 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;
}

void MediaSessionDescriptionFactory::ComputeAudioCodecsIntersectionAndUnion() {
  audio_sendrecv_codecs_.clear();
  all_audio_codecs_.clear();
  // Compute the audio codecs union.
  for (const AudioCodec& send : audio_send_codecs_) {
    all_audio_codecs_.push_back(send);
    if (!FindMatchingCodec<AudioCodec>(audio_send_codecs_, audio_recv_codecs_,
                                       send, nullptr)) {
      // It doesn't make sense to have an RTX codec we support sending but not
      // receiving.
      RTC_DCHECK(!IsRtxCodec(send));
    }
  }
  for (const AudioCodec& recv : audio_recv_codecs_) {
    if (!FindMatchingCodec<AudioCodec>(audio_recv_codecs_, audio_send_codecs_,
                                       recv, nullptr)) {
      all_audio_codecs_.push_back(recv);
    }
  }
  // Use NegotiateCodecs to merge our codec lists, since the operation is
  // essentially the same. Put send_codecs as the offered_codecs, which is the
  // order we'd like to follow. The reasoning is that encoding is usually more
  // expensive than decoding, and prioritizing a codec in the send list probably
  // means it's a codec we can handle efficiently.
  NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_,
                  &audio_sendrecv_codecs_, true);
}

void MediaSessionDescriptionFactory::ComputeVideoCodecsIntersectionAndUnion() {
  video_sendrecv_codecs_.clear();
  all_video_codecs_.clear();
  // Compute the video codecs union.
  for (const VideoCodec& send : video_send_codecs_) {
    all_video_codecs_.push_back(send);
    if (!FindMatchingCodec<VideoCodec>(video_send_codecs_, video_recv_codecs_,
                                       send, nullptr)) {
      // TODO(kron): This check is violated by the unit test:
      // MediaSessionDescriptionFactoryTest.RtxWithoutApt
      // Remove either the test or the check.

      // It doesn't make sense to have an RTX codec we support sending but not
      // receiving.
      // RTC_DCHECK(!IsRtxCodec(send));
    }
  }
  for (const VideoCodec& recv : video_recv_codecs_) {
    if (!FindMatchingCodec<VideoCodec>(video_recv_codecs_, video_send_codecs_,
                                       recv, nullptr)) {
      all_video_codecs_.push_back(recv);
    }
  }
  // Use NegotiateCodecs to merge our codec lists, since the operation is
  // essentially the same. Put send_codecs as the offered_codecs, which is the
  // order we'd like to follow. The reasoning is that encoding is usually more
  // expensive than decoding, and prioritizing a codec in the send list probably
  // means it's a codec we can handle efficiently.
  NegotiateCodecs(video_recv_codecs_, video_send_codecs_,
                  &video_sendrecv_codecs_, true);
}

bool IsMediaContent(const ContentInfo* content) {
  return (content && (content->type == MediaProtocolType::kRtp ||
                      content->type == MediaProtocolType::kSctp));
}

bool IsAudioContent(const ContentInfo* content) {
  return IsMediaContentOfType(content, MEDIA_TYPE_AUDIO);
}

bool IsVideoContent(const ContentInfo* content) {
  return IsMediaContentOfType(content, MEDIA_TYPE_VIDEO);
}

bool IsDataContent(const ContentInfo* content) {
  return IsMediaContentOfType(content, MEDIA_TYPE_DATA);
}

const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
                                        MediaType media_type) {
  for (const ContentInfo& content : contents) {
    if (IsMediaContentOfType(&content, media_type)) {
      return &content;
    }
  }
  return nullptr;
}

const ContentInfo* GetFirstAudioContent(const ContentInfos& contents) {
  return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
}

const ContentInfo* GetFirstVideoContent(const ContentInfos& contents) {
  return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
}

const ContentInfo* GetFirstDataContent(const ContentInfos& contents) {
  return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
}

const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
                                        MediaType media_type) {
  if (sdesc == nullptr) {
    return nullptr;
  }

  return GetFirstMediaContent(sdesc->contents(), media_type);
}

const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) {
  return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
}

const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) {
  return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
}

const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) {
  return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
}

const MediaContentDescription* GetFirstMediaContentDescription(
    const SessionDescription* sdesc,
    MediaType media_type) {
  const ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
  return (content ? content->media_description() : nullptr);
}

const AudioContentDescription* GetFirstAudioContentDescription(
    const SessionDescription* sdesc) {
  auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO);
  return desc ? desc->as_audio() : nullptr;
}

const VideoContentDescription* GetFirstVideoContentDescription(
    const SessionDescription* sdesc) {
  auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO);
  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);
  return desc ? desc->as_sctp() : nullptr;
}

//
// Non-const versions of the above functions.
//

ContentInfo* GetFirstMediaContent(ContentInfos* contents,
                                  MediaType media_type) {
  for (ContentInfo& content : *contents) {
    if (IsMediaContentOfType(&content, media_type)) {
      return &content;
    }
  }
  return nullptr;
}

ContentInfo* GetFirstAudioContent(ContentInfos* contents) {
  return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
}

ContentInfo* GetFirstVideoContent(ContentInfos* contents) {
  return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
}

ContentInfo* GetFirstDataContent(ContentInfos* contents) {
  return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
}

ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
                                  MediaType media_type) {
  if (sdesc == nullptr) {
    return nullptr;
  }

  return GetFirstMediaContent(&sdesc->contents(), media_type);
}

ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) {
  return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
}

ContentInfo* GetFirstVideoContent(SessionDescription* sdesc) {
  return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
}

ContentInfo* GetFirstDataContent(SessionDescription* sdesc) {
  return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
}

MediaContentDescription* GetFirstMediaContentDescription(
    SessionDescription* sdesc,
    MediaType media_type) {
  ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
  return (content ? content->media_description() : nullptr);
}

AudioContentDescription* GetFirstAudioContentDescription(
    SessionDescription* sdesc) {
  auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO);
  return desc ? desc->as_audio() : nullptr;
}

VideoContentDescription* GetFirstVideoContentDescription(
    SessionDescription* sdesc) {
  auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO);
  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);
  return desc ? desc->as_sctp() : nullptr;
}

}  // namespace cricket