Adding support for Unified Plan offer/answer negotiation.

webrtc/pc/mediasession.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.
  */
 
@@ skipping 170 matching lines @@
   for (std::vector<std::string>::const_iterator it = crypto_suites.begin();
        it != crypto_suites.end(); ++it) {
     if (!AddCryptoParams(*it, &cryptos)) {
       return false;
     }
   }
   AddMediaCryptos(cryptos, media);
   return true;
 }
 
-const CryptoParamsVec* GetCryptos(const MediaContentDescription* media) {
-  if (!media) {
-    return NULL;
+const CryptoParamsVec* GetCryptos(const ContentInfo* content) {
+  if (!content) {
+    return nullptr;
   }
-  return &media->cryptos();
+  return &(static_cast<const MediaContentDescription*>(content->description)
+               ->cryptos());
 }
 
 bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
                         const CryptoParams& crypto,
                         CryptoParams* out) {
   for (CryptoParamsVec::const_iterator it = cryptos.begin();
        it != cryptos.end(); ++it) {
     if (crypto.Matches(*it)) {
       *out = *it;
       return true;
@@ skipping 215 matching lines @@
 // audio and video extensions.
 class UsedRtpHeaderExtensionIds : public UsedIds<webrtc::RtpExtension> {
  public:
   UsedRtpHeaderExtensionIds()
       : UsedIds<webrtc::RtpExtension>(webrtc::RtpExtension::kMinId,
                                       webrtc::RtpExtension::kMaxId) {}
 
  private:
 };
 
-// Adds a StreamParams for each Stream in Streams with media type
-// media_type to content_description.
+// 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(MediaType media_type,
-                            const MediaSessionOptions& options,
-                            StreamParamsVec* current_streams,
-                            MediaContentDescriptionImpl<C>* content_description,
-                            const bool add_legacy_stream) {
+static bool AddStreamParams(
+    const std::vector<SenderOptions>& sender_options,
+    const std::string& rtcp_cname,
+    StreamParamsVec* current_streams,
+    MediaContentDescriptionImpl<C>* content_description) {
   // SCTP streams are not negotiated using SDP/ContentDescriptions.
   if (IsSctp(content_description->protocol())) {
     return true;
   }
 
   const bool include_rtx_streams =
       ContainsRtxCodec(content_description->codecs());
 
-  const MediaSessionOptions::Streams& streams = options.streams;
-  if (streams.empty() && add_legacy_stream) {

[Taylor Brandstetter, 2017/07/28 01:19:38]

Somewhere in the CL description, can you note that the "add_legacy_stream"
option is going away? Just in case anyone is relying on it and we didn't
realize.

[Zhi Huang, 2017/08/02 04:38:34]

Done.

-    // TODO(perkj): Remove this legacy stream when all apps use StreamParams.
-    std::vector<uint32_t> ssrcs;
-    int num_ssrcs = include_rtx_streams ? 2 : 1;
-    GenerateSsrcs(*current_streams, num_ssrcs, &ssrcs);
-    if (include_rtx_streams) {
-      content_description->AddLegacyStream(ssrcs[0], ssrcs[1]);
-      content_description->set_multistream(true);
-    } else {
-      content_description->AddLegacyStream(ssrcs[0]);
-    }
-    return true;
-  }
 
   const bool include_flexfec_stream =
       ContainsFlexfecCodec(content_description->codecs());
 
-  MediaSessionOptions::Streams::const_iterator stream_it;
-  for (stream_it = streams.begin();
-       stream_it != streams.end(); ++stream_it) {
-    if (stream_it->type != media_type)
-      continue;  // Wrong media type.
-
-    StreamParams* param = GetStreamByIds(*current_streams, "", stream_it->id);
+  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 stream.
+      // This is a new sender.
       std::vector<uint32_t> ssrcs;
-      GenerateSsrcs(*current_streams, stream_it->num_sim_layers, &ssrcs);
+      GenerateSsrcs(*current_streams, sender.num_sim_layers, &ssrcs);
       StreamParams stream_param;
-      stream_param.id = stream_it->id;
+      stream_param.id = sender.track_id;
       // Add the generated ssrc.
       for (size_t i = 0; i < ssrcs.size(); ++i) {
         stream_param.ssrcs.push_back(ssrcs[i]);
       }
-      if (stream_it->num_sim_layers > 1) {
+      if (sender.num_sim_layers > 1) {
         SsrcGroup group(kSimSsrcGroupSemantics, stream_param.ssrcs);
         stream_param.ssrc_groups.push_back(group);
       }
       // Generate extra ssrcs for include_rtx_streams case.
       if (include_rtx_streams) {
         // Generate an RTX ssrc for every ssrc in the group.
         std::vector<uint32_t> rtx_ssrcs;
         GenerateSsrcs(*current_streams, static_cast<int>(ssrcs.size()),
                       &rtx_ssrcs);
         for (size_t i = 0; i < ssrcs.size(); ++i) {
           stream_param.AddFidSsrc(ssrcs[i], rtx_ssrcs[i]);
         }
         content_description->set_multistream(true);
       }
       // Generate extra ssrc for include_flexfec_stream case.
       if (include_flexfec_stream) {
         // TODO(brandtr): Update when we support multistream protection.
         if (ssrcs.size() == 1) {
           std::vector<uint32_t> flexfec_ssrcs;
           GenerateSsrcs(*current_streams, 1, &flexfec_ssrcs);
           stream_param.AddFecFrSsrc(ssrcs[0], flexfec_ssrcs[0]);
           content_description->set_multistream(true);
         } else if (!ssrcs.empty()) {
           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.";
         }
       }
-      stream_param.cname = options.rtcp_cname;
-      stream_param.sync_label = stream_it->sync_label;
+      stream_param.cname = rtcp_cname;
+      stream_param.sync_label = sender.stream_id;
       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->sync_label = stream_it->sync_label;
+      param->sync_label = sender.stream_id;
       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|.
@@ skipping 191 matching lines @@
     }
   }
   return false;
 }
 
 template <class C>
 static bool IsFlexfecCodec(const C& codec) {
   return STR_CASE_CMP(codec.name.c_str(), kFlexfecCodecName) == 0;
 }
 
-static TransportOptions GetTransportOptions(const MediaSessionOptions& options,
-                                            const std::string& content_name) {
-  TransportOptions transport_options;
-  auto it = options.transport_options.find(content_name);
-  if (it != options.transport_options.end()) {
-    transport_options = it->second;
-  }
-  transport_options.enable_ice_renomination = options.enable_ice_renomination;
-  return transport_options;
-}
-
-// Create a media content to be offered in a session-initiate,
-// according to the given options.rtcp_mux, options.is_muc,
-// options.streams, codecs, secure_transport, crypto, and 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).  If
-// add_legacy_stream is true, and current_streams is empty, a legacy
-// stream is created.  The created content is added to the offer.
+// Create a media content to be offered in a session-initiate 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.
 template <class C>
 static bool CreateMediaContentOffer(
-    const MediaSessionOptions& options,
+    const std::vector<SenderOptions>& sender_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,
-    bool add_legacy_stream,
     StreamParamsVec* current_streams,
     MediaContentDescriptionImpl<C>* offer) {
   offer->AddCodecs(codecs);
 
-  offer->set_rtcp_mux(options.rtcp_mux_enabled);
+  offer->set_rtcp_mux(session_options.rtcp_mux_enabled);
   if (offer->type() == cricket::MEDIA_TYPE_VIDEO) {
     offer->set_rtcp_reduced_size(true);
   }
-  offer->set_multistream(options.is_muc);
+  offer->set_multistream(session_options.is_muc);
   offer->set_rtp_header_extensions(rtp_extensions);
 
-  if (!AddStreamParams(offer->type(), options, current_streams, offer,
-                       add_legacy_stream)) {
+  if (!AddStreamParams(sender_options, session_options.rtcp_cname,
+                       current_streams, offer)) {
     return false;
   }
 
   if (secure_policy != SEC_DISABLED) {
     if (current_cryptos) {
       AddMediaCryptos(*current_cryptos, offer);
     }
     if (offer->cryptos().empty()) {
       if (!CreateMediaCryptos(crypto_suites, offer)) {
         return false;
@@ skipping 87 matching lines @@
       }
       if (found_codec) {
         *found_codec = potential_match;
       }
       return true;
     }
   }
   return false;
 }
 
-// Adds all codecs from |reference_codecs| to |offered_codecs| that dont'
+// 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)) {
+    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)) {
+    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) {
+    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 FindCodecsToOffer(
-    const std::vector<C>& reference_codecs,
-    std::vector<C>* offered_codecs,
-    UsedPayloadTypes* used_pltypes) {
-
+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;
-
-      std::string associated_pt_str;
-      if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
-                              &associated_pt_str)) {
-        LOG(LS_WARNING) << "RTX codec " << rtx_codec.name
-                        << " is missing an associated payload type.";
+      const C* associated_codec =
+          GetAssociatedCodec(reference_codecs, rtx_codec);
+      if (!associated_codec) {
         continue;
       }
-
-      int associated_pt;
-      if (!rtc::FromString(associated_pt_str, &associated_pt)) {
-        LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str
-                        << " of RTX codec " << rtx_codec.name
-                        << " to an integer.";
-        continue;
-      }
-
-      // Find the associated reference codec for the reference RTX codec.
-      const C* associated_codec =
-          FindCodecById(reference_codecs, associated_pt);
-      if (!associated_codec) {
-        LOG(LS_WARNING) << "Couldn't find associated codec with payload type "
-                        << associated_pt << " for RTX codec " << rtx_codec.name
-                        << ".";
-        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)) {
         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);
     }
   }
 }
 
+static bool FindByUriAndEncryption(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);

[Taylor Brandstetter, 2017/07/28 01:19:39]

This doesn't seem like it would work... If "extensions" contains both the
encrypted and non-encrypted versions, FindHeaderExtensionByUri would just return
the first one found.

[Zhi Huang, 2017/08/02 04:38:34]

Agree. I'll update this.

+  if (!found || found->encrypt != ext_to_match.encrypt) {
+    return false;
+  }
+  if (found_extension) {
+    *found_extension = *found;
+  }
+  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;
   }
@@ skipping 23 matching lines @@
   }
   if (unencrypted_extension) {
     if (found_extension) {
       *found_extension = *unencrypted_extension;
     }
     return true;
   }
   return false;
 }
 
-// Iterates through |offered_extensions|, adding each one to
-// |regular_extensions| (or |encrypted_extensions| if encrypted) and |used_ids|,
-// and resolving ID conflicts.
-// If an offered extension has the same URI as one in |regular_extensions| or
-// |encrypted_extensions|, it will re-use the same ID and won't be treated as
-// a conflict.
-static void FindAndSetRtpHdrExtUsed(RtpHeaderExtensions* offered_extensions,
-                                    RtpHeaderExtensions* regular_extensions,
-                                    RtpHeaderExtensions* encrypted_extensions,
-                                    UsedRtpHeaderExtensionIds* used_ids) {
-  for (auto& extension : *offered_extensions) {
-    webrtc::RtpExtension existing;
-    if ((extension.encrypt &&
-        FindByUri(*encrypted_extensions, extension, &existing)) ||
-       (!extension.encrypt &&
-        FindByUri(*regular_extensions, extension, &existing))) {
-      extension.id = existing.id;
-    } else {
-      used_ids->FindAndSetIdUsed(&extension);
-      if (extension.encrypt) {
-        encrypted_extensions->push_back(extension);
+// 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 used.
+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,
+                                NULL)) {

[Taylor Brandstetter, 2017/07/28 01:19:38]

nullptr?

[Zhi Huang, 2017/08/02 04:38:34]

Done.

+      webrtc::RtpExtension existing;
+      if (reference_extension.encrypt) {
+        if (FindByUriAndEncryption(*encrypted_extensions, reference_extension,

[Taylor Brandstetter, 2017/07/28 01:19:39]

How could an extension be in encrypted_extensions but not offered_extensions? I
probably am overlooking something (sorry).

[Zhi Huang, 2017/08/02 04:38:34]

In current use case, both the regular and encrypted extensions are part of
offered extensions and this branch will never be executed.

It might still make sense when we use this method to merge the
|reference_extensions| to new |offered_extensions| with previously extracted
|regular_extensions| and |encrypted_extensions|.

+                                   &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 {
-        regular_extensions->push_back(extension);
+        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);
+        }
       }
     }
   }
-}
-
-// Adds |reference_extensions| to |offered_extensions|, while updating
-// |all_extensions| and |used_ids|.
-static void FindRtpHdrExtsToOffer(
-    const RtpHeaderExtensions& reference_extensions,
-    RtpHeaderExtensions* offered_extensions,
-    RtpHeaderExtensions* all_extensions,
-    UsedRtpHeaderExtensionIds* used_ids) {
-  for (auto reference_extension : reference_extensions) {
-    if (!FindByUri(*offered_extensions, reference_extension, NULL)) {
-      webrtc::RtpExtension existing;
-      if (FindByUri(*all_extensions, reference_extension, &existing)) {
-        offered_extensions->push_back(existing);
-      } else {
-        used_ids->FindAndSetIdUsed(&reference_extension);
-        all_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
@@ skipping 40 matching lines @@
   AudioCodecs::iterator iter = audio_codecs->begin();
   while (iter != audio_codecs->end()) {
     if (STR_CASE_CMP(iter->name.c_str(), kComfortNoiseCodecName) == 0) {
       iter = audio_codecs->erase(iter);
     } else {
       ++iter;
     }
   }
 }
 
-// Create a media content to be answered in a session-accept,
-// according to the given options.rtcp_mux, options.streams, codecs,
-// crypto, and 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).  If add_legacy_stream is
-// true, and current_streams is empty, a legacy stream is created.
-// The codecs, rtcp_mux, and crypto are all negotiated with the offer
-// from the incoming session-initiate.  If the negotiation fails, this
-// method returns false.  The created content is added to the offer.
+// Create a media content to be answered in a session-accept 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 from the incoming
+// session-initiate.  If the negotiation fails, this method returns false.  The
+// created content is added to the offer.
 template <class C>
 static bool CreateMediaContentAnswer(
     const MediaContentDescriptionImpl<C>* offer,
-    const MediaSessionOptions& options,
+    const std::vector<SenderOptions>& sender_options,
+    // TODO: combine these two params into mediadescriptionoptions

[Taylor Brandstetter, 2017/07/28 01:19:38]

Can this be done, since there's now MediaDescriptionOptions? I don't think there
was when I wrote this TODO to myself.

[Zhi Huang, 2017/08/02 04:38:34]

Done. I should be able be fix this before.

+    RtpTransceiverDirection desired_direction,
+    const MediaSessionOptions& session_options,
     const std::vector<C>& local_codecs,
     const SecurePolicy& sdes_policy,
     const CryptoParamsVec* current_cryptos,
     const RtpHeaderExtensions& local_rtp_extenstions,
     bool enable_encrypted_rtp_header_extensions,
     StreamParamsVec* current_streams,
-    bool add_legacy_stream,
     bool bundle_enabled,
     MediaContentDescriptionImpl<C>* answer) {
   std::vector<C> negotiated_codecs;
   NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs);
   answer->AddCodecs(negotiated_codecs);
   answer->set_protocol(offer->protocol());
   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(options.rtcp_mux_enabled && offer->rtcp_mux());
+  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());
   }
 
   if (sdes_policy != SEC_DISABLED) {
     CryptoParams crypto;
-    if (SelectCrypto(offer, bundle_enabled, options.crypto_options, &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;
   }
 
-  if (!AddStreamParams(answer->type(), options, current_streams, answer,
-                       add_legacy_stream)) {
+  if (!AddStreamParams(sender_options, session_options.rtcp_cname,
+                       current_streams, answer)) {
     return false;  // Something went seriously wrong.
   }
 
-  // Make sure the answer media content direction is per default set as
-  // described in RFC3264 section 6.1.
-  const bool is_data = !IsRtpProtocol(answer->protocol());
-  const bool has_send_streams = !answer->streams().empty();
-  const bool wants_send = has_send_streams || is_data;
-  const bool recv_audio =
-      answer->type() == cricket::MEDIA_TYPE_AUDIO && options.recv_audio;
-  const bool recv_video =
-      answer->type() == cricket::MEDIA_TYPE_VIDEO && options.recv_video;
-  const bool recv_data =
-      answer->type() == cricket::MEDIA_TYPE_DATA;
-  const bool wants_receive = recv_audio || recv_video || recv_data;
-
   auto offer_rtd =
       RtpTransceiverDirection::FromMediaContentDirection(offer->direction());
-  auto wants_rtd = RtpTransceiverDirection(wants_send, wants_receive);
-  answer->set_direction(NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd)
-                        .ToMediaContentDirection());
+
+  answer->set_direction(
+      NegotiateRtpTransceiverDirection(offer_rtd, desired_direction)
+          .ToMediaContentDirection());
   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;
@@ skipping 41 matching lines @@
     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 std::string& content_name,
-    const SessionDescription* current_description) {
-  if (!current_description)
+static bool IsDtlsActive(const ContentInfo* content,
+                         const SessionDescription* current_description) {
+  if (!content) {
     return false;
-
-  const ContentInfo* content =
-      current_description->GetContentByName(content_name);
-  if (!content)
-    return false;
+  }
 
   const TransportDescription* current_tdesc =
-      GetTransportDescription(content_name, current_description);
-  if (!current_tdesc)
+      GetTransportDescription(content->name, current_description);
+  if (!current_tdesc) {
     return false;
+  }
 
   return current_tdesc->secure();
 }
 
 std::string MediaContentDirectionToString(MediaContentDirection direction) {
   std::string dir_str;
   switch (direction) {
     case MD_INACTIVE:
       dir_str = "inactive";
       break;
     case MD_SENDONLY:
       dir_str = "sendonly";
       break;
     case MD_RECVONLY:
       dir_str = "recvonly";
       break;
     case MD_SENDRECV:
       dir_str = "sendrecv";
       break;
     default:
       RTC_NOTREACHED();
       break;
   }
 
   return dir_str;
 }
 
-void MediaSessionOptions::AddSendStream(MediaType type,
-                                    const std::string& id,
-                                    const std::string& sync_label) {
-  AddSendStreamInternal(type, id, sync_label, 1);
+void MediaDescriptionOptions::AddAudioSender(const std::string& track_id,
+                                             const std::string& stream_id) {
+  RTC_DCHECK(type == MEDIA_TYPE_AUDIO);
+  AddSenderInternal(track_id, stream_id, 1);
 }
 
-void MediaSessionOptions::AddSendVideoStream(
-    const std::string& id,
-    const std::string& sync_label,
-    int num_sim_layers) {
-  AddSendStreamInternal(MEDIA_TYPE_VIDEO, id, sync_label, num_sim_layers);
+void MediaDescriptionOptions::AddVideoSender(const std::string& track_id,
+                                             const std::string& stream_id,
+                                             int num_sim_layers) {
+  RTC_DCHECK(type == MEDIA_TYPE_VIDEO);
+  AddSenderInternal(track_id, stream_id, num_sim_layers);
 }
 
-void MediaSessionOptions::AddSendStreamInternal(
-    MediaType type,
-    const std::string& id,
-    const std::string& sync_label,
-    int num_sim_layers) {
-  streams.push_back(Stream(type, id, sync_label, num_sim_layers));
-
-  // If we haven't already set the data_channel_type, and we add a
-  // stream, we assume it's an RTP data stream.
-  if (type == MEDIA_TYPE_DATA && data_channel_type == DCT_NONE)
-    data_channel_type = DCT_RTP;
+void MediaDescriptionOptions::AddRtpDataChannel(const std::string& track_id,
+                                                const std::string& stream_id) {
+  RTC_DCHECK(type == MEDIA_TYPE_DATA);
+  AddSenderInternal(track_id, stream_id, 1);
 }
 
-void MediaSessionOptions::RemoveSendStream(MediaType type,
-                                       const std::string& id) {
-  Streams::iterator stream_it = streams.begin();
-  for (; stream_it != streams.end(); ++stream_it) {
-    if (stream_it->type == type && stream_it->id == id) {
-      streams.erase(stream_it);
-      return;
-    }
-  }
-  RTC_NOTREACHED();
+void MediaDescriptionOptions::AddSenderInternal(const std::string& track_id,
+                                                const std::string& stream_id,
+                                                int num_sim_layers) {
+  sender_options.push_back(SenderOptions{track_id, stream_id, num_sim_layers});
 }
 
-bool MediaSessionOptions::HasSendMediaStream(MediaType type) const {
-  Streams::const_iterator stream_it = streams.begin();
-  for (; stream_it != streams.end(); ++stream_it) {
-    if (stream_it->type == type) {
-      return true;
-    }
-  }
-  return false;
+bool MediaSessionOptions::HasMediaDescription(MediaType type) const {
+  return std::find_if(media_description_options.begin(),
+                      media_description_options.end(),
+                      [type](const MediaDescriptionOptions& t) {
+                        return t.type == type;
+                      }) != media_description_options.end();
+}
+
+std::vector<MediaDescriptionOptions>::iterator
+MediaSessionOptions::FindMediaDescription(const std::string& mid) {
+  return std::find_if(
+      media_description_options.begin(), media_description_options.end(),
+      [mid](const MediaDescriptionOptions& t) { return t.mid == mid; });
 }
 
 MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
     const TransportDescriptionFactory* transport_desc_factory)
-    : secure_(SEC_DISABLED),
-      add_legacy_(true),
-      transport_desc_factory_(transport_desc_factory) {
-}
+    : transport_desc_factory_(transport_desc_factory) {}
 
 MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
     ChannelManager* channel_manager,
     const TransportDescriptionFactory* transport_desc_factory)
-    : secure_(SEC_DISABLED),
-      add_legacy_(true),
-      transport_desc_factory_(transport_desc_factory) {
+    : transport_desc_factory_(transport_desc_factory) {
   channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
   channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_);
   channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_);
   channel_manager->GetSupportedVideoCodecs(&video_codecs_);
   channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_);
   channel_manager->GetSupportedDataCodecs(&data_codecs_);
-  NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_,
-                  &audio_sendrecv_codecs_);
+  ComputeAudioCodecsIntersectionAndUnion();
 }
 
 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;
   audio_sendrecv_codecs_.clear();
-  // 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(recv_codecs, send_codecs, &audio_sendrecv_codecs_);
+  all_audio_codecs_.clear();

[Taylor Brandstetter, 2017/07/28 01:19:38]

Maybe the "clear" should happen inside ComputeAudioCodecsIntersectionAndUnion?

[Zhi Huang, 2017/08/02 04:38:34]

Done.

+  ComputeAudioCodecsIntersectionAndUnion();
 }
 
 SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
-    const MediaSessionOptions& options,
+    const MediaSessionOptions& session_options,
     const SessionDescription* current_description) const {
   std::unique_ptr<SessionDescription> offer(new SessionDescription());
 
   StreamParamsVec current_streams;
   GetCurrentStreamParams(current_description, &current_streams);
 
-  const bool wants_send =
-      options.HasSendMediaStream(MEDIA_TYPE_AUDIO) || add_legacy_;
-  const AudioCodecs& supported_audio_codecs =
-      GetAudioCodecsForOffer({wants_send, options.recv_audio});
+  AudioCodecs offer_audio_codecs;
+  VideoCodecs offer_video_codecs;
+  DataCodecs offer_data_codecs;
+  GetCodecsForOffer(current_description, &offer_audio_codecs,
+                    &offer_video_codecs, &offer_data_codecs);
 
-  AudioCodecs audio_codecs;
-  VideoCodecs video_codecs;
-  DataCodecs data_codecs;
-  GetCodecsToOffer(current_description, supported_audio_codecs,
-                   video_codecs_, data_codecs_,
-                   &audio_codecs, &video_codecs, &data_codecs);
-
-  if (!options.vad_enabled) {
+  if (!session_options.vad_enabled) {
     // If application doesn't want CN codecs in offer.
-    StripCNCodecs(&audio_codecs);
+    StripCNCodecs(&offer_audio_codecs);
   }
+  FilterDataCodecs(&offer_data_codecs,
+                   session_options.data_channel_type == DCT_SCTP);
 
   RtpHeaderExtensions audio_rtp_extensions;
   RtpHeaderExtensions video_rtp_extensions;
   GetRtpHdrExtsToOffer(current_description, &audio_rtp_extensions,
                        &video_rtp_extensions);
 
-  bool audio_added = false;
-  bool video_added = false;
-  bool data_added = false;
-
-  // Iterate through the contents of |current_description| to maintain the order
-  // of the m-lines in the new offer.
+  // Must have options for each existing section.

[Taylor Brandstetter, 2017/07/28 01:19:38]

This conflicts with the comment in the header file that says:

// The m= section with index X will use media_description_options[X]. If no such
// MediaDescriptionOptions exists, the m= section will just be rejected (port of
// 0).

I think that comment is just out of date.

[Zhi Huang, 2017/08/02 04:38:34]

Done.

   if (current_description) {
-    ContentInfos::const_iterator it = current_description->contents().begin();
-    for (; it != current_description->contents().end(); ++it) {
-      if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) {
-        if (!AddAudioContentForOffer(options, current_description,
-                                     audio_rtp_extensions, audio_codecs,
-                                     &current_streams, offer.get())) {
-          return NULL;
-        }
-        audio_added = true;
-      } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) {
-        if (!AddVideoContentForOffer(options, current_description,
-                                     video_rtp_extensions, video_codecs,
-                                     &current_streams, offer.get())) {
-          return NULL;
-        }
-        video_added = true;
-      } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA)) {
-        MediaSessionOptions options_copy(options);
-        if (IsSctp(static_cast<const MediaContentDescription*>(it->description)
-                       ->protocol())) {
-          options_copy.data_channel_type = DCT_SCTP;
-        }
-        if (!AddDataContentForOffer(options_copy, current_description,
-                                    &data_codecs, &current_streams,
-                                    offer.get())) {
-          return NULL;
-        }
-        data_added = true;
-      } else {
-        RTC_NOTREACHED();
-      }
-    }
+    RTC_DCHECK(current_description->contents().size() <=
+               session_options.media_description_options.size());
   }
 
-  // Append contents that are not in |current_description|.
-  if (!audio_added && options.has_audio() &&
-      !AddAudioContentForOffer(options, current_description,
-                               audio_rtp_extensions, audio_codecs,
-                               &current_streams, offer.get())) {
-    return NULL;
-  }
-  if (!video_added && options.has_video() &&
-      !AddVideoContentForOffer(options, current_description,
-                               video_rtp_extensions, video_codecs,
-                               &current_streams, offer.get())) {
-    return NULL;
-  }
-  if (!data_added && options.has_data() &&
-      !AddDataContentForOffer(options, current_description, &data_codecs,
-                              &current_streams, offer.get())) {
-    return NULL;
+  // Iterate through the media description options, matching with existing media
+  // description in |current_description|.

[Taylor Brandstetter, 2017/07/28 01:19:38]

"descriptions", plural

[Zhi Huang, 2017/08/02 04:38:34]

Done.

+  int msection_index = 0;
+  for (const MediaDescriptionOptions& media_description_options :
+       session_options.media_description_options) {
+    const ContentInfo* current_content = nullptr;
+    if (current_description &&
+        msection_index <
+            static_cast<int>(current_description->contents().size())) {
+      current_content = &current_description->contents()[msection_index];
+      // Media type must match.
+      RTC_DCHECK(IsMediaContentOfType(current_content,
+                                      media_description_options.type));
+      // This restriction doesn't exist before.

[Taylor Brandstetter, 2017/07/28 01:19:38]

Hmmm... We may want to remove this restriction. The MID is allowed to change
when an m= section is recycled. And existing apps may be relying on this being
allowed.

[Taylor Brandstetter, 2017/07/28 01:23:14]

In case you're not familiar with the terminology, when I say "recycled", I mean
when it's rejected in one offer/answer, and its index is used for something else
in a subsequent offer/answer. Look for the word "recycle" in JSEP.

[Zhi Huang, 2017/08/02 04:38:34]

Thanks for clarifying it. 

+      RTC_DCHECK(media_description_options.mid == current_content->name);
+    }
+    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())) {
+          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())) {
+          return nullptr;
+        }
+        break;
+      case MEDIA_TYPE_DATA:
+        if (!AddDataContentForOffer(media_description_options, session_options,
+                                    current_content, current_description,
+                                    offer_data_codecs, &current_streams,
+                                    offer.get())) {
+          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 (options.bundle_enabled) {
+  if (session_options.bundle_enabled && offer->contents().size() > 0u) {
     ContentGroup offer_bundle(GROUP_TYPE_BUNDLE);
-    for (ContentInfos::const_iterator content = offer->contents().begin();
-       content != offer->contents().end(); ++content) {
-      offer_bundle.AddContentName(content->name);
+    for (const ContentInfo& content : offer->contents()) {
+      // 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);
     }
     offer->AddGroup(offer_bundle);
     if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) {
       LOG(LS_ERROR) << "CreateOffer failed to UpdateTransportInfoForBundle.";
-      return NULL;
+      return nullptr;
     }
     if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) {
       LOG(LS_ERROR) << "CreateOffer failed to UpdateCryptoParamsForBundle.";
-      return NULL;
+      return nullptr;
     }
   }
 
   return offer.release();
 }
 
 SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
-    const SessionDescription* offer, const MediaSessionOptions& options,
+    const SessionDescription* offer,
+    const MediaSessionOptions& session_options,
     const SessionDescription* current_description) const {
   if (!offer) {
     return nullptr;
   }
   // The answer contains the intersection of the codecs in the offer with the
   // codecs we support. As indicated by XEP-0167, we retain the same payload ids
   // from the offer in the answer.
   std::unique_ptr<SessionDescription> answer(new SessionDescription());
 
   StreamParamsVec current_streams;
   GetCurrentStreamParams(current_description, &current_streams);
 
   // 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;
 
-  ContentInfos::const_iterator it = offer->contents().begin();
-  for (; it != offer->contents().end(); ++it) {
-    if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) {
-      if (!AddAudioContentForAnswer(offer, options, current_description,
-                                    bundle_transport.get(), &current_streams,
-                                    answer.get())) {
-        return NULL;
-      }
-    } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) {
-      if (!AddVideoContentForAnswer(offer, options, current_description,
-                                    bundle_transport.get(), &current_streams,
-                                    answer.get())) {
-        return NULL;
-      }
-    } else {
-      RTC_DCHECK(IsMediaContentOfType(&*it, MEDIA_TYPE_DATA));
-      if (!AddDataContentForAnswer(offer, options, current_description,
-                                   bundle_transport.get(), &current_streams,
-                                   answer.get())) {
-        return NULL;
-      }
+  // 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;
+  DataCodecs answer_data_codecs;
+  GetCodecsForAnswer(current_description, offer, &answer_audio_codecs,
+                     &answer_video_codecs, &answer_data_codecs);
+
+  if (!session_options.vad_enabled) {
+    // If application doesn't want CN codecs in answer.
+    StripCNCodecs(&answer_audio_codecs);
+  }
+  FilterDataCodecs(&answer_data_codecs,
+                   session_options.data_channel_type == DCT_SCTP);
+
+  // Must have options for exactly as many sections as in the offer.
+  RTC_DCHECK(offer->contents().size() ==
+             session_options.media_description_options.size());
+  // Iterate through the media description options, matching with existing
+  // media descriptions in |current_description|.
+  int 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.

[Taylor Brandstetter, 2017/07/28 01:19:38]

nit: When referring to MIDs in comments, we should use "MID", all-caps. Like
"RTP" or "SDP".

[Zhi Huang, 2017/08/02 04:38:34]

Done.

+    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 <
+            static_cast<int>(current_description->contents().size())) {
+      current_content = &current_description->contents()[msection_index];
+      // Media types and mids must match.

[Taylor Brandstetter, 2017/07/28 01:19:38]

This is the same as the comment above; maybe the comments should say:

// Media types and MIDs must match between the remote offer and the
MediaDescriptionOptions.

And then:

// Media types and MIDs must match between the existing local description, if
one exists, and the MediaDescriptionOptions.

This second DCHECK may also be too strict for the "recycling" case though, so we
may want to remove it.

[Zhi Huang, 2017/08/02 04:38:34]

Done.

+      RTC_DCHECK(IsMediaContentOfType(current_content,
+                                      media_description_options.type));
+      RTC_DCHECK(media_description_options.mid == current_content->name);
     }
+    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())) {
+          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())) {
+          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_data_codecs,
+                                     &current_streams, answer.get())) {
+          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 && options.bundle_enabled && offer_bundle &&
+    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)));
     }
   }
 
   // Only put BUNDLE group in answer if nonempty.
   if (answer_bundle.FirstContentName()) {
     answer->AddGroup(answer_bundle);
@@ skipping 39 matching lines @@
     } else {
       return audio_send_codecs_;
     }
   } else if (answer.send) {
     return audio_send_codecs_;
   } else {
     return audio_recv_codecs_;
   }
 }
 
-void MediaSessionDescriptionFactory::GetCodecsToOffer(
+void MergeCodecsFromDescription(const SessionDescription* description,
+                                AudioCodecs* audio_codecs,
+                                VideoCodecs* video_codecs,
+                                DataCodecs* data_codecs,
+                                UsedPayloadTypes* used_pltypes) {
+  RTC_DCHECK(description);
+  for (const ContentInfo& content : description->contents()) {
+    if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) {
+      const AudioContentDescription* audio =
+          static_cast<AudioContentDescription*>(content.description);
+      MergeCodecs<AudioCodec>(audio->codecs(), audio_codecs, used_pltypes);
+    } else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) {
+      const VideoContentDescription* video =
+          static_cast<VideoContentDescription*>(content.description);
+      MergeCodecs<VideoCodec>(video->codecs(), video_codecs, used_pltypes);
+    } else if (IsMediaContentOfType(&content, MEDIA_TYPE_DATA)) {
+      const DataContentDescription* data =
+          static_cast<DataContentDescription*>(content.description);
+      MergeCodecs<DataCodec>(data->codecs(), 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 SessionDescription* current_description,
-    const AudioCodecs& supported_audio_codecs,
-    const VideoCodecs& supported_video_codecs,
-    const DataCodecs& supported_data_codecs,
     AudioCodecs* audio_codecs,
     VideoCodecs* video_codecs,
     DataCodecs* data_codecs) const {
   UsedPayloadTypes used_pltypes;
   audio_codecs->clear();
   video_codecs->clear();
   data_codecs->clear();
 
+  // 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.
+  if (current_description) {
+    MergeCodecsFromDescription(current_description, audio_codecs, video_codecs,
+                               data_codecs, &used_pltypes);
+  }
+
+  // Add our codecs that are not in |current_description|.
+  MergeCodecs<AudioCodec>(all_audio_codecs_, audio_codecs, &used_pltypes);
+  MergeCodecs<VideoCodec>(video_codecs_, video_codecs, &used_pltypes);
+  MergeCodecs<DataCodec>(data_codecs_, 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 SessionDescription* current_description,
+    const SessionDescription* remote_offer,
+    AudioCodecs* audio_codecs,
+    VideoCodecs* video_codecs,
+    DataCodecs* data_codecs) const {
+  UsedPayloadTypes used_pltypes;
+  audio_codecs->clear();
+  video_codecs->clear();
+  data_codecs->clear();
 
   // First - get all codecs from the current description if the media type
-  // is used.
-  // Add them to |used_pltypes| so the payloadtype is not reused if a new media
-  // type is added.
+  // is used. Add them to |used_pltypes| so the payload type is not reused if a
+  // new media type is added.
   if (current_description) {
-    const AudioContentDescription* audio =
-        GetFirstAudioContentDescription(current_description);
-    if (audio) {
-      *audio_codecs = audio->codecs();
-      used_pltypes.FindAndSetIdUsed<AudioCodec>(audio_codecs);
-    }
-    const VideoContentDescription* video =
-        GetFirstVideoContentDescription(current_description);
-    if (video) {
-      *video_codecs = video->codecs();
-      used_pltypes.FindAndSetIdUsed<VideoCodec>(video_codecs);
-    }
-    const DataContentDescription* data =
-        GetFirstDataContentDescription(current_description);
-    if (data) {
-      *data_codecs = data->codecs();
-      used_pltypes.FindAndSetIdUsed<DataCodec>(data_codecs);
+    MergeCodecsFromDescription(current_description, audio_codecs, video_codecs,
+                               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;
+  DataCodecs filtered_offered_data_codecs;
+  for (const ContentInfo& content : remote_offer->contents()) {
+    if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) {
+      const AudioContentDescription* audio =
+          static_cast<AudioContentDescription*>(content.description);
+      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 =
+          static_cast<VideoContentDescription*>(content.description);
+      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(), video_codecs_,
+                                          offered_video_codec, nullptr)) {
+          filtered_offered_video_codecs.push_back(offered_video_codec);
+        }
+      }
+    } else if (IsMediaContentOfType(&content, MEDIA_TYPE_DATA)) {
+      const DataContentDescription* data =
+          static_cast<DataContentDescription*>(content.description);
+      for (const DataCodec& offered_data_codec : data->codecs()) {
+        if (!FindMatchingCodec<DataCodec>(data->codecs(),
+                                          filtered_offered_data_codecs,
+                                          offered_data_codec, nullptr) &&
+            FindMatchingCodec<DataCodec>(data->codecs(), data_codecs_,
+                                         offered_data_codec, nullptr)) {
+          filtered_offered_data_codecs.push_back(offered_data_codec);
+        }
+      }
     }
   }
 
-  // Add our codecs that are not in |current_description|.
-  FindCodecsToOffer<AudioCodec>(supported_audio_codecs, audio_codecs,
-                                &used_pltypes);
-  FindCodecsToOffer<VideoCodec>(supported_video_codecs, video_codecs,
-                                &used_pltypes);
-  FindCodecsToOffer<DataCodec>(supported_data_codecs, data_codecs,
-                               &used_pltypes);
+  // Add codecs that are not in |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_data_codecs, data_codecs,
+                         &used_pltypes);
 }
 
 void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
     const SessionDescription* current_description,
-    RtpHeaderExtensions* audio_extensions,
-    RtpHeaderExtensions* video_extensions) const {
+    RtpHeaderExtensions* offer_audio_extensions,
+    RtpHeaderExtensions* offer_video_extensions) const {
   // All header extensions allocated from the same range to avoid potential
   // issues when using BUNDLE.
   UsedRtpHeaderExtensionIds used_ids;
   RtpHeaderExtensions all_regular_extensions;
   RtpHeaderExtensions all_encrypted_extensions;
-  audio_extensions->clear();
-  video_extensions->clear();
+  offer_audio_extensions->clear();
+  offer_video_extensions->clear();
 
   // 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.
   if (current_description) {
-    const AudioContentDescription* audio =
-        GetFirstAudioContentDescription(current_description);
-    if (audio) {
-      *audio_extensions = audio->rtp_header_extensions();
-      FindAndSetRtpHdrExtUsed(audio_extensions, &all_regular_extensions,
-          &all_encrypted_extensions, &used_ids);
-    }
-    const VideoContentDescription* video =
-        GetFirstVideoContentDescription(current_description);
-    if (video) {
-      *video_extensions = video->rtp_header_extensions();
-      FindAndSetRtpHdrExtUsed(video_extensions, &all_regular_extensions,
-          &all_encrypted_extensions, &used_ids);
+    for (const ContentInfo& content : current_description->contents()) {
+      if (IsMediaContentOfType(&content, MEDIA_TYPE_DATA)) {

[Taylor Brandstetter, 2017/07/28 01:19:38]

This "if (data) { continue; }" block can actually be removed, since nothing
happens in this loop outside the "if (audio) {} else if (video) {}" blocks.

[Zhi Huang, 2017/08/02 04:38:34]

Done.

+        continue;
+      }
+      if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) {
+        const AudioContentDescription* audio =
+            static_cast<const AudioContentDescription*>(content.description);
+        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 =
+            static_cast<const VideoContentDescription*>(content.description);
+        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
   // |current_description|.
-  FindRtpHdrExtsToOffer(audio_rtp_header_extensions(), audio_extensions,
-                        &all_regular_extensions, &used_ids);
-  FindRtpHdrExtsToOffer(video_rtp_header_extensions(), video_extensions,
-                        &all_regular_extensions, &used_ids);
+  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_description) {
-    AddEncryptedVersionsOfHdrExts(audio_extensions, &all_encrypted_extensions,
-        &used_ids);
-    AddEncryptedVersionsOfHdrExts(video_extensions, &all_encrypted_extensions,
-        &used_ids);
+    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) const {
   if (!transport_desc_factory_)
      return false;
@@ skipping 34 matching lines @@
   if (!answer_desc->AddTransportInfo(TransportInfo(content_name,
                                                    transport_desc))) {
     LOG(LS_ERROR)
         << "Failed to AddTransportAnswer, content name=" << content_name;
     return false;
   }
   return true;
 }
 
 bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
-    const MediaSessionOptions& options,
+    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) const {
-  const ContentInfo* current_audio_content =
-      GetFirstAudioContent(current_description);
-  std::string content_name =
-      current_audio_content ? current_audio_content->name : CN_AUDIO;
+  // Filter audio_codecs (which includes all codecs, with correctly remapped
+  // payload types) based on transceiver direction.
+  AudioCodecs supported_audio_codecs =

[Taylor Brandstetter, 2017/07/28 01:19:38]

This can be a const ref to avoid a copy.

[Zhi Huang, 2017/08/02 04:38:34]

Done.

+      GetAudioCodecsForOffer(media_description_options.direction);
 
+  AudioCodecs filtered_codecs;
+  if (current_description) {
+    for (const AudioCodec& codec : audio_codecs) {
+      if (FindMatchingCodec<AudioCodec>(audio_codecs, supported_audio_codecs,
+                                        codec, nullptr)) {
+        filtered_codecs.push_back(codec);
+      }
+    }
+  } else {
+    for (const AudioCodec& supported_audio_codec : supported_audio_codecs) {
+      if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs,
+                                        supported_audio_codec, nullptr)) {
+        filtered_codecs.push_back(supported_audio_codec);
+      }
+    }
+  }
   cricket::SecurePolicy sdes_policy =
-      IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
-                                                      : secure();
+      IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
+                                                         : secure();
 
   std::unique_ptr<AudioContentDescription> audio(new AudioContentDescription());
   std::vector<std::string> crypto_suites;
-  GetSupportedAudioSdesCryptoSuiteNames(options.crypto_options, &crypto_suites);
+  GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options,
+                                        &crypto_suites);
   if (!CreateMediaContentOffer(
-          options,
-          audio_codecs,
-          sdes_policy,
-          GetCryptos(GetFirstAudioContentDescription(current_description)),
-          crypto_suites,
-          audio_rtp_extensions,
-          add_legacy_,
-          current_streams,
-          audio.get())) {
+          media_description_options.sender_options, session_options,
+          filtered_codecs, sdes_policy, GetCryptos(current_content),
+          crypto_suites, audio_rtp_extensions, current_streams, audio.get())) {
     return false;
   }
   audio->set_lang(lang_);
 
   bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
   SetMediaProtocol(secure_transport, audio.get());
 
-  auto offer_rtd =
-      RtpTransceiverDirection(!audio->streams().empty(), options.recv_audio);
-  audio->set_direction(offer_rtd.ToMediaContentDirection());
+  audio->set_direction(
+      media_description_options.direction.ToMediaContentDirection());
 
-  desc->AddContent(content_name, NS_JINGLE_RTP, audio.release());
-  if (!AddTransportOffer(content_name,
-                         GetTransportOptions(options, content_name),
+  desc->AddContent(media_description_options.mid, NS_JINGLE_RTP,
+                   audio.release());
+  if (!AddTransportOffer(media_description_options.mid,
+                         media_description_options.transport_options,
                          current_description, desc)) {
     return false;
   }
 
   return true;
 }
 
 bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
-    const MediaSessionOptions& options,
+    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) const {
-  const ContentInfo* current_video_content =
-      GetFirstVideoContent(current_description);
-  std::string content_name =
-      current_video_content ? current_video_content->name : CN_VIDEO;
+  // Filter video_codecs (which includes all codecs, with correctly remapped
+  // payload types).
+  VideoCodecs filtered_codecs;
+  for (const VideoCodec& codec : video_codecs) {
+    if (FindMatchingCodec<VideoCodec>(video_codecs, video_codecs_, codec,
+                                      nullptr)) {
+      filtered_codecs.push_back(codec);
+    }
+  }
 
   cricket::SecurePolicy sdes_policy =
-      IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
-                                                      : secure();
+      IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
+                                                         : secure();
 
   std::unique_ptr<VideoContentDescription> video(new VideoContentDescription());
   std::vector<std::string> crypto_suites;
-  GetSupportedVideoSdesCryptoSuiteNames(options.crypto_options, &crypto_suites);
+  GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options,
+                                        &crypto_suites);
   if (!CreateMediaContentOffer(
-          options,
-          video_codecs,
-          sdes_policy,
-          GetCryptos(GetFirstVideoContentDescription(current_description)),
-          crypto_suites,
-          video_rtp_extensions,
-          add_legacy_,
-          current_streams,
-          video.get())) {
+          media_description_options.sender_options, session_options,
+          filtered_codecs, sdes_policy, GetCryptos(current_content),
+          crypto_suites, video_rtp_extensions, current_streams, video.get())) {
     return false;
   }
 
-  video->set_bandwidth(options.video_bandwidth);
+  video->set_bandwidth(kAutoBandwidth);
 
   bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
   SetMediaProtocol(secure_transport, video.get());
 
-  if (!video->streams().empty()) {
-    if (options.recv_video) {
-      video->set_direction(MD_SENDRECV);
-    } else {
-      video->set_direction(MD_SENDONLY);
-    }
-  } else {
-    if (options.recv_video) {
-      video->set_direction(MD_RECVONLY);
-    } else {
-      video->set_direction(MD_INACTIVE);
-    }
-  }
+  video->set_direction(
+      media_description_options.direction.ToMediaContentDirection());
 
-  desc->AddContent(content_name, NS_JINGLE_RTP, video.release());
-  if (!AddTransportOffer(content_name,
-                         GetTransportOptions(options, content_name),
+  desc->AddContent(media_description_options.mid, NS_JINGLE_RTP,
+                   video.release());
+  if (!AddTransportOffer(media_description_options.mid,
+                         media_description_options.transport_options,
                          current_description, desc)) {
     return false;
   }
-
   return true;
 }
 
 bool MediaSessionDescriptionFactory::AddDataContentForOffer(
-    const MediaSessionOptions& options,
+    const MediaDescriptionOptions& media_description_options,
+    const MediaSessionOptions& session_options,
+    const ContentInfo* current_content,
     const SessionDescription* current_description,
-    DataCodecs* data_codecs,
+    const DataCodecs& data_codecs,
     StreamParamsVec* current_streams,
     SessionDescription* desc) const {
   bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
 
   std::unique_ptr<DataContentDescription> data(new DataContentDescription());
-  bool is_sctp = (options.data_channel_type == DCT_SCTP);
-
-  FilterDataCodecs(data_codecs, is_sctp);
-
-  const ContentInfo* current_data_content =
-      GetFirstDataContent(current_description);
-  std::string content_name =
-      current_data_content ? current_data_content->name : CN_DATA;
+  bool is_sctp = (session_options.data_channel_type == DCT_SCTP);
+  // If the DataChannel type is not specified, use the DataChannel type in
+  // the current description.
+  if (session_options.data_channel_type == DCT_NONE && current_content) {
+    is_sctp = (static_cast<const DataContentDescription*>(
+                   current_content->description)
+                   ->protocol() == kMediaProtocolSctp);
+  }
 
   cricket::SecurePolicy sdes_policy =
-      IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
-                                                      : secure();
+      IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
+                                                         : secure();
   std::vector<std::string> crypto_suites;
   if (is_sctp) {
     // 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.
     // TODO(deadbeef): Offer kMediaProtocolUdpDtlsSctp (or TcpDtlsSctp), once
     // it's safe to do so. Older versions of webrtc would reject these
     // protocols; see https://bugs.chromium.org/p/webrtc/issues/detail?id=7706.
     data->set_protocol(
         secure_transport ? kMediaProtocolDtlsSctp : kMediaProtocolSctp);
   } else {
-    GetSupportedDataSdesCryptoSuiteNames(options.crypto_options,
+    GetSupportedDataSdesCryptoSuiteNames(session_options.crypto_options,
                                          &crypto_suites);
   }
 
+  // Even SCTP uses a "codec".
   if (!CreateMediaContentOffer(
-          options,
-          *data_codecs,
-          sdes_policy,
-          GetCryptos(GetFirstDataContentDescription(current_description)),
-          crypto_suites,
-          RtpHeaderExtensions(),
-          add_legacy_,
-          current_streams,
-          data.get())) {
+          media_description_options.sender_options, session_options,
+          data_codecs, sdes_policy, GetCryptos(current_content), crypto_suites,
+          RtpHeaderExtensions(), current_streams, data.get())) {
     return false;
   }
 
   if (is_sctp) {
-    desc->AddContent(content_name, NS_JINGLE_DRAFT_SCTP, data.release());
+    desc->AddContent(media_description_options.mid, NS_JINGLE_DRAFT_SCTP,
+                     data.release());
   } else {
-    data->set_bandwidth(options.data_bandwidth);
+    data->set_bandwidth(kDataMaxBandwidth);
     SetMediaProtocol(secure_transport, data.get());
-    desc->AddContent(content_name, NS_JINGLE_RTP, data.release());
+    desc->AddContent(media_description_options.mid, NS_JINGLE_RTP,
+                     data.release());
   }
-  if (!AddTransportOffer(content_name,
-                         GetTransportOptions(options, content_name),
+  if (!AddTransportOffer(media_description_options.mid,
+                         media_description_options.transport_options,
                          current_description, desc)) {
     return false;
   }
   return true;
 }
 
 bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
-    const SessionDescription* offer,
-    const MediaSessionOptions& options,
+    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) const {
-  const ContentInfo* audio_content = GetFirstAudioContent(offer);
-  const AudioContentDescription* offer_audio =
-      static_cast<const AudioContentDescription*>(audio_content->description);
+  const AudioContentDescription* offer_audio_description =
+      static_cast<const AudioContentDescription*>(offer_content->description);
 
   std::unique_ptr<TransportDescription> audio_transport(
-      CreateTransportAnswer(audio_content->name, offer,
-                            GetTransportOptions(options, audio_content->name),
+      CreateTransportAnswer(media_description_options.mid, offer_description,
+                            media_description_options.transport_options,
                             current_description, bundle_transport != nullptr));
   if (!audio_transport) {
     return false;
   }
 
-  // Pick codecs based on the requested communications direction in the offer.
-  const bool wants_send =
-      options.HasSendMediaStream(MEDIA_TYPE_AUDIO) || add_legacy_;
-  auto wants_rtd = RtpTransceiverDirection(wants_send, options.recv_audio);
-  auto offer_rtd =
-      RtpTransceiverDirection::FromMediaContentDirection(
-          offer_audio->direction());
+  // 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 = RtpTransceiverDirection::FromMediaContentDirection(
+      offer_audio_description->direction());
   auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
-  AudioCodecs audio_codecs = GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
-  if (!options.vad_enabled) {
-    StripCNCodecs(&audio_codecs);
-  }
-
-  bool bundle_enabled =
-      offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
+  AudioCodecs supported_audio_codecs =
+      GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
+
+  AudioCodecs filtered_codecs;
+  if (current_description) {
+    for (const AudioCodec& codec : audio_codecs) {
+      if (FindMatchingCodec<AudioCodec>(audio_codecs, supported_audio_codecs,
+                                        codec, nullptr)) {
+        filtered_codecs.push_back(codec);
+      }
+    }
+  } else {
+    for (const AudioCodec& supported_audio_codec : supported_audio_codecs) {
+      if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs,
+                                        supported_audio_codec, nullptr)) {
+        filtered_codecs.push_back(supported_audio_codec);
+      }
+    }
+  }
+
+  bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
+                        session_options.bundle_enabled;
   std::unique_ptr<AudioContentDescription> audio_answer(
       new AudioContentDescription());
   // Do not require or create SDES cryptos if DTLS is used.
   cricket::SecurePolicy sdes_policy =
       audio_transport->secure() ? cricket::SEC_DISABLED : secure();
   if (!CreateMediaContentAnswer(
-          offer_audio,
-          options,
-          audio_codecs,
-          sdes_policy,
-          GetCryptos(GetFirstAudioContentDescription(current_description)),
-          audio_rtp_extensions_,
-          enable_encrypted_rtp_header_extensions_,
-          current_streams,
-          add_legacy_,
-          bundle_enabled,
-          audio_answer.get())) {
+          offer_audio_description, media_description_options.sender_options,
+          media_description_options.direction, session_options, filtered_codecs,
+          sdes_policy, GetCryptos(current_content), audio_rtp_extensions_,
+          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 = !options.has_audio() || audio_content->rejected ||
+  bool rejected = media_description_options.stopped ||
+                  offer_content->rejected ||
                   !IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
                                             audio_answer->protocol(), secure);
   if (!rejected) {
-    AddTransportAnswer(audio_content->name, *(audio_transport.get()), answer);
-  } else {
-    // RFC 3264
-    // The answer MUST contain the same number of m-lines as the offer.
-    LOG(LS_INFO) << "Audio is not supported in the answer.";
-  }
-
-  answer->AddContent(audio_content->name, audio_content->type, rejected,
-                     audio_answer.release());
+    AddTransportAnswer(media_description_options.mid, *(audio_transport.get()),
+                       answer);
+  } else {
+    LOG(LS_INFO) << "Audio m= section '" << media_description_options.mid
+                 << "' being rejected in answer.";
+  }
+
+  answer->AddContent(media_description_options.mid, offer_content->type,
+                     rejected, audio_answer.release());
   return true;
 }
 
 bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
-    const SessionDescription* offer,
-    const MediaSessionOptions& options,
+    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) const {
-  const ContentInfo* video_content = GetFirstVideoContent(offer);
+  const VideoContentDescription* offer_video_description =
+      static_cast<const VideoContentDescription*>(offer_content->description);
+
   std::unique_ptr<TransportDescription> video_transport(
-      CreateTransportAnswer(video_content->name, offer,
-                            GetTransportOptions(options, video_content->name),
+      CreateTransportAnswer(media_description_options.mid, offer_description,
+                            media_description_options.transport_options,
                             current_description, bundle_transport != nullptr));
   if (!video_transport) {
     return false;
   }
 
+  bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
+                        session_options.bundle_enabled;
+
   std::unique_ptr<VideoContentDescription> video_answer(
       new VideoContentDescription());
   // Do not require or create SDES cryptos if DTLS is used.
   cricket::SecurePolicy sdes_policy =
       video_transport->secure() ? cricket::SEC_DISABLED : secure();
-  bool bundle_enabled =
-      offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
   if (!CreateMediaContentAnswer(
-          static_cast<const VideoContentDescription*>(
-              video_content->description),
-          options,
-          video_codecs_,
-          sdes_policy,
-          GetCryptos(GetFirstVideoContentDescription(current_description)),
-          video_rtp_extensions_,
-          enable_encrypted_rtp_header_extensions_,
-          current_streams,
-          add_legacy_,
-          bundle_enabled,
-          video_answer.get())) {
-    return false;
+          offer_video_description, media_description_options.sender_options,
+          media_description_options.direction, session_options, video_codecs,
+          sdes_policy, GetCryptos(current_content), video_rtp_extensions_,
+          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 = !options.has_video() || video_content->rejected ||
+  bool rejected = media_description_options.stopped ||
+                  offer_content->rejected ||
                   !IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
                                             video_answer->protocol(), secure);
   if (!rejected) {
-    if (!AddTransportAnswer(video_content->name, *(video_transport.get()),
-                            answer)) {
+    if (!AddTransportAnswer(media_description_options.mid,
+                            *(video_transport.get()), answer)) {
       return false;
     }
-    video_answer->set_bandwidth(options.video_bandwidth);
-  } else {
-    // RFC 3264
-    // The answer MUST contain the same number of m-lines as the offer.
-    LOG(LS_INFO) << "Video is not supported in the answer.";
-  }
-  answer->AddContent(video_content->name, video_content->type, rejected,
-                     video_answer.release());
+    video_answer->set_bandwidth(kAutoBandwidth);
+  } else {
+    LOG(LS_INFO) << "Video m= section '" << media_description_options.mid
+                 << "' being rejected in answer.";
+  }
+  answer->AddContent(media_description_options.mid, offer_content->type,
+                     rejected, video_answer.release());
   return true;
 }
 
 bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
-    const SessionDescription* offer,
-    const MediaSessionOptions& options,
+    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 DataCodecs& data_codecs,
     StreamParamsVec* current_streams,
     SessionDescription* answer) const {
-  const ContentInfo* data_content = GetFirstDataContent(offer);
   std::unique_ptr<TransportDescription> data_transport(
-      CreateTransportAnswer(data_content->name, offer,
-                            GetTransportOptions(options, data_content->name),
+      CreateTransportAnswer(media_description_options.mid, offer_description,
+                            media_description_options.transport_options,
                             current_description, bundle_transport != nullptr));
   if (!data_transport) {
     return false;
   }
-  bool is_sctp = (options.data_channel_type == DCT_SCTP);
-  std::vector<DataCodec> data_codecs(data_codecs_);
-  FilterDataCodecs(&data_codecs, is_sctp);
 
   std::unique_ptr<DataContentDescription> data_answer(
       new DataContentDescription());
   // 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->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
+  bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
+                        session_options.bundle_enabled;
   if (!CreateMediaContentAnswer(
           static_cast<const DataContentDescription*>(
-              data_content->description),
-          options,
-          data_codecs_,
-          sdes_policy,
-          GetCryptos(GetFirstDataContentDescription(current_description)),
-          RtpHeaderExtensions(),
-          enable_encrypted_rtp_header_extensions_,
-          current_streams,
-          add_legacy_,
-          bundle_enabled,
-          data_answer.get())) {
+              offer_content->description),
+          media_description_options.sender_options,
+          media_description_options.direction, session_options, data_codecs,
+          sdes_policy, GetCryptos(current_content), RtpHeaderExtensions(),
+          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.
   const DataContentDescription* offer_data_description =
-      static_cast<const DataContentDescription*>(data_content->description);
+      static_cast<const DataContentDescription*>(offer_content->description);
   bool offer_uses_sctpmap = offer_data_description->use_sctpmap();
   data_answer->set_use_sctpmap(offer_uses_sctpmap);
 
   bool secure = bundle_transport ? bundle_transport->description.secure()
                                  : data_transport->secure();
 
-  bool rejected = !options.has_data() || data_content->rejected ||
+  bool rejected = media_description_options.stopped ||
+                  offer_content->rejected ||
                   !IsMediaProtocolSupported(MEDIA_TYPE_DATA,
                                             data_answer->protocol(), secure);
   if (!rejected) {
-    data_answer->set_bandwidth(options.data_bandwidth);
-    if (!AddTransportAnswer(data_content->name, *(data_transport.get()),
-                            answer)) {
+    data_answer->set_bandwidth(kDataMaxBandwidth);
+    if (!AddTransportAnswer(media_description_options.mid,
+                            *(data_transport.get()), answer)) {
       return false;
     }
   } else {
     // RFC 3264
     // The answer MUST contain the same number of m-lines as the offer.
     LOG(LS_INFO) << "Data is not supported in the answer.";
   }
-  answer->AddContent(data_content->name, data_content->type, rejected,
-                     data_answer.release());
+  answer->AddContent(media_description_options.mid, offer_content->type,
+                     rejected, data_answer.release());
   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_);
+}
+
 bool IsMediaContent(const ContentInfo* content) {
   return (content &&
           (content->type == NS_JINGLE_RTP ||
            content->type == NS_JINGLE_DRAFT_SCTP));
 }
 
 bool IsAudioContent(const ContentInfo* content) {
   return IsMediaContentOfType(content, MEDIA_TYPE_AUDIO);
 }
 
@@ skipping 140 matching lines @@
       GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO));
 }
 
 DataContentDescription* GetFirstDataContentDescription(
     SessionDescription* sdesc) {
   return static_cast<DataContentDescription*>(
       GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA));
 }
 
 }  // namespace cricket