Initial asymmetric codec support in MediaSessionDescription

webrtc/pc/mediasession.h

 /*
  *  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 34 matching lines @@
 
 std::string MediaTypeToString(MediaType type);
 
 enum MediaContentDirection {
   MD_INACTIVE,
   MD_SENDONLY,
   MD_RECVONLY,
   MD_SENDRECV
 };
 
+std::string MediaContentDirectionToString(MediaContentDirection direction);
+
 enum CryptoType {
   CT_NONE,
   CT_SDES,
   CT_DTLS
 };
 
 // RTC4585 RTP/AVPF
 extern const char kMediaProtocolAvpf[];
 // RFC5124 RTP/SAVPF
 extern const char kMediaProtocolSavpf[];
@@ skipping 323 matching lines @@
   // Default ctor; use methods below to set configuration.
   // The TransportDescriptionFactory is not owned by MediaSessionDescFactory,
   // so it must be kept alive by the user of this class.
   explicit MediaSessionDescriptionFactory(
       const TransportDescriptionFactory* factory);
   // This helper automatically sets up the factory to get its configuration
   // from the specified ChannelManager.
   MediaSessionDescriptionFactory(ChannelManager* cmanager,
                                  const TransportDescriptionFactory* factory);
 
-  const AudioCodecs& audio_codecs() const { return audio_codecs_; }
-  void set_audio_codecs(const AudioCodecs& codecs) { audio_codecs_ = codecs; }
+  const AudioCodecs& audio_codecs() const;

[pthatcher1, 2016/05/31 21:52:36]

I'd prefer audio_sendrecv_codecs()

[ossu, 2016/06/02 13:28:02]

Sure! I've addressed it in the second CL.

+  const AudioCodecs& audio_send_codecs() const;
+  const AudioCodecs& audio_recv_codecs() const;
+  void set_audio_codecs(const AudioCodecs& send_and_recv_codecs);
+  void set_audio_codecs(const AudioCodecs& send_codecs,
+                        const AudioCodecs& recv_codecs);

[pthatcher1, 2016/05/31 21:52:36]

This should probably both be SetAudioCodecs.

Also, why do we need the first version?  Can we just get rid of it?  It would be
much more clear to always provide send and recv and let this class do the
merging rather than get in a mix of letting the caller do the merge or letting
the class do the merge.

[ossu, 2016/06/02 13:28:02]

I've kept it while we're redoing the rest of the code to provide two lists and
separate factories for encode and decode. However, looking more closely these
setters are only called by unit tests. I'll take a look at changing them to use
the two-parameter version.

   void set_audio_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
     audio_rtp_extensions_ = extensions;
   }
   const RtpHeaderExtensions& audio_rtp_header_extensions() const {
     return audio_rtp_extensions_;
   }
   const VideoCodecs& video_codecs() const { return video_codecs_; }
   void set_video_codecs(const VideoCodecs& codecs) { video_codecs_ = codecs; }
   void set_video_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
     video_rtp_extensions_ = extensions;
@@ skipping 14 matching lines @@
   SessionDescription* CreateOffer(
       const MediaSessionOptions& options,
       const SessionDescription* current_description) const;
   SessionDescription* CreateAnswer(
         const SessionDescription* offer,
         const MediaSessionOptions& options,
         const SessionDescription* current_description) const;
 
  private:
   void GetCodecsToOffer(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;
   void GetRtpHdrExtsToOffer(const SessionDescription* current_description,
                             RtpHeaderExtensions* audio_extensions,
                             RtpHeaderExtensions* video_extensions) const;
   bool AddTransportOffer(
       const std::string& content_name,
       const TransportOptions& transport_options,
       const SessionDescription* current_desc,
@@ skipping 51 matching lines @@
       StreamParamsVec* current_streams,
       SessionDescription* answer) const;
 
   bool AddDataContentForAnswer(
       const SessionDescription* offer,
       const MediaSessionOptions& options,
       const SessionDescription* current_description,
       StreamParamsVec* current_streams,
       SessionDescription* answer) const;
 
-  AudioCodecs audio_codecs_;
+  AudioCodecs audio_send_codecs_;
+  AudioCodecs audio_recv_codecs_;
+  AudioCodecs audio_sendrecv_codecs_;
   RtpHeaderExtensions audio_rtp_extensions_;
   VideoCodecs video_codecs_;
   RtpHeaderExtensions video_rtp_extensions_;
   DataCodecs data_codecs_;
   SecurePolicy secure_;
   bool add_legacy_;
   std::string lang_;
   const TransportDescriptionFactory* transport_desc_factory_;
 };
 
@@ skipping 26 matching lines @@
 void GetSupportedVideoCryptoSuiteNames(
     std::vector<std::string>* crypto_suite_names);
 void GetSupportedDataCryptoSuiteNames(
     std::vector<std::string>* crypto_suite_names);
 void GetDefaultSrtpCryptoSuiteNames(
     std::vector<std::string>* crypto_suite_names);
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_MEDIASESSION_H_