Move some send stream configuration into webrtc::AudioSendStream.

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 170 matching lines @@
           max_encoding_bandwidth(0),
           opus_dtx(false),
           red(false),
           nack(false),
           cn8_type(13),
           cn16_type(105),
           dtmf_type(106),
           red_type(117),
           nack_max_packets(0),
           send_ssrc(0),
-          send_audio_level_ext_(-1),
           receive_audio_level_ext_(-1),
-          send_absolute_sender_time_ext_(-1),
           receive_absolute_sender_time_ext_(-1),
           associate_send_channel(-1),
           neteq_capacity(-1),
           neteq_fast_accelerate(false) {
       memset(&send_codec, 0, sizeof(send_codec));
     }
     bool external_transport;
     bool send;
     bool playout;
     float volume_scale;
     bool vad;
     bool codec_fec;
     int max_encoding_bandwidth;
     bool opus_dtx;
     bool red;
     bool nack;
     int cn8_type;
     int cn16_type;
     int dtmf_type;
     int red_type;
     int nack_max_packets;
     uint32_t send_ssrc;
-    int send_audio_level_ext_;
     int receive_audio_level_ext_;
-    int send_absolute_sender_time_ext_;
     int receive_absolute_sender_time_ext_;
     int associate_send_channel;
     DtmfInfo dtmf_info;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
     webrtc::PacketTime last_rtp_packet_time;
     std::list<std::string> packets;
     int neteq_capacity;
     bool neteq_fast_accelerate;
   };
@@ skipping 31 matching lines @@
     for (std::map<int, Channel*>::const_iterator i = channels_.begin();
          i != channels_.end(); ++i) {
       delete i->second;
     }
   }
 
   bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
 
   bool IsInited() const { return inited_; }
   int GetLastChannel() const { return last_channel_; }
-  int GetChannelFromLocalSsrc(uint32_t local_ssrc) const {
-    for (std::map<int, Channel*>::const_iterator iter = channels_.begin();
-         iter != channels_.end(); ++iter) {
-      if (local_ssrc == iter->second->send_ssrc)
-        return iter->first;
-    }
-    return -1;
-  }
   int GetNumChannels() const { return static_cast<int>(channels_.size()); }
   uint32_t GetLocalSSRC(int channel) {
     return channels_[channel]->send_ssrc;
   }
   bool GetPlayout(int channel) {
     return channels_[channel]->playout;
   }
   bool GetSend(int channel) {
     return channels_[channel]->send;
   }
@@ skipping 69 matching lines @@
       ch->recv_codecs.push_back(codec);
     }
     if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
       ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
     }
     ch->neteq_fast_accelerate =
         config.Get<webrtc::NetEqFastAccelerate>().enabled;
     channels_[++last_channel_] = ch;
     return last_channel_;
   }
-  int GetSendRtpExtensionId(int channel, const std::string& extension) {
-    WEBRTC_ASSERT_CHANNEL(channel);
-    if (extension == kRtpAudioLevelHeaderExtension) {
-      return channels_[channel]->send_audio_level_ext_;
-    } else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) {
-      return channels_[channel]->send_absolute_sender_time_ext_;
-    }
-    return -1;
-  }
   int GetReceiveRtpExtensionId(int channel, const std::string& extension) {
     WEBRTC_ASSERT_CHANNEL(channel);
     if (extension == kRtpAudioLevelHeaderExtension) {
       return channels_[channel]->receive_audio_level_ext_;
     } else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) {
       return channels_[channel]->receive_absolute_sender_time_ext_;
     }
     return -1;
   }
 
@@ skipping 336 matching lines @@
                                    size_t length));
 
   // webrtc::VoERTP_RTCP
   WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->send_ssrc = ssrc;
     return 0;
   }
   WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc));
   WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
-  WEBRTC_FUNC(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
-      unsigned char id)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
-    channels_[channel]->send_audio_level_ext_ = (enable) ? id : -1;
-    return 0;
-  }
+  WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
+      unsigned char id));
   WEBRTC_FUNC(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
       unsigned char id)) {
     WEBRTC_CHECK_CHANNEL(channel);
     WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
     channels_[channel]->receive_audio_level_ext_ = (enable) ? id : -1;
    return 0;
   }
-  WEBRTC_FUNC(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
-      unsigned char id)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
-    channels_[channel]->send_absolute_sender_time_ext_ = (enable) ? id : -1;
-    return 0;
-  }
+  WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
+      unsigned char id));
   WEBRTC_FUNC(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable,
       unsigned char id)) {
     WEBRTC_CHECK_CHANNEL(channel);
     WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
     channels_[channel]->receive_absolute_sender_time_ext_ = (enable) ? id : -1;
     return 0;
   }
 
   WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable));
   WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled));
@@ skipping 247 matching lines @@
   int playout_sample_rate_;
   DtmfInfo dtmf_info_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_