Removed double-special-casing of ISAC in libjingle and WebRtcVoE.

webrtc/api/webrtcsdp.cc

 /*
  *  Copyright 2011 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 187 matching lines @@
 // address to IP6 ::.
 // TODO(perkj): FF can not parse IP6 ::. See http://crbug/430333
 // Use IPV4 per default.
 static const char kDummyAddress[] = "0.0.0.0";
 static const char kDummyPort[] = "9";
 // RFC 3556
 static const char kApplicationSpecificMaximum[] = "AS";
 
 static const int kDefaultVideoClockrate = 90000;
 
-// ISAC special-case.
-static const char kIsacCodecName[] = "ISAC";  // From webrtcvoiceengine.cc
-static const int kIsacWbDefaultRate = 32000;  // From acm_common_defs.h
-static const int kIsacSwbDefaultRate = 56000;  // From acm_common_defs.h
-
 static const char kDefaultSctpmapProtocol[] = "webrtc-datachannel";
 
 // RTP payload type is in the 0-127 range. Use -1 to indicate "all" payload
 // types.
 const int kWildcardPayloadType = -1;
 
 struct SsrcInfo {
   uint32_t ssrc_id;
   std::string cname;
   std::string stream_id;
@@ skipping 2862 matching lines @@
     // For audio streams, <encoding parameters> indicates the number
     // of audio channels.  This parameter is OPTIONAL and may be
     // omitted if the number of channels is one, provided that no
     // additional parameters are needed.
     size_t channels = 1;
     if (codec_params.size() == 3) {
       if (!GetValueFromString(line, codec_params[2], &channels, error)) {
         return false;
       }
     }
-    int bitrate = 0;
-    // The default behavior for ISAC (bitrate == 0) in webrtcvoiceengine.cc
-    // (specifically FindWebRtcCodec) is bandwidth-adaptive variable bitrate.
-    // The bandwidth adaptation doesn't always work well, so this code
-    // sets a fixed target bitrate instead.
-    if (_stricmp(encoding_name.c_str(), kIsacCodecName) == 0) {
-      if (clock_rate <= 16000) {
-        bitrate = kIsacWbDefaultRate;
-      } else {
-        bitrate = kIsacSwbDefaultRate;
-      }
-    }

[kwiberg-webrtc, 2017/01/23 15:18:11]

Hmm. I don't quite see how this piece of code didn't have any effect. Doesn't it
set the bitrate to one of two constants, a value which would be passed down to
the encoder without being changed along the way?

[ossu, 2017/01/23 15:33:18]

It did have an effect. It worked around the block of code I removed from
webrtcvoiceengine.cc. These two constants are already set in the ACMCodecDB.

     AudioContentDescription* audio_desc =
         static_cast<AudioContentDescription*>(media_desc);
-    UpdateCodec(payload_type, encoding_name, clock_rate, bitrate, channels,
+    UpdateCodec(payload_type, encoding_name, clock_rate, 0, channels,
                 audio_desc);
   } else if (media_type == cricket::MEDIA_TYPE_DATA) {
     DataContentDescription* data_desc =
         static_cast<DataContentDescription*>(media_desc);
     data_desc->AddCodec(cricket::DataCodec(payload_type, encoding_name));
   }
   return true;
 }
 
 bool ParseFmtpParam(const std::string& line, std::string* parameter,
@@ skipping 105 matching lines @@
     UpdateCodec<AudioContentDescription, cricket::AudioCodec>(
         media_desc, payload_type, feedback_param);
   } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
     UpdateCodec<VideoContentDescription, cricket::VideoCodec>(
         media_desc, payload_type, feedback_param);
   }
   return true;
 }
 
 }  // namespace webrtc