Chromium Code Reviews| Index: webrtc/api/webrtcsdp.cc |
| diff --git a/webrtc/api/webrtcsdp.cc b/webrtc/api/webrtcsdp.cc |
| index e19840bdf2f94a417d049b01a0f0a620d8bc9aa0..009527d157a0acb8ebfa75908c8d4da2f9fcab7f 100644 |
| --- a/webrtc/api/webrtcsdp.cc |
| +++ b/webrtc/api/webrtcsdp.cc |
| @@ -205,11 +205,6 @@ 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 |
| @@ -3092,21 +3087,9 @@ bool ParseRtpmapAttribute(const std::string& line, |
| 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
ossu
2017/01/23 15:33:18
It did have an effect. It worked around the block
|
| 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 = |