commit 8034614b817810f33ad193d8fa5c460bc6c28276
author deadbeef <deadbeef@webrtc.org> Wed Apr 27 21:17:10 2016
committer Commit bot <commit-bot@chromium.org> Wed Apr 27 21:17:15 2016
tree 5a2d7737b416255b18030eac3f31f54f61ccf299
parent 081254f2c62037d016f9fc961764c6f01cb095da

Cap the send bitrate for opus and iSAC before passing down to VoE.

The voice engine expects send bitrates no more than the maximum for the
codec. For example, 510kbps for opus. So if "b=AS" sets a maximum above
the codec maximum, WebRtcVoiceEngine needs to cap it.

BUG=603690

Review-Url: https://codereview.webrtc.org/1920123002
Cr-Commit-Position: refs/heads/master@{#12537}

webrtc/media/engine/webrtcvoiceengine.h

diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index fd46dee..1cc5504 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -228,10 +228,10 @@
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
-  bool SetSendBitrate(int bps);
+  bool SetMaxSendBitrate(int bps);
   bool SetChannelParameters(int channel,
                             const webrtc::RtpParameters& parameters);
-  bool SetSendBitrate(int channel, int bps);
+  bool SetMaxSendBitrate(int channel, int bps);
   bool HasSendCodec() const {
     return send_codec_spec_.codec_inst.pltype != -1;
   }
@@ -243,7 +243,7 @@
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> send_codecs_;
   std::vector<AudioCodec> recv_codecs_;
-  int send_bitrate_bps_ = 0;
+  int max_send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;