Upstream version 9.38.198.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / audio_coding / main / interface / audio_coding_module.h

diff --git a/src/third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/src/third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index cb0953a..aae7ead 100644 (file)
--- a/src/third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/src/third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -330,6 +330,7 @@ class AudioCodingModule: public Module {
   //   -1 if error occurred in setting the bandwidth,
   //    0 bandwidth is set successfully.
   //
+  // TODO(henrik.lundin) Unused. Remove?
   virtual int32_t SetReceivedEstimatedBandwidth(
       const int32_t bw) = 0;
 
@@ -915,6 +916,23 @@ class AudioCodingModule: public Module {
       bool enforce_frame_size = false) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
+  // int SetOpusMaxBandwidth()
+  // If current send codec is Opus, informs it about maximum audio bandwidth
+  // needs to be encoded. A use case of this is when the receiver can only play
+  // audio up to frequency limit. Opus can use this information to optimize
+  // the bit rate and increase the computation efficiency.
+  //
+  // Input:
+  //   -banbwidth_hz            : maximum bandwidth in Hz.
+  //
+  // Return value:
+  //   -1 if current send codec is not Opus or
+  //      error occurred in setting the bandwidth,
+  //    0 maximum bandwidth is set successfully.
+  //
+  virtual int SetOpusMaxBandwidth(int banbwidth_hz) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
   //   statistics
   //