Index: webrtc/modules/audio_coding/main/interface/audio_coding_module.h |
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h |
index f2972e712c9072ca02d93f6ca495e422b747dca1..77b4c9d3aa8070e61b60553d4db74255854d96ed 100644 |
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h |
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h |
@@ -720,40 +720,6 @@ class AudioCodingModule { |
// |
/////////////////////////////////////////////////////////////////////////// |
- // int32_t SetISACMaxRate() |
- // Set the maximum instantaneous rate of iSAC. For a payload of B bits |
- // with a frame-size of T sec the instantaneous rate is B/T bits per |
- // second. Therefore, (B/T < |max_rate_bps|) and |
- // (B < |max_payload_len_bytes| * 8) are always satisfied for iSAC payloads, |
- // c.f SetISACMaxPayloadSize(). |
- // |
- // Input: |
- // -max_rate_bps : maximum instantaneous bit-rate given in bits/sec. |
- // |
- // Return value: |
- // -1 if failed to set the maximum rate. |
- // 0 if the maximum rate is set successfully. |
- // |
- virtual int SetISACMaxRate(int max_rate_bps) = 0; |
- |
- /////////////////////////////////////////////////////////////////////////// |
- // int32_t SetISACMaxPayloadSize() |
- // Set the maximum payload size of iSAC packets. No iSAC payload, |
- // regardless of its frame-size, may exceed the given limit. For |
- // an iSAC payload of size B bits and frame-size T seconds we have; |
- // (B < |max_payload_len_bytes| * 8) and (B/T < |max_rate_bps|), c.f. |
- // SetISACMaxRate(). |
- // |
- // Input: |
- // -max_payload_len_bytes : maximum payload size in bytes. |
- // |
- // Return value: |
- // -1 if failed to set the maximum payload-size. |
- // 0 if the given length is set successfully. |
- // |
- virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0; |
- |
- /////////////////////////////////////////////////////////////////////////// |
// int SetOpusApplication() |
// Sets the intended application if current send codec is Opus. Opus uses this |
// to optimize the encoding for applications like VOIP and music. Currently, |