git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

modules/audio_coding/codecs/iSAC/fix/source/decode_bwe.c

@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 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.
+ */
+
+/*
+ * decode_bwe.c
+ *
+ * This C file contains the internal decode bandwidth estimate function.
+ *
+ */
+
+
+#include "bandwidth_estimator.h"
+#include "codec.h"
+#include "entropy_coding.h"
+#include "structs.h"
+
+
+
+
+int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
+                                    Bitstr_dec  *streamdata,
+                                    WebRtc_Word32  packet_size,
+                                    WebRtc_UWord16 rtp_seq_number,
+                                    WebRtc_UWord32 send_ts,
+                                    WebRtc_UWord32 arr_ts)
+{
+  WebRtc_Word16 index;
+  WebRtc_Word16 frame_samples;
+  int err;
+
+  /* decode framelength */
+  err = WebRtcIsacfix_DecodeFrameLen(streamdata, &frame_samples);
+  /* error check */
+  if (err<0) {
+    return err;
+  }
+
+  /* decode BW estimation */
+  err = WebRtcIsacfix_DecodeSendBandwidth(streamdata, &index);
+  /* error check */
+  if (err<0) {
+    return err;
+  }
+
+  /* Update BWE with received data */
+  err = WebRtcIsacfix_UpdateUplinkBwImpl(
+      bwest_str,
+      rtp_seq_number,
+      (WebRtc_UWord16)WEBRTC_SPL_UDIV(WEBRTC_SPL_UMUL(frame_samples,1000), FS),
+      send_ts,
+      arr_ts,
+      (WebRtc_Word16) packet_size,  /* in bytes */
+      index);
+
+  /* error check */
+  if (err<0) {
+    return err;
+  }
+
+  /* Succesful */
+  return 0;
+}