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

2011-05-30 11:22:19 +00:00
parent 01813fe945
commit 77ae29bc81
1153 changed files with 404089 additions and 0 deletions
--- a/modules/audio_coding/codecs/iSAC/main/source/decode_bwe.c
+++ b/modules/audio_coding/codecs/iSAC/main/source/decode_bwe.c
@@ -0,0 +1,88 @@
+/*
+ *  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.
+ */
+
+#include "structs.h"
+#include "bandwidth_estimator.h"
+#include "entropy_coding.h"
+#include "codec.h"
+
+
+int
+WebRtcIsac_EstimateBandwidth(
+    BwEstimatorstr*           bwest_str,
+    Bitstr*                   streamdata,
+    WebRtc_Word32               packet_size,
+    WebRtc_UWord16              rtp_seq_number,
+    WebRtc_UWord32              send_ts,
+    WebRtc_UWord32              arr_ts,
+    enum IsacSamplingRate encoderSampRate,
+    enum IsacSamplingRate decoderSampRate)
+{
+  WebRtc_Word16  index;
+  WebRtc_Word16  frame_samples;
+  WebRtc_UWord32 sendTimestampIn16kHz;
+  WebRtc_UWord32 arrivalTimestampIn16kHz;
+  WebRtc_UWord32 diffSendTime;
+  WebRtc_UWord32 diffArrivalTime;
+  int err;
+
+  /* decode framelength and BW estimation */
+  err = WebRtcIsac_DecodeFrameLen(streamdata, &frame_samples);
+  if(err < 0)  // error check
+  {
+    return err;
+  }
+  err = WebRtcIsac_DecodeSendBW(streamdata, &index);
+  if(err < 0)  // error check
+  {
+    return err;
+  }
+
+  /* UPDATE ESTIMATES FROM OTHER SIDE */
+  err = WebRtcIsac_UpdateUplinkBwImpl(bwest_str, index, encoderSampRate);
+  if(err < 0)
+  {
+    return err;
+  }
+
+  // We like BWE to work at 16 kHz sampling rate,
+  // therefore, we have to change the timestamps accordingly.
+  // translate the send timestamp if required
+  diffSendTime = (WebRtc_UWord32)((WebRtc_UWord32)send_ts -
+                                  (WebRtc_UWord32)bwest_str->senderTimestamp);
+  bwest_str->senderTimestamp = send_ts;
+
+  diffArrivalTime = (WebRtc_UWord32)((WebRtc_UWord32)arr_ts -
+                                     (WebRtc_UWord32)bwest_str->receiverTimestamp);
+  bwest_str->receiverTimestamp = arr_ts;
+
+  if(decoderSampRate == kIsacSuperWideband)
+  {
+    diffArrivalTime = (WebRtc_UWord32)diffArrivalTime >> 1;
+    diffSendTime = (WebRtc_UWord32)diffSendTime >> 1;
+  }
+  // arrival timestamp in 16 kHz
+  arrivalTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
+                                             bwest_str->prev_rec_arr_ts + (WebRtc_UWord32)diffArrivalTime);
+  // send timestamp in 16 kHz
+  sendTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
+                                          bwest_str->prev_rec_send_ts + (WebRtc_UWord32)diffSendTime);
+
+  err = WebRtcIsac_UpdateBandwidthEstimator(bwest_str, rtp_seq_number,
+                                            (frame_samples * 1000) / FS, sendTimestampIn16kHz,
+                                            arrivalTimestampIn16kHz, packet_size);
+  // error check
+  if(err < 0)
+  {
+    return err;
+  }
+
+  return 0;
+}