Lock scheme #5: Applied the render queueing to the agc

webrtc/modules/audio_processing/agc/legacy/gain_control.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 32 matching lines @@
     int16_t compressionGaindB; // default 9 dB
     uint8_t limiterEnable;     // default kAgcTrue (on)
 } WebRtcAgcConfig;
 
 #if defined(__cplusplus)
 extern "C"
 {
 #endif
 
 /*
+ * This function analyses the number of samples passed to
+ * farend and produces any error code that could arise.
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ *      - samples           : Number of samples in input vector.
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error.
+ */
+int WebRtcAgc_GetAddFarendError(void* state, size_t samples);
+
+/*
  * This function processes a 10 ms frame of far-end speech to determine
  * if there is active speech. The length of the input speech vector must be
  * given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
  * FS=48000).
  *
  * Input:
  *      - agcInst           : AGC instance.
  *      - inFar             : Far-end input speech vector
  *      - samples           : Number of samples in input vector
  *
@@ skipping 170 matching lines @@
                    int32_t minLevel,
                    int32_t maxLevel,
                    int16_t agcMode,
                    uint32_t fs);
 
 #if defined(__cplusplus)
 }
 #endif
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_