Removing the use of the soon-to-be-removed echo_cancellation_impl

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  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 1362 matching lines @@
           "WebRTC.Audio.PlatformReportedStreamDelayJump", diff_stream_delay_ms,
           kMinDiffDelayMs, 1000, 100);
       if (capture_.stream_delay_jumps == -1) {
         capture_.stream_delay_jumps = 0;  // Activate counter if needed.
       }
       capture_.stream_delay_jumps++;
     }
     capture_.last_stream_delay_ms = capture_nonlocked_.stream_delay_ms;
 
     // Detect a jump in AEC system delay and log the difference.
-    const int frames_per_ms =
+    const int samples_per_ms =
         rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
     const int aec_system_delay_ms =
-        WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_per_ms;
+        public_submodules_->echo_cancellation->GetSystemDelayInSamples() /

[the sun, 2016/02/17 11:39:03]

any risk of a div by zero here?

[peah-webrtc, 2016/03/04 09:46:25]

Should be no risk, but adding a DCHECK just in case.

+        samples_per_ms;
     const int diff_aec_system_delay_ms =
         aec_system_delay_ms - capture_.last_aec_system_delay_ms;
     if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
         capture_.last_aec_system_delay_ms != 0) {
       RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.AecSystemDelayJump",
                                   diff_aec_system_delay_ms, kMinDiffDelayMs,
                                   1000, 100);
       if (capture_.aec_system_delay_jumps == -1) {
         capture_.aec_system_delay_jumps = 0;  // Activate counter if needed.
       }
@@ skipping 145 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc