Add UMA logging for target audio bitrate

webrtc/modules/audio_coding/main/acm2/audio_coding_module_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.
  */
 
 #include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
 
 #include <assert.h>
 #include <stdlib.h>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 namespace acm2 {
 
 enum {
   kACMToneEnd = 999
@@ skipping 245 matching lines @@
   //  }
   //
   //  WebRtcACMCodecParams encoder_param;
   //  codec_manager_.current_encoder()->EncoderParams(&encoder_param);
   //
   //  return encoder_param.codec_inst.rate;
 }
 
 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
   CriticalSectionScoped lock(acm_crit_sect_);
-
   if (codec_manager_.CurrentEncoder()) {
     codec_manager_.CurrentEncoder()->SetTargetBitrate(bitrate_bps);
+    RTC_HISTOGRAM_COUNTS_100(
+        HISTOGRAM_NAME_AUDIO_TARGET_BITRATE_IN_KBPS,
+        codec_manager_.CurrentEncoder()->GetTargetBitrate() / 1000);
   }
 }
 
 // Set available bandwidth, inform the encoder about the estimated bandwidth
 // received from the remote party.
 // TODO(henrik.lundin): Remove; not used.
 int AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(int bw) {
   CriticalSectionScoped lock(acm_crit_sect_);
   FATAL() << "Dead code?";
   return -1;
@@ skipping 954 matching lines @@
       *channels = 1;
       break;
 #endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc