Adding a new ChangeLogger class to handle UMA logging of bitrates

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 85 matching lines @@
 }
 }  // namespace
 
 AudioCodingModuleImpl::AudioCodingModuleImpl(
     const AudioCodingModule::Config& config)
     : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       id_(config.id),
       expected_codec_ts_(0xD87F3F9F),
       expected_in_ts_(0xD87F3F9F),
       receiver_(config),
+      bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
+      codec_manager_(&bitrate_logger_),
       previous_pltype_(255),
       aux_rtp_header_(NULL),
       receiver_initialized_(false),
       first_10ms_data_(false),
       first_frame_(true),
       callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       packetization_callback_(NULL),
       vad_callback_(NULL) {
   if (InitializeReceiverSafe() < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
@@ skipping 36 matching lines @@
                              static_cast<uint32_t>(rtc::CheckedDivExact(
                                  audio_encoder->SampleRateHz(),
                                  audio_encoder->RtpTimestampRateHz())));
   last_timestamp_ = input_data.input_timestamp;
   last_rtp_timestamp_ = rtp_timestamp;
   first_frame_ = false;
 
   encoded_info = audio_encoder->Encode(rtp_timestamp, input_data.audio,
                                        input_data.length_per_channel,
                                        sizeof(stream), stream);
+  bitrate_logger_.MaybeLogValue();

[kwiberg-webrtc, 2015/06/25 09:42:44]

Sorry to come with an alternate suggestion at this stage, but if you do want to
suppress logging of the same value more than once in a row (which I asked about
in another comment), a simpler approach would be to replace DelayedLogger with a
DedupLogger class, and call

  bitrate_logger_.MaybeLogValue(audio_encoder_->GetTargetBitrate() / 1000);

here. That way, you wouldn't have to log in all the places where the target
bitrate can change, which would eliminate the possibility of missing one of
them.

   if (encoded_info.encoded_bytes == 0 && !encoded_info.send_even_if_empty) {
     // Not enough data.
     return 0;
   }
   previous_pltype = previous_pltype_;  // Read it while we have the critsect.
 
   RTPFragmentationHeader my_fragmentation;
   ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
   FrameType frame_type;
   if (encoded_info.encoded_bytes == 0 && encoded_info.send_even_if_empty) {
@@ skipping 91 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,
+    bitrate_logger_.SetValue(
         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?";
@@ skipping 955 matching lines @@
       *channels = 1;
       break;
 #endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc