Adding a new ChangeLogger class to handle UMA logging of bitrates

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index c01466f8c02f970e577d7c799dbf2da1b1a5b7fe..570b001bc2590158aafc37017a7825bc5395b3ca 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -23,7 +23,6 @@
 #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"
@@ -129,6 +128,8 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
       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),
@@ -185,6 +186,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   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;
@@ -296,8 +298,7 @@ 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);
   }
 }