Add new UMA metric for NetEq target buffer delay

The UMA metric will log the same information that goes into the
googPreferredJitterBufferMs stat.

Bug: webrtc:8488
Change-Id: I4e4e1e362dd42377105d52d2c4cd49c1ecb1a90d
Reviewed-on: https://webrtc-review.googlesource.com/26740
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20923}

modules/audio_coding/acm2/acm_receiver.cc

@@ -300,6 +300,10 @@ int AcmReceiver::FilteredCurrentDelayMs() const {
   return neteq_->FilteredCurrentDelayMs();
 }
 
+int AcmReceiver::TargetDelayMs() const {
+  return neteq_->TargetDelayMs();
+}
+
 int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
   rtc::CritScope lock(&crit_sect_);
   if (!last_audio_decoder_) {

modules/audio_coding/acm2/acm_receiver.h

@@ -204,6 +204,10 @@ class AcmReceiver {
   //
   int FilteredCurrentDelayMs() const;
 
+  // Returns the current target delay for NetEq in ms.
+  //
+  int TargetDelayMs() const;
+
   //
   // Get the audio codec associated with the last non-CNG/non-DTMF received
   // payload. If no non-CNG/non-DTMF packet is received -1 is returned,

modules/audio_coding/acm2/audio_coding_module.cc

@@ -164,6 +164,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
 
   int FilteredCurrentDelayMs() const override;
 
+  int TargetDelayMs() const override;
+
   // Get 10 milliseconds of raw audio data to play out, and
   // automatic resample to the requested frequency if > 0.
   int PlayoutData10Ms(int desired_freq_hz,
@@ -1193,6 +1195,10 @@ int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
   return receiver_.FilteredCurrentDelayMs();
 }
 
+int AudioCodingModuleImpl::TargetDelayMs() const {
+  return receiver_.TargetDelayMs();
+}
+
 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
   if (!encoder_stack_) {
     RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";

modules/audio_coding/include/audio_coding_module.h

@@ -655,6 +655,12 @@ class AudioCodingModule {
   //
   virtual int FilteredCurrentDelayMs() const = 0;
 
+  ///////////////////////////////////////////////////////////////////////////
+  // int FilteredCurrentDelayMs()
+  // Returns the current target delay for NetEq in ms.
+  //
+  virtual int TargetDelayMs() const = 0;
+
   ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutData10Ms(
   // Get 10 milliseconds of raw audio data for playout, at the given sampling

modules/audio_coding/neteq/include/neteq.h

@@ -208,7 +208,7 @@ class NetEq {
 
   // Returns the current target delay in ms. This includes any extra delay
   // requested through SetMinimumDelay.
-  virtual int TargetDelayMs() = 0;
+  virtual int TargetDelayMs() const = 0;
 
   // Returns the current total delay (packet buffer and sync buffer) in ms.
   virtual int CurrentDelayMs() const = 0;

modules/audio_coding/neteq/neteq_impl.cc

@@ -311,7 +311,7 @@ int NetEqImpl::SetTargetDelay() {
   return kNotImplemented;
 }
 
-int NetEqImpl::TargetDelayMs() {
+int NetEqImpl::TargetDelayMs() const {
   rtc::CritScope lock(&crit_sect_);
   RTC_DCHECK(delay_manager_.get());
   // The value from TargetLevel() is in number of packets, represented in Q8.

modules/audio_coding/neteq/neteq_impl.h

@@ -161,7 +161,7 @@ class NetEqImpl : public webrtc::NetEq {
 
   int SetTargetDelay() override;
 
-  int TargetDelayMs() override;
+  int TargetDelayMs() const override;
 
   int CurrentDelayMs() const override;
 

voice_engine/channel.cc

@@ -556,6 +556,8 @@ AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
   }
 
   {
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.TargetJitterBufferDelayMs",
+                              audio_coding_->TargetDelayMs());
     const int jitter_buffer_delay = audio_coding_->FilteredCurrentDelayMs();
     rtc::CritScope lock(&video_sync_lock_);
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs",