Implement RTCOutboundRtpStreamStats.targetBitrate for audio.

Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-targetbitrate

Bug: webrtc:13377
Change-Id: I98dd263e0b9d6e2ca94969d2a91857b14cd65f70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237402
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35337}
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 8ba1b9f..b742a82 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -92,6 +92,8 @@
 
   ANAStats GetANAStats() const override;
 
+  int GetTargetBitrate() const override;
+
  private:
   struct InputData {
     InputData() : buffer(kInitialInputDataBufferSize) {}
@@ -603,6 +605,14 @@
   return ANAStats();
 }
 
+int AudioCodingModuleImpl::GetTargetBitrate() const {
+  MutexLock lock(&acm_mutex_);
+  if (!encoder_stack_) {
+    return -1;
+  }
+  return encoder_stack_->GetTargetBitrate();
+}
+
 }  // namespace
 
 AudioCodingModule::Config::Config(