Adds target rate to audio send stream stats.

Bug: webrtc:9510
Change-Id: I8bd74fc115e3006f477b289edc58fa1f9d7b6bc6
Reviewed-on: https://webrtc-review.googlesource.com/c/107652
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25370}

audio/audio_send_stream.cc

@@ -367,6 +367,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;
   stats.local_ssrc = config_.rtp.ssrc;
+  stats.target_bitrate_bps = channel_proxy_->GetBitrate();
 
   webrtc::CallSendStatistics call_stats = channel_proxy_->GetRTCPStatistics();
   stats.bytes_sent = call_stats.bytesSent;

audio/audio_send_stream_unittest.cc

@@ -278,6 +278,7 @@ struct ConfigHelper {
         .WillRepeatedly(Return(report_blocks));
     EXPECT_CALL(*channel_proxy_, GetANAStatistics())
         .WillRepeatedly(Return(ANAStats()));
+    EXPECT_CALL(*channel_proxy_, GetBitrate()).WillRepeatedly(Return(0));
 
     audio_processing_stats_.echo_return_loss = kEchoReturnLoss;
     audio_processing_stats_.echo_return_loss_enhancement =

audio/channel_send.cc

@@ -572,6 +572,11 @@ void ChannelSend::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) {
     }
   });
   retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
+  configured_bitrate_bps_ = bitrate_bps;
+}
+
+int ChannelSend::GetBitRate() const {
+  return configured_bitrate_bps_;
 }
 
 void ChannelSend::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {

audio/channel_send.h

@@ -139,6 +139,7 @@ class ChannelSend
 
   // Codecs
   void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
+  int GetBitRate() const;
   bool EnableAudioNetworkAdaptor(const std::string& config_string);
   void DisableAudioNetworkAdaptor();
 
@@ -302,6 +303,7 @@ class ChannelSend
   FrameEncryptorInterface* frame_encryptor_ = nullptr;
   // E2EE Frame Encryption Options
   webrtc::CryptoOptions crypto_options_;
+  int configured_bitrate_bps_ = 0;
 };
 
 }  // namespace voe

audio/channel_send_proxy.cc

@@ -147,6 +147,10 @@ void ChannelSendProxy::SetBitrate(int bitrate_bps,
   channel_->SetBitRate(bitrate_bps, probing_interval_ms);
 }
 
+int ChannelSendProxy::GetBitrate() const {
+  return channel_->GetBitRate();
+}
+
 void ChannelSendProxy::SetInputMute(bool muted) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   channel_->SetInputMute(muted);

audio/channel_send_proxy.h

@@ -70,6 +70,7 @@ class ChannelSendProxy {
                                                 int payload_frequency);
   virtual bool SendTelephoneEventOutband(int event, int duration_ms);
   virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
+  virtual int GetBitrate() const;
   virtual void SetInputMute(bool muted);
 
   virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame);

audio/mock_voe_channel_proxy.h

@@ -99,6 +99,7 @@ class MockChannelSendProxy : public voe::ChannelSendProxy {
                void(std::unique_ptr<AudioFrame>* audio_frame));
   MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
   MOCK_CONST_METHOD0(GetRtpRtcp, RtpRtcp*());
+  MOCK_CONST_METHOD0(GetBitrate, int());
   MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
   MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
                void(float recoverable_packet_loss_rate));

call/audio_send_stream.h

@@ -58,6 +58,8 @@ class AudioSendStream {
 
     ANAStats ana_statistics;
     AudioProcessingStats apm_statistics;
+
+    int64_t target_bitrate_bps = 0;
   };
 
   struct Config {

test/scenario/audio_stream.cc

@@ -149,6 +149,16 @@ void SendAudioStream::Start() {
   send_stream_->Start();
 }
 
+ColumnPrinter SendAudioStream::StatsPrinter() {
+  return ColumnPrinter::Lambda(
+      "audio_target_rate",
+      [this](rtc::SimpleStringBuilder& sb) {
+        AudioSendStream::Stats stats = send_stream_->GetStats();
+        sb.AppendFormat("%.0lf", stats.target_bitrate_bps / 8.0);
+      },
+      64);
+}
+
 ReceiveAudioStream::ReceiveAudioStream(
     CallClient* receiver,
     AudioStreamConfig config,

test/scenario/audio_stream.h

@@ -29,6 +29,7 @@ class SendAudioStream {
   RTC_DISALLOW_COPY_AND_ASSIGN(SendAudioStream);
   ~SendAudioStream();
   void Start();
+  ColumnPrinter StatsPrinter();
 
  private:
   friend class Scenario;