Add AudioEncoder::GetTargetBitrate

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index befb3558beb85ff8eb304fae5317bbe6f19cde52..1c386e0181a2e064ea4ee6b0db7db6c2e4bd913e 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -77,10 +77,12 @@ AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(const Config& config)
   CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
   const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
   if (config.adaptive_mode) {
+    target_bitrate_bps_ = -1;
     CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate,
                               config.frame_size_ms, config.enforce_frame_size));
 
   } else {
+    target_bitrate_bps_ = bit_rate;
     CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
   }
   // When config.sample_rate_hz is set to 48000 Hz (iSAC-fb), the decoder is
@@ -130,6 +132,22 @@ int AudioEncoderDecoderIsacT<T>::Max10MsFramesInAPacket() const {
 }
 
 template <typename T>
+int AudioEncoderDecoderIsacT<T>::SetTargetBitrate(int bits_per_second) {
+  CriticalSectionScoped cs_lock(lock_.get());
+  if (target_bitrate_bps_ == -1) {
+    // Adaptive mode. Cannot change the target bitrate manually.
+    return target_bitrate_bps_;

[kwiberg-webrtc, 2015/06/16 19:22:54]

Just "return -1" might be clearer. Also, document in the interface what -1
means.

+  }
+  // Clamp to at minimum 10000 bps and maximum 32000 bps or 56000 bps for SWB.
+  target_bitrate_bps_ = std::min(std::max(bits_per_second, 10000),
+                                 SampleRateHz() == 16000 ? 32000 : 56000);
+  CriticalSectionScoped cs(state_lock_.get());
+  CHECK_EQ(0, T::Control(isac_state_, target_bitrate_bps_,
+                         Num10MsFramesInNextPacket() * 10));
+  return target_bitrate_bps_;
+}
+
+template <typename T>
 AudioEncoder::EncodedInfo AudioEncoderDecoderIsacT<T>::EncodeInternal(
     uint32_t rtp_timestamp,
     const int16_t* audio,