Add AudioEncoder::GetTargetBitrate

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

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 52 matching lines @@
       return false;
   }
 }
 
 template <typename T>
 AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(const Config& config)
     : payload_type_(config.payload_type),
       state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
       decoder_sample_rate_hz_(0),
       lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      packet_in_progress_(false) {
+      packet_in_progress_(false),
+      target_bitrate_bps_(config.adaptive_mode ? -1 : (config.bit_rate == 0
+                                                           ? kDefaultBitRate
+                                                           : config.bit_rate)) {
   CHECK(config.IsOk());
   CHECK_EQ(0, T::Create(&isac_state_));
   CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
   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) {
-    CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate,
-                              config.frame_size_ms, config.enforce_frame_size));
-
+    CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms,
+                              config.enforce_frame_size));
   } else {
     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
   // still set to 32000 Hz, since there is no full-band mode in the decoder.
   CHECK_EQ(0, T::SetDecSampRate(isac_state_,
                                 std::min(config.sample_rate_hz, 32000)));
   if (config.max_payload_size_bytes != -1)
     CHECK_EQ(0,
              T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
@@ skipping 30 matching lines @@
   return rtc::CheckedDivExact(samples_in_next_packet,
                               rtc::CheckedDivExact(SampleRateHz(), 100));
 }
 
 template <typename T>
 int AudioEncoderDecoderIsacT<T>::Max10MsFramesInAPacket() const {
   return 6;  // iSAC puts at most 60 ms in a packet.
 }
 
 template <typename T>
+int AudioEncoderDecoderIsacT<T>::GetTargetBitrate() const {
+  return target_bitrate_bps_;
+}
+
+template <typename T>
 AudioEncoder::EncodedInfo AudioEncoderDecoderIsacT<T>::EncodeInternal(
     uint32_t rtp_timestamp,
     const int16_t* audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   CriticalSectionScoped cs_lock(lock_.get());
   if (!packet_in_progress_) {
     // Starting a new packet; remember the timestamp for later.
     packet_in_progress_ = true;
     packet_timestamp_ = rtp_timestamp;
@@ skipping 79 matching lines @@
 
 template <typename T>
 int AudioEncoderDecoderIsacT<T>::ErrorCode() {
   CriticalSectionScoped cs(state_lock_.get());
   return T::GetErrorCode(isac_state_);
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_