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 59 matching lines @@
       state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
       decoder_sample_rate_hz_(0),
       lock_(CriticalSectionWrapper::CreateCriticalSection()),
       packet_in_progress_(false) {
   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) {
+    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
   // 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));
   if (config.max_bit_rate != -1)
@@ skipping 29 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>::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,
     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_