NetEq: Add new method last_output_sample_rate_hz

webrtc/modules/audio_coding/main/acm2/acm_receiver.h

 /*
  *  Copyright (c) 2013 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 136 matching lines @@
   // this is before applying any user-defined limits (specified by calling
   // (SetMinimumDelay() and/or SetMaximumDelay()).
   //
   int LeastRequiredDelayMs() const;
 
   //
   // Resets the initial delay to zero.
   //
   void ResetInitialDelay();
 
-  //
-  // Get the current sampling frequency in Hz.
-  //
-  // Return value             : Sampling frequency in Hz.
-  //
-  int current_sample_rate_hz() const;
+  // Returns last_output_sample_rate_hz from the NetEq instance.
+  int last_output_sample_rate_hz() const;
 
   //
   // Get the current network statistics from NetEq.
   //
   // Output:
   //   - statistics           : The current network statistics.
   //
   void GetNetworkStatistics(NetworkStatistics* statistics);
 
   //
@@ skipping 107 matching lines @@
   const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
                                     uint8_t payload_type) const
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   int id_;  // TODO(henrik.lundin) Make const.
   const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
-  int current_sample_rate_hz_ GUARDED_BY(crit_sect_);
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
   // Used in GetAudio, declared as member to avoid allocating every 10ms.
   // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
   rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
   NetEq* neteq_;
   // Decoders map is keyed by payload type
   std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_);
   bool vad_enabled_;
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_