AudioDecoder: New method SampleRateHz, + implementations for our codecs

webrtc/modules/audio_coding/codecs/audio_decoder.h

 /*
  *  Copyright (c) 2012 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 75 matching lines @@
   // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
   // duration estimate is available, or -1 in case of an error.
   virtual int PacketDurationRedundant(const uint8_t* encoded,
                                       size_t encoded_len) const;
 
   // Detects whether a packet has forward error correction. The packet is
   // comprised of the samples in |encoded| which is |encoded_len| bytes long.
   // Returns true if the packet has FEC and false otherwise.
   virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
 
+  // Returns the actual sample rate of the decoder's output.
+  // NOTE: For now, this has a default implementation that returns an unusable
+  // value (-1). That default implementation will go away soon, and at the same
+  // time callers will start relying on the return value, so make sure you
+  // override it with something that returns a correct value!
+  // TODO(kwiberg): Remove the default implementation.
+  virtual int SampleRateHz() const;
+
   virtual size_t Channels() const = 0;
 
  protected:
   static SpeechType ConvertSpeechType(int16_t type);
 
   virtual int DecodeInternal(const uint8_t* encoded,
                              size_t encoded_len,
                              int sample_rate_hz,
                              int16_t* decoded,
                              SpeechType* speech_type) = 0;
 
   virtual int DecodeRedundantInternal(const uint8_t* encoded,
                                       size_t encoded_len,
                                       int sample_rate_hz,
                                       int16_t* decoded,
                                       SpeechType* speech_type);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_