Add aecdump support to audioproc_f.

webrtc/common_audio/wav_file.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 11 matching lines @@
 namespace webrtc {
 
 // Interface to provide access to WAV file parameters.
 class WavFile {
  public:
   virtual ~WavFile() {}
 
   virtual int sample_rate() const = 0;
   virtual int num_channels() const = 0;
   virtual uint32_t num_samples() const = 0;
+
+  // Returns a human-readable string containing the audio format.
+  std::string FormatAsString() const;
 };
 
 // Simple C++ class for writing 16-bit PCM WAV files. All error handling is
 // by calls to RTC_CHECK(), making it unsuitable for anything but debug code.
 class WavWriter final : public WavFile {
  public:
   // Open a new WAV file for writing.
   WavWriter(const std::string& filename, int sample_rate, int num_channels);
 
   // Close the WAV file, after writing its header.
@@ skipping 64 matching lines @@
                          size_t num_samples);
 int rtc_WavSampleRate(const rtc_WavWriter* wf);
 int rtc_WavNumChannels(const rtc_WavWriter* wf);
 uint32_t rtc_WavNumSamples(const rtc_WavWriter* wf);
 
 #ifdef __cplusplus
 }  // extern "C"
 #endif
 
 #endif  // WEBRTC_COMMON_AUDIO_WAV_FILE_H_