Update audio_processing module

Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these

webrtc/common_audio/wav_file.h

@@ -0,0 +1,115 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_WAV_FILE_H_
+#define WEBRTC_COMMON_AUDIO_WAV_FILE_H_
+
+#ifdef __cplusplus
+
+#include <stdint.h>
+#include <cstddef>
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+
+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;
+};
+
+// 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.
+  ~WavWriter();
+
+  // Write additional samples to the file. Each sample is in the range
+  // [-32768,32767], and there must be the previously specified number of
+  // interleaved channels.
+  void WriteSamples(const float* samples, size_t num_samples);
+  void WriteSamples(const int16_t* samples, size_t num_samples);
+
+  int sample_rate() const override { return sample_rate_; }
+  int num_channels() const override { return num_channels_; }
+  uint32_t num_samples() const override { return num_samples_; }
+
+ private:
+  void Close();
+  const int sample_rate_;
+  const int num_channels_;
+  uint32_t num_samples_;  // Total number of samples written to file.
+  FILE* file_handle_;  // Output file, owned by this class
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(WavWriter);
+};
+
+// Follows the conventions of WavWriter.
+class WavReader final : public WavFile {
+ public:
+  // Opens an existing WAV file for reading.
+  explicit WavReader(const std::string& filename);
+
+  // Close the WAV file.
+  ~WavReader();
+
+  // Returns the number of samples read. If this is less than requested,
+  // verifies that the end of the file was reached.
+  size_t ReadSamples(size_t num_samples, float* samples);
+  size_t ReadSamples(size_t num_samples, int16_t* samples);
+
+  int sample_rate() const override { return sample_rate_; }
+  int num_channels() const override { return num_channels_; }
+  uint32_t num_samples() const override { return num_samples_; }
+
+ private:
+  void Close();
+  int sample_rate_;
+  int num_channels_;
+  uint32_t num_samples_;  // Total number of samples in the file.
+  uint32_t num_samples_remaining_;
+  FILE* file_handle_;  // Input file, owned by this class.
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(WavReader);
+};
+
+}  // namespace webrtc
+
+extern "C" {
+#endif  // __cplusplus
+
+// C wrappers for the WavWriter class.
+typedef struct rtc_WavWriter rtc_WavWriter;
+rtc_WavWriter* rtc_WavOpen(const char* filename,
+                           int sample_rate,
+                           int num_channels);
+void rtc_WavClose(rtc_WavWriter* wf);
+void rtc_WavWriteSamples(rtc_WavWriter* wf,
+                         const float* samples,
+                         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_