Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

webrtc/modules/audio_coding/codecs/audio_encoder.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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 
 #include <algorithm>
 #include <vector>
 
 #include "webrtc/base/array_view.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/deprecation.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // This is the interface class for encoders in AudioCoding module. Each codec
 // type must have an implementation of this class.
 class AudioEncoder {
  public:
   struct EncodedInfoLeaf {
     size_t encoded_bytes = 0;
@@ skipping 50 matching lines @@
   // Num10MsFramesInNextPacket().
   virtual size_t Max10MsFramesInAPacket() const = 0;
 
   // Returns the current target bitrate in bits/s. The value -1 means that the
   // codec adapts the target automatically, and a current target cannot be
   // provided.
   virtual int GetTargetBitrate() const = 0;
 
   // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
   // NumChannels() samples). Multi-channel audio must be sample-interleaved.
-  // The encoder produces zero or more bytes of output in |encoded| and
-  // returns additional encoding information.
-  // The caller is responsible for making sure that |max_encoded_bytes| is
-  // not smaller than the number of bytes actually produced by the encoder.
-  // Encode() checks some preconditions, calls EncodeInternal() which does the
-  // actual work, and then checks some postconditions.
+  // The encoder appends zero or more bytes of output to |encoded| and returns
+  // additional encoding information.  Encode() checks some preconditions, calls
+  // EncodeInternal() which does the actual work, and then checks some
+  // postconditions.
   EncodedInfo Encode(uint32_t rtp_timestamp,
                      rtc::ArrayView<const int16_t> audio,
-                     size_t max_encoded_bytes,
-                     uint8_t* encoded);
+                     rtc::Buffer* encoded);
 
-  virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                                     rtc::ArrayView<const int16_t> audio,
-                                     size_t max_encoded_bytes,
-                                     uint8_t* encoded) = 0;
+  // Deprecated interface to Encode (remove eventually, bug 5591). May incur a
+  // copy. The encoder produces zero or more bytes of output in |encoded| and
+  // returns additional encoding information.  The caller is responsible for
+  // making sure that |max_encoded_bytes| is not smaller than the number of
+  // bytes actually produced by the encoder.
+  RTC_DEPRECATED EncodedInfo Encode(uint32_t rtp_timestamp,
+                                    rtc::ArrayView<const int16_t> audio,
+                                    size_t max_encoded_bytes,
+                                    uint8_t* encoded);
+
+  EncodedInfo DEPRECATED_Encode(uint32_t rtp_timestamp,
+                                rtc::ArrayView<const int16_t> audio,
+                                size_t max_encoded_bytes,
+                                uint8_t* encoded);
+
+  // Deprecated interface of EncodeInternal (also bug 5591). May incur a copy.
+  // Subclasses implement this to perform the actual encoding. Called by
+  // Encode(). By default, this is implemented as a call to the newer
+  // EncodeInternal() that accepts an rtc::Buffer instead of a raw pointer.
+  // That version is protected, so see below. At least one of the two
+  // interfaces of EncodeInternal _must_ be implemented by a subclass.
+  virtual EncodedInfo EncodeInternal(
+      uint32_t rtp_timestamp,
+      rtc::ArrayView<const int16_t> audio,
+      size_t max_encoded_bytes,
+      uint8_t* encoded);
 
   // Resets the encoder to its starting state, discarding any input that has
   // been fed to the encoder but not yet emitted in a packet.
   virtual void Reset() = 0;
 
   // Enables or disables codec-internal FEC (forward error correction). Returns
   // true if the codec was able to comply. The default implementation returns
   // true when asked to disable FEC and false when asked to enable it (meaning
   // that FEC isn't supported).
   virtual bool SetFec(bool enable);
@@ skipping 18 matching lines @@
   // Tells the encoder what the projected packet loss rate is. The rate is in
   // the range [0.0, 1.0]. The encoder would typically use this information to
   // adjust channel coding efforts, such as FEC. The default implementation
   // does nothing.
   virtual void SetProjectedPacketLossRate(double fraction);
 
   // Tells the encoder what average bitrate we'd like it to produce. The
   // encoder is free to adjust or disregard the given bitrate (the default
   // implementation does the latter).
   virtual void SetTargetBitrate(int target_bps);
+
+ protected:
+  // Subclasses implement this to perform the actual encoding. Called by
+  // Encode(). For compatibility reasons, this is implemented by default as a
+  // call to the older version of EncodeInternal(). At least one of the two
+  // interfaces of EncodeInternal _must_ be implemented by a subclass.
+  // Preferably this one.
+  virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
+                                     rtc::ArrayView<const int16_t> audio,
+                                     rtc::Buffer* encoded);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_