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

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

 /*
  *  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 74 matching lines @@
       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
       return 13333;
     default:
       FATAL();
   }
 }
 
 AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
-    size_t max_encoded_bytes,
-    uint8_t* encoded) {
-  RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
+    rtc::Buffer* encoded) {
 
   // Save timestamp if starting a new packet.
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Buffer input.
-  RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size());
   std::copy(audio.cbegin(), audio.cend(),
             input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
 
   // If we don't yet have enough buffered input for a whole packet, we're done
   // for now.
   if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
     return EncodedInfo();
   }
 
   // Encode buffered input.
   RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
-  const int output_len = WebRtcIlbcfix_Encode(
-      encoder_,
-      input_buffer_,
-      kSampleRateHz / 100 * num_10ms_frames_per_packet_,
-      encoded);
-  RTC_CHECK_GE(output_len, 0);
+  size_t encoded_bytes =
+      encoded->AppendData(
+          RequiredOutputSizeBytes(),
+          [&] (rtc::ArrayView<uint8_t> encoded) {
+            const int r = WebRtcIlbcfix_Encode(
+                encoder_,
+                input_buffer_,
+                kSampleRateHz / 100 * num_10ms_frames_per_packet_,
+                encoded.data());
+            RTC_CHECK_GE(r, 0);
+
+            return static_cast<size_t>(r);
+          });
+
+  RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
+
   EncodedInfo info;
-  info.encoded_bytes = static_cast<size_t>(output_len);
-  RTC_DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
+  info.encoded_bytes = encoded_bytes;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = config_.payload_type;
   return info;
 }
 
 void AudioEncoderIlbc::Reset() {
   if (encoder_)
     RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
   RTC_CHECK(config_.IsOk());
   RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
   const int encoder_frame_size_ms = config_.frame_size_ms > 30
                                         ? config_.frame_size_ms / 2
                                         : config_.frame_size_ms;
   RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
   num_10ms_frames_buffered_ = 0;
 }
 
 size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const {
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
     case 3:   return 50;
     case 4:   return 2 * 38;
     case 6:   return 2 * 50;
     default:  FATAL();
   }
 }
 
 }  // namespace webrtc