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

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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 112 matching lines @@
 }
 
 size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
   return Num10msFramesPerPacket();
 }
 
 int AudioEncoderOpus::GetTargetBitrate() const {
   return config_.bitrate_bps;
 }
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(

[hlundin-webrtc, 2016/02/29 12:46:47]

Order no longer matches that of the .h file. I do realize that the diff will be
much larger if you fix it in this CL. Consider doing it in a last step, or in a
follow-up.

[ossu, 2016/02/29 13:23:01]

Acknowledged.

     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
-    size_t max_encoded_bytes,
-    uint8_t* encoded) {
+    rtc::Buffer* encoded) {
+
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
-  RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size());
+
   input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
   if (input_buffer_.size() <
       (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
   RTC_CHECK_EQ(input_buffer_.size(),
                Num10msFramesPerPacket() * SamplesPer10msFrame());
-  int status = WebRtcOpus_Encode(
-      inst_, &input_buffer_[0],
-      rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
-      rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded);
-  RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
+
+  const size_t max_encoded_bytes = MaxEncodedBytes();
+  EncodedInfo info;
+  info.encoded_bytes =
+      encoded->AppendData(
+          max_encoded_bytes, [&] (rtc::ArrayView<uint8_t> encoded) {
+            int status = WebRtcOpus_Encode(
+                inst_, &input_buffer_[0],
+                rtc::CheckedDivExact(input_buffer_.size(),
+                                     config_.num_channels),
+                rtc::saturated_cast<int16_t>(max_encoded_bytes),
+                encoded.data());
+
+            RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
+
+            return static_cast<size_t>(status);
+          });
   input_buffer_.clear();
-  EncodedInfo info;
-  info.encoded_bytes = static_cast<size_t>(status);
+
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = config_.payload_type;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
-  info.speech = (status > 0);
+  info.speech = (info.encoded_bytes > 0);
   return info;
 }
 
 void AudioEncoderOpus::Reset() {
   RTC_CHECK(RecreateEncoderInstance(config_));
 }
 
 bool AudioEncoderOpus::SetFec(bool enable) {
   auto conf = config_;
   conf.fec_enabled = enable;
@@ skipping 77 matching lines @@
     RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
   RTC_CHECK_EQ(0,
                WebRtcOpus_SetPacketLossRate(
                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
   config_ = config;
   return true;
 }
 
 }  // namespace webrtc