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

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 707d6c24884683ce00b5fb849ebd91f0406c1b51..bd3af22b461fe0e764fa15b543cfc5f7b7a62dc5 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -133,11 +133,11 @@ int AudioEncoderOpus::GetTargetBitrate() const {
 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())) {
@@ -145,18 +145,29 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
   }
   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.
-  input_buffer_.clear();
+
+  const size_t max_encoded_bytes = MaxEncodedBytes();
   EncodedInfo info;
-  info.encoded_bytes = static_cast<size_t>(status);
+  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();
+
   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;
 }