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

Index: webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index ddd6dde31c9e3594346fc08f82a0ba6f63c6d087..61f7920d81b8c093cafabe1a758c0a95dcb1f585 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -92,16 +92,13 @@ int AudioEncoderIlbc::GetTargetBitrate() const {
 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_);
 
@@ -114,15 +111,24 @@ AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
   // 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 (r >= 0) ? static_cast<size_t>(r) : 0;

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

You just CHECKed that r >= 0; no need to take care of the negative case.

Alternative: just return rtc::checked_cast<size_t>(r);

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

Acknowledged.

+          });
+
+  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;