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

webrtc/modules/audio_coding/codecs/audio_encoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index e99fc30995995dea55ba1bcae4347b784e4691f3..766a3035b78a947952280b74fad033d86a4ef4fd 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -26,17 +26,67 @@ int AudioEncoder::RtpTimestampRateHz() const {
 AudioEncoder::EncodedInfo AudioEncoder::Encode(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
+  RTC_CHECK_EQ(audio.size(),
+               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
+
+  const size_t old_size = encoded->size();
+  EncodedInfo info = EncodeInternal(rtp_timestamp, audio, encoded);
+  RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
+  return info;
+}
+
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    size_t max_encoded_bytes,
+    uint8_t* encoded) {
+  return DEPRECATED_Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
+}
+
+AudioEncoder::EncodedInfo AudioEncoder::DEPRECATED_Encode(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
   RTC_CHECK_EQ(audio.size(),
                static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
   EncodedInfo info =
-      EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
+      EncodeInternal(rtp_timestamp, audio,

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

What was wrong with the old formatting?

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

I ... don't know. :(
Chances are it got wrecked while I was doing things and I missed putting it
back.
Will do!

+                     max_encoded_bytes, encoded);
   RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
   return info;
 }
 
+AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded)
+{
+  EncodedInfo info;
+  encoded->AppendData(MaxEncodedBytes(), [&] (rtc::ArrayView<uint8_t> encoded) {
+      info = EncodeInternal(rtp_timestamp, audio,
+                            encoded.size(), encoded.data());
+      return info.encoded_bytes;
+    });
+  return info;
+}
+
+AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    size_t max_encoded_bytes,
+    uint8_t* encoded)
+{
+  rtc::Buffer temp_buffer;
+  EncodedInfo info = EncodeInternal(rtp_timestamp, audio, &temp_buffer);
+  RTC_DCHECK_LE(temp_buffer.size(), max_encoded_bytes);
+  std::memcpy(encoded, temp_buffer.data(), info.encoded_bytes);
+  return info;
+}
+
 bool AudioEncoder::SetFec(bool enable) {
   return !enable;
 }