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

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h

Index: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index 415511705acc4665760767bbeb32c673b5db3f73..0b8baab2093d544d8be33da5244e76ac17938274 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -20,6 +20,8 @@ namespace webrtc {
 
 class AudioEncoderPcm : public AudioEncoder {
  public:
+  using AudioEncoder::EncodeInternal;
+
   struct Config {
    public:
     bool IsOk() const;
@@ -41,15 +43,15 @@ class AudioEncoderPcm : public AudioEncoder {
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
-  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             rtc::ArrayView<const int16_t> audio,
-                             size_t max_encoded_bytes,
-                             uint8_t* encoded) override;
   void Reset() override;
 
  protected:
   AudioEncoderPcm(const Config& config, int sample_rate_hz);
 
+  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
+                             rtc::ArrayView<const int16_t> audio,
+                             rtc::Buffer* encoded) override;
+
   virtual size_t EncodeCall(const int16_t* audio,
                             size_t input_len,
                             uint8_t* encoded) = 0;