Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/neteq_impl.h

Index: webrtc/modules/audio_coding/neteq/neteq_impl.h
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 78c678c00b82e59b16237eca6b0473253a1edabc..4575864c0d8655bbc368f57a0fad713e671de43f 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -96,19 +96,7 @@ class NetEqImpl : public webrtc::NetEq {
   int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                        uint32_t receive_timestamp) override;
 
-  // Instructs NetEq to deliver 10 ms of audio data. The data is written to
-  // |output_audio|, which can hold (at least) |max_length| elements.
-  // The number of channels that were written to the output is provided in
-  // the output variable |num_channels|, and each channel contains
-  // |samples_per_channel| elements. If more than one channel is written,
-  // the samples are interleaved.
-  // The speech type is written to |type|, if |type| is not NULL.
-  // Returns kOK on success, or kFail in case of an error.
-  int GetAudio(size_t max_length,
-               int16_t* output_audio,
-               size_t* samples_per_channel,
-               size_t* num_channels,
-               NetEqOutputType* type) override;
+  int GetAudio(AudioFrame* audio_frame, NetEqOutputType* type) override;
 
   int RegisterPayloadType(NetEqDecoder codec,
                           const std::string& codec_name,
@@ -211,16 +199,9 @@ class NetEqImpl : public webrtc::NetEq {
                            bool is_sync_packet)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
-  // Delivers 10 ms of audio data. The data is written to |output|, which can
-  // hold (at least) |max_length| elements. The number of channels that were
-  // written to the output is provided in the output variable |num_channels|,
-  // and each channel contains |samples_per_channel| elements. If more than one
-  // channel is written, the samples are interleaved.
+  // Delivers 10 ms of audio data. The data is written to |audio_frame|.
   // Returns 0 on success, otherwise an error code.
-  int GetAudioInternal(size_t max_length,
-                       int16_t* output,
-                       size_t* samples_per_channel,
-                       size_t* num_channels)
+  int GetAudioInternal(AudioFrame* audio_frame)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Provides a decision to the GetAudioInternal method. The decision what to