Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal()

Old review at:
https://webrtc-codereview.appspot.com/43839004/

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45769004

Cr-Commit-Position: refs/heads/master@{#8788}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8788 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 13 insertions, 13 deletions

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 ae08423db8..941e635b17 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -183,19 +183,18 @@ void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
   }
 }
 
-void AudioEncoderOpus::EncodeInternal(uint32_t rtp_timestamp,
-                                      const int16_t* audio,
-                                      size_t max_encoded_bytes,
-                                      uint8_t* encoded,
-                                      EncodedInfo* info) {
+AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
+    uint32_t rtp_timestamp,
+    const int16_t* audio,
+    size_t max_encoded_bytes,
+    uint8_t* encoded) {
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
   input_buffer_.insert(input_buffer_.end(), audio,
                        audio + samples_per_10ms_frame_);
   if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) *
                               samples_per_10ms_frame_)) {
-    info->encoded_bytes = 0;
-    return;
+    return EncodedInfo();
   }
   CHECK_EQ(input_buffer_.size(),
            static_cast<size_t>(num_10ms_frames_per_packet_) *
@@ -207,12 +206,13 @@ void AudioEncoderOpus::EncodeInternal(uint32_t rtp_timestamp,
       ClampInt16(max_encoded_bytes), encoded);
   CHECK_GE(r, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
-  info->encoded_bytes = r;
-  info->encoded_timestamp = first_timestamp_in_buffer_;
-  info->payload_type = payload_type_;
-  // Allows Opus to send empty packets.
-  info->send_even_if_empty = true;
-  info->speech = r > 0;
+  EncodedInfo info;
+  info.encoded_bytes = r;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.send_even_if_empty = true;  // Allows Opus to send empty packets.
+  info.speech = r > 0;
+  return info;
 }
 
 }  // namespace webrtc