Move audio frame memory handling inside AudioMixer.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 694 matching lines @@
         capture_start_ntp_time_ms_ =
             audioFrame->ntp_time_ms_ - audioFrame->elapsed_time_ms_;
       }
     }
   }
 
   return muted ? MixerParticipant::AudioFrameInfo::kMuted
                : MixerParticipant::AudioFrameInfo::kNormal;
 }
 
-AudioMixer::Source::AudioFrameWithInfo Channel::GetAudioFrameWithInfo(
-    int sample_rate_hz) {
-  mix_audio_frame_.sample_rate_hz_ = sample_rate_hz;
+AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
+    int sample_rate_hz,
+    AudioFrame* audio_frame) {
+  audio_frame->sample_rate_hz_ = sample_rate_hz;
 
-  const auto frame_info = GetAudioFrameWithMuted(-1, &mix_audio_frame_);
+  const auto frame_info = GetAudioFrameWithMuted(-1, audio_frame);
 
   using FrameInfo = AudioMixer::Source::AudioFrameInfo;
   FrameInfo new_audio_frame_info = FrameInfo::kError;
   switch (frame_info) {
     case MixerParticipant::AudioFrameInfo::kNormal:
       new_audio_frame_info = FrameInfo::kNormal;
       break;
     case MixerParticipant::AudioFrameInfo::kMuted:
       new_audio_frame_info = FrameInfo::kMuted;
       break;
     case MixerParticipant::AudioFrameInfo::kError:
       new_audio_frame_info = FrameInfo::kError;
       break;
   }
-  return {&mix_audio_frame_, new_audio_frame_info};
+  return new_audio_frame_info;
 }
 
 int32_t Channel::NeededFrequency(int32_t id) const {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::NeededFrequency(id=%d)", id);
 
   int highestNeeded = 0;
 
   // Determine highest needed receive frequency
   int32_t receiveFrequency = audio_coding_->ReceiveFrequency();
@@ skipping 2521 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc