Move audio frame memory handling inside AudioMixer.

webrtc/modules/audio_mixer/audio_mixer_impl.h

 /*
  *  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 11 matching lines @@
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/voice_engine_configurations.h"
 
 namespace webrtc {
 
 typedef std::vector<AudioFrame*> AudioFrameList;
 
 class AudioMixerImpl : public AudioMixer {
  public:
-  struct SourceStatus {
-    SourceStatus(Source* audio_source, bool is_mixed, float gain)
-        : audio_source(audio_source), is_mixed(is_mixed), gain(gain) {}
+  struct SourceStatusWithFrame {
+    SourceStatusWithFrame(Source* audio_source, bool is_mixed, float gain)
+        : audio_source(audio_source),
+          is_mixed(is_mixed),
+          gain(gain),
+          audio_frame() {}
     Source* audio_source = nullptr;
     bool is_mixed = false;
     float gain = 0.0f;
+
+    // A frame that will be passed to audio_source->GetAudioFrameWithInfo.
+    AudioFrame audio_frame{};
   };
 
-  typedef std::vector<SourceStatus> SourceStatusList;
+  typedef std::vector<std::unique_ptr<SourceStatusWithFrame>>
+      SourceStatusWithFrameList;
 
   // AudioProcessing only accepts 10 ms frames.
   static const int kFrameDurationInMs = 10;
   static const int kMaximumAmountOfMixedAudioSources = 3;
   static const int kDefaultFrequency = 48000;
 
   static rtc::scoped_refptr<AudioMixerImpl> Create();
 
   ~AudioMixerImpl() override;
 
@@ skipping 19 matching lines @@
   int OutputFrequency() const;
 
   // Compute what audio sources to mix from audio_source_list_. Ramp
   // in and out. Update mixed status. Mixes up to
   // kMaximumAmountOfMixedAudioSources audio sources.
   AudioFrameList GetAudioFromSources() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   // Add/remove the MixerAudioSource to the specified
   // MixerAudioSource list.
   bool AddAudioSourceToList(Source* audio_source,
-                            SourceStatusList* audio_source_list) const;
-  bool RemoveAudioSourceFromList(Source* remove_audio_source,
-                                 SourceStatusList* audio_source_list) const;
+                            SourceStatusWithFrameList* audio_source_list) const;
+  bool RemoveAudioSourceFromList(
+      Source* remove_audio_source,
+      SourceStatusWithFrameList* audio_source_list) const;
 
   bool LimitMixedAudio(AudioFrame* mixed_audio) const;
 
 
   rtc::CriticalSection crit_;
 
   // The current sample frequency and sample size when mixing.
   int output_frequency_ ACCESS_ON(&thread_checker_);
   size_t sample_size_ ACCESS_ON(&thread_checker_);
 
   // List of all audio sources. Note all lists are disjunct
-  SourceStatusList audio_source_list_ GUARDED_BY(crit_);  // May be mixed.
+  SourceStatusWithFrameList audio_source_list_
+      GUARDED_BY(crit_);  // May be mixed.
 
   // Determines if we will use a limiter for clipping protection during
   // mixing.
   bool use_limiter_ ACCESS_ON(&thread_checker_);
 
   uint32_t time_stamp_ ACCESS_ON(&thread_checker_);
 
   // Ensures that Mix is called from the same thread.
   rtc::ThreadChecker thread_checker_;
 
   // Used for inhibiting saturation in mixing.
   std::unique_ptr<AudioProcessing> limiter_ ACCESS_ON(&thread_checker_);
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioMixerImpl);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_