Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(498)

Unified Diff: webrtc/modules/audio_mixer/source/audio_conference_mixer_impl.h

Issue 2104363003: A simple copy of the old mixer to a new directory. I also plan to run 'git cl format'. In another C… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Run 'git cl format' on the mixer Created 4 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/modules/audio_mixer/source/audio_conference_mixer_impl.h
diff --git a/webrtc/modules/audio_mixer/source/audio_conference_mixer_impl.h b/webrtc/modules/audio_mixer/source/audio_conference_mixer_impl.h
new file mode 100644
index 0000000000000000000000000000000000000000..97e1393ffc0c65f2013f1ccadb32f9800effbd7d
--- /dev/null
+++ b/webrtc/modules/audio_mixer/source/audio_conference_mixer_impl.h
@@ -0,0 +1,188 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
+
+#include <list>
+#include <map>
+#include <memory>
+
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
+#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+class AudioProcessing;
+class CriticalSectionWrapper;
+
+struct FrameAndMuteInfo {
+ FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
+ AudioFrame* frame;
+ bool muted;
+};
+
+typedef std::list<FrameAndMuteInfo> AudioFrameList;
+typedef std::list<MixerParticipant*> MixerParticipantList;
+
+// Cheshire cat implementation of MixerParticipant's non virtual functions.
+class MixHistory {
+ public:
+ MixHistory();
+ ~MixHistory();
+
+ // Returns true if the participant is being mixed.
+ bool IsMixed() const;
+
+ // Returns true if the participant was mixed previous mix
+ // iteration.
+ bool WasMixed() const;
+
+ // Updates the mixed status.
+ int32_t SetIsMixed(bool mixed);
+
+ void ResetMixedStatus();
+
+ private:
+ bool _isMixed;
+};
+
+class AudioConferenceMixerImpl : public AudioConferenceMixer {
+ public:
+ // AudioProcessing only accepts 10 ms frames.
+ enum { kProcessPeriodicityInMs = 10 };
+
+ AudioConferenceMixerImpl(int id);
+ ~AudioConferenceMixerImpl();
+
+ // Must be called after ctor.
+ bool Init();
+
+ // Module functions
+ int64_t TimeUntilNextProcess() override;
+ void Process() override;
+
+ // AudioConferenceMixer functions
+ int32_t RegisterMixedStreamCallback(
+ AudioMixerOutputReceiver* mixReceiver) override;
+ int32_t UnRegisterMixedStreamCallback() override;
+ int32_t SetMixabilityStatus(MixerParticipant* participant,
+ bool mixable) override;
+ bool MixabilityStatus(const MixerParticipant& participant) const override;
+ int32_t SetMinimumMixingFrequency(Frequency freq) override;
+ int32_t SetAnonymousMixabilityStatus(MixerParticipant* participant,
+ bool mixable) override;
+ bool AnonymousMixabilityStatus(
+ const MixerParticipant& participant) const override;
+
+ private:
+ enum { DEFAULT_AUDIO_FRAME_POOLSIZE = 50 };
+
+ // Set/get mix frequency
+ int32_t SetOutputFrequency(const Frequency& frequency);
+ Frequency OutputFrequency() const;
+
+ // Fills mixList with the AudioFrames pointers that should be used when
+ // mixing.
+ // maxAudioFrameCounter both input and output specifies how many more
+ // AudioFrames that are allowed to be mixed.
+ // rampOutList contain AudioFrames corresponding to an audio stream that
+ // used to be mixed but shouldn't be mixed any longer. These AudioFrames
+ // should be ramped out over this AudioFrame to avoid audio discontinuities.
+ void UpdateToMix(AudioFrameList* mixList,
+ AudioFrameList* rampOutList,
+ std::map<int, MixerParticipant*>* mixParticipantList,
+ size_t* maxAudioFrameCounter) const;
+
+ // Return the lowest mixing frequency that can be used without having to
+ // downsample any audio.
+ int32_t GetLowestMixingFrequency() const;
+ int32_t GetLowestMixingFrequencyFromList(
+ const MixerParticipantList& mixList) const;
+
+ // Return the AudioFrames that should be mixed anonymously.
+ void GetAdditionalAudio(AudioFrameList* additionalFramesList) const;
+
+ // Update the MixHistory of all MixerParticipants. mixedParticipantsList
+ // should contain a map of MixerParticipants that have been mixed.
+ void UpdateMixedStatus(
+ const std::map<int, MixerParticipant*>& mixedParticipantsList) const;
+
+ // Clears audioFrameList and reclaims all memory associated with it.
+ void ClearAudioFrameList(AudioFrameList* audioFrameList) const;
+
+ // This function returns true if it finds the MixerParticipant in the
+ // specified list of MixerParticipants.
+ bool IsParticipantInList(const MixerParticipant& participant,
+ const MixerParticipantList& participantList) const;
+
+ // Add/remove the MixerParticipant to the specified
+ // MixerParticipant list.
+ bool AddParticipantToList(MixerParticipant* participant,
+ MixerParticipantList* participantList) const;
+ bool RemoveParticipantFromList(MixerParticipant* removeParticipant,
+ MixerParticipantList* participantList) const;
+
+ // Mix the AudioFrames stored in audioFrameList into mixedAudio.
+ int32_t MixFromList(AudioFrame* mixedAudio,
+ const AudioFrameList& audioFrameList) const;
+
+ // Mix the AudioFrames stored in audioFrameList into mixedAudio. No
+ // record will be kept of this mix (e.g. the corresponding MixerParticipants
+ // will not be marked as IsMixed()
+ int32_t MixAnonomouslyFromList(AudioFrame* mixedAudio,
+ const AudioFrameList& audioFrameList) const;
+
+ bool LimitMixedAudio(AudioFrame* mixedAudio) const;
+
+ std::unique_ptr<CriticalSectionWrapper> _crit;
+ std::unique_ptr<CriticalSectionWrapper> _cbCrit;
+
+ int32_t _id;
+
+ Frequency _minimumMixingFreq;
+
+ // Mix result callback
+ AudioMixerOutputReceiver* _mixReceiver;
+
+ // The current sample frequency and sample size when mixing.
+ Frequency _outputFrequency;
+ size_t _sampleSize;
+
+ // Memory pool to avoid allocating/deallocating AudioFrames
+ MemoryPool<AudioFrame>* _audioFramePool;
+
+ // List of all participants. Note all lists are disjunct
+ MixerParticipantList _participantList; // May be mixed.
+ // Always mixed, anonomously.
+ MixerParticipantList _additionalParticipantList;
+
+ size_t _numMixedParticipants;
+ // Determines if we will use a limiter for clipping protection during
+ // mixing.
+ bool use_limiter_;
+
+ uint32_t _timeStamp;
+
+ // Metronome class.
+ TimeScheduler _timeScheduler;
+
+ // Counter keeping track of concurrent calls to process.
+ // Note: should never be higher than 1 or lower than 0.
+ int16_t _processCalls;
+
+ // Used for inhibiting saturation in mixing.
+ std::unique_ptr<AudioProcessing> _limiter;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_

Powered by Google App Engine
This is Rietveld 408576698