1597 lines
69 KiB
C++
1597 lines
69 KiB
C++
/*
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**
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** Copyright 2007, The Android Open Source Project
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**
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** Licensed under the Apache License, Version 2.0 (the "License");
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** you may not use this file except in compliance with the License.
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** You may obtain a copy of the License at
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**
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** http://www.apache.org/licenses/LICENSE-2.0
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**
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** Unless required by applicable law or agreed to in writing, software
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** distributed under the License is distributed on an "AS IS" BASIS,
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** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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** See the License for the specific language governing permissions and
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** limitations under the License.
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*/
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#ifndef ANDROID_AUDIO_FLINGER_H
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#define ANDROID_AUDIO_FLINGER_H
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#include <stdint.h>
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#include <sys/types.h>
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#include <limits.h>
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#include <common_time/cc_helper.h>
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#include <media/IAudioFlinger.h>
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#include <media/IAudioFlingerClient.h>
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#include <media/IAudioTrack.h>
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#include <media/IAudioRecord.h>
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#include <media/AudioSystem.h>
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#include <media/AudioTrack.h>
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#include <utils/Atomic.h>
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#include <utils/Errors.h>
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#include <utils/threads.h>
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#include <utils/SortedVector.h>
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#include <utils/TypeHelpers.h>
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#include <utils/Vector.h>
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#include <binder/BinderService.h>
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#include <binder/MemoryDealer.h>
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#include <system/audio.h>
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#include <hardware/audio.h>
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#include "AudioBufferProvider.h"
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#include <powermanager/IPowerManager.h>
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namespace android {
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class audio_track_cblk_t;
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class effect_param_cblk_t;
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class AudioMixer;
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class AudioBuffer;
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class AudioResampler;
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// ----------------------------------------------------------------------------
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static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
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class AudioFlinger :
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public BinderService<AudioFlinger>,
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public BnAudioFlinger
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{
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friend class BinderService<AudioFlinger>;
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public:
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static const char* getServiceName() { return "media.audio_flinger"; }
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virtual status_t dump(int fd, const Vector<String16>& args);
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// IAudioFlinger interface, in binder opcode order
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virtual sp<IAudioTrack> createTrack(
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pid_t pid,
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audio_stream_type_t streamType,
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uint32_t sampleRate,
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audio_format_t format,
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uint32_t channelMask,
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int frameCount,
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uint32_t flags,
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const sp<IMemory>& sharedBuffer,
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audio_io_handle_t output,
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bool isTimed,
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int *sessionId,
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status_t *status);
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virtual sp<IAudioRecord> openRecord(
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pid_t pid,
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audio_io_handle_t input,
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uint32_t sampleRate,
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audio_format_t format,
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uint32_t channelMask,
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int frameCount,
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uint32_t flags,
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int *sessionId,
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status_t *status);
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virtual uint32_t sampleRate(audio_io_handle_t output) const;
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virtual int channelCount(audio_io_handle_t output) const;
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virtual audio_format_t format(audio_io_handle_t output) const;
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virtual size_t frameCount(audio_io_handle_t output) const;
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virtual uint32_t latency(audio_io_handle_t output) const;
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virtual status_t setMasterVolume(float value);
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virtual status_t setMasterMute(bool muted);
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virtual float masterVolume() const;
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virtual float masterVolumeSW() const;
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virtual bool masterMute() const;
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virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
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audio_io_handle_t output);
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virtual status_t setStreamMute(audio_stream_type_t stream, bool muted);
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virtual float streamVolume(audio_stream_type_t stream,
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audio_io_handle_t output) const;
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virtual bool streamMute(audio_stream_type_t stream) const;
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virtual status_t setMode(audio_mode_t mode);
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virtual status_t setMicMute(bool state);
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virtual bool getMicMute() const;
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virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
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virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
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virtual void registerClient(const sp<IAudioFlingerClient>& client);
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virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const;
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virtual audio_io_handle_t openOutput(uint32_t *pDevices,
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uint32_t *pSamplingRate,
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audio_format_t *pFormat,
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uint32_t *pChannels,
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uint32_t *pLatencyMs,
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uint32_t flags);
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virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
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audio_io_handle_t output2);
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virtual status_t closeOutput(audio_io_handle_t output);
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virtual status_t suspendOutput(audio_io_handle_t output);
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virtual status_t restoreOutput(audio_io_handle_t output);
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virtual audio_io_handle_t openInput(uint32_t *pDevices,
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uint32_t *pSamplingRate,
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audio_format_t *pFormat,
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uint32_t *pChannels,
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audio_in_acoustics_t acoustics);
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virtual status_t closeInput(audio_io_handle_t input);
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virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output);
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virtual status_t setVoiceVolume(float volume);
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virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
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audio_io_handle_t output) const;
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virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
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virtual int newAudioSessionId();
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virtual void acquireAudioSessionId(int audioSession);
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virtual void releaseAudioSessionId(int audioSession);
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virtual status_t queryNumberEffects(uint32_t *numEffects) const;
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virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
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virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
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effect_descriptor_t *descriptor) const;
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virtual sp<IEffect> createEffect(pid_t pid,
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effect_descriptor_t *pDesc,
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const sp<IEffectClient>& effectClient,
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int32_t priority,
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audio_io_handle_t io,
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int sessionId,
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status_t *status,
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int *id,
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int *enabled);
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virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
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audio_io_handle_t dstOutput);
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virtual status_t onTransact(
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uint32_t code,
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const Parcel& data,
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Parcel* reply,
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uint32_t flags);
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// end of IAudioFlinger interface
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private:
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audio_mode_t getMode() const { return mMode; }
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bool btNrecIsOff() const { return mBtNrecIsOff; }
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AudioFlinger();
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virtual ~AudioFlinger();
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// call in any IAudioFlinger method that accesses mPrimaryHardwareDev
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status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; }
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virtual void onFirstRef();
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audio_hw_device_t* findSuitableHwDev_l(uint32_t devices);
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void purgeStaleEffects_l();
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static nsecs_t mStandbyTimeInNsecs;
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// Internal dump utilites.
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status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
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status_t dumpClients(int fd, const Vector<String16>& args);
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status_t dumpInternals(int fd, const Vector<String16>& args);
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// --- Client ---
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class Client : public RefBase {
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public:
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Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
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virtual ~Client();
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sp<MemoryDealer> heap() const;
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pid_t pid() const { return mPid; }
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sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
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bool reserveTimedTrack();
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void releaseTimedTrack();
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private:
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Client(const Client&);
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Client& operator = (const Client&);
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const sp<AudioFlinger> mAudioFlinger;
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const sp<MemoryDealer> mMemoryDealer;
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const pid_t mPid;
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Mutex mTimedTrackLock;
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int mTimedTrackCount;
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};
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// --- Notification Client ---
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class NotificationClient : public IBinder::DeathRecipient {
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public:
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NotificationClient(const sp<AudioFlinger>& audioFlinger,
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const sp<IAudioFlingerClient>& client,
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pid_t pid);
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virtual ~NotificationClient();
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sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
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// IBinder::DeathRecipient
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virtual void binderDied(const wp<IBinder>& who);
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private:
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NotificationClient(const NotificationClient&);
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NotificationClient& operator = (const NotificationClient&);
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const sp<AudioFlinger> mAudioFlinger;
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const pid_t mPid;
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const sp<IAudioFlingerClient> mAudioFlingerClient;
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};
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class TrackHandle;
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class RecordHandle;
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class RecordThread;
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class PlaybackThread;
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class MixerThread;
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class DirectOutputThread;
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class DuplicatingThread;
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class Track;
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class RecordTrack;
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class EffectModule;
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class EffectHandle;
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class EffectChain;
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struct AudioStreamOut;
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struct AudioStreamIn;
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class ThreadBase : public Thread {
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public:
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enum type_t {
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MIXER, // Thread class is MixerThread
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DIRECT, // Thread class is DirectOutputThread
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DUPLICATING, // Thread class is DuplicatingThread
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RECORD // Thread class is RecordThread
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};
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ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type);
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virtual ~ThreadBase();
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status_t dumpBase(int fd, const Vector<String16>& args);
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status_t dumpEffectChains(int fd, const Vector<String16>& args);
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void clearPowerManager();
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// base for record and playback
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class TrackBase : public AudioBufferProvider, public RefBase {
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public:
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enum track_state {
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IDLE,
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TERMINATED,
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// These are order-sensitive; do not change order without reviewing the impact.
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// In particular there are assumptions about > STOPPED.
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STOPPED,
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RESUMING,
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ACTIVE,
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PAUSING,
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PAUSED
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};
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TrackBase(ThreadBase *thread,
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const sp<Client>& client,
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uint32_t sampleRate,
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audio_format_t format,
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uint32_t channelMask,
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int frameCount,
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const sp<IMemory>& sharedBuffer,
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int sessionId);
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virtual ~TrackBase();
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virtual status_t start(pid_t tid) = 0;
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virtual void stop() = 0;
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sp<IMemory> getCblk() const { return mCblkMemory; }
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audio_track_cblk_t* cblk() const { return mCblk; }
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int sessionId() const { return mSessionId; }
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protected:
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friend class ThreadBase;
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friend class RecordHandle;
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friend class PlaybackThread;
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friend class RecordThread;
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friend class MixerThread;
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friend class DirectOutputThread;
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TrackBase(const TrackBase&);
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TrackBase& operator = (const TrackBase&);
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// AudioBufferProvider interface
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virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0;
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virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
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audio_format_t format() const {
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return mFormat;
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}
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int channelCount() const { return mChannelCount; }
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uint32_t channelMask() const { return mChannelMask; }
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int sampleRate() const; // FIXME inline after cblk sr moved
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void* getBuffer(uint32_t offset, uint32_t frames) const;
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bool isStopped() const {
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return mState == STOPPED;
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}
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bool isTerminated() const {
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return mState == TERMINATED;
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}
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bool step();
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void reset();
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const wp<ThreadBase> mThread;
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/*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const
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sp<IMemory> mCblkMemory;
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audio_track_cblk_t* mCblk;
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void* mBuffer;
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void* mBufferEnd;
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uint32_t mFrameCount;
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// we don't really need a lock for these
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track_state mState;
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const audio_format_t mFormat;
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bool mStepServerFailed;
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const int mSessionId;
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uint8_t mChannelCount;
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uint32_t mChannelMask;
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};
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class ConfigEvent {
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public:
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ConfigEvent() : mEvent(0), mParam(0) {}
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int mEvent;
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int mParam;
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};
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class PMDeathRecipient : public IBinder::DeathRecipient {
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public:
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PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
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virtual ~PMDeathRecipient() {}
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// IBinder::DeathRecipient
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virtual void binderDied(const wp<IBinder>& who);
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private:
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PMDeathRecipient(const PMDeathRecipient&);
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PMDeathRecipient& operator = (const PMDeathRecipient&);
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wp<ThreadBase> mThread;
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};
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virtual status_t initCheck() const = 0;
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type_t type() const { return mType; }
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uint32_t sampleRate() const { return mSampleRate; }
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int channelCount() const { return mChannelCount; }
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audio_format_t format() const { return mFormat; }
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size_t frameCount() const { return mFrameCount; }
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void wakeUp() { mWaitWorkCV.broadcast(); }
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// Should be "virtual status_t requestExitAndWait()" and override same
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// method in Thread, but Thread::requestExitAndWait() is not yet virtual.
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void exit();
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virtual bool checkForNewParameters_l() = 0;
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virtual status_t setParameters(const String8& keyValuePairs);
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virtual String8 getParameters(const String8& keys) = 0;
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virtual void audioConfigChanged_l(int event, int param = 0) = 0;
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void sendConfigEvent(int event, int param = 0);
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void sendConfigEvent_l(int event, int param = 0);
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void processConfigEvents();
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audio_io_handle_t id() const { return mId;}
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bool standby() const { return mStandby; }
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uint32_t device() const { return mDevice; }
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virtual audio_stream_t* stream() = 0;
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sp<EffectHandle> createEffect_l(
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const sp<AudioFlinger::Client>& client,
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const sp<IEffectClient>& effectClient,
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int32_t priority,
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int sessionId,
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effect_descriptor_t *desc,
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int *enabled,
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status_t *status);
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void disconnectEffect(const sp< EffectModule>& effect,
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const wp<EffectHandle>& handle,
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bool unpinIfLast);
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// return values for hasAudioSession (bit field)
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enum effect_state {
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EFFECT_SESSION = 0x1, // the audio session corresponds to at least one
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// effect
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TRACK_SESSION = 0x2 // the audio session corresponds to at least one
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// track
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};
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// get effect chain corresponding to session Id.
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sp<EffectChain> getEffectChain(int sessionId);
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// same as getEffectChain() but must be called with ThreadBase mutex locked
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sp<EffectChain> getEffectChain_l(int sessionId);
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// add an effect chain to the chain list (mEffectChains)
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virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
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// remove an effect chain from the chain list (mEffectChains)
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virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0;
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// lock mall effect chains Mutexes. Must be called before releasing the
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// ThreadBase mutex before processing the mixer and effects. This guarantees the
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// integrity of the chains during the process.
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void lockEffectChains_l(Vector<sp <EffectChain> >& effectChains);
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// unlock effect chains after process
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void unlockEffectChains(const Vector<sp<EffectChain> >& effectChains);
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// set audio mode to all effect chains
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void setMode(audio_mode_t mode);
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// get effect module with corresponding ID on specified audio session
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sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
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// add and effect module. Also creates the effect chain is none exists for
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// the effects audio session
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status_t addEffect_l(const sp< EffectModule>& effect);
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// remove and effect module. Also removes the effect chain is this was the last
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// effect
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void removeEffect_l(const sp< EffectModule>& effect);
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// detach all tracks connected to an auxiliary effect
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virtual void detachAuxEffect_l(int effectId) {}
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// returns either EFFECT_SESSION if effects on this audio session exist in one
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// chain, or TRACK_SESSION if tracks on this audio session exist, or both
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virtual uint32_t hasAudioSession(int sessionId) = 0;
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// the value returned by default implementation is not important as the
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// strategy is only meaningful for PlaybackThread which implements this method
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virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
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// suspend or restore effect according to the type of effect passed. a NULL
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// type pointer means suspend all effects in the session
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void setEffectSuspended(const effect_uuid_t *type,
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bool suspend,
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int sessionId = AUDIO_SESSION_OUTPUT_MIX);
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// check if some effects must be suspended/restored when an effect is enabled
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// or disabled
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void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
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bool enabled,
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int sessionId = AUDIO_SESSION_OUTPUT_MIX);
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void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
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bool enabled,
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int sessionId = AUDIO_SESSION_OUTPUT_MIX);
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mutable Mutex mLock;
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protected:
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// entry describing an effect being suspended in mSuspendedSessions keyed vector
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class SuspendedSessionDesc : public RefBase {
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public:
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SuspendedSessionDesc() : mRefCount(0) {}
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int mRefCount; // number of active suspend requests
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effect_uuid_t mType; // effect type UUID
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};
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void acquireWakeLock();
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void acquireWakeLock_l();
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void releaseWakeLock();
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void releaseWakeLock_l();
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void setEffectSuspended_l(const effect_uuid_t *type,
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bool suspend,
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int sessionId = AUDIO_SESSION_OUTPUT_MIX);
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// updated mSuspendedSessions when an effect suspended or restored
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void updateSuspendedSessions_l(const effect_uuid_t *type,
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bool suspend,
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int sessionId);
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// check if some effects must be suspended when an effect chain is added
|
|
void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
|
|
|
|
friend class AudioFlinger;
|
|
friend class Track;
|
|
friend class TrackBase;
|
|
friend class PlaybackThread;
|
|
friend class MixerThread;
|
|
friend class DirectOutputThread;
|
|
friend class DuplicatingThread;
|
|
friend class RecordThread;
|
|
friend class RecordTrack;
|
|
|
|
const type_t mType;
|
|
Condition mWaitWorkCV;
|
|
const sp<AudioFlinger> mAudioFlinger;
|
|
uint32_t mSampleRate;
|
|
size_t mFrameCount;
|
|
uint32_t mChannelMask;
|
|
uint16_t mChannelCount;
|
|
size_t mFrameSize;
|
|
audio_format_t mFormat;
|
|
Condition mParamCond;
|
|
Vector<String8> mNewParameters;
|
|
status_t mParamStatus;
|
|
Vector<ConfigEvent> mConfigEvents;
|
|
bool mStandby;
|
|
const audio_io_handle_t mId;
|
|
Vector< sp<EffectChain> > mEffectChains;
|
|
uint32_t mDevice; // output device for PlaybackThread
|
|
// input + output devices for RecordThread
|
|
static const int kNameLength = 32;
|
|
char mName[kNameLength];
|
|
sp<IPowerManager> mPowerManager;
|
|
sp<IBinder> mWakeLockToken;
|
|
const sp<PMDeathRecipient> mDeathRecipient;
|
|
// list of suspended effects per session and per type. The first vector is
|
|
// keyed by session ID, the second by type UUID timeLow field
|
|
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions;
|
|
};
|
|
|
|
struct stream_type_t {
|
|
stream_type_t()
|
|
: volume(1.0f),
|
|
mute(false),
|
|
valid(true)
|
|
{
|
|
}
|
|
float volume;
|
|
bool mute;
|
|
bool valid;
|
|
};
|
|
|
|
// --- PlaybackThread ---
|
|
class PlaybackThread : public ThreadBase {
|
|
public:
|
|
|
|
enum mixer_state {
|
|
MIXER_IDLE,
|
|
MIXER_TRACKS_ENABLED,
|
|
MIXER_TRACKS_READY
|
|
};
|
|
|
|
// playback track
|
|
class Track : public TrackBase {
|
|
public:
|
|
Track( PlaybackThread *thread,
|
|
const sp<Client>& client,
|
|
audio_stream_type_t streamType,
|
|
uint32_t sampleRate,
|
|
audio_format_t format,
|
|
uint32_t channelMask,
|
|
int frameCount,
|
|
const sp<IMemory>& sharedBuffer,
|
|
int sessionId);
|
|
virtual ~Track();
|
|
|
|
void dump(char* buffer, size_t size);
|
|
virtual status_t start(pid_t tid);
|
|
virtual void stop();
|
|
void pause();
|
|
|
|
void flush();
|
|
void destroy();
|
|
void mute(bool);
|
|
int name() const {
|
|
return mName;
|
|
}
|
|
|
|
audio_stream_type_t streamType() const {
|
|
return mStreamType;
|
|
}
|
|
status_t attachAuxEffect(int EffectId);
|
|
void setAuxBuffer(int EffectId, int32_t *buffer);
|
|
int32_t *auxBuffer() const { return mAuxBuffer; }
|
|
void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
|
|
int16_t *mainBuffer() const { return mMainBuffer; }
|
|
int auxEffectId() const { return mAuxEffectId; }
|
|
|
|
protected:
|
|
friend class ThreadBase;
|
|
friend class TrackHandle;
|
|
friend class PlaybackThread;
|
|
friend class MixerThread;
|
|
friend class DirectOutputThread;
|
|
|
|
Track(const Track&);
|
|
Track& operator = (const Track&);
|
|
|
|
// AudioBufferProvider interface
|
|
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS);
|
|
// releaseBuffer() not overridden
|
|
|
|
virtual uint32_t framesReady() const;
|
|
|
|
bool isMuted() const { return mMute; }
|
|
bool isPausing() const {
|
|
return mState == PAUSING;
|
|
}
|
|
bool isPaused() const {
|
|
return mState == PAUSED;
|
|
}
|
|
bool isReady() const;
|
|
void setPaused() { mState = PAUSED; }
|
|
void reset();
|
|
|
|
bool isOutputTrack() const {
|
|
return (mStreamType == AUDIO_STREAM_CNT);
|
|
}
|
|
|
|
virtual bool isTimedTrack() const { return false; }
|
|
|
|
// we don't really need a lock for these
|
|
volatile bool mMute;
|
|
// FILLED state is used for suppressing volume ramp at begin of playing
|
|
enum {FS_FILLING, FS_FILLED, FS_ACTIVE};
|
|
mutable uint8_t mFillingUpStatus;
|
|
int8_t mRetryCount;
|
|
sp<IMemory> mSharedBuffer;
|
|
bool mResetDone;
|
|
audio_stream_type_t mStreamType;
|
|
int mName;
|
|
int16_t *mMainBuffer;
|
|
int32_t *mAuxBuffer;
|
|
int mAuxEffectId;
|
|
bool mHasVolumeController;
|
|
}; // end of Track
|
|
|
|
class TimedTrack : public Track {
|
|
public:
|
|
static sp<TimedTrack> create(PlaybackThread *thread,
|
|
const sp<Client>& client,
|
|
audio_stream_type_t streamType,
|
|
uint32_t sampleRate,
|
|
audio_format_t format,
|
|
uint32_t channelMask,
|
|
int frameCount,
|
|
const sp<IMemory>& sharedBuffer,
|
|
int sessionId);
|
|
~TimedTrack();
|
|
|
|
class TimedBuffer {
|
|
public:
|
|
TimedBuffer();
|
|
TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
|
|
const sp<IMemory>& buffer() const { return mBuffer; }
|
|
int64_t pts() const { return mPTS; }
|
|
int position() const { return mPosition; }
|
|
void setPosition(int pos) { mPosition = pos; }
|
|
private:
|
|
sp<IMemory> mBuffer;
|
|
int64_t mPTS;
|
|
int mPosition;
|
|
};
|
|
|
|
virtual bool isTimedTrack() const { return true; }
|
|
|
|
virtual uint32_t framesReady() const;
|
|
|
|
// AudioBufferProvider interface
|
|
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
|
|
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
|
|
|
|
void timedYieldSamples(AudioBufferProvider::Buffer* buffer);
|
|
void timedYieldSilence(uint32_t numFrames,
|
|
AudioBufferProvider::Buffer* buffer);
|
|
|
|
status_t allocateTimedBuffer(size_t size,
|
|
sp<IMemory>* buffer);
|
|
status_t queueTimedBuffer(const sp<IMemory>& buffer,
|
|
int64_t pts);
|
|
status_t setMediaTimeTransform(const LinearTransform& xform,
|
|
TimedAudioTrack::TargetTimeline target);
|
|
void trimTimedBufferQueue_l();
|
|
|
|
private:
|
|
TimedTrack(PlaybackThread *thread,
|
|
const sp<Client>& client,
|
|
audio_stream_type_t streamType,
|
|
uint32_t sampleRate,
|
|
audio_format_t format,
|
|
uint32_t channelMask,
|
|
int frameCount,
|
|
const sp<IMemory>& sharedBuffer,
|
|
int sessionId);
|
|
|
|
uint64_t mLocalTimeFreq;
|
|
LinearTransform mLocalTimeToSampleTransform;
|
|
sp<MemoryDealer> mTimedMemoryDealer;
|
|
Vector<TimedBuffer> mTimedBufferQueue;
|
|
uint8_t* mTimedSilenceBuffer;
|
|
uint32_t mTimedSilenceBufferSize;
|
|
mutable Mutex mTimedBufferQueueLock;
|
|
bool mTimedAudioOutputOnTime;
|
|
CCHelper mCCHelper;
|
|
|
|
Mutex mMediaTimeTransformLock;
|
|
LinearTransform mMediaTimeTransform;
|
|
bool mMediaTimeTransformValid;
|
|
TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
|
|
};
|
|
|
|
|
|
// playback track
|
|
class OutputTrack : public Track {
|
|
public:
|
|
|
|
class Buffer: public AudioBufferProvider::Buffer {
|
|
public:
|
|
int16_t *mBuffer;
|
|
};
|
|
|
|
OutputTrack(PlaybackThread *thread,
|
|
DuplicatingThread *sourceThread,
|
|
uint32_t sampleRate,
|
|
audio_format_t format,
|
|
uint32_t channelMask,
|
|
int frameCount);
|
|
virtual ~OutputTrack();
|
|
|
|
virtual status_t start(pid_t tid);
|
|
virtual void stop();
|
|
bool write(int16_t* data, uint32_t frames);
|
|
bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
|
|
bool isActive() const { return mActive; }
|
|
const wp<ThreadBase>& thread() const { return mThread; }
|
|
|
|
private:
|
|
|
|
enum {
|
|
NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value
|
|
};
|
|
|
|
status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
|
|
void clearBufferQueue();
|
|
|
|
// Maximum number of pending buffers allocated by OutputTrack::write()
|
|
static const uint8_t kMaxOverFlowBuffers = 10;
|
|
|
|
Vector < Buffer* > mBufferQueue;
|
|
AudioBufferProvider::Buffer mOutBuffer;
|
|
bool mActive;
|
|
DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
|
|
}; // end of OutputTrack
|
|
|
|
PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
|
|
audio_io_handle_t id, uint32_t device, type_t type);
|
|
virtual ~PlaybackThread();
|
|
|
|
virtual status_t dump(int fd, const Vector<String16>& args);
|
|
|
|
// Thread virtuals
|
|
virtual status_t readyToRun();
|
|
virtual void onFirstRef();
|
|
|
|
virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
|
|
|
|
virtual uint32_t latency() const;
|
|
|
|
void setMasterVolume(float value);
|
|
void setMasterMute(bool muted);
|
|
|
|
void setStreamVolume(audio_stream_type_t stream, float value);
|
|
void setStreamMute(audio_stream_type_t stream, bool muted);
|
|
|
|
float streamVolume(audio_stream_type_t stream) const;
|
|
|
|
sp<Track> createTrack_l(
|
|
const sp<AudioFlinger::Client>& client,
|
|
audio_stream_type_t streamType,
|
|
uint32_t sampleRate,
|
|
audio_format_t format,
|
|
uint32_t channelMask,
|
|
int frameCount,
|
|
const sp<IMemory>& sharedBuffer,
|
|
int sessionId,
|
|
bool isTimed,
|
|
status_t *status);
|
|
|
|
AudioStreamOut* getOutput() const;
|
|
AudioStreamOut* clearOutput();
|
|
virtual audio_stream_t* stream();
|
|
|
|
void suspend() { mSuspended++; }
|
|
void restore() { if (mSuspended) mSuspended--; }
|
|
bool isSuspended() const { return (mSuspended != 0); }
|
|
virtual String8 getParameters(const String8& keys);
|
|
virtual void audioConfigChanged_l(int event, int param = 0);
|
|
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
|
|
int16_t *mixBuffer() const { return mMixBuffer; };
|
|
|
|
virtual void detachAuxEffect_l(int effectId);
|
|
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
|
|
int EffectId);
|
|
status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track,
|
|
int EffectId);
|
|
|
|
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
|
|
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
|
|
virtual uint32_t hasAudioSession(int sessionId);
|
|
virtual uint32_t getStrategyForSession_l(int sessionId);
|
|
|
|
void setStreamValid(audio_stream_type_t streamType, bool valid);
|
|
|
|
protected:
|
|
int16_t* mMixBuffer;
|
|
int mSuspended;
|
|
int mBytesWritten;
|
|
private:
|
|
// mMasterMute is in both PlaybackThread and in AudioFlinger. When a
|
|
// PlaybackThread needs to find out if master-muted, it checks it's local
|
|
// copy rather than the one in AudioFlinger. This optimization saves a lock.
|
|
bool mMasterMute;
|
|
void setMasterMute_l(bool muted) { mMasterMute = muted; }
|
|
protected:
|
|
SortedVector< wp<Track> > mActiveTracks;
|
|
|
|
virtual int getTrackName_l() = 0;
|
|
virtual void deleteTrackName_l(int name) = 0;
|
|
virtual uint32_t activeSleepTimeUs();
|
|
virtual uint32_t idleSleepTimeUs() = 0;
|
|
virtual uint32_t suspendSleepTimeUs() = 0;
|
|
|
|
// Code snippets that are temporarily lifted up out of threadLoop() until the merge
|
|
void checkSilentMode_l();
|
|
|
|
private:
|
|
|
|
friend class AudioFlinger;
|
|
friend class OutputTrack;
|
|
friend class Track;
|
|
friend class TrackBase;
|
|
friend class MixerThread;
|
|
friend class DirectOutputThread;
|
|
friend class DuplicatingThread;
|
|
|
|
PlaybackThread(const Client&);
|
|
PlaybackThread& operator = (const PlaybackThread&);
|
|
|
|
status_t addTrack_l(const sp<Track>& track);
|
|
void destroyTrack_l(const sp<Track>& track);
|
|
void removeTrack_l(const sp<Track>& track);
|
|
|
|
void readOutputParameters();
|
|
|
|
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
|
|
status_t dumpTracks(int fd, const Vector<String16>& args);
|
|
|
|
SortedVector< sp<Track> > mTracks;
|
|
// mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread
|
|
stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1];
|
|
AudioStreamOut *mOutput;
|
|
float mMasterVolume;
|
|
nsecs_t mLastWriteTime;
|
|
int mNumWrites;
|
|
int mNumDelayedWrites;
|
|
bool mInWrite;
|
|
};
|
|
|
|
class MixerThread : public PlaybackThread {
|
|
public:
|
|
MixerThread (const sp<AudioFlinger>& audioFlinger,
|
|
AudioStreamOut* output,
|
|
audio_io_handle_t id,
|
|
uint32_t device,
|
|
type_t type = MIXER);
|
|
virtual ~MixerThread();
|
|
|
|
// Thread virtuals
|
|
virtual bool threadLoop();
|
|
|
|
void invalidateTracks(audio_stream_type_t streamType);
|
|
virtual bool checkForNewParameters_l();
|
|
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
|
|
|
|
protected:
|
|
mixer_state prepareTracks_l(const SortedVector< wp<Track> >& activeTracks,
|
|
Vector< sp<Track> > *tracksToRemove);
|
|
virtual int getTrackName_l();
|
|
virtual void deleteTrackName_l(int name);
|
|
virtual uint32_t idleSleepTimeUs();
|
|
virtual uint32_t suspendSleepTimeUs();
|
|
|
|
AudioMixer* mAudioMixer;
|
|
mixer_state mPrevMixerStatus; // previous status returned by prepareTracks_l()
|
|
};
|
|
|
|
class DirectOutputThread : public PlaybackThread {
|
|
public:
|
|
|
|
DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
|
|
audio_io_handle_t id, uint32_t device);
|
|
virtual ~DirectOutputThread();
|
|
|
|
// Thread virtuals
|
|
virtual bool threadLoop();
|
|
|
|
virtual bool checkForNewParameters_l();
|
|
|
|
protected:
|
|
virtual int getTrackName_l();
|
|
virtual void deleteTrackName_l(int name);
|
|
virtual uint32_t activeSleepTimeUs();
|
|
virtual uint32_t idleSleepTimeUs();
|
|
virtual uint32_t suspendSleepTimeUs();
|
|
|
|
private:
|
|
void applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp);
|
|
|
|
// volumes last sent to audio HAL with stream->set_volume()
|
|
// FIXME use standard representation and names
|
|
float mLeftVolFloat;
|
|
float mRightVolFloat;
|
|
uint16_t mLeftVolShort;
|
|
uint16_t mRightVolShort;
|
|
};
|
|
|
|
class DuplicatingThread : public MixerThread {
|
|
public:
|
|
DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
|
|
audio_io_handle_t id);
|
|
virtual ~DuplicatingThread();
|
|
|
|
// Thread virtuals
|
|
virtual bool threadLoop();
|
|
void addOutputTrack(MixerThread* thread);
|
|
void removeOutputTrack(MixerThread* thread);
|
|
uint32_t waitTimeMs() { return mWaitTimeMs; }
|
|
protected:
|
|
virtual uint32_t activeSleepTimeUs();
|
|
|
|
private:
|
|
bool outputsReady(const SortedVector<sp<OutputTrack> > &outputTracks);
|
|
void updateWaitTime();
|
|
|
|
SortedVector < sp<OutputTrack> > mOutputTracks;
|
|
uint32_t mWaitTimeMs;
|
|
};
|
|
|
|
PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
|
|
MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
|
|
RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
|
|
// no range check, AudioFlinger::mLock held
|
|
bool streamMute_l(audio_stream_type_t stream) const
|
|
{ return mStreamTypes[stream].mute; }
|
|
// no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
|
|
float streamVolume_l(audio_stream_type_t stream) const
|
|
{ return mStreamTypes[stream].volume; }
|
|
void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2);
|
|
|
|
// allocate an audio_io_handle_t, session ID, or effect ID
|
|
uint32_t nextUniqueId();
|
|
|
|
status_t moveEffectChain_l(int sessionId,
|
|
PlaybackThread *srcThread,
|
|
PlaybackThread *dstThread,
|
|
bool reRegister);
|
|
// return thread associated with primary hardware device, or NULL
|
|
PlaybackThread *primaryPlaybackThread_l() const;
|
|
uint32_t primaryOutputDevice_l() const;
|
|
|
|
friend class AudioBuffer;
|
|
|
|
// server side of the client's IAudioTrack
|
|
class TrackHandle : public android::BnAudioTrack {
|
|
public:
|
|
TrackHandle(const sp<PlaybackThread::Track>& track);
|
|
virtual ~TrackHandle();
|
|
virtual sp<IMemory> getCblk() const;
|
|
virtual status_t start(pid_t tid);
|
|
virtual void stop();
|
|
virtual void flush();
|
|
virtual void mute(bool);
|
|
virtual void pause();
|
|
virtual status_t attachAuxEffect(int effectId);
|
|
virtual status_t allocateTimedBuffer(size_t size,
|
|
sp<IMemory>* buffer);
|
|
virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
|
|
int64_t pts);
|
|
virtual status_t setMediaTimeTransform(const LinearTransform& xform,
|
|
int target);
|
|
virtual status_t onTransact(
|
|
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
|
|
private:
|
|
const sp<PlaybackThread::Track> mTrack;
|
|
};
|
|
|
|
friend class Client;
|
|
friend class PlaybackThread::Track;
|
|
|
|
|
|
void removeClient_l(pid_t pid);
|
|
void removeNotificationClient(pid_t pid);
|
|
|
|
|
|
// record thread
|
|
class RecordThread : public ThreadBase, public AudioBufferProvider
|
|
{
|
|
public:
|
|
|
|
// record track
|
|
class RecordTrack : public TrackBase {
|
|
public:
|
|
RecordTrack(RecordThread *thread,
|
|
const sp<Client>& client,
|
|
uint32_t sampleRate,
|
|
audio_format_t format,
|
|
uint32_t channelMask,
|
|
int frameCount,
|
|
int sessionId);
|
|
virtual ~RecordTrack();
|
|
|
|
virtual status_t start(pid_t tid);
|
|
virtual void stop();
|
|
|
|
bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
|
|
bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
|
|
|
|
void dump(char* buffer, size_t size);
|
|
|
|
private:
|
|
friend class AudioFlinger;
|
|
friend class RecordThread;
|
|
|
|
RecordTrack(const RecordTrack&);
|
|
RecordTrack& operator = (const RecordTrack&);
|
|
|
|
// AudioBufferProvider interface
|
|
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS);
|
|
// releaseBuffer() not overridden
|
|
|
|
bool mOverflow;
|
|
};
|
|
|
|
|
|
RecordThread(const sp<AudioFlinger>& audioFlinger,
|
|
AudioStreamIn *input,
|
|
uint32_t sampleRate,
|
|
uint32_t channels,
|
|
audio_io_handle_t id,
|
|
uint32_t device);
|
|
virtual ~RecordThread();
|
|
|
|
virtual bool threadLoop();
|
|
virtual status_t readyToRun();
|
|
virtual void onFirstRef();
|
|
|
|
virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
|
|
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
|
|
const sp<AudioFlinger::Client>& client,
|
|
uint32_t sampleRate,
|
|
audio_format_t format,
|
|
int channelMask,
|
|
int frameCount,
|
|
int sessionId,
|
|
status_t *status);
|
|
|
|
status_t start(RecordTrack* recordTrack);
|
|
status_t start(RecordTrack* recordTrack, pid_t tid);
|
|
void stop(RecordTrack* recordTrack);
|
|
status_t dump(int fd, const Vector<String16>& args);
|
|
AudioStreamIn* getInput() const;
|
|
AudioStreamIn* clearInput();
|
|
virtual audio_stream_t* stream();
|
|
|
|
// AudioBufferProvider interface
|
|
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
|
|
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
|
|
|
|
virtual bool checkForNewParameters_l();
|
|
virtual String8 getParameters(const String8& keys);
|
|
virtual void audioConfigChanged_l(int event, int param = 0);
|
|
void readInputParameters();
|
|
virtual unsigned int getInputFramesLost();
|
|
|
|
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
|
|
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
|
|
virtual uint32_t hasAudioSession(int sessionId);
|
|
RecordTrack* track();
|
|
|
|
private:
|
|
RecordThread();
|
|
AudioStreamIn *mInput;
|
|
RecordTrack* mTrack;
|
|
sp<RecordTrack> mActiveTrack;
|
|
Condition mStartStopCond;
|
|
AudioResampler *mResampler;
|
|
int32_t *mRsmpOutBuffer;
|
|
int16_t *mRsmpInBuffer;
|
|
size_t mRsmpInIndex;
|
|
size_t mInputBytes;
|
|
const int mReqChannelCount;
|
|
const uint32_t mReqSampleRate;
|
|
ssize_t mBytesRead;
|
|
};
|
|
|
|
// server side of the client's IAudioRecord
|
|
class RecordHandle : public android::BnAudioRecord {
|
|
public:
|
|
RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
|
|
virtual ~RecordHandle();
|
|
virtual sp<IMemory> getCblk() const;
|
|
virtual status_t start(pid_t tid);
|
|
virtual void stop();
|
|
virtual status_t onTransact(
|
|
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
|
|
private:
|
|
const sp<RecordThread::RecordTrack> mRecordTrack;
|
|
};
|
|
|
|
//--- Audio Effect Management
|
|
|
|
// EffectModule and EffectChain classes both have their own mutex to protect
|
|
// state changes or resource modifications. Always respect the following order
|
|
// if multiple mutexes must be acquired to avoid cross deadlock:
|
|
// AudioFlinger -> ThreadBase -> EffectChain -> EffectModule
|
|
|
|
// The EffectModule class is a wrapper object controlling the effect engine implementation
|
|
// in the effect library. It prevents concurrent calls to process() and command() functions
|
|
// from different client threads. It keeps a list of EffectHandle objects corresponding
|
|
// to all client applications using this effect and notifies applications of effect state,
|
|
// control or parameter changes. It manages the activation state machine to send appropriate
|
|
// reset, enable, disable commands to effect engine and provide volume
|
|
// ramping when effects are activated/deactivated.
|
|
// When controlling an auxiliary effect, the EffectModule also provides an input buffer used by
|
|
// the attached track(s) to accumulate their auxiliary channel.
|
|
class EffectModule: public RefBase {
|
|
public:
|
|
EffectModule(ThreadBase *thread,
|
|
const wp<AudioFlinger::EffectChain>& chain,
|
|
effect_descriptor_t *desc,
|
|
int id,
|
|
int sessionId);
|
|
virtual ~EffectModule();
|
|
|
|
enum effect_state {
|
|
IDLE,
|
|
RESTART,
|
|
STARTING,
|
|
ACTIVE,
|
|
STOPPING,
|
|
STOPPED,
|
|
DESTROYED
|
|
};
|
|
|
|
int id() const { return mId; }
|
|
void process();
|
|
void updateState();
|
|
status_t command(uint32_t cmdCode,
|
|
uint32_t cmdSize,
|
|
void *pCmdData,
|
|
uint32_t *replySize,
|
|
void *pReplyData);
|
|
|
|
void reset_l();
|
|
status_t configure();
|
|
status_t init();
|
|
effect_state state() const {
|
|
return mState;
|
|
}
|
|
uint32_t status() {
|
|
return mStatus;
|
|
}
|
|
int sessionId() const {
|
|
return mSessionId;
|
|
}
|
|
status_t setEnabled(bool enabled);
|
|
bool isEnabled() const;
|
|
bool isProcessEnabled() const;
|
|
|
|
void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; }
|
|
int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; }
|
|
void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; }
|
|
int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; }
|
|
void setChain(const wp<EffectChain>& chain) { mChain = chain; }
|
|
void setThread(const wp<ThreadBase>& thread) { mThread = thread; }
|
|
const wp<ThreadBase>& thread() { return mThread; }
|
|
|
|
status_t addHandle(const sp<EffectHandle>& handle);
|
|
void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast);
|
|
size_t removeHandle (const wp<EffectHandle>& handle);
|
|
|
|
effect_descriptor_t& desc() { return mDescriptor; }
|
|
wp<EffectChain>& chain() { return mChain; }
|
|
|
|
status_t setDevice(uint32_t device);
|
|
status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
|
|
status_t setMode(audio_mode_t mode);
|
|
status_t start();
|
|
status_t stop();
|
|
void setSuspended(bool suspended);
|
|
bool suspended() const;
|
|
|
|
sp<EffectHandle> controlHandle();
|
|
|
|
bool isPinned() const { return mPinned; }
|
|
void unPin() { mPinned = false; }
|
|
|
|
status_t dump(int fd, const Vector<String16>& args);
|
|
|
|
protected:
|
|
friend class EffectHandle;
|
|
friend class AudioFlinger;
|
|
bool mPinned;
|
|
|
|
// Maximum time allocated to effect engines to complete the turn off sequence
|
|
static const uint32_t MAX_DISABLE_TIME_MS = 10000;
|
|
|
|
EffectModule(const EffectModule&);
|
|
EffectModule& operator = (const EffectModule&);
|
|
|
|
status_t start_l();
|
|
status_t stop_l();
|
|
|
|
mutable Mutex mLock; // mutex for process, commands and handles list protection
|
|
wp<ThreadBase> mThread; // parent thread
|
|
wp<EffectChain> mChain; // parent effect chain
|
|
int mId; // this instance unique ID
|
|
int mSessionId; // audio session ID
|
|
effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
|
|
effect_config_t mConfig; // input and output audio configuration
|
|
effect_handle_t mEffectInterface; // Effect module C API
|
|
status_t mStatus; // initialization status
|
|
effect_state mState; // current activation state
|
|
Vector< wp<EffectHandle> > mHandles; // list of client handles
|
|
// First handle in mHandles has highest priority and controls the effect module
|
|
uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after
|
|
// sending disable command.
|
|
uint32_t mDisableWaitCnt; // current process() calls count during disable period.
|
|
bool mSuspended; // effect is suspended: temporarily disabled by framework
|
|
};
|
|
|
|
// The EffectHandle class implements the IEffect interface. It provides resources
|
|
// to receive parameter updates, keeps track of effect control
|
|
// ownership and state and has a pointer to the EffectModule object it is controlling.
|
|
// There is one EffectHandle object for each application controlling (or using)
|
|
// an effect module.
|
|
// The EffectHandle is obtained by calling AudioFlinger::createEffect().
|
|
class EffectHandle: public android::BnEffect {
|
|
public:
|
|
|
|
EffectHandle(const sp<EffectModule>& effect,
|
|
const sp<AudioFlinger::Client>& client,
|
|
const sp<IEffectClient>& effectClient,
|
|
int32_t priority);
|
|
virtual ~EffectHandle();
|
|
|
|
// IEffect
|
|
virtual status_t enable();
|
|
virtual status_t disable();
|
|
virtual status_t command(uint32_t cmdCode,
|
|
uint32_t cmdSize,
|
|
void *pCmdData,
|
|
uint32_t *replySize,
|
|
void *pReplyData);
|
|
virtual void disconnect();
|
|
private:
|
|
void disconnect(bool unpinIfLast);
|
|
public:
|
|
virtual sp<IMemory> getCblk() const { return mCblkMemory; }
|
|
virtual status_t onTransact(uint32_t code, const Parcel& data,
|
|
Parcel* reply, uint32_t flags);
|
|
|
|
|
|
// Give or take control of effect module
|
|
// - hasControl: true if control is given, false if removed
|
|
// - signal: true client app should be signaled of change, false otherwise
|
|
// - enabled: state of the effect when control is passed
|
|
void setControl(bool hasControl, bool signal, bool enabled);
|
|
void commandExecuted(uint32_t cmdCode,
|
|
uint32_t cmdSize,
|
|
void *pCmdData,
|
|
uint32_t replySize,
|
|
void *pReplyData);
|
|
void setEnabled(bool enabled);
|
|
bool enabled() const { return mEnabled; }
|
|
|
|
// Getters
|
|
int id() const { return mEffect->id(); }
|
|
int priority() const { return mPriority; }
|
|
bool hasControl() const { return mHasControl; }
|
|
sp<EffectModule> effect() const { return mEffect; }
|
|
|
|
void dump(char* buffer, size_t size);
|
|
|
|
protected:
|
|
friend class AudioFlinger;
|
|
friend class EffectModule;
|
|
EffectHandle(const EffectHandle&);
|
|
EffectHandle& operator =(const EffectHandle&);
|
|
|
|
sp<EffectModule> mEffect; // pointer to controlled EffectModule
|
|
sp<IEffectClient> mEffectClient; // callback interface for client notifications
|
|
/*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect()
|
|
sp<IMemory> mCblkMemory; // shared memory for control block
|
|
effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory
|
|
uint8_t* mBuffer; // pointer to parameter area in shared memory
|
|
int mPriority; // client application priority to control the effect
|
|
bool mHasControl; // true if this handle is controlling the effect
|
|
bool mEnabled; // cached enable state: needed when the effect is
|
|
// restored after being suspended
|
|
};
|
|
|
|
// the EffectChain class represents a group of effects associated to one audio session.
|
|
// There can be any number of EffectChain objects per output mixer thread (PlaybackThread).
|
|
// The EffecChain with session ID 0 contains global effects applied to the output mix.
|
|
// Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks)
|
|
// are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding
|
|
// in the effect process order. When attached to a track (session ID != 0), it also provide it's own
|
|
// input buffer used by the track as accumulation buffer.
|
|
class EffectChain: public RefBase {
|
|
public:
|
|
EffectChain(const wp<ThreadBase>& wThread, int sessionId);
|
|
EffectChain(ThreadBase *thread, int sessionId);
|
|
virtual ~EffectChain();
|
|
|
|
// special key used for an entry in mSuspendedEffects keyed vector
|
|
// corresponding to a suspend all request.
|
|
static const int kKeyForSuspendAll = 0;
|
|
|
|
// minimum duration during which we force calling effect process when last track on
|
|
// a session is stopped or removed to allow effect tail to be rendered
|
|
static const int kProcessTailDurationMs = 1000;
|
|
|
|
void process_l();
|
|
|
|
void lock() {
|
|
mLock.lock();
|
|
}
|
|
void unlock() {
|
|
mLock.unlock();
|
|
}
|
|
|
|
status_t addEffect_l(const sp<EffectModule>& handle);
|
|
size_t removeEffect_l(const sp<EffectModule>& handle);
|
|
|
|
int sessionId() const { return mSessionId; }
|
|
void setSessionId(int sessionId) { mSessionId = sessionId; }
|
|
|
|
sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
|
|
sp<EffectModule> getEffectFromId_l(int id);
|
|
sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
|
|
bool setVolume_l(uint32_t *left, uint32_t *right);
|
|
void setDevice_l(uint32_t device);
|
|
void setMode_l(audio_mode_t mode);
|
|
|
|
void setInBuffer(int16_t *buffer, bool ownsBuffer = false) {
|
|
mInBuffer = buffer;
|
|
mOwnInBuffer = ownsBuffer;
|
|
}
|
|
int16_t *inBuffer() const {
|
|
return mInBuffer;
|
|
}
|
|
void setOutBuffer(int16_t *buffer) {
|
|
mOutBuffer = buffer;
|
|
}
|
|
int16_t *outBuffer() const {
|
|
return mOutBuffer;
|
|
}
|
|
|
|
void incTrackCnt() { android_atomic_inc(&mTrackCnt); }
|
|
void decTrackCnt() { android_atomic_dec(&mTrackCnt); }
|
|
int32_t trackCnt() const { return mTrackCnt;}
|
|
|
|
void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt);
|
|
mTailBufferCount = mMaxTailBuffers; }
|
|
void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); }
|
|
int32_t activeTrackCnt() const { return mActiveTrackCnt;}
|
|
|
|
uint32_t strategy() const { return mStrategy; }
|
|
void setStrategy(uint32_t strategy)
|
|
{ mStrategy = strategy; }
|
|
|
|
// suspend effect of the given type
|
|
void setEffectSuspended_l(const effect_uuid_t *type,
|
|
bool suspend);
|
|
// suspend all eligible effects
|
|
void setEffectSuspendedAll_l(bool suspend);
|
|
// check if effects should be suspend or restored when a given effect is enable or disabled
|
|
void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
|
|
bool enabled);
|
|
|
|
status_t dump(int fd, const Vector<String16>& args);
|
|
|
|
protected:
|
|
friend class AudioFlinger;
|
|
EffectChain(const EffectChain&);
|
|
EffectChain& operator =(const EffectChain&);
|
|
|
|
class SuspendedEffectDesc : public RefBase {
|
|
public:
|
|
SuspendedEffectDesc() : mRefCount(0) {}
|
|
|
|
int mRefCount;
|
|
effect_uuid_t mType;
|
|
wp<EffectModule> mEffect;
|
|
};
|
|
|
|
// get a list of effect modules to suspend when an effect of the type
|
|
// passed is enabled.
|
|
void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects);
|
|
|
|
// get an effect module if it is currently enable
|
|
sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type);
|
|
// true if the effect whose descriptor is passed can be suspended
|
|
// OEMs can modify the rules implemented in this method to exclude specific effect
|
|
// types or implementations from the suspend/restore mechanism.
|
|
bool isEffectEligibleForSuspend(const effect_descriptor_t& desc);
|
|
|
|
wp<ThreadBase> mThread; // parent mixer thread
|
|
Mutex mLock; // mutex protecting effect list
|
|
Vector<sp<EffectModule> > mEffects; // list of effect modules
|
|
int mSessionId; // audio session ID
|
|
int16_t *mInBuffer; // chain input buffer
|
|
int16_t *mOutBuffer; // chain output buffer
|
|
volatile int32_t mActiveTrackCnt; // number of active tracks connected
|
|
volatile int32_t mTrackCnt; // number of tracks connected
|
|
int32_t mTailBufferCount; // current effect tail buffer count
|
|
int32_t mMaxTailBuffers; // maximum effect tail buffers
|
|
bool mOwnInBuffer; // true if the chain owns its input buffer
|
|
int mVolumeCtrlIdx; // index of insert effect having control over volume
|
|
uint32_t mLeftVolume; // previous volume on left channel
|
|
uint32_t mRightVolume; // previous volume on right channel
|
|
uint32_t mNewLeftVolume; // new volume on left channel
|
|
uint32_t mNewRightVolume; // new volume on right channel
|
|
uint32_t mStrategy; // strategy for this effect chain
|
|
// mSuspendedEffects lists all effect currently suspended in the chain
|
|
// use effect type UUID timelow field as key. There is no real risk of identical
|
|
// timeLow fields among effect type UUIDs.
|
|
KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
|
|
};
|
|
|
|
// AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
|
|
// For emphasis, we could also make all pointers to them be "const *",
|
|
// but that would clutter the code unnecessarily.
|
|
|
|
struct AudioStreamOut {
|
|
audio_hw_device_t* const hwDev;
|
|
audio_stream_out_t* const stream;
|
|
|
|
AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) :
|
|
hwDev(dev), stream(out) {}
|
|
};
|
|
|
|
struct AudioStreamIn {
|
|
audio_hw_device_t* const hwDev;
|
|
audio_stream_in_t* const stream;
|
|
|
|
AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) :
|
|
hwDev(dev), stream(in) {}
|
|
};
|
|
|
|
// for mAudioSessionRefs only
|
|
struct AudioSessionRef {
|
|
// FIXME rename parameter names when fields get "m" prefix
|
|
AudioSessionRef(int sessionid_, pid_t pid_) :
|
|
sessionid(sessionid_), pid(pid_), cnt(1) {}
|
|
const int sessionid;
|
|
const pid_t pid;
|
|
int cnt;
|
|
};
|
|
|
|
friend class RecordThread;
|
|
friend class PlaybackThread;
|
|
|
|
enum master_volume_support {
|
|
// MVS_NONE:
|
|
// Audio HAL has no support for master volume, either setting or
|
|
// getting. All master volume control must be implemented in SW by the
|
|
// AudioFlinger mixing core.
|
|
MVS_NONE,
|
|
|
|
// MVS_SETONLY:
|
|
// Audio HAL has support for setting master volume, but not for getting
|
|
// master volume (original HAL design did not include a getter).
|
|
// AudioFlinger needs to keep track of the last set master volume in
|
|
// addition to needing to set an initial, default, master volume at HAL
|
|
// load time.
|
|
MVS_SETONLY,
|
|
|
|
// MVS_FULL:
|
|
// Audio HAL has support both for setting and getting master volume.
|
|
// AudioFlinger should send all set and get master volume requests
|
|
// directly to the HAL.
|
|
MVS_FULL,
|
|
};
|
|
|
|
mutable Mutex mLock;
|
|
|
|
DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client()
|
|
|
|
mutable Mutex mHardwareLock;
|
|
|
|
// These two fields are immutable after onFirstRef(), so no lock needed to access
|
|
audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
|
|
Vector<audio_hw_device_t*> mAudioHwDevs;
|
|
|
|
// for dump, indicates which hardware operation is currently in progress (but not stream ops)
|
|
enum hardware_call_state {
|
|
AUDIO_HW_IDLE = 0, // no operation in progress
|
|
AUDIO_HW_INIT, // init_check
|
|
AUDIO_HW_OUTPUT_OPEN, // open_output_stream
|
|
AUDIO_HW_OUTPUT_CLOSE, // unused
|
|
AUDIO_HW_INPUT_OPEN, // unused
|
|
AUDIO_HW_INPUT_CLOSE, // unused
|
|
AUDIO_HW_STANDBY, // unused
|
|
AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume
|
|
AUDIO_HW_GET_ROUTING, // unused
|
|
AUDIO_HW_SET_ROUTING, // unused
|
|
AUDIO_HW_GET_MODE, // unused
|
|
AUDIO_HW_SET_MODE, // set_mode
|
|
AUDIO_HW_GET_MIC_MUTE, // get_mic_mute
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AUDIO_HW_SET_MIC_MUTE, // set_mic_mute
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AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume
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AUDIO_HW_SET_PARAMETER, // set_parameters
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AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size
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AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume
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AUDIO_HW_GET_PARAMETER, // get_parameters
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};
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mutable hardware_call_state mHardwareStatus; // for dump only
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DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads;
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stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
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// both are protected by mLock
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float mMasterVolume;
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float mMasterVolumeSW;
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master_volume_support mMasterVolumeSupportLvl;
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bool mMasterMute;
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DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
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DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
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volatile int32_t mNextUniqueId; // updated by android_atomic_inc
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audio_mode_t mMode;
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bool mBtNrecIsOff;
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// protected by mLock
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Vector<AudioSessionRef*> mAudioSessionRefs;
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float masterVolume_l() const;
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float masterVolumeSW_l() const { return mMasterVolumeSW; }
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bool masterMute_l() const { return mMasterMute; }
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private:
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sp<Client> registerPid_l(pid_t pid); // always returns non-0
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};
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// ----------------------------------------------------------------------------
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}; // namespace android
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#endif // ANDROID_AUDIO_FLINGER_H
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