775 lines
28 KiB
C++
775 lines
28 KiB
C++
/* //device/include/server/AudioFlinger/AudioFlinger.h
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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 <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/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 <binder/MemoryDealer.h>
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#include <utils/SortedVector.h>
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#include <utils/Vector.h>
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#include <hardware_legacy/AudioHardwareInterface.h>
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#include "AudioBufferProvider.h"
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namespace android {
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class audio_track_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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#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
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#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
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// ----------------------------------------------------------------------------
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static const nsecs_t kStandbyTimeInNsecs = seconds(3);
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class AudioFlinger : public BnAudioFlinger, public IBinder::DeathRecipient
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{
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public:
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static void instantiate();
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virtual status_t dump(int fd, const Vector<String16>& args);
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// IAudioFlinger interface
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virtual sp<IAudioTrack> createTrack(
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pid_t pid,
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int streamType,
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uint32_t sampleRate,
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int format,
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int channelCount,
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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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int output,
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status_t *status);
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virtual uint32_t sampleRate(int output) const;
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virtual int channelCount(int output) const;
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virtual int format(int output) const;
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virtual size_t frameCount(int output) const;
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virtual uint32_t latency(int 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 bool masterMute() const;
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virtual status_t setStreamVolume(int stream, float value, int output);
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virtual status_t setStreamMute(int stream, bool muted);
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virtual float streamVolume(int stream, int output) const;
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virtual bool streamMute(int stream) const;
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virtual status_t setMode(int mode);
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virtual status_t setMicMute(bool state);
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virtual bool getMicMute() const;
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virtual bool isMusicActive() const;
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virtual status_t setParameters(int ioHandle, const String8& keyValuePairs);
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virtual String8 getParameters(int ioHandle, const String8& keys);
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virtual void registerClient(const sp<IAudioFlingerClient>& client);
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virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
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virtual int openOutput(uint32_t *pDevices,
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uint32_t *pSamplingRate,
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uint32_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 int openDuplicateOutput(int output1, int output2);
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virtual status_t closeOutput(int output);
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virtual status_t suspendOutput(int output);
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virtual status_t restoreOutput(int output);
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virtual int openInput(uint32_t *pDevices,
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uint32_t *pSamplingRate,
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uint32_t *pFormat,
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uint32_t *pChannels,
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uint32_t acoustics);
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virtual status_t closeInput(int input);
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virtual status_t setStreamOutput(uint32_t stream, int output);
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virtual status_t setVoiceVolume(float volume);
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// IBinder::DeathRecipient
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virtual void binderDied(const wp<IBinder>& who);
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enum hardware_call_state {
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AUDIO_HW_IDLE = 0,
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AUDIO_HW_INIT,
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AUDIO_HW_OUTPUT_OPEN,
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AUDIO_HW_OUTPUT_CLOSE,
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AUDIO_HW_INPUT_OPEN,
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AUDIO_HW_INPUT_CLOSE,
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AUDIO_HW_STANDBY,
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AUDIO_HW_SET_MASTER_VOLUME,
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AUDIO_HW_GET_ROUTING,
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AUDIO_HW_SET_ROUTING,
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AUDIO_HW_GET_MODE,
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AUDIO_HW_SET_MODE,
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AUDIO_HW_GET_MIC_MUTE,
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AUDIO_HW_SET_MIC_MUTE,
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AUDIO_SET_VOICE_VOLUME,
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AUDIO_SET_PARAMETER,
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};
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// record interface
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virtual sp<IAudioRecord> openRecord(
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pid_t pid,
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int input,
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uint32_t sampleRate,
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int format,
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int channelCount,
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int frameCount,
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uint32_t flags,
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status_t *status);
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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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private:
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AudioFlinger();
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virtual ~AudioFlinger();
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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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const sp<MemoryDealer>& heap() const;
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pid_t pid() const { return mPid; }
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sp<AudioFlinger> audioFlinger() { return mAudioFlinger; }
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private:
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Client(const Client&);
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Client& operator = (const Client&);
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sp<AudioFlinger> mAudioFlinger;
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sp<MemoryDealer> mMemoryDealer;
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pid_t mPid;
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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 Track;
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class RecordTrack;
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class ThreadBase : public Thread {
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public:
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ThreadBase (const sp<AudioFlinger>& audioFlinger);
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virtual ~ThreadBase();
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status_t dumpBase(int fd, const Vector<String16>& args);
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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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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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enum track_flags {
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STEPSERVER_FAILED = 0x01, // StepServer could not acquire cblk->lock mutex
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SYSTEM_FLAGS_MASK = 0x0000ffffUL,
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// The upper 16 bits are used for track-specific flags.
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};
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TrackBase(const wp<ThreadBase>& thread,
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const sp<Client>& client,
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uint32_t sampleRate,
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int format,
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int channelCount,
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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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~TrackBase();
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virtual status_t start() = 0;
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virtual void stop() = 0;
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sp<IMemory> getCblk() const;
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audio_track_cblk_t* cblk() const { return mCblk; }
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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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virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
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virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
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int format() const {
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return mFormat;
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}
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int channelCount() const ;
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int sampleRate() const;
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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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wp<ThreadBase> mThread;
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sp<Client> mClient;
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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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int mState;
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int mClientTid;
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uint8_t mFormat;
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uint32_t mFlags;
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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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uint32_t sampleRate() const;
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int channelCount() const;
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int format() const;
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size_t frameCount() const;
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void wakeUp() { mWaitWorkCV.broadcast(); }
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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(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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mutable Mutex mLock;
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protected:
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friend class Track;
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friend class TrackBase;
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friend class PlaybackThread;
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friend class MixerThread;
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friend class DirectOutputThread;
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friend class DuplicatingThread;
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friend class RecordThread;
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friend class RecordTrack;
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Condition mWaitWorkCV;
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sp<AudioFlinger> mAudioFlinger;
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uint32_t mSampleRate;
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size_t mFrameCount;
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int mChannelCount;
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int mFormat;
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uint32_t mFrameSize;
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Condition mParamCond;
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Vector<String8> mNewParameters;
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status_t mParamStatus;
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Vector<ConfigEvent *> mConfigEvents;
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bool mStandby;
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};
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// --- PlaybackThread ---
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class PlaybackThread : public ThreadBase {
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public:
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enum type {
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MIXER,
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DIRECT,
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DUPLICATING
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};
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// playback track
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class Track : public TrackBase {
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public:
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Track( const wp<ThreadBase>& thread,
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const sp<Client>& client,
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int streamType,
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uint32_t sampleRate,
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int format,
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int channelCount,
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int frameCount,
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const sp<IMemory>& sharedBuffer);
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~Track();
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void dump(char* buffer, size_t size);
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virtual status_t start();
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virtual void stop();
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void pause();
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void flush();
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void destroy();
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void mute(bool);
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void setVolume(float left, float right);
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int name() const {
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return mName;
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}
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int type() const {
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return mStreamType;
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}
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protected:
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friend class ThreadBase;
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friend class AudioFlinger;
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friend class TrackHandle;
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friend class PlaybackThread;
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friend class MixerThread;
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friend class DirectOutputThread;
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Track(const Track&);
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Track& operator = (const Track&);
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virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
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bool isMuted() { return mMute; }
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bool isPausing() const {
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return mState == PAUSING;
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}
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bool isPaused() const {
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return mState == PAUSED;
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}
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bool isReady() const;
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void setPaused() { mState = PAUSED; }
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void reset();
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// we don't really need a lock for these
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float mVolume[2];
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volatile bool mMute;
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// FILLED state is used for suppressing volume ramp at begin of playing
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enum {FS_FILLING, FS_FILLED, FS_ACTIVE};
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mutable uint8_t mFillingUpStatus;
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int8_t mRetryCount;
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sp<IMemory> mSharedBuffer;
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bool mResetDone;
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int mStreamType;
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int mName;
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}; // end of Track
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// playback track
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class OutputTrack : public Track {
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public:
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class Buffer: public AudioBufferProvider::Buffer {
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public:
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int16_t *mBuffer;
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};
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OutputTrack( const wp<ThreadBase>& thread,
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uint32_t sampleRate,
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int format,
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int channelCount,
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int frameCount);
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~OutputTrack();
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virtual status_t start();
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virtual void stop();
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bool write(int16_t* data, uint32_t frames);
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bool bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; }
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bool isActive() { return mActive; }
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wp<ThreadBase>& thread() { return mThread; }
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private:
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status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
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void clearBufferQueue();
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// Maximum number of pending buffers allocated by OutputTrack::write()
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static const uint8_t kMaxOverFlowBuffers = 3;
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Vector < Buffer* > mBufferQueue;
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AudioBufferProvider::Buffer mOutBuffer;
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uint32_t mWaitTimeMs;
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bool mActive;
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}; // end of OutputTrack
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PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
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virtual ~PlaybackThread();
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virtual status_t dump(int fd, const Vector<String16>& args);
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// Thread virtuals
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virtual status_t readyToRun();
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virtual void onFirstRef();
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virtual uint32_t latency() 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 bool masterMute() const;
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virtual status_t setStreamVolume(int stream, float value);
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virtual status_t setStreamMute(int stream, bool muted);
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virtual float streamVolume(int stream) const;
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virtual bool streamMute(int stream) const;
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bool isMusicActive() const;
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sp<Track> createTrack_l(
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const sp<AudioFlinger::Client>& client,
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int streamType,
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uint32_t sampleRate,
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int format,
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int channelCount,
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int frameCount,
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const sp<IMemory>& sharedBuffer,
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status_t *status);
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AudioStreamOut* getOutput() { return mOutput; }
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virtual int type() const { return mType; }
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void suspend() { mSuspended++; }
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void restore() { if (mSuspended) mSuspended--; }
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virtual String8 getParameters(const String8& keys);
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virtual void audioConfigChanged(int event, int param = 0);
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struct stream_type_t {
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stream_type_t()
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: volume(1.0f),
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mute(false)
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{
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}
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float volume;
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bool mute;
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};
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protected:
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int mType;
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int16_t* mMixBuffer;
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int mSuspended;
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int mBytesWritten;
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bool mMasterMute;
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SortedVector< wp<Track> > mActiveTracks;
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virtual int getTrackName_l() = 0;
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virtual void deleteTrackName_l(int name) = 0;
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virtual uint32_t getMaxBufferRecoveryInUsecs() = 0;
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private:
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friend class AudioFlinger;
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friend class OutputTrack;
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friend class Track;
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friend class TrackBase;
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friend class MixerThread;
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friend class DirectOutputThread;
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friend class DuplicatingThread;
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PlaybackThread(const Client&);
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PlaybackThread& operator = (const PlaybackThread&);
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status_t addTrack_l(const sp<Track>& track);
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void destroyTrack_l(const sp<Track>& track);
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void readOutputParameters();
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virtual status_t dumpInternals(int fd, const Vector<String16>& args);
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status_t dumpTracks(int fd, const Vector<String16>& args);
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SortedVector< sp<Track> > mTracks;
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// mStreamTypes[] uses 1 additionnal stream type internally for the OutputTrack used by DuplicatingThread
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stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES + 1];
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AudioStreamOut* mOutput;
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float mMasterVolume;
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nsecs_t mLastWriteTime;
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int mNumWrites;
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int mNumDelayedWrites;
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bool mInWrite;
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int mMinBytesToWrite;
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};
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class MixerThread : public PlaybackThread {
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public:
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MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
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virtual ~MixerThread();
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// Thread virtuals
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virtual bool threadLoop();
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void getTracks(SortedVector < sp<Track> >& tracks,
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SortedVector < wp<Track> >& activeTracks,
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int streamType);
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void putTracks(SortedVector < sp<Track> >& tracks,
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SortedVector < wp<Track> >& activeTracks);
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virtual bool checkForNewParameters_l();
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virtual status_t dumpInternals(int fd, const Vector<String16>& args);
|
|
|
|
protected:
|
|
size_t prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove);
|
|
virtual int getTrackName_l();
|
|
virtual void deleteTrackName_l(int name);
|
|
virtual uint32_t getMaxBufferRecoveryInUsecs();
|
|
|
|
AudioMixer* mAudioMixer;
|
|
};
|
|
|
|
class DirectOutputThread : public PlaybackThread {
|
|
public:
|
|
|
|
DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
|
|
~DirectOutputThread();
|
|
|
|
// Thread virtuals
|
|
virtual bool threadLoop();
|
|
|
|
virtual bool checkForNewParameters_l();
|
|
|
|
protected:
|
|
virtual int getTrackName_l();
|
|
virtual void deleteTrackName_l(int name);
|
|
virtual uint32_t getMaxBufferRecoveryInUsecs();
|
|
|
|
private:
|
|
float mLeftVolume;
|
|
float mRightVolume;
|
|
};
|
|
|
|
class DuplicatingThread : public MixerThread {
|
|
public:
|
|
DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread);
|
|
~DuplicatingThread();
|
|
|
|
// Thread virtuals
|
|
virtual bool threadLoop();
|
|
void addOutputTrack(MixerThread* thread);
|
|
void removeOutputTrack(MixerThread* thread);
|
|
|
|
private:
|
|
SortedVector < sp<OutputTrack> > mOutputTracks;
|
|
};
|
|
|
|
PlaybackThread *checkPlaybackThread_l(int output) const;
|
|
MixerThread *checkMixerThread_l(int output) const;
|
|
RecordThread *checkRecordThread_l(int input) const;
|
|
float streamVolumeInternal(int stream) const { return mStreamTypes[stream].volume; }
|
|
void audioConfigChanged_l(int event, const sp<ThreadBase>& thread, void *param2);
|
|
|
|
friend class AudioBuffer;
|
|
|
|
class TrackHandle : public android::BnAudioTrack {
|
|
public:
|
|
TrackHandle(const sp<PlaybackThread::Track>& track);
|
|
virtual ~TrackHandle();
|
|
virtual status_t start();
|
|
virtual void stop();
|
|
virtual void flush();
|
|
virtual void mute(bool);
|
|
virtual void pause();
|
|
virtual void setVolume(float left, float right);
|
|
virtual sp<IMemory> getCblk() const;
|
|
virtual status_t onTransact(
|
|
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
|
|
private:
|
|
sp<PlaybackThread::Track> mTrack;
|
|
};
|
|
|
|
friend class Client;
|
|
friend class PlaybackThread::Track;
|
|
|
|
|
|
void removeClient_l(pid_t pid);
|
|
|
|
|
|
// record thread
|
|
class RecordThread : public ThreadBase, public AudioBufferProvider
|
|
{
|
|
public:
|
|
|
|
// record track
|
|
class RecordTrack : public TrackBase {
|
|
public:
|
|
RecordTrack(const wp<ThreadBase>& thread,
|
|
const sp<Client>& client,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags);
|
|
~RecordTrack();
|
|
|
|
virtual status_t start();
|
|
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&);
|
|
|
|
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
|
|
|
|
bool mOverflow;
|
|
};
|
|
|
|
|
|
RecordThread(const sp<AudioFlinger>& audioFlinger,
|
|
AudioStreamIn *input,
|
|
uint32_t sampleRate,
|
|
uint32_t channels);
|
|
~RecordThread();
|
|
|
|
virtual bool threadLoop();
|
|
virtual status_t readyToRun() { return NO_ERROR; }
|
|
virtual void onFirstRef();
|
|
|
|
status_t start(RecordTrack* recordTrack);
|
|
void stop(RecordTrack* recordTrack);
|
|
status_t dump(int fd, const Vector<String16>& args);
|
|
AudioStreamIn* getInput() { return mInput; }
|
|
|
|
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
|
|
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
|
|
virtual bool checkForNewParameters_l();
|
|
virtual String8 getParameters(const String8& keys);
|
|
virtual void audioConfigChanged(int event, int param = 0);
|
|
void readInputParameters();
|
|
|
|
private:
|
|
RecordThread();
|
|
AudioStreamIn *mInput;
|
|
sp<RecordTrack> mActiveTrack;
|
|
Condition mStartStopCond;
|
|
AudioResampler *mResampler;
|
|
int32_t *mRsmpOutBuffer;
|
|
int16_t *mRsmpInBuffer;
|
|
size_t mRsmpInIndex;
|
|
size_t mInputBytes;
|
|
int mReqChannelCount;
|
|
uint32_t mReqSampleRate;
|
|
};
|
|
|
|
class RecordHandle : public android::BnAudioRecord {
|
|
public:
|
|
RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
|
|
virtual ~RecordHandle();
|
|
virtual status_t start();
|
|
virtual void stop();
|
|
virtual sp<IMemory> getCblk() const;
|
|
virtual status_t onTransact(
|
|
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
|
|
private:
|
|
sp<RecordThread::RecordTrack> mRecordTrack;
|
|
};
|
|
|
|
friend class RecordThread;
|
|
friend class PlaybackThread;
|
|
|
|
|
|
mutable Mutex mLock;
|
|
|
|
DefaultKeyedVector< pid_t, wp<Client> > mClients;
|
|
|
|
mutable Mutex mHardwareLock;
|
|
AudioHardwareInterface* mAudioHardware;
|
|
mutable int mHardwareStatus;
|
|
|
|
|
|
DefaultKeyedVector< int, sp<PlaybackThread> > mPlaybackThreads;
|
|
PlaybackThread::stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES];
|
|
float mMasterVolume;
|
|
bool mMasterMute;
|
|
|
|
DefaultKeyedVector< int, sp<RecordThread> > mRecordThreads;
|
|
|
|
SortedVector< sp<IBinder> > mNotificationClients;
|
|
int mNextThreadId;
|
|
};
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
}; // namespace android
|
|
|
|
#endif // ANDROID_AUDIO_FLINGER_H
|