Mike J. Chen 7bce396226 Media framework changes for Tungsten.
Squashed merge from master-tungsten of the following changes:

commit 73d09e18c4557e583a1684d44d598a1a02fd0cf2
Author: John Grossman <johngro@google.com>
Date:   Mon Jun 20 13:57:44 2011 -0700

    Remove TungstenMisc and rename LinearTransform

    Change-Id: Ie8aa3e24e09fdbf6ef8996c26deb9c5640e20d1b

commit 3114aabe76ad733b59929d87e49c68229f5ae2e8
Author: John Grossman <johngro@google.com>
Date:   Fri Jun 3 10:47:16 2011 -0700

    Name changes and spelling fixes.

    + Replace the term TungstenTime with the Eugene-approved term CommonTime.
    + Fix a spelling error in a comment I noticed.

    Change-Id: I8c10d618206826d16055f78c7724e24443bb03fd

commit cbf2903ab6893b6e662514e2f6d670e268a419df
Author: John Grossman <johngro@google.com>
Date:   Fri Apr 15 09:27:54 2011 -0700

    Migrate Tungsten code from the HC-Tungsten to the Master-Tungsten branch.

    Change-Id: I95372d913a0761d90168edb4016f5ece0ea74502

commit bc7c46aa629f9883e959ef23de8da297f9eb508b
Author: Jason Simmons <jsimmons@google.com>
Date:   Mon Jun 20 13:59:17 2011 -0700

    Create a separate class for timed AudioTracks

commit 43be3231034ff8537fdd84422a7954780038671f
Author: John Grossman <johngro@google.com>
Date:   Mon Jun 27 18:59:12 2011 -0700

    Move libaah_rtp over from the vendor directory.

    Also move factor PipeEvent out into utils.

    Change-Id: Id3877c66efe22d771cf3ef4877107e431b828e37

commit 17526eb3148c9c3d4365b6d5b47e8dc13bca71b6
Author: John Grossman <johngro@google.com>
Date:   Mon Jun 27 17:06:49 2011 -0700

    Name changes for the TRTP Players s/tungsten/aah/g

    Change-Id: I55e9ad13003f6aa6a36955b54426a7efbe31ac51

commit 423fc1bfc0fda799c421a650c83c4b9293b1a08c
Author: Jason Simmons <jsimmons@google.com>
Date:   Mon Jun 20 17:56:09 2011 -0700

    More timed AudioFlinger changes requested by code review:
    * change trimTimedBufferQueue to trimTimedBufferQueue_l
    * create one timed audio buffer heap per client process instead of one per track
    * grow the silence buffer on demand
    * some error handling fixes in timed getNextBuffer
    * calculate the next output PTS in all mixer and track hooks

    Change-Id: Ifc51a08b55029b7c48902ab2f22933ad7bafe1ad

commit a148e2674b1d3cb73289b82b85c333f0a66824a9
Author: John Grossman <johngro@google.com>
Date:   Mon Jun 20 17:02:24 2011 -0700

    Move the A@H time service into frameworks/base

    Change-Id: I5c570cde70e8931e205516cb33517585804ce841

commit dfa438fa49bdaeeb2ec5fd0d17b30d881608b6b1
Author: John Grossman <johngro@google.com>
Date:   Mon Jun 20 11:55:36 2011 -0700

    Fix the build after Mike's code moving.

    Change-Id: Ia883643ded252168bcc5a70584ab6ce97bb05266

commit 04489474ec8e73efe1bf52918831f41659033162
Author: John Grossman <johngro@google.com>
Date:   Fri Jun 17 14:19:50 2011 -0700

    Refactor the local/common clock services.

    This change is one of a set of 5 changes made to different repositories.  Look
    for this comment in all of them.

    Refactor the local/common clock services in tungsten to match android best
    practice.  Notable changes include

    + The kernel no longer knows anything about common time.  Common time has been
      moved completely up into user land.  This has an impact on the accuracy of the
      timesync debugging code, and the netfilter assisted approach to network based
      timesync is going to have to be modified.
    + The timesync driver used by A@H is now just local time driver.
    + The kernel no longer needs access to the linear transform math code, and it
      has been removed.
    + A new HAL has been introduced to expose the concept of local time to the
      system.
    + A non-slewable stub implementation of the local time HAL based on
      CLOCK_MONOTONIC has been added.
    + The TungstenTime library has been eliminated.  Its functionality has been
      distributed among the common time binder service, the local time hal and the
      linear transform utility code.
    + All clients of the old TungstenTime library have been changed to be clients of
      the binder service, the hal and the utility code.
    + The reset_tt utilities have been removed, they no longer have a purpose in the
      system.
    + more progress has been made in eliminating the word "tungsten" from the code.

    Things left to do include
    + Finish getting rid of tungsten from the time service.
    + Move the time service into the framework; AudioFlinger's new timed mode
      depends on it and the service cannot continue to live in vendor tungsten.

    Change-Id: I999b6cfb4a9d267818a86d747c35eecfc6693101

commit d48194545eed1116a84d81e2fb53315d2b0701a7
Author: Jason Simmons <jsimmons@google.com>
Date:   Thu Jun 16 14:22:46 2011 -0700

    Change the interface of the AudioMixer and AudioBufferProvider to accept a presentation timestamp

    Change-Id: Ice2df5628d45a7f77100e7008103b35b3d3160a4

commit 02561419db82b01ffb28df38000716c612988427
Author: John Grossman <johngro@google.com>
Date:   Tue May 10 14:00:21 2011 -0700

    Put in a hack for controling master volume in the policy manager.
    Fix initial master volume reporting.

    Change-Id: Ia6caf2bbc6083c5f99fab852baa40fff10fc5fc7

commit 549cdc3ba115dc654cdade261fb055c72c6cdb79
Author: John Grossman <johngro@google.com>
Date:   Wed May 4 11:46:17 2011 -0700

    Make certain the logic for computing the output stream mixing point is hardened
    against underflow and overflow when input and output sample rates don't match.

    Change-Id: I5ebea07c9938107b435bec7413418622767e4e16

commit 8043d8ed63f51e76d452d22be7d453d4a7794530
Author: Jason Simmons <jsimmons@google.com>
Date:   Wed Apr 27 18:06:27 2011 -0700

    Add the patch for timed audio support to the mono resampler

    Change-Id: I526f34ae9d1e8e3b0ed2fb05af3d024d5c5fe711

commit 2be89486ef23f0b0b0cc2dc25a4c0ee691043f00
Author: John Grossman <johngro@google.com>
Date:   Wed Apr 27 10:38:57 2011 -0700

    Extend the AudioHWInterface to allow it to specify the initial master volume used by AudioFlinger.

    Change-Id: I8823330801c927494cf7ca31a6b8f9264fbfbb26

commit ff89a4d5e37e6a05a2b03f79ab4e97833dd66393
Author: John Grossman <johngro@google.com>
Date:   Wed Apr 27 09:07:14 2011 -0700

    Fix an issue with inconsistent volume reporting.

    Changed masterVolume() to return the same value as the last call
    to setMasterVolume when the HW layer is implementing master
    volume control.  The masterVolume/setMasterVolume API seems to be
    an idea which was abandonded a long time ago; as of today the
    system only ever sets it to 1.0 at startup and then never changes
    it.  Until we can figure out how the concept of external
    amplifier gain control fits into the Android audio framework,
    Tungsten is exposing this API via a hack-tastic invoke back door
    in the TungstenRXPlayer and needs the getter/setter results to be
    consistent.

    Change-Id: I2ac730fa8fc9ee28c88f1a8e6f2e493eb5b65544

commit 086511b2d19cceb976747ac23e12b73fc7c28bea
Author: Jason Simmons <jsimmons@google.com>
Date:   Mon Apr 25 16:07:19 2011 -0700

    Add handling of timed audio tracks in the generic resampling mixer

    Change-Id: Ic3be1d21b1117f1b233808be543c28a0dcec4792

Change-Id: I6ec5d2bca9b8ebc0acd395a7dd92e1a48fcdfa9b
Signed-off-by: Mike J. Chen <mjchen@google.com>
Signed-off-by: John Grossman <johngro@google.com>
Signed-off-by: Jason Simmons <jsimmons@google.com>
2011-10-28 10:14:48 -04:00

515 lines
20 KiB
C++

/*
* Copyright (C) 2007 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#ifndef ANDROID_AUDIOTRACK_H
#define ANDROID_AUDIOTRACK_H
#include <stdint.h>
#include <sys/types.h>
#include <media/IAudioFlinger.h>
#include <media/IAudioTrack.h>
#include <media/AudioSystem.h>
#include <utils/RefBase.h>
#include <utils/Errors.h>
#include <binder/IInterface.h>
#include <binder/IMemory.h>
#include <utils/threads.h>
namespace android {
// ----------------------------------------------------------------------------
class audio_track_cblk_t;
// ----------------------------------------------------------------------------
class AudioTrack
{
public:
enum channel_index {
MONO = 0,
LEFT = 0,
RIGHT = 1
};
/* Events used by AudioTrack callback function (audio_track_cblk_t).
*/
enum event_type {
EVENT_MORE_DATA = 0, // Request to write more data to PCM buffer.
EVENT_UNDERRUN = 1, // PCM buffer underrun occured.
EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from loop start if loop count was not 0.
EVENT_MARKER = 3, // Playback head is at the specified marker position (See setMarkerPosition()).
EVENT_NEW_POS = 4, // Playback head is at a new position (See setPositionUpdatePeriod()).
EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer.
};
/* Create Buffer on the stack and pass it to obtainBuffer()
* and releaseBuffer().
*/
class Buffer
{
public:
enum {
MUTE = 0x00000001
};
uint32_t flags;
int format;
int channelCount; // will be removed in the future, do not use
size_t frameCount;
size_t size;
union {
void* raw;
short* i16;
int8_t* i8;
};
};
/* As a convenience, if a callback is supplied, a handler thread
* is automatically created with the appropriate priority. This thread
* invokes the callback when a new buffer becomes availlable or an underrun condition occurs.
* Parameters:
*
* event: type of event notified (see enum AudioTrack::event_type).
* user: Pointer to context for use by the callback receiver.
* info: Pointer to optional parameter according to event type:
* - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
* more bytes than indicated by 'size' field and update 'size' if less bytes are
* written.
* - EVENT_UNDERRUN: unused.
* - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
* - EVENT_MARKER: pointer to an uin32_t containing the marker position in frames.
* - EVENT_NEW_POS: pointer to an uin32_t containing the new position in frames.
* - EVENT_BUFFER_END: unused.
*/
typedef void (*callback_t)(int event, void* user, void *info);
/* Returns the minimum frame count required for the successful creation of
* an AudioTrack object.
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
*/
static status_t getMinFrameCount(int* frameCount,
int streamType =-1,
uint32_t sampleRate = 0);
/* Constructs an uninitialized AudioTrack. No connection with
* AudioFlinger takes place.
*/
AudioTrack();
/* Creates an audio track and registers it with AudioFlinger.
* Once created, the track needs to be started before it can be used.
* Unspecified values are set to the audio hardware's current
* values.
*
* Parameters:
*
* streamType: Select the type of audio stream this track is attached to
* (e.g. AUDIO_STREAM_MUSIC).
* sampleRate: Track sampling rate in Hz.
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
* channelMask: Channel mask: see audio_channels_t.
* frameCount: Total size of track PCM buffer in frames. This defines the
* latency of the track.
* flags: Reserved for future use.
* cbf: Callback function. If not null, this function is called periodically
* to request new PCM data.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames have been comsumed from track input buffer.
* user Context for use by the callback receiver.
*/
AudioTrack( int streamType,
uint32_t sampleRate = 0,
int format = 0,
int channelMask = 0,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
void* user = 0,
int notificationFrames = 0,
int sessionId = 0);
/* Creates an audio track and registers it with AudioFlinger. With this constructor,
* The PCM data to be rendered by AudioTrack is passed in a shared memory buffer
* identified by the argument sharedBuffer. This prototype is for static buffer playback.
* PCM data must be present into memory before the AudioTrack is started.
* The Write() and Flush() methods are not supported in this case.
* It is recommented to pass a callback function to be notified of playback end by an
* EVENT_UNDERRUN event.
*/
AudioTrack( int streamType,
uint32_t sampleRate = 0,
int format = 0,
int channelMask = 0,
const sp<IMemory>& sharedBuffer = 0,
uint32_t flags = 0,
callback_t cbf = 0,
void* user = 0,
int notificationFrames = 0,
int sessionId = 0);
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources assotiated with the AudioTrack.
*/
~AudioTrack();
/* Initialize an uninitialized AudioTrack.
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful intialization
* - INVALID_OPERATION: AudioTrack is already intitialized
* - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* */
status_t set(int streamType =-1,
uint32_t sampleRate = 0,
int format = 0,
int channelMask = 0,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
void* user = 0,
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
int sessionId = 0);
/* Result of constructing the AudioTrack. This must be checked
* before using any AudioTrack API (except for set()), using
* an uninitialized AudioTrack produces undefined results.
* See set() method above for possible return codes.
*/
status_t initCheck() const;
/* Returns this track's latency in milliseconds.
* This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
* and audio hardware driver.
*/
uint32_t latency() const;
/* getters, see constructor */
int streamType() const;
int format() const;
int channelCount() const;
uint32_t frameCount() const;
int frameSize() const;
sp<IMemory>& sharedBuffer();
/* After it's created the track is not active. Call start() to
* make it active. If set, the callback will start being called.
*/
void start();
/* Stop a track. If set, the callback will cease being called and
* obtainBuffer returns STOPPED. Note that obtainBuffer() still works
* and will fill up buffers until the pool is exhausted.
*/
void stop();
bool stopped() const;
/* flush a stopped track. All pending buffers are discarded.
* This function has no effect if the track is not stoped.
*/
void flush();
/* Pause a track. If set, the callback will cease being called and
* obtainBuffer returns STOPPED. Note that obtainBuffer() still works
* and will fill up buffers until the pool is exhausted.
*/
void pause();
/* mute or unmutes this track.
* While mutted, the callback, if set, is still called.
*/
void mute(bool);
bool muted() const;
/* set volume for this track, mostly used for games' sound effects
* left and right volumes. Levels must be <= 1.0.
*/
status_t setVolume(float left, float right);
void getVolume(float* left, float* right);
/* set the send level for this track. An auxiliary effect should be attached
* to the track with attachEffect(). Level must be <= 1.0.
*/
status_t setAuxEffectSendLevel(float level);
void getAuxEffectSendLevel(float* level);
/* set sample rate for this track, mostly used for games' sound effects
*/
status_t setSampleRate(int sampleRate);
uint32_t getSampleRate();
/* Enables looping and sets the start and end points of looping.
*
* Parameters:
*
* loopStart: loop start expressed as the number of PCM frames played since AudioTrack start.
* loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start.
* loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any pending or
* active loop. loopCount = -1 means infinite looping.
*
* For proper operation the following condition must be respected:
* (loopEnd-loopStart) <= framecount()
*/
status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
status_t getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount);
/* Sets marker position. When playback reaches the number of frames specified, a callback with event
* type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker notification
* callback.
* If the AudioTrack has been opened with no callback function associated, the operation will fail.
*
* Parameters:
*
* marker: marker position expressed in frames.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the AudioTrack has no callback installed.
*/
status_t setMarkerPosition(uint32_t marker);
status_t getMarkerPosition(uint32_t *marker);
/* Sets position update period. Every time the number of frames specified has been played,
* a callback with event type EVENT_NEW_POS is called.
* Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
* callback.
* If the AudioTrack has been opened with no callback function associated, the operation will fail.
*
* Parameters:
*
* updatePeriod: position update notification period expressed in frames.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the AudioTrack has no callback installed.
*/
status_t setPositionUpdatePeriod(uint32_t updatePeriod);
status_t getPositionUpdatePeriod(uint32_t *updatePeriod);
/* Sets playback head position within AudioTrack buffer. The new position is specified
* in number of frames.
* This method must be called with the AudioTrack in paused or stopped state.
* Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
* Therefore using this method makes sense only when playing a "static" audio buffer
* as opposed to streaming.
* The getPosition() method on the other hand returns the total number of frames played since
* playback start.
*
* Parameters:
*
* position: New playback head position within AudioTrack buffer.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the AudioTrack is not stopped.
* - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer
*/
status_t setPosition(uint32_t position);
status_t getPosition(uint32_t *position);
/* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
* rewriting the buffer before restarting playback after a stop.
* This method must be called with the AudioTrack in paused or stopped state.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the AudioTrack is not stopped.
*/
status_t reload();
/* returns a handle on the audio output used by this AudioTrack.
*
* Parameters:
* none.
*
* Returned value:
* handle on audio hardware output
*/
audio_io_handle_t getOutput();
/* returns the unique ID associated to this track.
*
* Parameters:
* none.
*
* Returned value:
* AudioTrack ID.
*/
int getSessionId();
/* Attach track auxiliary output to specified effect. Used effectId = 0
* to detach track from effect.
*
* Parameters:
*
* effectId: effectId obtained from AudioEffect::id().
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the effect is not an auxiliary effect.
* - BAD_VALUE: The specified effect ID is invalid
*/
status_t attachAuxEffect(int effectId);
/* obtains a buffer of "frameCount" frames. The buffer must be
* filled entirely. If the track is stopped, obtainBuffer() returns
* STOPPED instead of NO_ERROR as long as there are buffers availlable,
* at which point NO_MORE_BUFFERS is returned.
* Buffers will be returned until the pool (buffercount())
* is exhausted, at which point obtainBuffer() will either block
* or return WOULD_BLOCK depending on the value of the "blocking"
* parameter.
*/
enum {
NO_MORE_BUFFERS = 0x80000001,
STOPPED = 1
};
status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
void releaseBuffer(Buffer* audioBuffer);
/* As a convenience we provide a write() interface to the audio buffer.
* This is implemented on top of lockBuffer/unlockBuffer. For best
* performance
*
*/
ssize_t write(const void* buffer, size_t size);
/*
* Dumps the state of an audio track.
*/
status_t dump(int fd, const Vector<String16>& args) const;
protected:
/* copying audio tracks is not allowed */
AudioTrack(const AudioTrack& other);
AudioTrack& operator = (const AudioTrack& other);
/* a small internal class to handle the callback */
class AudioTrackThread : public Thread
{
public:
AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false);
private:
friend class AudioTrack;
virtual bool threadLoop();
virtual status_t readyToRun();
virtual void onFirstRef();
AudioTrack& mReceiver;
Mutex mLock;
};
bool processAudioBuffer(const sp<AudioTrackThread>& thread);
status_t createTrack_l(int streamType,
uint32_t sampleRate,
uint32_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool enforceFrameCount);
void flush_l();
status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
audio_io_handle_t getOutput_l();
status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
sp<AudioTrackThread> mAudioTrackThread;
float mVolume[2];
float mSendLevel;
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
uint32_t mFormat;
uint8_t mStreamType;
uint8_t mChannelCount;
uint8_t mMuted;
uint8_t mReserved;
uint32_t mChannelMask;
status_t mStatus;
uint32_t mLatency;
volatile int32_t mActive;
callback_t mCbf;
void* mUserData;
uint32_t mNotificationFramesReq; // requested number of frames between each notification callback
uint32_t mNotificationFramesAct; // actual number of frames between each notification callback
sp<IMemory> mSharedBuffer;
int mLoopCount;
uint32_t mRemainingFrames;
uint32_t mMarkerPosition;
bool mMarkerReached;
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
bool mFlushed; // FIXME will be made obsolete by making flush() synchronous
uint32_t mFlags;
int mSessionId;
int mAuxEffectId;
Mutex mLock;
status_t mRestoreStatus;
bool mIsTimed;
};
class TimedAudioTrack : public AudioTrack
{
public:
TimedAudioTrack();
/* allocate a shared memory buffer that can be passed to queueTimedBuffer */
status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
/* queue a buffer obtained via allocateTimedBuffer for playback at the
given timestamp */
status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
/* define a transform between media time and either common time or
local time */
enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
status_t setMediaTimeTransform(const LinearTransform& xform,
TargetTimeline target);
};
}; // namespace android
#endif // ANDROID_AUDIOTRACK_H