diff options
Diffstat (limited to 'include/media')
46 files changed, 1229 insertions, 174 deletions
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index f70d981..7be2c3e 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -42,8 +42,7 @@ public: EVENT_MORE_DATA = 0, // Request to read available data from buffer. // If this event is delivered but the callback handler // does not want to read the available data, the handler must - // explicitly - // ignore the event by setting frameCount to zero. + // explicitly ignore the event by setting frameCount to zero. EVENT_OVERRUN = 1, // Buffer overrun occurred. EVENT_MARKER = 2, // Record head is at the specified marker position // (See setMarkerPosition()). @@ -53,7 +52,7 @@ public: // voluntary invalidation by mediaserver, or mediaserver crash. }; - /* Client should declare Buffer on the stack and pass address to obtainBuffer() + /* Client should declare a Buffer and pass address to obtainBuffer() * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. */ @@ -62,20 +61,25 @@ public: public: // FIXME use m prefix size_t frameCount; // number of sample frames corresponding to size; - // on input it is the number of frames available, - // on output is the number of frames actually drained - // (currently ignored but will make the primary field in future) + // on input to obtainBuffer() it is the number of frames desired + // on output from obtainBuffer() it is the number of available + // frames to be read + // on input to releaseBuffer() it is currently ignored size_t size; // input/output in bytes == frameCount * frameSize - // on output is the number of bytes actually drained - // FIXME this is redundant with respect to frameCount, - // and TRANSFER_OBTAIN mode is broken for 8-bit data - // since we don't define the frame format + // on input to obtainBuffer() it is ignored + // on output from obtainBuffer() it is the number of available + // bytes to be read, which is frameCount * frameSize + // on input to releaseBuffer() it is the number of bytes to + // release + // FIXME This is redundant with respect to frameCount. Consider + // removing size and making frameCount the primary field. union { void* raw; short* i16; // signed 16-bit int8_t* i8; // unsigned 8-bit, offset by 0x80 + // input to obtainBuffer(): unused, output: pointer to buffer }; }; @@ -88,8 +92,8 @@ public: * user: Pointer to context for use by the callback receiver. * info: Pointer to optional parameter according to event type: * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read - * more bytes than indicated by 'size' field and update 'size' if fewer bytes are - * consumed. + * more bytes than indicated by 'size' field and update 'size' if + * fewer bytes are consumed. * - EVENT_OVERRUN: unused. * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. @@ -106,6 +110,7 @@ public: * - BAD_VALUE: unsupported configuration * frameCount is guaranteed to be non-zero if status is NO_ERROR, * and is undefined otherwise. + * FIXME This API assumes a route, and so should be deprecated. */ static status_t getMinFrameCount(size_t* frameCount, @@ -118,7 +123,7 @@ public: enum transfer_type { TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters TRANSFER_CALLBACK, // callback EVENT_MORE_DATA - TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() + TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() TRANSFER_SYNC, // synchronous read() }; @@ -144,15 +149,16 @@ public: * be larger if the requested size is not compatible with current audio HAL * latency. Zero means to use a default value. * cbf: Callback function. If not null, this function is called periodically - * to consume new data and inform of marker, position updates, etc. + * to consume new data in TRANSFER_CALLBACK mode + * and inform of marker, position updates, etc. * user: Context for use by the callback receiver. * notificationFrames: The callback function is called each time notificationFrames PCM * frames are ready in record track output buffer. * sessionId: Not yet supported. * transferType: How data is transferred from AudioRecord. * flags: See comments on audio_input_flags_t in <system/audio.h> + * pAttributes: If not NULL, supersedes inputSource for use case selection. * threadCanCallJava: Not present in parameter list, and so is fixed at false. - * pAttributes: if not NULL, supersedes inputSource for use case selection */ AudioRecord(audio_source_t inputSource, @@ -177,6 +183,7 @@ public: /* Initialize an AudioRecord that was created using the AudioRecord() constructor. * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. + * set() is not multi-thread safe. * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful intialization * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use @@ -211,7 +218,7 @@ public: status_t initCheck() const { return mStatus; } /* Returns this track's estimated latency in milliseconds. - * This includes the latency due to AudioRecord buffer size, + * This includes the latency due to AudioRecord buffer size, resampling if applicable, * and audio hardware driver. */ uint32_t latency() const { return mLatency; } @@ -243,11 +250,6 @@ public: */ uint32_t getSampleRate() const { return mSampleRate; } - /* Return the notification frame count. - * This is approximately how often the callback is invoked, for transfer type TRANSFER_CALLBACK. - */ - size_t notificationFrames() const { return mNotificationFramesAct; } - /* Sets marker position. When record reaches the number of frames specified, * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition * with marker == 0 cancels marker notification callback. @@ -309,7 +311,12 @@ public: * Returned value: * handle on audio hardware input */ - audio_io_handle_t getInput() const; +// FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp + audio_io_handle_t getInput() const __attribute__((__deprecated__)) + { return getInputPrivate(); } +private: + audio_io_handle_t getInputPrivate() const; +public: /* Returns the audio session ID associated with this AudioRecord. * @@ -323,7 +330,8 @@ public: */ int getSessionId() const { return mSessionId; } - /* Obtains a buffer of up to "audioBuffer->frameCount" full frames. + /* Public API for TRANSFER_OBTAIN mode. + * Obtains a buffer of up to "audioBuffer->frameCount" full frames. * After draining these frames of data, the caller should release them with releaseBuffer(). * If the track buffer is not empty, obtainBuffer() returns as many contiguous * full frames as are available immediately. @@ -336,9 +344,6 @@ public: * or return WOULD_BLOCK depending on the value of the "waitCount" * parameter. * - * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, - * which should use read() or callback EVENT_MORE_DATA instead. - * * Interpretation of waitCount: * +n limits wait time to n * WAIT_PERIOD_MS, * -1 causes an (almost) infinite wait time, @@ -347,6 +352,8 @@ public: * Buffer fields * On entry: * frameCount number of frames requested + * size ignored + * raw ignored * After error return: * frameCount 0 * size 0 @@ -357,9 +364,7 @@ public: * raw pointer to the buffer */ - /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ - status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) - __attribute__((__deprecated__)); + status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); private: /* If nonContig is non-NULL, it is an output parameter that will be set to the number of @@ -372,9 +377,15 @@ private: struct timespec *elapsed = NULL, size_t *nonContig = NULL); public: - /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */ - // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed - void releaseBuffer(Buffer* audioBuffer); + /* Public API for TRANSFER_OBTAIN mode. + * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. + * + * Buffer fields: + * frameCount currently ignored but recommend to set to actual number of frames consumed + * size actual number of bytes consumed, must be multiple of frameSize + * raw ignored + */ + void releaseBuffer(const Buffer* audioBuffer); /* As a convenience we provide a read() interface to the audio buffer. * Input parameter 'size' is in byte units. @@ -386,8 +397,11 @@ public: * WOULD_BLOCK when obtainBuffer() returns same, or * AudioRecord was stopped during the read * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). + * Default behavior is to only return when all data has been transferred. Set 'blocking' to + * false for the method to return immediately without waiting to try multiple times to read + * the full content of the buffer. */ - ssize_t read(void* buffer, size_t size); + ssize_t read(void* buffer, size_t size, bool blocking = true); /* Return the number of input frames lost in the audio driver since the last call of this * function. Audio driver is expected to reset the value to 0 and restart counting upon @@ -416,6 +430,7 @@ private: void pause(); // suspend thread from execution at next loop boundary void resume(); // allow thread to execute, if not requested to exit + void wake(); // wake to handle changed notification conditions. private: void pauseInternal(nsecs_t ns = 0LL); @@ -430,7 +445,9 @@ private: bool mPaused; // whether thread is requested to pause at next loop entry bool mPausedInt; // whether thread internally requests pause nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored - bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request + bool mIgnoreNextPausedInt; // skip any internal pause and go immediately + // to processAudioBuffer() as state may have changed + // since pause time calculated. }; // body of AudioRecordThread::threadLoop() @@ -458,7 +475,7 @@ private: bool mActive; // for client callback handler - callback_t mCbf; // callback handler for events, or NULL + callback_t mCbf; // callback handler for events, or NULL void* mUserData; // for notification APIs @@ -475,10 +492,10 @@ private: bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() uint32_t mObservedSequence; // last observed value of mSequence - uint32_t mMarkerPosition; // in wrapping (overflow) frame units + uint32_t mMarkerPosition; // in wrapping (overflow) frame units bool mMarkerReached; - uint32_t mNewPosition; // in frames - uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS + uint32_t mNewPosition; // in frames + uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS status_t mStatus; diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h index 97847a0..b705efa 100644 --- a/include/media/AudioResamplerPublic.h +++ b/include/media/AudioResamplerPublic.h @@ -26,4 +26,17 @@ // TODO: replace with an API #define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256 +// Returns the source frames needed to resample to destination frames. This is not a precise +// value and depends on the resampler (and possibly how it handles rounding internally). +// Nevertheless, this should be an upper bound on the requirements of the resampler. +// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which +// may not be true if the resampler is asynchronous. +static inline size_t sourceFramesNeeded( + uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) { + // +1 for rounding - always do this even if matched ratio (resampler may use phases not ratio) + // +1 for additional sample needed for interpolation + return srcSampleRate == dstSampleRate ? dstFramesRequired : + size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1); +} + #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index 843a354..f5db1bb 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -98,10 +98,13 @@ public: // Returned samplingRate and frameCount output values are guaranteed // to be non-zero if status == NO_ERROR + // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t stream); + // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputFrameCount(size_t* frameCount, audio_stream_type_t stream); + // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream); static status_t getSamplingRate(audio_io_handle_t output, @@ -110,19 +113,20 @@ public: // audio_stream->get_buffer_size()/audio_stream_out_frame_size() static status_t getFrameCount(audio_io_handle_t output, size_t* frameCount); - // returns the audio output stream latency in ms. Corresponds to + // returns the audio output latency in ms. Corresponds to // audio_stream_out->get_latency() static status_t getLatency(audio_io_handle_t output, uint32_t* latency); // return status NO_ERROR implies *buffSize > 0 + // FIXME This API assumes a route, and so should deprecated. static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t* buffSize); static status_t setVoiceVolume(float volume); // return the number of audio frames written by AudioFlinger to audio HAL and - // audio dsp to DAC since the specified output I/O handle has exited standby. + // audio dsp to DAC since the specified output has exited standby. // returned status (from utils/Errors.h) can be: // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data // - INVALID_OPERATION: Not supported on current hardware platform @@ -201,7 +205,7 @@ public: // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) // static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, - const char *device_address); + const char *device_address, const char *device_name); static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address); static status_t setPhoneState(audio_mode_t state); @@ -342,7 +346,8 @@ public: }; - static void setAudioPortCallback(sp<AudioPortCallback> callBack); + static status_t addAudioPortCallback(const sp<AudioPortCallback>& callBack); + static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callBack); private: @@ -369,12 +374,19 @@ private: AudioPolicyServiceClient() { } + status_t addAudioPortCallback(const sp<AudioPortCallback>& callBack); + status_t removeAudioPortCallback(const sp<AudioPortCallback>& callBack); + // DeathRecipient virtual void binderDied(const wp<IBinder>& who); // IAudioPolicyServiceClient virtual void onAudioPortListUpdate(); virtual void onAudioPatchListUpdate(); + + private: + Mutex mLock; + Vector <sp <AudioPortCallback> > mAudioPortCallbacks; }; static sp<AudioFlingerClient> gAudioFlingerClient; @@ -386,7 +398,6 @@ private: static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat, // gPrevInChannelMask and gInBuffSize static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient - static Mutex gLockAPC; // protects gAudioPortCallback static sp<IAudioFlinger> gAudioFlinger; static audio_error_callback gAudioErrorCallback; @@ -401,8 +412,6 @@ private: // list of output descriptors containing cached parameters // (sampling rate, framecount, channel count...) static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; - - static sp<AudioPortCallback> gAudioPortCallback; }; }; // namespace android diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index fd51b8f..d9b7057 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -63,7 +63,7 @@ public: // See AudioTimestamp for the information included with event. }; - /* Client should declare Buffer on the stack and pass address to obtainBuffer() + /* Client should declare a Buffer and pass the address to obtainBuffer() * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. */ @@ -72,22 +72,26 @@ public: public: // FIXME use m prefix size_t frameCount; // number of sample frames corresponding to size; - // on input it is the number of frames desired, - // on output is the number of frames actually filled - // (currently ignored, but will make the primary field in future) + // on input to obtainBuffer() it is the number of frames desired, + // on output from obtainBuffer() it is the number of available + // [empty slots for] frames to be filled + // on input to releaseBuffer() it is currently ignored size_t size; // input/output in bytes == frameCount * frameSize - // on input it is unused - // on output is the number of bytes actually filled - // FIXME this is redundant with respect to frameCount, - // and TRANSFER_OBTAIN mode is broken for 8-bit data - // since we don't define the frame format + // on input to obtainBuffer() it is ignored + // on output from obtainBuffer() it is the number of available + // [empty slots for] bytes to be filled, + // which is frameCount * frameSize + // on input to releaseBuffer() it is the number of bytes to + // release + // FIXME This is redundant with respect to frameCount. Consider + // removing size and making frameCount the primary field. union { void* raw; short* i16; // signed 16-bit int8_t* i8; // unsigned 8-bit, offset by 0x80 - }; // input: unused, output: pointer to buffer + }; // input to obtainBuffer(): unused, output: pointer to buffer }; /* As a convenience, if a callback is supplied, a handler thread @@ -121,6 +125,7 @@ public: * - BAD_VALUE: unsupported configuration * frameCount is guaranteed to be non-zero if status is NO_ERROR, * and is undefined otherwise. + * FIXME This API assumes a route, and so should be deprecated. */ static status_t getMinFrameCount(size_t* frameCount, @@ -132,7 +137,7 @@ public: enum transfer_type { TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters TRANSFER_CALLBACK, // callback EVENT_MORE_DATA - TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() + TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() TRANSFER_SYNC, // synchronous write() TRANSFER_SHARED, // shared memory }; @@ -145,18 +150,15 @@ public: /* Creates an AudioTrack object and registers it with AudioFlinger. * Once created, the track needs to be started before it can be used. * Unspecified values are set to appropriate default values. - * With this constructor, the track is configured for streaming mode. - * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. - * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. * * Parameters: * * streamType: Select the type of audio stream this track is attached to * (e.g. AUDIO_STREAM_MUSIC). * sampleRate: Data source sampling rate in Hz. - * format: Audio format. For mixed tracks, any PCM format supported by server is OK - * or AUDIO_FORMAT_PCM_8_BIT which is handled on client side. For direct - * and offloaded tracks, the possible format(s) depends on the output sink. + * format: Audio format. For mixed tracks, any PCM format supported by server is OK. + * For direct and offloaded tracks, the possible format(s) depends on the + * output sink. * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. * frameCount: Minimum size of track PCM buffer in frames. This defines the * application's contribution to the @@ -165,20 +167,28 @@ public: * configuration. Zero means to use a default value. * flags: See comments on audio_output_flags_t in <system/audio.h>. * cbf: Callback function. If not null, this function is called periodically - * to provide new data and inform of marker, position updates, etc. + * to provide new data in TRANSFER_CALLBACK mode + * and inform of marker, position updates, etc. * user: Context for use by the callback receiver. * notificationFrames: The callback function is called each time notificationFrames PCM * frames have been consumed from track input buffer. * This is expressed in units of frames at the initial source sample rate. * sessionId: Specific session ID, or zero to use default. * transferType: How data is transferred to AudioTrack. + * offloadInfo: If not NULL, provides offload parameters for + * AudioSystem::getOutputForAttr(). + * uid: User ID of the app which initially requested this AudioTrack + * for power management tracking, or -1 for current user ID. + * pid: Process ID of the app which initially requested this AudioTrack + * for power management tracking, or -1 for current process ID. + * pAttributes: If not NULL, supersedes streamType for use case selection. * threadCanCallJava: Not present in parameter list, and so is fixed at false. */ AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, - audio_channel_mask_t, + audio_channel_mask_t channelMask, size_t frameCount = 0, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, callback_t cbf = NULL, @@ -193,9 +203,10 @@ public: /* Creates an audio track and registers it with AudioFlinger. * With this constructor, the track is configured for static buffer mode. - * The format must not be 8-bit linear PCM. * Data to be rendered is passed in a shared memory buffer - * identified by the argument sharedBuffer, which must be non-0. + * identified by the argument sharedBuffer, which should be non-0. + * If sharedBuffer is zero, this constructor is equivalent to the previous constructor + * but without the ability to specify a non-zero value for the frameCount parameter. * The memory should be initialized to the desired data before calling start(). * The write() method is not supported in this case. * It is recommended to pass a callback function to be notified of playback end by an @@ -227,6 +238,7 @@ public: /* Initialize an AudioTrack that was created using the AudioTrack() constructor. * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. + * set() is not multi-thread safe. * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful initialization * - INVALID_OPERATION: AudioTrack is already initialized @@ -461,7 +473,9 @@ public: * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the * track needed to be re-created but that failed */ +private: audio_io_handle_t getOutput() const; +public: /* Returns the unique session ID associated with this track. * @@ -487,10 +501,18 @@ public: */ status_t attachAuxEffect(int effectId); - /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. + /* Public API for TRANSFER_OBTAIN mode. + * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. * After filling these slots with data, the caller should release them with releaseBuffer(). * If the track buffer is not full, obtainBuffer() returns as many contiguous * [empty slots for] frames as are available immediately. + * + * If nonContig is non-NULL, it is an output parameter that will be set to the number of + * additional non-contiguous frames that are predicted to be available immediately, + * if the client were to release the first frames and then call obtainBuffer() again. + * This value is only a prediction, and needs to be confirmed. + * It will be set to zero for an error return. + * * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK * regardless of the value of waitCount. * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a @@ -499,10 +521,6 @@ public: * is exhausted, at which point obtainBuffer() will either block * or return WOULD_BLOCK depending on the value of the "waitCount" * parameter. - * Each sample is 16-bit signed PCM. - * - * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, - * which should use write() or callback EVENT_MORE_DATA instead. * * Interpretation of waitCount: * +n limits wait time to n * WAIT_PERIOD_MS, @@ -511,24 +529,27 @@ public: * * Buffer fields * On entry: - * frameCount number of frames requested + * frameCount number of [empty slots for] frames requested + * size ignored + * raw ignored * After error return: * frameCount 0 * size 0 * raw undefined * After successful return: - * frameCount actual number of frames available, <= number requested + * frameCount actual number of [empty slots for] frames available, <= number requested * size actual number of bytes available * raw pointer to the buffer */ - - /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ - status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) - __attribute__((__deprecated__)); + status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, + size_t *nonContig = NULL); private: /* If nonContig is non-NULL, it is an output parameter that will be set to the number of - * additional non-contiguous frames that are available immediately. + * additional non-contiguous frames that are predicted to be available immediately, + * if the client were to release the first frames and then call obtainBuffer() again. + * This value is only a prediction, and needs to be confirmed. + * It will be set to zero for an error return. * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), * in case the requested amount of frames is in two or more non-contiguous regions. * FIXME requested and elapsed are both relative times. Consider changing to absolute time. @@ -537,9 +558,15 @@ private: struct timespec *elapsed = NULL, size_t *nonContig = NULL); public: - /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ - // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed - void releaseBuffer(Buffer* audioBuffer); + /* Public API for TRANSFER_OBTAIN mode. + * Release a filled buffer of frames for AudioFlinger to process. + * + * Buffer fields: + * frameCount currently ignored but recommend to set to actual number of frames filled + * size actual number of bytes filled, must be multiple of frameSize + * raw ignored + */ + void releaseBuffer(const Buffer* audioBuffer); /* As a convenience we provide a write() interface to the audio buffer. * Input parameter 'size' is in byte units. @@ -551,7 +578,7 @@ public: * WOULD_BLOCK when obtainBuffer() returns same, or * AudioTrack was stopped during the write * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). - * Default behavior is to only return until all data has been transferred. Set 'blocking' to + * Default behavior is to only return when all data has been transferred. Set 'blocking' to * false for the method to return immediately without waiting to try multiple times to write * the full content of the buffer. */ @@ -559,6 +586,7 @@ public: /* * Dumps the state of an audio track. + * Not a general-purpose API; intended only for use by media player service to dump its tracks. */ status_t dump(int fd, const Vector<String16>& args) const; @@ -600,8 +628,6 @@ protected: AudioTrack(const AudioTrack& other); AudioTrack& operator = (const AudioTrack& other); - void setAttributesFromStreamType(audio_stream_type_t streamType); - /* a small internal class to handle the callback */ class AudioTrackThread : public Thread { @@ -614,6 +640,7 @@ protected: void pause(); // suspend thread from execution at next loop boundary void resume(); // allow thread to execute, if not requested to exit + void wake(); // wake to handle changed notification conditions. private: void pauseInternal(nsecs_t ns = 0LL); @@ -628,7 +655,9 @@ protected: bool mPaused; // whether thread is requested to pause at next loop entry bool mPausedInt; // whether thread internally requests pause nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored - bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request + bool mIgnoreNextPausedInt; // skip any internal pause and go immediately + // to processAudioBuffer() as state may have changed + // since pause time calculated. }; // body of AudioTrackThread::threadLoop() @@ -641,10 +670,6 @@ protected: static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; nsecs_t processAudioBuffer(); - bool isOffloaded() const; - bool isDirect() const; - bool isOffloadedOrDirect() const; - // caller must hold lock on mLock for all _l methods status_t createTrack_l(); @@ -657,6 +682,10 @@ protected: // FIXME enum is faster than strcmp() for parameter 'from' status_t restoreTrack_l(const char *from); + bool isOffloaded() const; + bool isDirect() const; + bool isOffloadedOrDirect() const; + bool isOffloaded_l() const { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } @@ -680,7 +709,7 @@ protected: float mVolume[2]; float mSendLevel; - mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. + mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it size_t mFrameCount; // corresponds to current IAudioTrack, value is // reported back by AudioFlinger to the client size_t mReqFrameCount; // frame count to request the first or next time @@ -698,10 +727,7 @@ protected: const audio_offload_info_t* mOffloadInfo; audio_attributes_t mAttributes; - // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's - // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. - size_t mFrameSize; // app-level frame size - size_t mFrameSizeAF; // AudioFlinger frame size + size_t mFrameSize; // frame size in bytes status_t mStatus; @@ -732,17 +758,25 @@ protected: bool mRefreshRemaining; // processAudioBuffer() should refresh // mRemainingFrames and mRetryOnPartialBuffer + // used for static track cbf and restoration + int32_t mLoopCount; // last setLoop loopCount; zero means disabled + uint32_t mLoopStart; // last setLoop loopStart + uint32_t mLoopEnd; // last setLoop loopEnd + int32_t mLoopCountNotified; // the last loopCount notified by callback. + // mLoopCountNotified counts down, matching + // the remaining loop count for static track + // playback. + // These are private to processAudioBuffer(), and are not protected by a lock uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() uint32_t mObservedSequence; // last observed value of mSequence - uint32_t mLoopPeriod; // in frames, zero means looping is disabled - uint32_t mMarkerPosition; // in wrapping (overflow) frame units bool mMarkerReached; uint32_t mNewPosition; // in frames uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS + uint32_t mServer; // in frames, last known mProxy->getPosition() // which is count of frames consumed by server, // reset by new IAudioTrack, diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h index b1ed7b0..64a3212 100644 --- a/include/media/EffectsFactoryApi.h +++ b/include/media/EffectsFactoryApi.h @@ -171,6 +171,8 @@ int EffectGetDescriptor(const effect_uuid_t *pEffectUuid, effect_descriptor_t *p //////////////////////////////////////////////////////////////////////////////// int EffectIsNullUuid(const effect_uuid_t *pEffectUuid); +int EffectDumpEffects(int fd); + #if __cplusplus } // extern "C" #endif diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h index 31a14f0..f927a80 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -94,6 +94,8 @@ public: sp<IMemory>& buffers, // return value 0 means it follows cblk status_t *status) = 0; + // FIXME Surprisingly, sampleRate/format/frameCount/latency don't work for input handles + /* query the audio hardware state. This state never changes, * and therefore can be cached. */ @@ -142,6 +144,7 @@ public: virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0; // retrieve the audio recording buffer size + // FIXME This API assumes a route, and so should be deprecated. virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) const = 0; diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h index c98c475..fecc6f1 100644 --- a/include/media/IAudioPolicyService.h +++ b/include/media/IAudioPolicyService.h @@ -44,7 +44,8 @@ public: // virtual status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, - const char *device_address) = 0; + const char *device_address, + const char *device_name) = 0; virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address) = 0; virtual status_t setPhoneState(audio_mode_t state) = 0; diff --git a/include/media/IDrm.h b/include/media/IDrm.h index affcbd7..9449beb 100644 --- a/include/media/IDrm.h +++ b/include/media/IDrm.h @@ -47,7 +47,8 @@ struct IDrm : public IInterface { Vector<uint8_t> const &initData, String8 const &mimeType, DrmPlugin::KeyType keyType, KeyedVector<String8, String8> const &optionalParameters, - Vector<uint8_t> &request, String8 &defaultUrl) = 0; + Vector<uint8_t> &request, String8 &defaultUrl, + DrmPlugin::KeyRequestType *keyRequestType) = 0; virtual status_t provideKeyResponse(Vector<uint8_t> const &sessionId, Vector<uint8_t> const &response, diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h index db62cd5..4153c25 100644 --- a/include/media/IMediaPlayer.h +++ b/include/media/IMediaPlayer.h @@ -56,6 +56,7 @@ public: virtual status_t stop() = 0; virtual status_t pause() = 0; virtual status_t isPlaying(bool* state) = 0; + virtual status_t setPlaybackRate(float rate) = 0; virtual status_t seekTo(int msec) = 0; virtual status_t getCurrentPosition(int* msec) = 0; virtual status_t getDuration(int* msec) = 0; diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h index 67b599a..49a3d61 100644 --- a/include/media/IMediaPlayerService.h +++ b/include/media/IMediaPlayerService.h @@ -49,7 +49,8 @@ public: virtual sp<IMediaRecorder> createMediaRecorder() = 0; virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0; - virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0; + virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) + = 0; virtual sp<IOMX> getOMX() = 0; virtual sp<ICrypto> makeCrypto() = 0; diff --git a/include/media/IMediaRecorder.h b/include/media/IMediaRecorder.h index 3e67550..509c06b 100644 --- a/include/media/IMediaRecorder.h +++ b/include/media/IMediaRecorder.h @@ -41,7 +41,6 @@ public: virtual status_t setOutputFormat(int of) = 0; virtual status_t setVideoEncoder(int ve) = 0; virtual status_t setAudioEncoder(int ae) = 0; - virtual status_t setOutputFile(const char* path) = 0; virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0; virtual status_t setVideoSize(int width, int height) = 0; virtual status_t setVideoFrameRate(int frames_per_second) = 0; diff --git a/include/media/IOMX.h b/include/media/IOMX.h index 627f23b..6def65b 100644 --- a/include/media/IOMX.h +++ b/include/media/IOMX.h @@ -147,6 +147,7 @@ public: INTERNAL_OPTION_SUSPEND, // data is a bool INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY, // data is an int64_t INTERNAL_OPTION_MAX_TIMESTAMP_GAP, // data is int64_t + INTERNAL_OPTION_MAX_FPS, // data is float INTERNAL_OPTION_START_TIME, // data is an int64_t INTERNAL_OPTION_TIME_LAPSE, // data is an int64_t[2] }; diff --git a/include/media/IResourceManagerClient.h b/include/media/IResourceManagerClient.h new file mode 100644 index 0000000..3587aea --- /dev/null +++ b/include/media/IResourceManagerClient.h @@ -0,0 +1,47 @@ +/* + * Copyright 2015 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_IRESOURCEMANAGERCLIENT_H +#define ANDROID_IRESOURCEMANAGERCLIENT_H + +#include <utils/RefBase.h> +#include <binder/IInterface.h> +#include <binder/Parcel.h> + +namespace android { + +class IResourceManagerClient: public IInterface +{ +public: + DECLARE_META_INTERFACE(ResourceManagerClient); + + virtual bool reclaimResource() = 0; +}; + +// ---------------------------------------------------------------------------- + +class BnResourceManagerClient: public BnInterface<IResourceManagerClient> +{ +public: + virtual status_t onTransact(uint32_t code, + const Parcel &data, + Parcel *reply, + uint32_t flags = 0); +}; + +}; // namespace android + +#endif // ANDROID_IRESOURCEMANAGERCLIENT_H diff --git a/include/media/IResourceManagerService.h b/include/media/IResourceManagerService.h new file mode 100644 index 0000000..067392c --- /dev/null +++ b/include/media/IResourceManagerService.h @@ -0,0 +1,66 @@ +/* + * Copyright 2015 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_IRESOURCEMANAGERSERVICE_H +#define ANDROID_IRESOURCEMANAGERSERVICE_H + +#include <utils/Errors.h> // for status_t +#include <utils/KeyedVector.h> +#include <utils/RefBase.h> +#include <utils/String8.h> +#include <binder/IInterface.h> +#include <binder/Parcel.h> + +#include <media/IResourceManagerClient.h> +#include <media/MediaResource.h> +#include <media/MediaResourcePolicy.h> + +namespace android { + +class IResourceManagerService: public IInterface +{ +public: + DECLARE_META_INTERFACE(ResourceManagerService); + + virtual void config(const Vector<MediaResourcePolicy> &policies) = 0; + + virtual void addResource( + int pid, + int64_t clientId, + const sp<IResourceManagerClient> client, + const Vector<MediaResource> &resources) = 0; + + virtual void removeResource(int64_t clientId) = 0; + + virtual bool reclaimResource( + int callingPid, + const Vector<MediaResource> &resources) = 0; +}; + +// ---------------------------------------------------------------------------- + +class BnResourceManagerService: public BnInterface<IResourceManagerService> +{ +public: + virtual status_t onTransact(uint32_t code, + const Parcel &data, + Parcel *reply, + uint32_t flags = 0); +}; + +}; // namespace android + +#endif // ANDROID_IRESOURCEMANAGERSERVICE_H diff --git a/include/media/IStreamSource.h b/include/media/IStreamSource.h index 677119b..149bd49 100644 --- a/include/media/IStreamSource.h +++ b/include/media/IStreamSource.h @@ -81,6 +81,13 @@ struct IStreamListener : public IInterface { // with the next PTS occuring in the stream. The value is of type int64_t. static const char *const kKeyMediaTimeUs; + // Optionally signalled as part of a discontinuity that includes + // DISCONTINUITY_TIME. It indicates the media time (in us) of a recent + // sample from the same content, and is used as a hint for the parser to + // handle PTS wraparound. This is required when a new parser is created + // to continue parsing content from the same timeline. + static const char *const kKeyRecentMediaTimeUs; + virtual void issueCommand( Command cmd, bool synchronous, const sp<AMessage> &msg = NULL) = 0; }; diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h index 4a6bf28..d6fe390 100644 --- a/include/media/MediaPlayerInterface.h +++ b/include/media/MediaPlayerInterface.h @@ -113,7 +113,19 @@ public: const audio_offload_info_t *offloadInfo = NULL) = 0; virtual status_t start() = 0; - virtual ssize_t write(const void* buffer, size_t size) = 0; + + /* Input parameter |size| is in byte units stored in |buffer|. + * Data is copied over and actual number of bytes written (>= 0) + * is returned, or no data is copied and a negative status code + * is returned (even when |blocking| is true). + * When |blocking| is false, AudioSink will immediately return after + * part of or full |buffer| is copied over. + * When |blocking| is true, AudioSink will wait to copy the entire + * buffer, unless an error occurs or the copy operation is + * prematurely stopped. + */ + virtual ssize_t write(const void* buffer, size_t size, bool blocking = true) = 0; + virtual void stop() = 0; virtual void flush() = 0; virtual void pause() = 0; @@ -156,6 +168,7 @@ public: virtual status_t stop() = 0; virtual status_t pause() = 0; virtual bool isPlaying() = 0; + virtual status_t setPlaybackRate(float rate) { return INVALID_OPERATION; } virtual status_t seekTo(int msec) = 0; virtual status_t getCurrentPosition(int *msec) = 0; virtual status_t getDuration(int *msec) = 0; diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h index d7ac302..f55063e 100644 --- a/include/media/MediaRecorderBase.h +++ b/include/media/MediaRecorderBase.h @@ -43,7 +43,6 @@ struct MediaRecorderBase { virtual status_t setCamera(const sp<ICamera>& camera, const sp<ICameraRecordingProxy>& proxy) = 0; virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface) = 0; - virtual status_t setOutputFile(const char *path) = 0; virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0; virtual status_t setOutputFileAuxiliary(int fd) {return INVALID_OPERATION;} virtual status_t setParameters(const String8& params) = 0; diff --git a/include/media/MediaResource.h b/include/media/MediaResource.h new file mode 100644 index 0000000..0b57c84 --- /dev/null +++ b/include/media/MediaResource.h @@ -0,0 +1,51 @@ +/* + * Copyright 2015 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_MEDIA_RESOURCE_H +#define ANDROID_MEDIA_RESOURCE_H + +#include <binder/Parcel.h> +#include <utils/String8.h> + +namespace android { + +extern const char kResourceSecureCodec[]; +extern const char kResourceNonSecureCodec[]; +extern const char kResourceGraphicMemory[]; + +class MediaResource { +public: + MediaResource(); + MediaResource(String8 type, uint64_t value); + MediaResource(String8 type, String8 subType, uint64_t value); + + void readFromParcel(const Parcel &parcel); + void writeToParcel(Parcel *parcel) const; + + String8 toString() const; + + bool operator==(const MediaResource &other) const; + bool operator!=(const MediaResource &other) const; + + String8 mType; + String8 mSubType; + uint64_t mValue; +}; + +}; // namespace android + +#endif // ANDROID_MEDIA_RESOURCE_H diff --git a/include/media/MediaResourcePolicy.h b/include/media/MediaResourcePolicy.h new file mode 100644 index 0000000..1e1c341 --- /dev/null +++ b/include/media/MediaResourcePolicy.h @@ -0,0 +1,45 @@ +/* + * Copyright 2015 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_MEDIA_RESOURCE_POLICY_H +#define ANDROID_MEDIA_RESOURCE_POLICY_H + +#include <binder/Parcel.h> +#include <utils/String8.h> + +namespace android { + +extern const char kPolicySupportsMultipleSecureCodecs[]; +extern const char kPolicySupportsSecureWithNonSecureCodec[]; + +class MediaResourcePolicy { +public: + MediaResourcePolicy(); + MediaResourcePolicy(String8 type, uint64_t value); + + void readFromParcel(const Parcel &parcel); + void writeToParcel(Parcel *parcel) const; + + String8 toString() const; + + String8 mType; + uint64_t mValue; +}; + +}; // namespace android + +#endif // ANDROID_MEDIA_RESOURCE_POLICY_H diff --git a/include/media/SingleStateQueue.h b/include/media/SingleStateQueue.h index 04c5fd0..d423962 100644 --- a/include/media/SingleStateQueue.h +++ b/include/media/SingleStateQueue.h @@ -21,6 +21,7 @@ // Non-blocking single-reader / single-writer multi-word atomic load / store #include <stdint.h> +#include <cutils/atomic.h> namespace android { @@ -31,6 +32,12 @@ public: class Mutator; class Observer; + enum SSQ_STATUS { + SSQ_PENDING, /* = 0 */ + SSQ_READ, + SSQ_DONE, + }; + struct Shared { // needs to be part of a union so don't define constructor or destructor @@ -41,28 +48,56 @@ private: void init() { mAck = 0; mSequence = 0; } volatile int32_t mAck; -#if 0 - int mPad[7]; - // cache line boundary -#endif volatile int32_t mSequence; T mValue; }; class Mutator { public: - Mutator(Shared *shared); - /*virtual*/ ~Mutator() { } + Mutator(Shared *shared) + : mSequence(0), mShared(shared) + { + // exactly one of Mutator and Observer must initialize, currently it is Observer + // shared->init(); + } // push new value onto state queue, overwriting previous value; // returns a sequence number which can be used with ack() - int32_t push(const T& value); - - // return true if most recent push has been observed - bool ack(); + int32_t push(const T& value) + { + Shared *shared = mShared; + int32_t sequence = mSequence; + sequence++; + android_atomic_acquire_store(sequence, &shared->mSequence); + shared->mValue = value; + sequence++; + android_atomic_release_store(sequence, &shared->mSequence); + mSequence = sequence; + // consider signalling a futex here, if we know that observer is waiting + return sequence; + } + + // returns the status of the last state push. This may be a stale value. + // + // SSQ_PENDING, or 0, means it has not been observed + // SSQ_READ means it has been read + // SSQ_DONE means it has been acted upon, after Observer::done() is called + enum SSQ_STATUS ack() const + { + // in the case of SSQ_DONE, prevent any subtle data-races of subsequent reads + // being performed (out-of-order) before the ack read, should the caller be + // depending on sequentiality of reads. + const int32_t ack = android_atomic_acquire_load(&mShared->mAck); + return ack - mSequence & ~1 ? SSQ_PENDING /* seq differ */ : + ack & 1 ? SSQ_DONE : SSQ_READ; + } // return true if a push with specified sequence number or later has been observed - bool ack(int32_t sequence); + bool ack(int32_t sequence) const + { + // this relies on 2's complement rollover to detect an ancient sequence number + return mShared->mAck - sequence >= 0; + } private: int32_t mSequence; @@ -71,11 +106,54 @@ private: class Observer { public: - Observer(Shared *shared); - /*virtual*/ ~Observer() { } + Observer(Shared *shared) + : mSequence(0), mSeed(1), mShared(shared) + { + // exactly one of Mutator and Observer must initialize, currently it is Observer + shared->init(); + } // return true if value has changed - bool poll(T& value); + bool poll(T& value) + { + Shared *shared = mShared; + int32_t before = shared->mSequence; + if (before == mSequence) { + return false; + } + for (int tries = 0; ; ) { + const int MAX_TRIES = 5; + if (before & 1) { + if (++tries >= MAX_TRIES) { + return false; + } + before = shared->mSequence; + } else { + android_memory_barrier(); + T temp = shared->mValue; + int32_t after = android_atomic_release_load(&shared->mSequence); + if (after == before) { + value = temp; + shared->mAck = before; + mSequence = before; // mSequence is even after poll success + return true; + } + if (++tries >= MAX_TRIES) { + return false; + } + before = after; + } + } + } + + // (optional) used to indicate to the Mutator that the state that has been polled + // has also been acted upon. + void done() + { + const int32_t ack = mShared->mAck + 1; + // ensure all previous writes have been performed. + android_atomic_release_store(ack, &mShared->mAck); // mSequence is odd after "done" + } private: int32_t mSequence; diff --git a/include/media/StringArray.h b/include/media/StringArray.h index ae47085..48d98bf 100644 --- a/include/media/StringArray.h +++ b/include/media/StringArray.h @@ -16,7 +16,7 @@ // // Sortable array of strings. STL-ish, but STL-free. -// +// #ifndef _LIBS_MEDIA_STRING_ARRAY_H #define _LIBS_MEDIA_STRING_ARRAY_H diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h index 5830933..808e893 100644 --- a/include/media/mediaplayer.h +++ b/include/media/mediaplayer.h @@ -220,6 +220,7 @@ public: status_t stop(); status_t pause(); bool isPlaying(); + status_t setPlaybackRate(float rate); status_t getVideoWidth(int *w); status_t getVideoHeight(int *h); status_t seekTo(int msec); @@ -274,6 +275,7 @@ private: int mVideoWidth; int mVideoHeight; int mAudioSessionId; + float mPlaybackRate; float mSendLevel; struct sockaddr_in mRetransmitEndpoint; bool mRetransmitEndpointValid; diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h index b0a62a7..74a6469 100644 --- a/include/media/mediarecorder.h +++ b/include/media/mediarecorder.h @@ -221,7 +221,6 @@ public: status_t setOutputFormat(int of); status_t setVideoEncoder(int ve); status_t setAudioEncoder(int ae); - status_t setOutputFile(const char* path); status_t setOutputFile(int fd, int64_t offset, int64_t length); status_t setVideoSize(int width, int height); status_t setVideoFrameRate(int frames_per_second); diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h index d422576..d9bbc8d 100644 --- a/include/media/nbaio/NBAIO.h +++ b/include/media/nbaio/NBAIO.h @@ -231,7 +231,8 @@ public: virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; } protected: - NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { } + NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) + { } virtual ~NBAIO_Sink() { } // Implementations are free to ignore these if they don't need them @@ -322,7 +323,8 @@ public: virtual void onTimestamp(const AudioTimestamp& timestamp) { } protected: - NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { } + NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) + { } virtual ~NBAIO_Source() { } // Implementations are free to ignore these if they don't need them diff --git a/include/media/nbaio/NBLog.h b/include/media/nbaio/NBLog.h index bcbbc04..1297b51 100644 --- a/include/media/nbaio/NBLog.h +++ b/include/media/nbaio/NBLog.h @@ -21,7 +21,7 @@ #include <binder/IMemory.h> #include <utils/Mutex.h> -#include <media/nbaio/roundup.h> +#include <audio_utils/roundup.h> namespace android { diff --git a/include/media/stagefright/AACWriter.h b/include/media/stagefright/AACWriter.h index d22707a..86417a5 100644 --- a/include/media/stagefright/AACWriter.h +++ b/include/media/stagefright/AACWriter.h @@ -27,7 +27,6 @@ struct MediaSource; struct MetaData; struct AACWriter : public MediaWriter { - AACWriter(const char *filename); AACWriter(int fd); status_t initCheck() const; diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h index cd2bd27..c1483f3 100644 --- a/include/media/stagefright/ACodec.h +++ b/include/media/stagefright/ACodec.h @@ -214,6 +214,7 @@ private: int64_t mRepeatFrameDelayUs; int64_t mMaxPtsGapUs; + float mMaxFps; int64_t mTimePerFrameUs; int64_t mTimePerCaptureUs; @@ -298,6 +299,8 @@ private: status_t setupRawAudioFormat( OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels); + status_t setPriority(int32_t priority); + status_t setMinBufferSize(OMX_U32 portIndex, size_t size); status_t setupMPEG4EncoderParameters(const sp<AMessage> &msg); diff --git a/include/media/stagefright/AMRWriter.h b/include/media/stagefright/AMRWriter.h index 392f968..bac878b 100644 --- a/include/media/stagefright/AMRWriter.h +++ b/include/media/stagefright/AMRWriter.h @@ -29,7 +29,6 @@ struct MediaSource; struct MetaData; struct AMRWriter : public MediaWriter { - AMRWriter(const char *filename); AMRWriter(int fd); status_t initCheck() const; diff --git a/include/media/stagefright/BufferProducerWrapper.h b/include/media/stagefright/BufferProducerWrapper.h index d8acf30..4caa2c6 100644 --- a/include/media/stagefright/BufferProducerWrapper.h +++ b/include/media/stagefright/BufferProducerWrapper.h @@ -19,6 +19,7 @@ #define BUFFER_PRODUCER_WRAPPER_H_ #include <gui/IGraphicBufferProducer.h> +#include <media/stagefright/foundation/ABase.h> namespace android { diff --git a/include/media/stagefright/MPEG2TSWriter.h b/include/media/stagefright/MPEG2TSWriter.h index 2e2922e..3d7960b 100644 --- a/include/media/stagefright/MPEG2TSWriter.h +++ b/include/media/stagefright/MPEG2TSWriter.h @@ -29,7 +29,6 @@ struct ABuffer; struct MPEG2TSWriter : public MediaWriter { MPEG2TSWriter(int fd); - MPEG2TSWriter(const char *filename); MPEG2TSWriter( void *cookie, diff --git a/include/media/stagefright/MPEG4Writer.h b/include/media/stagefright/MPEG4Writer.h index 26ce5f9..a195fe8 100644 --- a/include/media/stagefright/MPEG4Writer.h +++ b/include/media/stagefright/MPEG4Writer.h @@ -26,13 +26,13 @@ namespace android { +class AMessage; class MediaBuffer; class MediaSource; class MetaData; class MPEG4Writer : public MediaWriter { public: - MPEG4Writer(const char *filename); MPEG4Writer(int fd); // Limitations @@ -49,6 +49,7 @@ public: virtual status_t dump(int fd, const Vector<String16>& args); void beginBox(const char *fourcc); + void beginBox(uint32_t id); void writeInt8(int8_t x); void writeInt16(int16_t x); void writeInt32(int32_t x); @@ -63,6 +64,7 @@ public: int32_t getTimeScale() const { return mTimeScale; } status_t setGeoData(int latitudex10000, int longitudex10000); + status_t setCaptureRate(float captureFps); virtual void setStartTimeOffsetMs(int ms) { mStartTimeOffsetMs = ms; } virtual int32_t getStartTimeOffsetMs() const { return mStartTimeOffsetMs; } @@ -89,6 +91,7 @@ private: off64_t mFreeBoxOffset; bool mStreamableFile; off64_t mEstimatedMoovBoxSize; + off64_t mMoovExtraSize; uint32_t mInterleaveDurationUs; int32_t mTimeScale; int64_t mStartTimestampUs; @@ -103,6 +106,8 @@ private: List<off64_t> mBoxes; + sp<AMessage> mMetaKeys; + void setStartTimestampUs(int64_t timeUs); int64_t getStartTimestampUs(); // Not const status_t startTracks(MetaData *params); @@ -196,6 +201,12 @@ private: void writeGeoDataBox(); void writeLatitude(int degreex10000); void writeLongitude(int degreex10000); + + void addDeviceMeta(); + void writeHdlr(); + void writeKeys(); + void writeIlst(); + void writeMetaBox(); void sendSessionSummary(); void release(); status_t reset(); diff --git a/include/media/stagefright/MediaClock.h b/include/media/stagefright/MediaClock.h new file mode 100644 index 0000000..e9c09a1 --- /dev/null +++ b/include/media/stagefright/MediaClock.h @@ -0,0 +1,79 @@ +/* + * Copyright (C) 2015 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 MEDIA_CLOCK_H_ + +#define MEDIA_CLOCK_H_ + +#include <media/stagefright/foundation/ABase.h> +#include <utils/Mutex.h> +#include <utils/RefBase.h> + +namespace android { + +struct AMessage; + +struct MediaClock : public RefBase { + MediaClock(); + + void setStartingTimeMedia(int64_t startingTimeMediaUs); + + void clearAnchor(); + // It's required to use timestamp of just rendered frame as + // anchor time in paused state. + void updateAnchor( + int64_t anchorTimeMediaUs, + int64_t anchorTimeRealUs, + int64_t maxTimeMediaUs = INT64_MAX); + + void updateMaxTimeMedia(int64_t maxTimeMediaUs); + + void setPlaybackRate(float rate); + + // query media time corresponding to real time |realUs|, and save the + // result in |outMediaUs|. + status_t getMediaTime( + int64_t realUs, + int64_t *outMediaUs, + bool allowPastMaxTime = false) const; + // query real time corresponding to media time |targetMediaUs|. + // The result is saved in |outRealUs|. + status_t getRealTimeFor(int64_t targetMediaUs, int64_t *outRealUs) const; + +protected: + virtual ~MediaClock(); + +private: + status_t getMediaTime_l( + int64_t realUs, + int64_t *outMediaUs, + bool allowPastMaxTime) const; + + mutable Mutex mLock; + + int64_t mAnchorTimeMediaUs; + int64_t mAnchorTimeRealUs; + int64_t mMaxTimeMediaUs; + int64_t mStartingTimeMediaUs; + + float mPlaybackRate; + + DISALLOW_EVIL_CONSTRUCTORS(MediaClock); +}; + +} // namespace android + +#endif // MEDIA_CLOCK_H_ diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h index d448097..8241e19 100644 --- a/include/media/stagefright/MediaCodec.h +++ b/include/media/stagefright/MediaCodec.h @@ -27,6 +27,7 @@ namespace android { struct ABuffer; struct AMessage; +struct AReplyToken; struct AString; struct CodecBase; struct ICrypto; @@ -222,7 +223,7 @@ private: sp<ALooper> mCodecLooper; sp<CodecBase> mCodec; AString mComponentName; - uint32_t mReplyID; + sp<AReplyToken> mReplyID; uint32_t mFlags; status_t mStickyError; sp<Surface> mNativeWindow; @@ -249,10 +250,10 @@ private: Vector<BufferInfo> mPortBuffers[2]; int32_t mDequeueInputTimeoutGeneration; - uint32_t mDequeueInputReplyID; + sp<AReplyToken> mDequeueInputReplyID; int32_t mDequeueOutputTimeoutGeneration; - uint32_t mDequeueOutputReplyID; + sp<AReplyToken> mDequeueOutputReplyID; sp<ICrypto> mCrypto; @@ -267,7 +268,7 @@ private: static status_t PostAndAwaitResponse( const sp<AMessage> &msg, sp<AMessage> *response); - static void PostReplyWithError(int32_t replyID, int32_t err); + static void PostReplyWithError(const sp<AReplyToken> &replyID, int32_t err); status_t init(const AString &name, bool nameIsType, bool encoder); @@ -283,8 +284,8 @@ private: size_t portIndex, size_t index, sp<ABuffer> *buffer, sp<AMessage> *format); - bool handleDequeueInputBuffer(uint32_t replyID, bool newRequest = false); - bool handleDequeueOutputBuffer(uint32_t replyID, bool newRequest = false); + bool handleDequeueInputBuffer(const sp<AReplyToken> &replyID, bool newRequest = false); + bool handleDequeueOutputBuffer(const sp<AReplyToken> &replyID, bool newRequest = false); void cancelPendingDequeueOperations(); void extractCSD(const sp<AMessage> &format); diff --git a/include/media/stagefright/MediaCodecSource.h b/include/media/stagefright/MediaCodecSource.h index 0970b2b..7b8f59d 100644 --- a/include/media/stagefright/MediaCodecSource.h +++ b/include/media/stagefright/MediaCodecSource.h @@ -25,6 +25,7 @@ namespace android { class ALooper; class AMessage; +struct AReplyToken; class IGraphicBufferProducer; class MediaCodec; class MetaData; @@ -99,7 +100,7 @@ private: sp<Puller> mPuller; sp<MediaCodec> mEncoder; uint32_t mFlags; - List<uint32_t> mStopReplyIDQueue; + List<sp<AReplyToken>> mStopReplyIDQueue; bool mIsVideo; bool mStarted; bool mStopping; diff --git a/include/media/stagefright/MediaFilter.h b/include/media/stagefright/MediaFilter.h new file mode 100644 index 0000000..7b3f700 --- /dev/null +++ b/include/media/stagefright/MediaFilter.h @@ -0,0 +1,167 @@ +/* + * Copyright (C) 2014 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 MEDIA_FILTER_H_ +#define MEDIA_FILTER_H_ + +#include <media/stagefright/CodecBase.h> + +namespace android { + +struct ABuffer; +struct GraphicBufferListener; +struct MemoryDealer; +struct SimpleFilter; + +struct MediaFilter : public CodecBase { + MediaFilter(); + + virtual void setNotificationMessage(const sp<AMessage> &msg); + + virtual void initiateAllocateComponent(const sp<AMessage> &msg); + virtual void initiateConfigureComponent(const sp<AMessage> &msg); + virtual void initiateCreateInputSurface(); + virtual void initiateStart(); + virtual void initiateShutdown(bool keepComponentAllocated = false); + + virtual void signalFlush(); + virtual void signalResume(); + + virtual void signalRequestIDRFrame(); + virtual void signalSetParameters(const sp<AMessage> &msg); + virtual void signalEndOfInputStream(); + + virtual void onMessageReceived(const sp<AMessage> &msg); + + struct PortDescription : public CodecBase::PortDescription { + virtual size_t countBuffers(); + virtual IOMX::buffer_id bufferIDAt(size_t index) const; + virtual sp<ABuffer> bufferAt(size_t index) const; + + protected: + PortDescription(); + + private: + friend struct MediaFilter; + + Vector<IOMX::buffer_id> mBufferIDs; + Vector<sp<ABuffer> > mBuffers; + + void addBuffer(IOMX::buffer_id id, const sp<ABuffer> &buffer); + + DISALLOW_EVIL_CONSTRUCTORS(PortDescription); + }; + +protected: + virtual ~MediaFilter(); + +private: + struct BufferInfo { + enum Status { + OWNED_BY_US, + OWNED_BY_UPSTREAM, + }; + + IOMX::buffer_id mBufferID; + int32_t mGeneration; + int32_t mOutputFlags; + Status mStatus; + + sp<ABuffer> mData; + }; + + enum State { + UNINITIALIZED, + INITIALIZED, + CONFIGURED, + STARTED, + }; + + enum { + kWhatInputBufferFilled = 'inpF', + kWhatOutputBufferDrained = 'outD', + kWhatShutdown = 'shut', + kWhatFlush = 'flus', + kWhatResume = 'resm', + kWhatAllocateComponent = 'allo', + kWhatConfigureComponent = 'conf', + kWhatCreateInputSurface = 'cisf', + kWhatSignalEndOfInputStream = 'eois', + kWhatStart = 'star', + kWhatSetParameters = 'setP', + kWhatProcessBuffers = 'proc', + }; + + enum { + kPortIndexInput = 0, + kPortIndexOutput = 1 + }; + + // member variables + AString mComponentName; + State mState; + status_t mInputEOSResult; + int32_t mWidth, mHeight; + int32_t mStride, mSliceHeight; + int32_t mColorFormatIn, mColorFormatOut; + size_t mMaxInputSize, mMaxOutputSize; + int32_t mGeneration; + sp<AMessage> mNotify; + sp<AMessage> mInputFormat; + sp<AMessage> mOutputFormat; + + sp<MemoryDealer> mDealer[2]; + Vector<BufferInfo> mBuffers[2]; + Vector<BufferInfo*> mAvailableInputBuffers; + Vector<BufferInfo*> mAvailableOutputBuffers; + bool mPortEOS[2]; + + sp<SimpleFilter> mFilter; + sp<GraphicBufferListener> mGraphicBufferListener; + + // helper functions + void signalProcessBuffers(); + void signalError(status_t error); + + status_t allocateBuffersOnPort(OMX_U32 portIndex); + BufferInfo *findBufferByID( + uint32_t portIndex, IOMX::buffer_id bufferID, + ssize_t *index = NULL); + void postFillThisBuffer(BufferInfo *info); + void postDrainThisBuffer(BufferInfo *info); + void postEOS(); + void sendFormatChange(); + void requestFillEmptyInput(); + void processBuffers(); + + void onAllocateComponent(const sp<AMessage> &msg); + void onConfigureComponent(const sp<AMessage> &msg); + void onStart(); + void onInputBufferFilled(const sp<AMessage> &msg); + void onOutputBufferDrained(const sp<AMessage> &msg); + void onShutdown(const sp<AMessage> &msg); + void onFlush(); + void onSetParameters(const sp<AMessage> &msg); + void onCreateInputSurface(); + void onInputFrameAvailable(); + void onSignalEndOfInputStream(); + + DISALLOW_EVIL_CONSTRUCTORS(MediaFilter); +}; + +} // namespace android + +#endif // MEDIA_FILTER_H_ diff --git a/include/media/stagefright/MediaMuxer.h b/include/media/stagefright/MediaMuxer.h index 9da98d9..e6538d1 100644 --- a/include/media/stagefright/MediaMuxer.h +++ b/include/media/stagefright/MediaMuxer.h @@ -50,9 +50,6 @@ public: OUTPUT_FORMAT_LIST_END // must be last - used to validate format type }; - // Construct the muxer with the output file path. - MediaMuxer(const char *path, OutputFormat format); - // Construct the muxer with the file descriptor. Note that the MediaMuxer // will close this file at stop(). MediaMuxer(int fd, OutputFormat format); diff --git a/include/media/stagefright/MediaSync.h b/include/media/stagefright/MediaSync.h new file mode 100644 index 0000000..8bb8c7f --- /dev/null +++ b/include/media/stagefright/MediaSync.h @@ -0,0 +1,239 @@ +/* + * Copyright 2015 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 MEDIA_SYNC_H +#define MEDIA_SYNC_H + +#include <gui/IConsumerListener.h> +#include <gui/IProducerListener.h> + +#include <media/stagefright/foundation/AHandler.h> + +#include <utils/Condition.h> +#include <utils/Mutex.h> + +namespace android { + +class AudioTrack; +class BufferItem; +class Fence; +class GraphicBuffer; +class IGraphicBufferConsumer; +class IGraphicBufferProducer; +struct MediaClock; + +// MediaSync manages media playback and its synchronization to a media clock +// source. It can be also used for video-only playback. +// +// For video playback, it requires an output surface and provides an input +// surface. It then controls the rendering of input buffers (buffer queued to +// the input surface) on the output surface to happen at the appropriate time. +// +// For audio playback, it requires an audio track and takes updates of +// information of rendered audio data to maintain media clock when audio track +// serves as media clock source. (TODO: move audio rendering from JAVA to +// native code). +// +// It can use the audio or video track as media clock source, as well as an +// external clock. (TODO: actually support external clock as media clock +// sources; use video track as media clock source for audio-and-video stream). +// +// In video-only mode, MediaSync will playback every video frame even though +// a video frame arrives late based on its timestamp and last frame's. +// +// The client needs to configure surface (for output video rendering) and audio +// track (for querying information of audio rendering) for MediaSync. +// +// Then the client needs to obtain a surface from MediaSync and render video +// frames onto that surface. Internally, the MediaSync will receive those video +// frames and render them onto the output surface at the appropriate time. +// +// The client needs to call updateQueuedAudioData() immediately after it writes +// audio data to the audio track. Such information will be used to update media +// clock. +// +class MediaSync : public AHandler { +public: + // Create an instance of MediaSync. + static sp<MediaSync> create(); + + // Called when MediaSync is used to render video. It should be called + // before createInputSurface(). + status_t configureSurface(const sp<IGraphicBufferProducer> &output); + + // Called when audio track is used as media clock source. It should be + // called before updateQueuedAudioData(). + // |nativeSampleRateInHz| is the sample rate of audio data fed into audio + // track. It's the same number used to create AudioTrack. + status_t configureAudioTrack( + const sp<AudioTrack> &audioTrack, uint32_t nativeSampleRateInHz); + + // Create a surface for client to render video frames. This is the surface + // on which the client should render video frames. Those video frames will + // be internally directed to output surface for rendering at appropriate + // time. + status_t createInputSurface(sp<IGraphicBufferProducer> *outBufferProducer); + + // Update just-rendered audio data size and the presentation timestamp of + // the first frame of that audio data. It should be called immediately + // after the client write audio data into AudioTrack. + // This function assumes continous audio stream. + // TODO: support gap or backwards updates. + status_t updateQueuedAudioData( + size_t sizeInBytes, int64_t presentationTimeUs); + + // Set the consumer name of the input queue. + void setName(const AString &name); + + // Set the playback in a desired speed. + // This method can be called any time. + // |rate| is the ratio between desired speed and the normal one, and should + // be non-negative. The meaning of rate values: + // 1.0 -- normal playback + // 0.0 -- stop or pause + // larger than 1.0 -- faster than normal speed + // between 0.0 and 1.0 -- slower than normal speed + status_t setPlaybackRate(float rate); + + // Get the media clock used by the MediaSync so that the client can obtain + // corresponding media time or real time via + // MediaClock::getMediaTime() and MediaClock::getRealTimeFor(). + sp<const MediaClock> getMediaClock(); + +protected: + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kWhatDrainVideo = 'dVid', + }; + + static const int MAX_OUTSTANDING_BUFFERS = 2; + + // This is a thin wrapper class that lets us listen to + // IConsumerListener::onFrameAvailable from mInput. + class InputListener : public BnConsumerListener, + public IBinder::DeathRecipient { + public: + InputListener(const sp<MediaSync> &sync); + virtual ~InputListener(); + + // From IConsumerListener + virtual void onFrameAvailable(const BufferItem &item); + + // From IConsumerListener + // We don't care about released buffers because we detach each buffer as + // soon as we acquire it. See the comment for onBufferReleased below for + // some clarifying notes about the name. + virtual void onBuffersReleased() {} + + // From IConsumerListener + // We don't care about sideband streams, since we won't relay them. + virtual void onSidebandStreamChanged(); + + // From IBinder::DeathRecipient + virtual void binderDied(const wp<IBinder> &who); + + private: + sp<MediaSync> mSync; + }; + + // This is a thin wrapper class that lets us listen to + // IProducerListener::onBufferReleased from mOutput. + class OutputListener : public BnProducerListener, + public IBinder::DeathRecipient { + public: + OutputListener(const sp<MediaSync> &sync); + virtual ~OutputListener(); + + // From IProducerListener + virtual void onBufferReleased(); + + // From IBinder::DeathRecipient + virtual void binderDied(const wp<IBinder> &who); + + private: + sp<MediaSync> mSync; + }; + + // mIsAbandoned is set to true when the input or output dies. + // Once the MediaSync has been abandoned by one side, it will disconnect + // from the other side and not attempt to communicate with it further. + bool mIsAbandoned; + + mutable Mutex mMutex; + Condition mReleaseCondition; + size_t mNumOutstandingBuffers; + sp<IGraphicBufferConsumer> mInput; + sp<IGraphicBufferProducer> mOutput; + + sp<AudioTrack> mAudioTrack; + uint32_t mNativeSampleRateInHz; + int64_t mNumFramesWritten; + bool mHasAudio; + + int64_t mNextBufferItemMediaUs; + List<BufferItem> mBufferItems; + sp<ALooper> mLooper; + float mPlaybackRate; + + sp<MediaClock> mMediaClock; + + MediaSync(); + + // Must be accessed through RefBase + virtual ~MediaSync(); + + int64_t getRealTime(int64_t mediaTimeUs, int64_t nowUs); + int64_t getDurationIfPlayedAtNativeSampleRate_l(int64_t numFrames); + int64_t getPlayedOutAudioDurationMedia_l(int64_t nowUs); + + void onDrainVideo_l(); + + // This implements the onFrameAvailable callback from IConsumerListener. + // It gets called from an InputListener. + // During this callback, we detach the buffer from the input, and queue + // it for rendering on the output. This call can block if there are too + // many outstanding buffers. If it blocks, it will resume when + // onBufferReleasedByOutput releases a buffer back to the input. + void onFrameAvailableFromInput(); + + // Send |bufferItem| to the output for rendering. + void renderOneBufferItem_l(const BufferItem &bufferItem); + + // This implements the onBufferReleased callback from IProducerListener. + // It gets called from an OutputListener. + // During this callback, we detach the buffer from the output, and release + // it to the input. A blocked onFrameAvailable call will be allowed to proceed. + void onBufferReleasedByOutput(); + + // Return |buffer| back to the input. + void returnBufferToInput_l(const sp<GraphicBuffer> &buffer, const sp<Fence> &fence); + + // When this is called, the MediaSync disconnects from (i.e., abandons) its + // input or output, and signals any waiting onFrameAvailable calls to wake + // up. This must be called with mMutex locked. + void onAbandoned_l(bool isInput); + + // helper. + bool isPlaying() { return mPlaybackRate != 0.0; } + + DISALLOW_EVIL_CONSTRUCTORS(MediaSync); +}; + +} // namespace android + +#endif diff --git a/include/media/stagefright/ProcessInfo.h b/include/media/stagefright/ProcessInfo.h new file mode 100644 index 0000000..ec0cdff --- /dev/null +++ b/include/media/stagefright/ProcessInfo.h @@ -0,0 +1,40 @@ +/* + * Copyright (C) 2015 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 PROCESS_INFO_H_ + +#define PROCESS_INFO_H_ + +#include <media/stagefright/foundation/ABase.h> +#include <media/stagefright/ProcessInfoInterface.h> + +namespace android { + +struct ProcessInfo : public ProcessInfoInterface { + ProcessInfo(); + + virtual bool getPriority(int pid, int* priority); + +protected: + virtual ~ProcessInfo(); + +private: + DISALLOW_EVIL_CONSTRUCTORS(ProcessInfo); +}; + +} // namespace android + +#endif // PROCESS_INFO_H_ diff --git a/include/media/nbaio/roundup.h b/include/media/stagefright/ProcessInfoInterface.h index 4c3cc25..222f92d 100644 --- a/include/media/nbaio/roundup.h +++ b/include/media/stagefright/ProcessInfoInterface.h @@ -1,5 +1,5 @@ /* - * Copyright (C) 2012 The Android Open Source Project + * Copyright (C) 2015 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. @@ -14,18 +14,20 @@ * limitations under the License. */ -#ifndef ROUNDUP_H -#define ROUNDUP_H +#ifndef PROCESS_INFO_INTERFACE_H_ +#define PROCESS_INFO_INTERFACE_H_ -#ifdef __cplusplus -extern "C" { -#endif +#include <utils/RefBase.h> -// Round up to the next highest power of 2 -unsigned roundup(unsigned v); +namespace android { -#ifdef __cplusplus -} -#endif +struct ProcessInfoInterface : public RefBase { + virtual bool getPriority(int pid, int* priority) = 0; -#endif // ROUNDUP_H +protected: + virtual ~ProcessInfoInterface() {} +}; + +} // namespace android + +#endif // PROCESS_INFO_INTERFACE_H_ diff --git a/include/media/stagefright/RenderScriptWrapper.h b/include/media/stagefright/RenderScriptWrapper.h new file mode 100644 index 0000000..b42649e --- /dev/null +++ b/include/media/stagefright/RenderScriptWrapper.h @@ -0,0 +1,42 @@ +/* + * Copyright (C) 2014 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 RENDERSCRIPT_WRAPPER_H_ +#define RENDERSCRIPT_WRAPPER_H_ + +#include <RenderScript.h> + +namespace android { + +struct RenderScriptWrapper : public RefBase { +public: + struct RSFilterCallback : public RefBase { + public: + // called by RSFilter to process each input buffer + virtual status_t processBuffers( + RSC::Allocation* inBuffer, + RSC::Allocation* outBuffer) = 0; + + virtual status_t handleSetParameters(const sp<AMessage> &msg) = 0; + }; + + sp<RSFilterCallback> mCallback; + RSC::sp<RSC::RS> mContext; +}; + +} // namespace android + +#endif // RENDERSCRIPT_WRAPPER_H_ diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h index a795c80..ec3a10e 100644 --- a/include/media/stagefright/Utils.h +++ b/include/media/stagefright/Utils.h @@ -65,6 +65,17 @@ bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, AString uriDebugString(const AString &uri, bool incognito = false); +struct HLSTime { + int32_t mSeq; + int64_t mTimeUs; + sp<AMessage> mMeta; + + HLSTime(const sp<AMessage> &meta = NULL); + int64_t getSegmentTimeUs(bool midpoint = false) const; +}; + +bool operator <(const HLSTime &t0, const HLSTime &t1); + } // namespace android #endif // UTILS_H_ diff --git a/include/media/stagefright/foundation/ABase.h b/include/media/stagefright/foundation/ABase.h index 72e3d87..ef1e010 100644 --- a/include/media/stagefright/foundation/ABase.h +++ b/include/media/stagefright/foundation/ABase.h @@ -18,7 +18,9 @@ #define A_BASE_H_ +#ifndef ARRAY_SIZE #define ARRAY_SIZE(a) (sizeof(a) / sizeof(*(a))) +#endif #define DISALLOW_EVIL_CONSTRUCTORS(name) \ name(const name &); \ diff --git a/include/media/stagefright/foundation/AHandler.h b/include/media/stagefright/foundation/AHandler.h index 41ade77..fe02a86 100644 --- a/include/media/stagefright/foundation/AHandler.h +++ b/include/media/stagefright/foundation/AHandler.h @@ -29,6 +29,7 @@ struct AMessage; struct AHandler : public RefBase { AHandler() : mID(0), + mVerboseStats(false), mMessageCounter(0) { } @@ -36,23 +37,40 @@ struct AHandler : public RefBase { return mID; } - sp<ALooper> looper(); + sp<ALooper> looper() const { + return mLooper.promote(); + } + + wp<ALooper> getLooper() const { + return mLooper; + } + + wp<AHandler> getHandler() const { + // allow getting a weak reference to a const handler + return const_cast<AHandler *>(this); + } protected: virtual void onMessageReceived(const sp<AMessage> &msg) = 0; private: - friend struct ALooperRoster; + friend struct AMessage; // deliverMessage() + friend struct ALooperRoster; // setID() ALooper::handler_id mID; + wp<ALooper> mLooper; - void setID(ALooper::handler_id id) { + inline void setID(ALooper::handler_id id, wp<ALooper> looper) { mID = id; + mLooper = looper; } + bool mVerboseStats; uint32_t mMessageCounter; KeyedVector<uint32_t, uint32_t> mMessages; + void deliverMessage(const sp<AMessage> &msg); + DISALLOW_EVIL_CONSTRUCTORS(AHandler); }; diff --git a/include/media/stagefright/foundation/ALooper.h b/include/media/stagefright/foundation/ALooper.h index 70e0c5e..09c469b 100644 --- a/include/media/stagefright/foundation/ALooper.h +++ b/include/media/stagefright/foundation/ALooper.h @@ -30,6 +30,7 @@ namespace android { struct AHandler; struct AMessage; +struct AReplyToken; struct ALooper : public RefBase { typedef int32_t event_id; @@ -53,11 +54,15 @@ struct ALooper : public RefBase { static int64_t GetNowUs(); + const char *getName() const { + return mName.c_str(); + } + protected: virtual ~ALooper(); private: - friend struct ALooperRoster; + friend struct AMessage; // post() struct Event { int64_t mWhenUs; @@ -75,12 +80,32 @@ private: sp<LooperThread> mThread; bool mRunningLocally; + // use a separate lock for reply handling, as it is always on another thread + // use a central lock, however, to avoid creating a mutex for each reply + Mutex mRepliesLock; + Condition mRepliesCondition; + + // START --- methods used only by AMessage + + // posts a message on this looper with the given timeout void post(const sp<AMessage> &msg, int64_t delayUs); + + // creates a reply token to be used with this looper + sp<AReplyToken> createReplyToken(); + // waits for a response for the reply token. If status is OK, the response + // is stored into the supplied variable. Otherwise, it is unchanged. + status_t awaitResponse(const sp<AReplyToken> &replyToken, sp<AMessage> *response); + // posts a reply for a reply token. If the reply could be successfully posted, + // it returns OK. Otherwise, it returns an error value. + status_t postReply(const sp<AReplyToken> &replyToken, const sp<AMessage> &msg); + + // END --- methods used only by AMessage + bool loop(); DISALLOW_EVIL_CONSTRUCTORS(ALooper); }; -} // namespace android +} // namespace android #endif // A_LOOPER_H_ diff --git a/include/media/stagefright/foundation/ALooperRoster.h b/include/media/stagefright/foundation/ALooperRoster.h index a0be8eb..9912455 100644 --- a/include/media/stagefright/foundation/ALooperRoster.h +++ b/include/media/stagefright/foundation/ALooperRoster.h @@ -33,16 +33,6 @@ struct ALooperRoster { void unregisterHandler(ALooper::handler_id handlerID); void unregisterStaleHandlers(); - status_t postMessage(const sp<AMessage> &msg, int64_t delayUs = 0); - void deliverMessage(const sp<AMessage> &msg); - - status_t postAndAwaitResponse( - const sp<AMessage> &msg, sp<AMessage> *response); - - void postReply(uint32_t replyID, const sp<AMessage> &reply); - - sp<ALooper> findLooper(ALooper::handler_id handlerID); - void dump(int fd, const Vector<String16>& args); private: @@ -54,10 +44,6 @@ private: Mutex mLock; KeyedVector<ALooper::handler_id, HandlerInfo> mHandlers; ALooper::handler_id mNextHandlerID; - uint32_t mNextReplyID; - Condition mRepliesCondition; - - KeyedVector<uint32_t, sp<AMessage> > mReplies; DISALLOW_EVIL_CONSTRUCTORS(ALooperRoster); }; diff --git a/include/media/stagefright/foundation/AMessage.h b/include/media/stagefright/foundation/AMessage.h index a9e235b..4c6bd21 100644 --- a/include/media/stagefright/foundation/AMessage.h +++ b/include/media/stagefright/foundation/AMessage.h @@ -26,11 +26,41 @@ namespace android { struct ABuffer; +struct AHandler; struct AString; struct Parcel; +struct AReplyToken : public RefBase { + AReplyToken(const sp<ALooper> &looper) + : mLooper(looper), + mReplied(false) { + } + +private: + friend struct AMessage; + friend struct ALooper; + wp<ALooper> mLooper; + sp<AMessage> mReply; + bool mReplied; + + sp<ALooper> getLooper() const { + return mLooper.promote(); + } + // if reply is not set, returns false; otherwise, it retrieves the reply and returns true + bool retrieveReply(sp<AMessage> *reply) { + if (mReplied) { + *reply = mReply; + mReply.clear(); + } + return mReplied; + } + // sets the reply for this token. returns OK or error + status_t setReply(const sp<AMessage> &reply); +}; + struct AMessage : public RefBase { - AMessage(uint32_t what = 0, ALooper::handler_id target = 0); + AMessage(); + AMessage(uint32_t what, const sp<const AHandler> &handler); static sp<AMessage> FromParcel(const Parcel &parcel); void writeToParcel(Parcel *parcel) const; @@ -38,8 +68,7 @@ struct AMessage : public RefBase { void setWhat(uint32_t what); uint32_t what() const; - void setTarget(ALooper::handler_id target); - ALooper::handler_id target() const; + void setTarget(const sp<const AHandler> &handler); void clear(); @@ -76,18 +105,22 @@ struct AMessage : public RefBase { const char *name, int32_t *left, int32_t *top, int32_t *right, int32_t *bottom) const; - void post(int64_t delayUs = 0); + status_t post(int64_t delayUs = 0); // Posts the message to its target and waits for a response (or error) // before returning. status_t postAndAwaitResponse(sp<AMessage> *response); // If this returns true, the sender of this message is synchronously - // awaiting a response, the "replyID" can be used to send the response - // via "postReply" below. - bool senderAwaitsResponse(uint32_t *replyID) const; + // awaiting a response and the reply token is consumed from the message + // and stored into replyID. The reply token must be used to send the response + // using "postReply" below. + bool senderAwaitsResponse(sp<AReplyToken> *replyID); - void postReply(uint32_t replyID); + // Posts the message as a response to a reply token. A reply token can + // only be used once. Returns OK if the response could be posted; otherwise, + // an error. + status_t postReply(const sp<AReplyToken> &replyID); // Performs a deep-copy of "this", contained messages are in turn "dup'ed". // Warning: RefBase items, i.e. "objects" are _not_ copied but only have @@ -117,9 +150,16 @@ protected: virtual ~AMessage(); private: + friend struct ALooper; // deliver() + uint32_t mWhat; + + // used only for debugging ALooper::handler_id mTarget; + wp<AHandler> mHandler; + wp<ALooper> mLooper; + struct Rect { int32_t mLeft, mTop, mRight, mBottom; }; @@ -157,6 +197,8 @@ private: size_t findItemIndex(const char *name, size_t len) const; + void deliver(); + DISALLOW_EVIL_CONSTRUCTORS(AMessage); }; |