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-rw-r--r--include/media/AudioBufferProvider.h11
-rw-r--r--include/media/AudioEffect.h2
-rw-r--r--include/media/AudioRecord.h58
-rw-r--r--include/media/AudioSystem.h41
-rw-r--r--include/media/AudioTimestamp.h4
-rw-r--r--include/media/AudioTrack.h60
-rw-r--r--include/media/IAudioFlinger.h11
-rw-r--r--include/media/IMediaHTTPConnection.h48
-rw-r--r--include/media/IMediaHTTPService.h41
-rw-r--r--include/media/IMediaMetadataRetriever.h3
-rw-r--r--include/media/IMediaPlayer.h8
-rw-r--r--include/media/IMediaPlayerService.h15
-rw-r--r--include/media/IOMX.h2
-rw-r--r--include/media/MediaMetadataRetrieverInterface.h3
-rw-r--r--include/media/MediaPlayerInterface.h6
-rw-r--r--include/media/mediametadataretriever.h2
-rw-r--r--include/media/mediaplayer.h17
-rw-r--r--include/media/mediarecorder.h2
-rw-r--r--include/media/mediascanner.h14
-rw-r--r--include/media/nbaio/AudioBufferProviderSource.h2
-rw-r--r--include/media/nbaio/MonoPipe.h2
-rw-r--r--include/media/nbaio/NBAIO.h32
-rw-r--r--include/media/nbaio/Pipe.h2
-rw-r--r--include/media/stagefright/ACodec.h6
-rw-r--r--include/media/stagefright/CameraSource.h3
-rw-r--r--include/media/stagefright/DataSource.h2
-rw-r--r--include/media/stagefright/MediaCodecSource.h134
-rw-r--r--include/media/stagefright/MediaHTTP.h77
-rw-r--r--include/media/stagefright/NuMediaExtractor.h2
-rw-r--r--include/media/stagefright/SkipCutBuffer.h1
-rw-r--r--include/media/stagefright/timedtext/TimedTextDriver.h6
-rw-r--r--include/private/media/AudioTrackShared.h22
-rw-r--r--include/private/media/StaticAudioTrackState.h10
33 files changed, 526 insertions, 123 deletions
diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h
index ef392f0..7be449c 100644
--- a/include/media/AudioBufferProvider.h
+++ b/include/media/AudioBufferProvider.h
@@ -61,6 +61,17 @@ public:
// buffer->frameCount 0
virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0;
+ // Release (a portion of) the buffer previously obtained by getNextBuffer().
+ // It is permissible to call releaseBuffer() multiple times per getNextBuffer().
+ // On entry:
+ // buffer->frameCount number of frames to release, must be <= number of frames
+ // obtained but not yet released
+ // buffer->raw unused
+ // On return:
+ // buffer->frameCount 0; implementation MUST set to zero
+ // buffer->raw undefined; implementation is PERMITTED to set to any value,
+ // so if caller needs to continue using this buffer it must
+ // keep track of the pointer itself
virtual void releaseBuffer(Buffer* buffer) = 0;
};
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 05d834d..f3024b7 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -36,7 +36,7 @@ namespace android {
// ----------------------------------------------------------------------------
-class effect_param_cblk_t;
+struct effect_param_cblk_t;
// ----------------------------------------------------------------------------
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 052064d..7054fd4 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -26,7 +26,7 @@ namespace android {
// ----------------------------------------------------------------------------
-class audio_track_cblk_t;
+struct audio_track_cblk_t;
class AudioRecordClientProxy;
// ----------------------------------------------------------------------------
@@ -39,8 +39,12 @@ public:
* Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
*/
enum event_type {
- EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer.
- EVENT_OVERRUN = 1, // PCM buffer overrun occurred.
+ 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.
+ EVENT_OVERRUN = 1, // Buffer overrun occurred.
EVENT_MARKER = 2, // Record head is at the specified marker position
// (See setMarkerPosition()).
EVENT_NEW_POS = 3, // Record head is at a new position
@@ -60,9 +64,10 @@ public:
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)
+ // (currently ignored but will make the primary field in future)
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
@@ -76,7 +81,7 @@ public:
/* 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 ready or various conditions occur.
+ * invokes the callback when a new buffer becomes available or various conditions occur.
* Parameters:
*
* event: type of event notified (see enum AudioRecord::event_type).
@@ -99,6 +104,8 @@ public:
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
* - BAD_VALUE: unsupported configuration
+ * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+ * and is undefined otherwise.
*/
static status_t getMinFrameCount(size_t* frameCount,
@@ -109,7 +116,7 @@ public:
/* How data is transferred from AudioRecord
*/
enum transfer_type {
- TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters
+ 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_SYNC, // synchronous read()
@@ -137,7 +144,7 @@ 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 PCM data and inform of marker, position updates, etc.
+ * to consume new data 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.
@@ -155,7 +162,7 @@ public:
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
@@ -171,9 +178,10 @@ public:
* 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
- * - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
+ * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
+ * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
*
* Parameters not listed in the AudioRecord constructors above:
*
@@ -188,11 +196,11 @@ public:
void* user = NULL,
int notificationFrames = 0,
bool threadCanCallJava = false,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
- /* Result of constructing the AudioRecord. This must be checked
+ /* Result of constructing the AudioRecord. This must be checked for successful initialization
* before using any AudioRecord API (except for set()), because using
* an uninitialized AudioRecord produces undefined results.
* See set() method above for possible return codes.
@@ -221,7 +229,7 @@ public:
status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
- /* Stop a track. If set, the callback will cease being called. Note that obtainBuffer() still
+ /* Stop a track. The callback will cease being called. Note that obtainBuffer() still
* works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
*/
void stop();
@@ -236,7 +244,7 @@ public:
* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
* with marker == 0 cancels marker notification callback.
* To set a marker at a position which would compute as 0,
- * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+ * a workaround is to set the marker at a nearby position such as ~0 or 1.
* If the AudioRecord has been opened with no callback function associated,
* the operation will fail.
*
@@ -378,8 +386,10 @@ public:
* returning the current value by this function call. Such loss typically occurs when the
* user space process is blocked longer than the capacity of audio driver buffers.
* Units: the number of input audio frames.
+ * FIXME The side-effect of resetting the counter may be incompatible with multi-client.
+ * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects.
*/
- unsigned int getInputFramesLost() const;
+ uint32_t getInputFramesLost() const;
private:
/* copying audio record objects is not allowed */
@@ -412,6 +422,7 @@ 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
};
// body of AudioRecordThread::threadLoop()
@@ -422,9 +433,10 @@ private:
// NS_INACTIVE inactive so don't run again until re-started
// NS_NEVER never again
static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
- nsecs_t processAudioBuffer(const sp<AudioRecordThread>& thread);
+ nsecs_t processAudioBuffer();
// caller must hold lock on mLock for all _l methods
+
status_t openRecord_l(size_t epoch);
// FIXME enum is faster than strcmp() for parameter 'from'
@@ -446,12 +458,13 @@ private:
// notification callback
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback
- bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
+ bool mRefreshRemaining; // processAudioBuffer() should refresh
+ // mRemainingFrames and mRetryOnPartialBuffer
// 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()
- int mObservedSequence; // last observed value of mSequence
+ uint32_t mObservedSequence; // last observed value of mSequence
uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
@@ -460,9 +473,13 @@ private:
status_t mStatus;
+ size_t mFrameCount; // corresponds to current IAudioRecord, value is
+ // reported back by AudioFlinger to the client
+ size_t mReqFrameCount; // frame count to request the first or next time
+ // a new IAudioRecord is needed, non-decreasing
+
// constant after constructor or set()
uint32_t mSampleRate;
- size_t mFrameCount;
audio_format_t mFormat;
uint32_t mChannelCount;
size_t mFrameSize; // app-level frame size == AudioFlinger frame size
@@ -473,12 +490,11 @@ private:
int mSessionId;
transfer_type mTransfer;
- audio_io_handle_t mInput; // returned by AudioSystem::getInput()
-
- // may be changed if IAudioRecord object is re-created
+ // Next 4 fields may be changed if IAudioRecord is re-created, but always != 0
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ audio_io_handle_t mInput; // returned by AudioSystem::getInput()
int mPreviousPriority; // before start()
SchedPolicy mPreviousSchedulingGroup;
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 4c22412..dfc2066 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -67,20 +67,24 @@ public:
// returns true in *state if tracks are active on the specified stream or have been active
// in the past inPastMs milliseconds
- static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0);
+ static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs);
// returns true in *state if tracks are active for what qualifies as remote playback
// on the specified stream or have been active in the past inPastMs milliseconds. Remote
// playback isn't mutually exclusive with local playback.
static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state,
- uint32_t inPastMs = 0);
+ uint32_t inPastMs);
// returns true in *state if a recorder is currently recording with the specified source
static status_t isSourceActive(audio_source_t source, bool *state);
// set/get audio hardware parameters. The function accepts a list of parameters
// key value pairs in the form: key1=value1;key2=value2;...
// Some keys are reserved for standard parameters (See AudioParameter class).
+ // The versions with audio_io_handle_t are intended for internal media framework use only.
static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
+ // The versions without audio_io_handle_t are intended for JNI.
+ static status_t setParameters(const String8& keyValuePairs);
+ static String8 getParameters(const String8& keys);
static void setErrorCallback(audio_error_callback cb);
@@ -90,12 +94,14 @@ public:
static float linearToLog(int volume);
static int logToLinear(float volume);
+ // Returned samplingRate and frameCount output values are guaranteed
+ // to be non-zero if status == NO_ERROR
static status_t getOutputSamplingRate(uint32_t* samplingRate,
- audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ audio_stream_type_t stream);
static status_t getOutputFrameCount(size_t* frameCount,
- audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ audio_stream_type_t stream);
static status_t getOutputLatency(uint32_t* latency,
- audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ audio_stream_type_t stream);
static status_t getSamplingRate(audio_io_handle_t output,
audio_stream_type_t streamType,
uint32_t* samplingRate);
@@ -132,7 +138,7 @@ public:
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
// return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
- static size_t getInputFramesLost(audio_io_handle_t ioHandle);
+ static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
static int newAudioSessionId();
static void acquireAudioSessionId(int audioSession);
@@ -155,7 +161,8 @@ public:
class OutputDescriptor {
public:
OutputDescriptor()
- : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) {}
+ : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)
+ {}
uint32_t samplingRate;
audio_format_t format;
@@ -193,24 +200,32 @@ public:
static status_t setPhoneState(audio_mode_t state);
static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+
+ // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
+ // or release it with releaseOutput().
static audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
const audio_offload_info_t *offloadInfo = NULL);
+
static status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
- int session = 0);
+ int session);
static status_t stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
- int session = 0);
+ int session);
static void releaseOutput(audio_io_handle_t output);
+
+ // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
+ // or release it with releaseInput().
static audio_io_handle_t getInput(audio_source_t inputSource,
- uint32_t samplingRate = 0,
- audio_format_t format = AUDIO_FORMAT_DEFAULT,
- audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO,
- int sessionId = 0);
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ int sessionId);
+
static status_t startInput(audio_io_handle_t input);
static status_t stopInput(audio_io_handle_t input);
static void releaseInput(audio_io_handle_t input);
diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
index c29c7e5..99e9c3e 100644
--- a/include/media/AudioTimestamp.h
+++ b/include/media/AudioTimestamp.h
@@ -19,6 +19,8 @@
#include <time.h>
+namespace android {
+
class AudioTimestamp {
public:
AudioTimestamp() : mPosition(0) {
@@ -30,4 +32,6 @@ public:
struct timespec mTime; // corresponding CLOCK_MONOTONIC when frame is expected to present
};
+} // namespace
+
#endif // ANDROID_AUDIO_TIMESTAMP_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index f6646ab..5a50280 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -27,7 +27,7 @@ namespace android {
// ----------------------------------------------------------------------------
-class audio_track_cblk_t;
+struct audio_track_cblk_t;
class AudioTrackClientProxy;
class StaticAudioTrackClientProxy;
@@ -123,6 +123,8 @@ public:
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
* - BAD_VALUE: unsupported configuration
+ * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+ * and is undefined otherwise.
*/
static status_t getMinFrameCount(size_t* frameCount,
@@ -158,7 +160,7 @@ public:
* sampleRate: Data source sampling rate in Hz.
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
- * channelMask: Channel mask.
+ * 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
* latency of the track. The actual size selected by the AudioTrack could be
@@ -185,7 +187,7 @@ public:
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1);
@@ -210,7 +212,7 @@ public:
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1);
@@ -248,7 +250,7 @@ public:
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1);
@@ -336,7 +338,7 @@ public:
*/
status_t setSampleRate(uint32_t sampleRate);
- /* Return current source sample rate in Hz, or 0 if unknown */
+ /* Return current source sample rate in Hz */
uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
@@ -361,7 +363,7 @@ public:
/* 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. To set a marker at a position which would compute as 0,
- * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+ * a workaround is to set the marker at a nearby position such as ~0 or 1.
* If the AudioTrack has been opened with no callback function associated, the operation will
* fail.
*
@@ -452,7 +454,7 @@ public:
* Returned value:
* handle on audio hardware output
*/
- audio_io_handle_t getOutput();
+ audio_io_handle_t getOutput() const;
/* Returns the unique session ID associated with this track.
*
@@ -566,7 +568,7 @@ public:
uint32_t getUnderrunFrames() const;
/* Get the flags */
- audio_output_flags_t getFlags() const { return mFlags; }
+ audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; }
/* Set parameters - only possible when using direct output */
status_t setParameters(const String8& keyValuePairs);
@@ -626,53 +628,50 @@ protected:
// NS_INACTIVE inactive so don't run again until re-started
// NS_NEVER never again
static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
- nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
- status_t processStreamEnd(int32_t waitCount);
+ nsecs_t processAudioBuffer();
+ bool isOffloaded() const;
// caller must hold lock on mLock for all _l methods
- status_t createTrack_l(audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- size_t frameCount,
- audio_output_flags_t flags,
- const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output,
- size_t epoch);
+ status_t createTrack_l(size_t epoch);
// can only be called when mState != STATE_ACTIVE
void flush_l();
void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
- audio_io_handle_t getOutput_l();
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreTrack_l(const char *from);
- bool isOffloaded() const
+ bool isOffloaded_l() const
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
- // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
+ // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ audio_io_handle_t mOutput; // returned by AudioSystem::getOutput()
sp<AudioTrackThread> mAudioTrackThread;
+
float mVolume[2];
float mSendLevel;
mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it.
- size_t mFrameCount; // corresponds to current IAudioTrack
- size_t mReqFrameCount; // frame count to request the next time a new
- // IAudioTrack is needed
-
+ 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
+ // a new IAudioTrack is needed, non-decreasing
// constant after constructor or set()
audio_format_t mFormat; // as requested by client, not forced to 16-bit
audio_stream_type_t mStreamType;
uint32_t mChannelCount;
audio_channel_mask_t mChannelMask;
+ sp<IMemory> mSharedBuffer;
transfer_type mTransfer;
+ audio_offload_info_t mOffloadInfoCopy;
+ const audio_offload_info_t* mOffloadInfo;
// 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.
@@ -705,21 +704,25 @@ protected:
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback,
// at initial source sample rate
- bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
+ bool mRefreshRemaining; // processAudioBuffer() should refresh
+ // mRemainingFrames and mRetryOnPartialBuffer
// 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
- sp<IMemory> mSharedBuffer;
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
audio_output_flags_t mFlags;
+ // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD.
+ // mLock must be held to read or write those bits reliably.
+
int mSessionId;
int mAuxEffectId;
@@ -753,7 +756,6 @@ private:
sp<DeathNotifier> mDeathNotifier;
uint32_t mSequence; // incremented for each new IAudioTrack attempt
- audio_io_handle_t mOutput; // cached output io handle
int mClientUid;
};
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 282f275..ea225ac 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -64,9 +64,12 @@ public:
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
+ // On successful return, AudioFlinger takes over the handle
+ // reference and will release it when the track is destroyed.
+ // However on failure, the client is responsible for release.
audio_io_handle_t output,
pid_t tid, // -1 means unused, otherwise must be valid non-0
int *sessionId,
@@ -78,11 +81,14 @@ public:
status_t *status) = 0;
virtual sp<IAudioRecord> openRecord(
+ // On successful return, AudioFlinger takes over the handle
+ // reference and will release it when the track is destroyed.
+ // However on failure, the client is responsible for release.
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
pid_t tid, // -1 means unused, otherwise must be valid non-0
int *sessionId,
@@ -188,6 +194,7 @@ public:
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
+ // AudioFlinger doesn't take over handle reference from client
audio_io_handle_t output,
int sessionId,
status_t *status,
diff --git a/include/media/IMediaHTTPConnection.h b/include/media/IMediaHTTPConnection.h
new file mode 100644
index 0000000..e048b64
--- /dev/null
+++ b/include/media/IMediaHTTPConnection.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2013 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 I_MEDIA_HTTP_CONNECTION_H_
+
+#define I_MEDIA_HTTP_CONNECTION_H_
+
+#include <binder/IInterface.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+/** MUST stay in sync with IMediaHTTPConnection.aidl */
+
+struct IMediaHTTPConnection : public IInterface {
+ DECLARE_META_INTERFACE(MediaHTTPConnection);
+
+ virtual bool connect(
+ const char *uri, const KeyedVector<String8, String8> *headers) = 0;
+
+ virtual void disconnect() = 0;
+ virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
+ virtual off64_t getSize() = 0;
+ virtual status_t getMIMEType(String8 *mimeType) = 0;
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPConnection);
+};
+
+} // namespace android
+
+#endif // I_MEDIA_HTTP_CONNECTION_H_
diff --git a/include/media/IMediaHTTPService.h b/include/media/IMediaHTTPService.h
new file mode 100644
index 0000000..f66d6c8
--- /dev/null
+++ b/include/media/IMediaHTTPService.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2013 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 I_MEDIA_HTTP_SERVICE_H_
+
+#define I_MEDIA_HTTP_SERVICE_H_
+
+#include <binder/IInterface.h>
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+/** MUST stay in sync with IMediaHTTPService.aidl */
+
+struct IMediaHTTPService : public IInterface {
+ DECLARE_META_INTERFACE(MediaHTTPService);
+
+ virtual sp<IMediaHTTPConnection> makeHTTPConnection() = 0;
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPService);
+};
+
+} // namespace android
+
+#endif // I_MEDIA_HTTP_SERVICE_H_
diff --git a/include/media/IMediaMetadataRetriever.h b/include/media/IMediaMetadataRetriever.h
index 6dbb2d7..2529800 100644
--- a/include/media/IMediaMetadataRetriever.h
+++ b/include/media/IMediaMetadataRetriever.h
@@ -26,6 +26,8 @@
namespace android {
+struct IMediaHTTPService;
+
class IMediaMetadataRetriever: public IInterface
{
public:
@@ -33,6 +35,7 @@ public:
virtual void disconnect() = 0;
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *srcUrl,
const KeyedVector<String8, String8> *headers = NULL) = 0;
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
index 0cbd269..db62cd5 100644
--- a/include/media/IMediaPlayer.h
+++ b/include/media/IMediaPlayer.h
@@ -33,6 +33,7 @@ class Parcel;
class Surface;
class IStreamSource;
class IGraphicBufferProducer;
+struct IMediaHTTPService;
class IMediaPlayer: public IInterface
{
@@ -41,8 +42,11 @@ public:
virtual void disconnect() = 0;
- virtual status_t setDataSource(const char *url,
- const KeyedVector<String8, String8>* headers) = 0;
+ virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8>* headers) = 0;
+
virtual status_t setDataSource(int fd, int64_t offset, int64_t length) = 0;
virtual status_t setDataSource(const sp<IStreamSource>& source) = 0;
virtual status_t setVideoSurfaceTexture(
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 2998b37..5b45376 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -34,6 +34,7 @@ namespace android {
struct ICrypto;
struct IDrm;
struct IHDCP;
+struct IMediaHTTPService;
class IMediaRecorder;
class IOMX;
class IRemoteDisplay;
@@ -49,9 +50,14 @@ public:
virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
- virtual status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+ virtual status_t decode(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ uint32_t *pSampleRate,
+ int* pNumChannels,
+ audio_format_t* pFormat,
+ const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+
virtual status_t decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
int* pNumChannels, audio_format_t* pFormat,
const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
@@ -93,9 +99,6 @@ public:
virtual void addBatteryData(uint32_t params) = 0;
virtual status_t pullBatteryData(Parcel* reply) = 0;
-
- virtual status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index 9c8451c..3db2c38 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -142,6 +142,8 @@ public:
enum InternalOptionType {
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_START_TIME, // data is an int64_t
};
virtual status_t setInternalOption(
node_id node,
diff --git a/include/media/MediaMetadataRetrieverInterface.h b/include/media/MediaMetadataRetrieverInterface.h
index ecc3b65..bb6b97b 100644
--- a/include/media/MediaMetadataRetrieverInterface.h
+++ b/include/media/MediaMetadataRetrieverInterface.h
@@ -24,6 +24,8 @@
namespace android {
+struct IMediaHTTPService;
+
// Abstract base class
class MediaMetadataRetrieverBase : public RefBase
{
@@ -32,6 +34,7 @@ public:
virtual ~MediaMetadataRetrieverBase() {}
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers = NULL) = 0;
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 26d8729..87717da 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -137,6 +137,7 @@ public:
}
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers = NULL) = 0;
@@ -213,11 +214,6 @@ public:
return INVALID_OPERATION;
}
- virtual status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- return INVALID_OPERATION;
- }
-
private:
friend class MediaPlayerService;
diff --git a/include/media/mediametadataretriever.h b/include/media/mediametadataretriever.h
index 0df77c1..b35cf32 100644
--- a/include/media/mediametadataretriever.h
+++ b/include/media/mediametadataretriever.h
@@ -25,6 +25,7 @@
namespace android {
+struct IMediaHTTPService;
class IMediaPlayerService;
class IMediaMetadataRetriever;
@@ -68,6 +69,7 @@ public:
void disconnect();
status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *dataSourceUrl,
const KeyedVector<String8, String8> *headers = NULL);
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 4c05fc3..f8e4e3b 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -189,6 +189,8 @@ public:
virtual void notify(int msg, int ext1, int ext2, const Parcel *obj) = 0;
};
+struct IMediaHTTPService;
+
class MediaPlayer : public BnMediaPlayerClient,
public virtual IMediaDeathNotifier
{
@@ -199,6 +201,7 @@ public:
void disconnect();
status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers);
@@ -224,9 +227,14 @@ public:
bool isLooping();
status_t setVolume(float leftVolume, float rightVolume);
void notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
- static status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize);
+ static status_t decode(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ uint32_t *pSampleRate,
+ int* pNumChannels,
+ audio_format_t* pFormat,
+ const sp<IMemoryHeap>& heap,
+ size_t *pSize);
static status_t decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
int* pNumChannels, audio_format_t* pFormat,
const sp<IMemoryHeap>& heap, size_t *pSize);
@@ -242,9 +250,6 @@ public:
status_t setRetransmitEndpoint(const char* addrString, uint16_t port);
status_t setNextMediaPlayer(const sp<MediaPlayer>& player);
- status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList);
-
private:
void clear_l();
status_t seekTo_l(int msec);
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index 88a42a0..142cb90 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -39,7 +39,7 @@ typedef void (*media_completion_f)(status_t status, void *cookie);
enum video_source {
VIDEO_SOURCE_DEFAULT = 0,
VIDEO_SOURCE_CAMERA = 1,
- VIDEO_SOURCE_GRALLOC_BUFFER = 2,
+ VIDEO_SOURCE_SURFACE = 2,
VIDEO_SOURCE_LIST_END // must be last - used to validate audio source type
};
diff --git a/include/media/mediascanner.h b/include/media/mediascanner.h
index a73403b..4537679 100644
--- a/include/media/mediascanner.h
+++ b/include/media/mediascanner.h
@@ -21,6 +21,7 @@
#include <utils/threads.h>
#include <utils/List.h>
#include <utils/Errors.h>
+#include <utils/String8.h>
#include <pthread.h>
struct dirent;
@@ -29,6 +30,7 @@ namespace android {
class MediaScannerClient;
class StringArray;
+class CharacterEncodingDetector;
enum MediaScanResult {
// This file or directory was scanned successfully.
@@ -94,15 +96,9 @@ public:
virtual status_t setMimeType(const char* mimeType) = 0;
protected:
- void convertValues(uint32_t encoding);
-
-protected:
- // cached name and value strings, for native encoding support.
- StringArray* mNames;
- StringArray* mValues;
-
- // default encoding based on MediaScanner::mLocale string
- uint32_t mLocaleEncoding;
+ // default encoding from MediaScanner::mLocale
+ String8 mLocale;
+ CharacterEncodingDetector *mEncodingDetector;
};
}; // namespace android
diff --git a/include/media/nbaio/AudioBufferProviderSource.h b/include/media/nbaio/AudioBufferProviderSource.h
index 2c4aaff..b16e20a 100644
--- a/include/media/nbaio/AudioBufferProviderSource.h
+++ b/include/media/nbaio/AudioBufferProviderSource.h
@@ -27,7 +27,7 @@ namespace android {
class AudioBufferProviderSource : public NBAIO_Source {
public:
- AudioBufferProviderSource(AudioBufferProvider *provider, NBAIO_Format format);
+ AudioBufferProviderSource(AudioBufferProvider *provider, const NBAIO_Format& format);
virtual ~AudioBufferProviderSource();
// NBAIO_Port interface
diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
index d3802fe..b09b35f 100644
--- a/include/media/nbaio/MonoPipe.h
+++ b/include/media/nbaio/MonoPipe.h
@@ -41,7 +41,7 @@ public:
// Note: whatever shares this object with another thread needs to do so in an SMP-safe way (like
// creating it the object before creating the other thread, or storing the object with a
// release_store). Otherwise the other thread could see a partially-constructed object.
- MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock = false);
+ MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock = false);
virtual ~MonoPipe();
// NBAIO_Port interface
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index 1da0c73..56896b9 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -52,31 +52,41 @@ enum {
// the combinations that are actually needed within AudioFlinger. If the list of combinations grows
// too large, then this decision should be re-visited.
// Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed.
-typedef unsigned NBAIO_Format;
-enum {
- Format_Invalid
+struct NBAIO_Format {
+//private:
+ unsigned mPacked;
};
+extern const NBAIO_Format Format_Invalid;
+
// Return the frame size of an NBAIO_Format in bytes
-size_t Format_frameSize(NBAIO_Format format);
+size_t Format_frameSize(const NBAIO_Format& format);
// Return the frame size of an NBAIO_Format as a bit shift
-size_t Format_frameBitShift(NBAIO_Format format);
+// or -1 if frame size is not a power of 2
+int Format_frameBitShift(const NBAIO_Format& format);
// Convert a sample rate in Hz and channel count to an NBAIO_Format
+// FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount);
// Return the sample rate in Hz of an NBAIO_Format
-unsigned Format_sampleRate(NBAIO_Format format);
+unsigned Format_sampleRate(const NBAIO_Format& format);
// Return the channel count of an NBAIO_Format
-unsigned Format_channelCount(NBAIO_Format format);
+unsigned Format_channelCount(const NBAIO_Format& format);
// Callbacks used by NBAIO_Sink::writeVia() and NBAIO_Source::readVia() below.
typedef ssize_t (*writeVia_t)(void *user, void *buffer, size_t count);
typedef ssize_t (*readVia_t)(void *user, const void *buffer,
size_t count, int64_t readPTS);
+// Check whether an NBAIO_Format is valid
+bool Format_isValid(const NBAIO_Format& format);
+
+// Compare two NBAIO_Format values
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2);
+
// Abstract class (interface) representing a data port.
class NBAIO_Port : public RefBase {
@@ -115,8 +125,8 @@ public:
virtual NBAIO_Format format() const { return mNegotiated ? mFormat : Format_Invalid; }
protected:
- NBAIO_Port(NBAIO_Format format) : mNegotiated(false), mFormat(format),
- mBitShift(Format_frameBitShift(format)) { }
+ NBAIO_Port(const NBAIO_Format& format) : mNegotiated(false), mFormat(format),
+ mBitShift(Format_frameBitShift(format)) { }
virtual ~NBAIO_Port() { }
// Implementations are free to ignore these if they don't need them
@@ -220,7 +230,7 @@ public:
virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
protected:
- NBAIO_Sink(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
@@ -311,7 +321,7 @@ public:
virtual void onTimestamp(const AudioTimestamp& timestamp) { }
protected:
- NBAIO_Source(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/Pipe.h b/include/media/nbaio/Pipe.h
index 79a4eee..c784129 100644
--- a/include/media/nbaio/Pipe.h
+++ b/include/media/nbaio/Pipe.h
@@ -30,7 +30,7 @@ class Pipe : public NBAIO_Sink {
public:
// maxFrames will be rounded up to a power of 2, and all slots are available. Must be >= 2.
- Pipe(size_t maxFrames, NBAIO_Format format);
+ Pipe(size_t maxFrames, const NBAIO_Format& format);
virtual ~Pipe();
// NBAIO_Port interface
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index f1636e6..e284109 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -67,6 +67,8 @@ struct ACodec : public AHierarchicalStateMachine {
void signalRequestIDRFrame();
+ bool isConfiguredForAdaptivePlayback() { return mIsConfiguredForAdaptivePlayback; }
+
struct PortDescription : public RefBase {
size_t countBuffers();
IOMX::buffer_id bufferIDAt(size_t index) const;
@@ -187,6 +189,7 @@ private:
bool mIsEncoder;
bool mUseMetadataOnEncoderOutput;
bool mShutdownInProgress;
+ bool mIsConfiguredForAdaptivePlayback;
// If "mKeepComponentAllocated" we only transition back to Loaded state
// and do not release the component instance.
@@ -202,6 +205,9 @@ private:
int32_t mMetaDataBuffersToSubmit;
int64_t mRepeatFrameDelayUs;
+ int64_t mMaxPtsGapUs;
+
+ bool mCreateInputBuffersSuspended;
status_t setCyclicIntraMacroblockRefresh(const sp<AMessage> &msg, int32_t mode);
status_t allocateBuffersOnPort(OMX_U32 portIndex);
diff --git a/include/media/stagefright/CameraSource.h b/include/media/stagefright/CameraSource.h
index a829916..69cfbd0 100644
--- a/include/media/stagefright/CameraSource.h
+++ b/include/media/stagefright/CameraSource.h
@@ -185,6 +185,8 @@ protected:
virtual void dataCallbackTimestamp(int64_t timestampUs, int32_t msgType,
const sp<IMemory> &data);
+ void releaseCamera();
+
private:
friend class CameraSourceListener;
@@ -233,7 +235,6 @@ private:
int32_t frameRate);
void stopCameraRecording();
- void releaseCamera();
status_t reset();
CameraSource(const CameraSource &);
diff --git a/include/media/stagefright/DataSource.h b/include/media/stagefright/DataSource.h
index 157b1aa..f8787dd 100644
--- a/include/media/stagefright/DataSource.h
+++ b/include/media/stagefright/DataSource.h
@@ -31,6 +31,7 @@
namespace android {
struct AMessage;
+struct IMediaHTTPService;
class String8;
class DataSource : public RefBase {
@@ -43,6 +44,7 @@ public:
};
static sp<DataSource> CreateFromURI(
+ const sp<IMediaHTTPService> &httpService,
const char *uri,
const KeyedVector<String8, String8> *headers = NULL);
diff --git a/include/media/stagefright/MediaCodecSource.h b/include/media/stagefright/MediaCodecSource.h
new file mode 100644
index 0000000..4b18a0b
--- /dev/null
+++ b/include/media/stagefright/MediaCodecSource.h
@@ -0,0 +1,134 @@
+/*
+ * 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 MediaCodecSource_H_
+#define MediaCodecSource_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/AHandlerReflector.h>
+#include <media/stagefright/MediaSource.h>
+
+namespace android {
+
+class ALooper;
+class AMessage;
+class IGraphicBufferProducer;
+class MediaCodec;
+class MetaData;
+
+struct MediaCodecSource : public MediaSource,
+ public MediaBufferObserver {
+ enum FlagBits {
+ FLAG_USE_SURFACE_INPUT = 1,
+ FLAG_USE_METADATA_INPUT = 2,
+ };
+
+ static sp<MediaCodecSource> Create(
+ const sp<ALooper> &looper,
+ const sp<AMessage> &format,
+ const sp<MediaSource> &source,
+ uint32_t flags = 0);
+
+ bool isVideo() const { return mIsVideo; }
+ sp<IGraphicBufferProducer> getGraphicBufferProducer();
+
+ // MediaSource
+ virtual status_t start(MetaData *params = NULL);
+ virtual status_t stop();
+ virtual status_t pause();
+ virtual sp<MetaData> getFormat() { return mMeta; }
+ virtual status_t read(
+ MediaBuffer **buffer,
+ const ReadOptions *options = NULL);
+
+ // MediaBufferObserver
+ virtual void signalBufferReturned(MediaBuffer *buffer);
+
+ // for AHandlerReflector
+ void onMessageReceived(const sp<AMessage> &msg);
+
+protected:
+ virtual ~MediaCodecSource();
+
+private:
+ struct Puller;
+
+ enum {
+ kWhatPullerNotify,
+ kWhatEncoderActivity,
+ kWhatStart,
+ kWhatStop,
+ kWhatPause,
+ };
+
+ MediaCodecSource(
+ const sp<ALooper> &looper,
+ const sp<AMessage> &outputFormat,
+ const sp<MediaSource> &source,
+ uint32_t flags = 0);
+
+ status_t onStart(MetaData *params);
+ status_t init();
+ status_t initEncoder();
+ void releaseEncoder();
+ status_t feedEncoderInputBuffers();
+ void scheduleDoMoreWork();
+ status_t doMoreWork();
+ void suspend();
+ void resume(int64_t skipFramesBeforeUs = -1ll);
+ void signalEOS(status_t err = ERROR_END_OF_STREAM);
+ bool reachedEOS();
+ status_t postSynchronouslyAndReturnError(const sp<AMessage> &msg);
+
+ sp<ALooper> mLooper;
+ sp<ALooper> mCodecLooper;
+ sp<AHandlerReflector<MediaCodecSource> > mReflector;
+ sp<AMessage> mOutputFormat;
+ sp<MetaData> mMeta;
+ sp<Puller> mPuller;
+ sp<MediaCodec> mEncoder;
+ uint32_t mFlags;
+ List<uint32_t> mStopReplyIDQueue;
+ bool mIsVideo;
+ bool mStarted;
+ bool mStopping;
+ bool mDoMoreWorkPending;
+ bool mPullerReachedEOS;
+ sp<AMessage> mEncoderActivityNotify;
+ sp<IGraphicBufferProducer> mGraphicBufferProducer;
+ Vector<sp<ABuffer> > mEncoderInputBuffers;
+ Vector<sp<ABuffer> > mEncoderOutputBuffers;
+ List<MediaBuffer *> mInputBufferQueue;
+ List<size_t> mAvailEncoderInputIndices;
+ List<int64_t> mDecodingTimeQueue; // decoding time (us) for video
+
+ // audio drift time
+ int64_t mFirstSampleTimeUs;
+ List<int64_t> mDriftTimeQueue;
+
+ // following variables are protected by mOutputBufferLock
+ Mutex mOutputBufferLock;
+ Condition mOutputBufferCond;
+ List<MediaBuffer*> mOutputBufferQueue;
+ bool mEncodedReachedEOS;
+ status_t mErrorCode;
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaCodecSource);
+};
+
+} // namespace android
+
+#endif /* MediaCodecSource_H_ */
diff --git a/include/media/stagefright/MediaHTTP.h b/include/media/stagefright/MediaHTTP.h
new file mode 100644
index 0000000..006d8d8
--- /dev/null
+++ b/include/media/stagefright/MediaHTTP.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2013 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_HTTP_H_
+
+#define MEDIA_HTTP_H_
+
+#include <media/stagefright/foundation/AString.h>
+
+#include "include/HTTPBase.h"
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+struct MediaHTTP : public HTTPBase {
+ MediaHTTP(const sp<IMediaHTTPConnection> &conn);
+
+ virtual status_t connect(
+ const char *uri,
+ const KeyedVector<String8, String8> *headers,
+ off64_t offset);
+
+ virtual void disconnect();
+
+ virtual status_t initCheck() const;
+
+ virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+
+ virtual status_t getSize(off64_t *size);
+
+ virtual uint32_t flags();
+
+ virtual status_t reconnectAtOffset(off64_t offset);
+
+protected:
+ virtual ~MediaHTTP();
+
+ virtual sp<DecryptHandle> DrmInitialization(const char* mime);
+ virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
+ virtual String8 getUri();
+ virtual String8 getMIMEType() const;
+
+private:
+ status_t mInitCheck;
+ sp<IMediaHTTPConnection> mHTTPConnection;
+
+ KeyedVector<String8, String8> mLastHeaders;
+ AString mLastURI;
+
+ bool mCachedSizeValid;
+ off64_t mCachedSize;
+
+ sp<DecryptHandle> mDecryptHandle;
+ DrmManagerClient *mDrmManagerClient;
+
+ void clearDRMState_l();
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaHTTP);
+};
+
+} // namespace android
+
+#endif // MEDIA_HTTP_H_
diff --git a/include/media/stagefright/NuMediaExtractor.h b/include/media/stagefright/NuMediaExtractor.h
index 5ae6f6b..402e7f8 100644
--- a/include/media/stagefright/NuMediaExtractor.h
+++ b/include/media/stagefright/NuMediaExtractor.h
@@ -31,6 +31,7 @@ namespace android {
struct ABuffer;
struct AMessage;
struct DataSource;
+struct IMediaHTTPService;
struct MediaBuffer;
struct MediaExtractor;
struct MediaSource;
@@ -45,6 +46,7 @@ struct NuMediaExtractor : public RefBase {
NuMediaExtractor();
status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *path,
const KeyedVector<String8, String8> *headers = NULL);
diff --git a/include/media/stagefright/SkipCutBuffer.h b/include/media/stagefright/SkipCutBuffer.h
index 2653b53..098aa69 100644
--- a/include/media/stagefright/SkipCutBuffer.h
+++ b/include/media/stagefright/SkipCutBuffer.h
@@ -47,6 +47,7 @@ class SkipCutBuffer: public RefBase {
private:
void write(const char *src, size_t num);
size_t read(char *dst, size_t num);
+ int32_t mSkip;
int32_t mFrontPadding;
int32_t mBackPadding;
int32_t mWriteHead;
diff --git a/include/media/stagefright/timedtext/TimedTextDriver.h b/include/media/stagefright/timedtext/TimedTextDriver.h
index f23c337..37ef674 100644
--- a/include/media/stagefright/timedtext/TimedTextDriver.h
+++ b/include/media/stagefright/timedtext/TimedTextDriver.h
@@ -25,6 +25,7 @@
namespace android {
class ALooper;
+struct IMediaHTTPService;
class MediaPlayerBase;
class MediaSource;
class Parcel;
@@ -34,7 +35,9 @@ class DataSource;
class TimedTextDriver {
public:
- TimedTextDriver(const wp<MediaPlayerBase> &listener);
+ TimedTextDriver(
+ const wp<MediaPlayerBase> &listener,
+ const sp<IMediaHTTPService> &httpService);
~TimedTextDriver();
@@ -77,6 +80,7 @@ private:
sp<ALooper> mLooper;
sp<TimedTextPlayer> mPlayer;
wp<MediaPlayerBase> mListener;
+ sp<IMediaHTTPService> mHTTPService;
// Variables to be guarded by mLock.
State mState;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 7fd9379..3901e79 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -48,7 +48,7 @@ namespace android {
#define CBLK_STREAM_END_DONE 0x400 // set by server on render completion, cleared by client
//EL_FIXME 20 seconds may not be enough and must be reconciled with new obtainBuffer implementation
-#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 //assuming upto a maximum of 20 seconds of offloaded
+#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 // assuming up to a maximum of 20 seconds of offloaded
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
@@ -65,7 +65,9 @@ typedef SingleStateQueue<StaticAudioTrackState> StaticAudioTrackSingleStateQueue
struct AudioTrackSharedStatic {
StaticAudioTrackSingleStateQueue::Shared
mSingleStateQueue;
- size_t mBufferPosition; // updated asynchronously by server,
+ // This field should be a size_t, but since it is located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mBufferPosition; // updated asynchronously by server,
// "for entertainment purposes only"
};
@@ -96,11 +98,7 @@ struct audio_track_cblk_t
// The value should be used "for entertainment purposes only",
// which means don't make important decisions based on it.
- size_t frameCount_; // used during creation to pass actual track buffer size
- // from AudioFlinger to client, and not referenced again
- // FIXME remove here and replace by createTrack() in/out
- // parameter
- // renamed to "_" to detect incorrect use
+ uint32_t mPad1; // unused
volatile int32_t mFutex; // event flag: down (P) by client,
// up (V) by server or binderDied() or interrupt()
@@ -108,7 +106,9 @@ struct audio_track_cblk_t
private:
- size_t mMinimum; // server wakes up client if available >= mMinimum
+ // This field should be a size_t, but since it is located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mMinimum; // server wakes up client if available >= mMinimum
// Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
// Left channel is in [0:15], right channel is in [16:31].
@@ -245,7 +245,11 @@ public:
}
void setMinimum(size_t minimum) {
- mCblk->mMinimum = minimum;
+ // This can only happen on a 64-bit client
+ if (minimum > UINT32_MAX) {
+ minimum = UINT32_MAX;
+ }
+ mCblk->mMinimum = (uint32_t) minimum;
}
// Return the number of frames that would need to be obtained and released
diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h
index 46a5946..d483061 100644
--- a/include/private/media/StaticAudioTrackState.h
+++ b/include/private/media/StaticAudioTrackState.h
@@ -25,9 +25,13 @@ namespace android {
// state is wrapped by a SingleStateQueue.
struct StaticAudioTrackState {
// do not define constructors, destructors, or virtual methods
- size_t mLoopStart;
- size_t mLoopEnd;
- int mLoopCount;
+
+ // These fields should both be size_t, but since they are located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mLoopStart;
+ uint32_t mLoopEnd;
+
+ int mLoopCount;
};
} // namespace android