Public API changes for audio offload support.
NOTE: this does _not_ include all private member variables added
to classes as part of offload support. Only public/protected functions
and stubs functions/variables needed to make the changes buildable.
- isOffloadSupported() added to audio policy service
A stub implementation is required to build, this always returns false
- setParameters() added to IAudioTrack
A stub implementation is required to build, this always returns
INVALID_OPERATION
- CBlk flag for stream end
- Change AudioSystem::getRenderPosition() to take an audio_output_t
so caller can specify which output to query
- Add AudioSystem::isOffloadSupported()
This is fully implemented down to the AudioFlinger function
AudioPolicyServer::isOffloadSupported() which is just a stub
that always returns false.
- Add EVENT_STREAM_END to AudioTrack interface.
STREAM_END is used to signal when the hardware has actually finished
playing all the data it was sent.
- Add event type enumeration to media player interface AudioSink callbacks
so that the same callback can be used to handle multiple types of
event. For offloaded tracks we also have to handle STREAM_END and
TEAR_DOWN events
- Pass audio_offload_info_t to various functions used for opening outputs,
tracks and audio players. This passes additional information about the
compressed stream down to the HAL when using offload.
For publicly-available APIs this is an optional parameter (for some of
the internal and low-level APIs around the HAL interface it is mandatory)
- Add getParameters() and setParameters() API to AudioTrack
Currently dummy implementations.
- Change AudioPlayer contructor so that it takes a set of bitflags defining what
options are required. This replaces the original bool which only specified
whether to use deep buffering.
- Changes to StageFright class definition related to handling tearing-down of
an offloaded track when we need to switch back to software decode
- Define new StageFright utility functions used for offloaded tracks
Currently dummy implementations.
- AudioFlinger changes to use extended audio_config_t.
Fills in audio_offload_info_t member if this info is passed in when
opening an output.
- libvideoeditor changes required to add the new event type parameter
to AudioSink callback functions
- libmediaplayerservice changes required to add the new event type parameter
to AudioSink callback functions
Change-Id: I3ab41138aa1083d81fe83b886a9b1021ec7320f1
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index e9bb76a..6727601 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -60,6 +60,8 @@
// Not currently used by android.media.AudioTrack.
EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and
// voluntary invalidation by mediaserver, or mediaserver crash.
+ EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played
+ // back (after stop is called)
};
/* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -175,7 +177,8 @@
void* user = NULL,
int notificationFrames = 0,
int sessionId = 0,
- transfer_type transferType = TRANSFER_DEFAULT);
+ transfer_type transferType = TRANSFER_DEFAULT,
+ const audio_offload_info_t *offloadInfo = NULL);
/* Creates an audio track and registers it with AudioFlinger.
* With this constructor, the track is configured for static buffer mode.
@@ -198,7 +201,8 @@
void* user = NULL,
int notificationFrames = 0,
int sessionId = 0,
- transfer_type transferType = TRANSFER_DEFAULT);
+ transfer_type transferType = TRANSFER_DEFAULT,
+ const audio_offload_info_t *offloadInfo = NULL);
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioTrack.
@@ -233,7 +237,8 @@
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
int sessionId = 0,
- transfer_type transferType = TRANSFER_DEFAULT);
+ transfer_type transferType = TRANSFER_DEFAULT,
+ const audio_offload_info_t *offloadInfo = NULL);
/* Result of constructing the AudioTrack. This must be checked
* before using any AudioTrack API (except for set()), because using
@@ -521,6 +526,15 @@
struct timespec *elapsed = NULL, size_t *nonContig = NULL);
public:
+//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy
+// enum {
+// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value
+// TEAR_DOWN = 0x80000002,
+// STOPPED = 1,
+// STREAM_END_WAIT,
+// STREAM_END
+// };
+
/* 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);
@@ -550,6 +564,15 @@
*/
uint32_t getUnderrunFrames() const;
+ /* Get the flags */
+ audio_output_flags_t getFlags() const { return mFlags; }
+
+ /* Set parameters - only possible when using direct output */
+ status_t setParameters(const String8& keyValuePairs);
+
+ /* Get parameters */
+ String8 getParameters(const String8& keys);
+
protected:
/* copying audio tracks is not allowed */
AudioTrack(const AudioTrack& other);
@@ -590,8 +613,11 @@
// 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);
+
// 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,
@@ -607,6 +633,8 @@
void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
audio_io_handle_t getOutput_l();
+ status_t getPosition_l(uint32_t *position);
+
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreTrack_l(const char *from);