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-rw-r--r--include/media/AudioSystem.h13
-rw-r--r--include/media/AudioTrack.h34
-rw-r--r--include/media/IAudioFlinger.h4
-rw-r--r--include/media/IAudioPolicyService.h6
-rw-r--r--include/media/IAudioTrack.h4
-rw-r--r--include/media/MediaPlayerInterface.h17
-rw-r--r--include/media/stagefright/AudioPlayer.h13
-rw-r--r--include/media/stagefright/Utils.h11
-rw-r--r--include/private/media/AudioTrackShared.h4
9 files changed, 94 insertions, 12 deletions
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index b11c812..09160cc 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -128,8 +128,10 @@ public:
// - BAD_VALUE: invalid parameter
// NOTE: this feature is not supported on all hardware platforms and it is
// necessary to check returned status before using the returned values.
- static status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
- audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ static status_t getRenderPosition(audio_io_handle_t output,
+ size_t *halFrames,
+ size_t *dspFrames,
+ 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);
@@ -197,7 +199,8 @@ public:
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);
+ 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);
@@ -245,6 +248,10 @@ public:
static uint32_t getPrimaryOutputSamplingRate();
static size_t getPrimaryOutputFrameCount();
+ // Check if hw offload is possible for given format, stream type, sample rate,
+ // bit rate, duration, video and streaming or offload property is enabled
+ static bool isOffloadSupported(const audio_offload_info_t& info);
+
// ----------------------------------------------------------------------------
private:
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 @@ public:
// 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 @@ public:
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 @@ public:
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 @@ public:
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 @@ private:
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 @@ public:
*/
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 @@ protected:
// 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 @@ protected:
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);
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 9c3067e..f8a9f2b 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -49,6 +49,7 @@ public:
TRACK_DEFAULT = 0, // client requests a default AudioTrack
TRACK_TIMED = 1, // client requests a TimedAudioTrack
TRACK_FAST = 2, // client requests a fast AudioTrack or AudioRecord
+ TRACK_OFFLOAD = 4, // client requests offload to hw codec
};
typedef uint32_t track_flags_t;
@@ -137,7 +138,8 @@ public:
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
uint32_t *pLatencyMs,
- audio_output_flags_t flags) = 0;
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2) = 0;
virtual status_t closeOutput(audio_io_handle_t output) = 0;
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index b5ad4ef..09b9ea6 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -53,7 +53,8 @@ public:
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = 0,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0) = 0;
@@ -95,6 +96,9 @@ public:
virtual status_t queryDefaultPreProcessing(int audioSession,
effect_descriptor_t *descriptors,
uint32_t *count) = 0;
+ // Check if offload is possible for given format, stream type, sample rate,
+ // bit rate, duration, video and streaming or offload property is enabled
+ virtual bool isOffloadSupported(const audio_offload_info_t& info) = 0;
};
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 144be0e..1014403 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -25,6 +25,7 @@
#include <binder/IInterface.h>
#include <binder/IMemory.h>
#include <utils/LinearTransform.h>
+#include <utils/String8.h>
namespace android {
@@ -82,6 +83,9 @@ public:
or Tungsten time. The values for target are defined in AudioTrack.h */
virtual status_t setMediaTimeTransform(const LinearTransform& xform,
int target) = 0;
+
+ /* Send parameters to the audio hardware */
+ virtual status_t setParameters(const String8& keyValuePairs) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 9a75f81..61f7dc7 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -74,9 +74,18 @@ public:
// AudioSink: abstraction layer for audio output
class AudioSink : public RefBase {
public:
+ enum cb_event_t {
+ CB_EVENT_FILL_BUFFER, // Request to write more data to buffer.
+ CB_EVENT_STREAM_END, // Sent after all the buffers queued in AF and HW are played
+ // back (after stop is called)
+ CB_EVENT_TEAR_DOWN // The AudioTrack was invalidated due to use case change:
+ // Need to re-evaluate offloading options
+ };
+
// Callback returns the number of bytes actually written to the buffer.
typedef size_t (*AudioCallback)(
- AudioSink *audioSink, void *buffer, size_t size, void *cookie);
+ AudioSink *audioSink, void *buffer, size_t size, void *cookie,
+ cb_event_t event);
virtual ~AudioSink() {}
virtual bool ready() const = 0; // audio output is open and ready
@@ -99,7 +108,8 @@ public:
int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT,
AudioCallback cb = NULL,
void *cookie = NULL,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
virtual void start() = 0;
virtual ssize_t write(const void* buffer, size_t size) = 0;
@@ -110,6 +120,9 @@ public:
virtual status_t setPlaybackRatePermille(int32_t rate) { return INVALID_OPERATION; }
virtual bool needsTrailingPadding() { return true; }
+
+ virtual status_t setParameters(const String8& keyValuePairs) { return NO_ERROR; };
+ virtual String8 getParameters(const String8& keys) { return String8::empty(); };
};
MediaPlayerBase() : mCookie(0), mNotify(0) {}
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 3bf046d..ec9f2df 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -36,8 +36,13 @@ public:
SEEK_COMPLETE
};
+ enum {
+ ALLOW_DEEP_BUFFERING = 0x01,
+ USE_OFFLOAD = 0x02
+ };
+
AudioPlayer(const sp<MediaPlayerBase::AudioSink> &audioSink,
- bool allowDeepBuffering = false,
+ uint32_t flags = 0,
AwesomePlayer *audioObserver = NULL);
virtual ~AudioPlayer();
@@ -67,6 +72,8 @@ public:
status_t setPlaybackRatePermille(int32_t ratePermille);
+ void notifyAudioEOS();
+
private:
friend class VideoEditorAudioPlayer;
sp<MediaSource> mSource;
@@ -107,7 +114,8 @@ private:
static size_t AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
- void *data, size_t size, void *me);
+ void *data, size_t size, void *me,
+ MediaPlayerBase::AudioSink::cb_event_t event);
size_t fillBuffer(void *data, size_t size);
@@ -116,6 +124,7 @@ private:
void reset();
uint32_t getNumFramesPendingPlayout() const;
+ int64_t getOutputPlayPositionUs_l() const;
AudioPlayer(const AudioPlayer &);
AudioPlayer &operator=(const AudioPlayer &);
diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h
index 73940d3..c24f612 100644
--- a/include/media/stagefright/Utils.h
+++ b/include/media/stagefright/Utils.h
@@ -22,6 +22,8 @@
#include <stdint.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
+#include <system/audio.h>
+#include <media/MediaPlayerInterface.h>
namespace android {
@@ -48,6 +50,15 @@ void convertMessageToMetaData(
AString MakeUserAgent();
+// Convert a MIME type to a AudioSystem::audio_format
+status_t mapMimeToAudioFormat(audio_format_t& format, const char* mime);
+
+// Send information from MetaData to the HAL via AudioSink
+status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink, const sp<MetaData>& meta);
+
+// Check whether the stream defined by meta can be offloaded to hardware
+bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, bool isStreaming);
+
} // namespace android
#endif // UTILS_H_
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index ef5bb8d..b41684a 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -44,6 +44,10 @@ namespace android {
#define CBLK_BUFFER_END 0x80 // set by server when the position reaches end of buffer if not looping
#define CBLK_OVERRUN 0x100 // set by server immediately on input overrun, cleared by client
#define CBLK_INTERRUPT 0x200 // set by client on interrupt(), cleared by client in obtainBuffer()
+#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
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
generated by cgit v1.1 at 2024-06-07 09:04:37 +0000