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>

8 files changed, 90 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_