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>

11 files changed, 129 insertions, 25 deletions

diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 693df60..a6dedec 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -361,8 +361,8 @@ status_t AudioSystem::setVoiceVolume(float value)
     return af->setVoiceVolume(value);
 }
 
-status_t AudioSystem::getRenderPosition(size_t *halFrames, size_t *dspFrames,
-        audio_stream_type_t stream)
+status_t AudioSystem::getRenderPosition(audio_io_handle_t output, size_t *halFrames,
+                                        size_t *dspFrames, audio_stream_type_t stream)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
@@ -371,7 +371,11 @@ status_t AudioSystem::getRenderPosition(size_t *halFrames, size_t *dspFrames,
         stream = AUDIO_STREAM_MUSIC;
     }
 
-    return af->getRenderPosition(halFrames, dspFrames, getOutput(stream));
+    if (output == 0) {
+        output = getOutput(stream);
+    }
+
+    return af->getRenderPosition(halFrames, dspFrames, output);
 }
 
 size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
@@ -585,11 +589,12 @@ audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
                                     uint32_t samplingRate,
                                     audio_format_t format,
                                     audio_channel_mask_t channelMask,
-                                    audio_output_flags_t flags)
+                                    audio_output_flags_t flags,
+                                    const audio_offload_info_t *offloadInfo)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return 0;
-    return aps->getOutput(stream, samplingRate, format, channelMask, flags);
+    return aps->getOutput(stream, samplingRate, format, channelMask, flags, offloadInfo);
 }
 
 status_t AudioSystem::startOutput(audio_io_handle_t output,
@@ -771,6 +776,14 @@ void AudioSystem::clearAudioConfigCache()
     gOutputs.clear();
 }
 
+bool AudioSystem::isOffloadSupported(const audio_offload_info_t& info)
+{
+    ALOGV("isOffloadSupported()");
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return false;
+    return aps->isOffloadSupported(info);
+}
+
 // ---------------------------------------------------------------------------
 
 void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who) {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index faca054..2af162c 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -97,7 +97,8 @@ AudioTrack::AudioTrack(
         void* user,
         int notificationFrames,
         int sessionId,
-        transfer_type transferType)
+        transfer_type transferType,
+        const audio_offload_info_t *offloadInfo)
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -105,7 +106,7 @@ AudioTrack::AudioTrack(
 {
     mStatus = set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
-            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType);
+            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo);
 }
 
 AudioTrack::AudioTrack(
@@ -119,7 +120,8 @@ AudioTrack::AudioTrack(
         void* user,
         int notificationFrames,
         int sessionId,
-        transfer_type transferType)
+        transfer_type transferType,
+        const audio_offload_info_t *offloadInfo)
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -127,7 +129,7 @@ AudioTrack::AudioTrack(
 {
     mStatus = set(streamType, sampleRate, format, channelMask,
             0 /*frameCount*/, flags, cbf, user, notificationFrames,
-            sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType);
+            sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo);
 }
 
 AudioTrack::~AudioTrack()
@@ -164,7 +166,8 @@ status_t AudioTrack::set(
         const sp<IMemory>& sharedBuffer,
         bool threadCanCallJava,
         int sessionId,
-        transfer_type transferType)
+        transfer_type transferType,
+        const audio_offload_info_t *offloadInfo)
 {
     switch (transferType) {
     case TRANSFER_DEFAULT:
@@ -284,7 +287,8 @@ status_t AudioTrack::set(
     audio_io_handle_t output = AudioSystem::getOutput(
                                     streamType,
                                     sampleRate, format, channelMask,
-                                    flags);
+                                    flags,
+                                    offloadInfo);
 
     if (output == 0) {
         ALOGE("Could not get audio output for stream type %d", streamType);
@@ -1543,6 +1547,21 @@ status_t AudioTrack::restoreTrack_l(const char *from)
     return result;
 }
 
+status_t AudioTrack::setParameters(const String8& keyValuePairs)
+{
+    AutoMutex lock(mLock);
+    if (mAudioTrack != 0) {
+        return mAudioTrack->setParameters(keyValuePairs);
+    } else {
+        return NO_INIT;
+    }
+}
+
+String8 AudioTrack::getParameters(const String8& keys)
+{
+    return String8::empty();
+}
+
 status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
 {
 
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 2f18680..e4df77d 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -361,7 +361,8 @@ public:
                                          audio_format_t *pFormat,
                                          audio_channel_mask_t *pChannelMask,
                                          uint32_t *pLatencyMs,
-                                         audio_output_flags_t flags)
+                                         audio_output_flags_t flags,
+                                         const audio_offload_info_t *offloadInfo)
     {
         Parcel data, reply;
         audio_devices_t devices = pDevices ? *pDevices : (audio_devices_t)0;
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 386c351..57de58f 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -56,7 +56,8 @@ enum {
     GET_DEVICES_FOR_STREAM,
     QUERY_DEFAULT_PRE_PROCESSING,
     SET_EFFECT_ENABLED,
-    IS_STREAM_ACTIVE_REMOTELY
+    IS_STREAM_ACTIVE_REMOTELY,
+    IS_OFFLOAD_SUPPORTED
 };
 
 class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
@@ -126,7 +127,8 @@ public:
                                         uint32_t samplingRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        audio_output_flags_t flags)
+                                        audio_output_flags_t flags,
+                                        const audio_offload_info_t *offloadInfo)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -374,6 +376,12 @@ public:
         *count = retCount;
         return status;
     }
+
+    virtual bool isOffloadSupported(const audio_offload_info_t& info)
+    {
+        // stub function
+        return false;
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioPolicyService, "android.media.IAudioPolicyService");
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index e92f8aa..a2b49a3 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -39,6 +39,7 @@ enum {
     ALLOCATE_TIMED_BUFFER,
     QUEUE_TIMED_BUFFER,
     SET_MEDIA_TIME_TRANSFORM,
+    SET_PARAMETERS
 };
 
 class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -154,6 +155,17 @@ public:
         }
         return status;
     }
+
+    virtual status_t setParameters(const String8& keyValuePairs) {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+        data.writeString8(keyValuePairs);
+        status_t status = remote()->transact(SET_PARAMETERS, data, &reply);
+        if (status == NO_ERROR) {
+            status = reply.readInt32();
+        }
+        return status;
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -223,6 +235,12 @@ status_t BnAudioTrack::onTransact(
             reply->writeInt32(setMediaTimeTransform(xform, target));
             return NO_ERROR;
         } break;
+        case SET_PARAMETERS: {
+            CHECK_INTERFACE(IAudioTrack, data, reply);
+            String8 keyValuePairs(data.readString8());
+            reply->writeInt32(setParameters(keyValuePairs));
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index fa1ff36..53dce65 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1385,7 +1385,8 @@ status_t MediaPlayerService::AudioOutput::open(
         uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
         audio_format_t format, int bufferCount,
         AudioCallback cb, void *cookie,
-        audio_output_flags_t flags)
+        audio_output_flags_t flags,
+        const audio_offload_info_t *offloadInfo)
 {
     mCallback = cb;
     mCallbackCookie = cookie;
@@ -1661,7 +1662,8 @@ void MediaPlayerService::AudioOutput::CallbackWrapper(
     }
 
     size_t actualSize = (*me->mCallback)(
-            me, buffer->raw, buffer->size, me->mCallbackCookie);
+            me, buffer->raw, buffer->size, me->mCallbackCookie,
+            CB_EVENT_FILL_BUFFER);
 
     if (actualSize == 0 && buffer->size > 0 && me->mNextOutput == NULL) {
         // We've reached EOS but the audio track is not stopped yet,
@@ -1767,7 +1769,8 @@ bool CallbackThread::threadLoop() {
     }
 
     size_t actualSize =
-        (*mCallback)(sink.get(), mBuffer, mBufferSize, mCookie);
+        (*mCallback)(sink.get(), mBuffer, mBufferSize, mCookie,
+                MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER);
 
     if (actualSize > 0) {
         sink->write(mBuffer, actualSize);
@@ -1781,7 +1784,8 @@ bool CallbackThread::threadLoop() {
 status_t MediaPlayerService::AudioCache::open(
         uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
         audio_format_t format, int bufferCount,
-        AudioCallback cb, void *cookie, audio_output_flags_t flags)
+        AudioCallback cb, void *cookie, audio_output_flags_t flags,
+        const audio_offload_info_t *offloadInfo)
 {
     ALOGV("open(%u, %d, 0x%x, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount);
     if (mHeap->getHeapID() < 0) {
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index e586156..1f8bcc7 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -94,7 +94,8 @@ class MediaPlayerService : public BnMediaPlayerService
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
                 audio_format_t format, int bufferCount,
                 AudioCallback cb, void *cookie,
-                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);
 
         virtual void            start();
         virtual ssize_t         write(const void* buffer, size_t size);
@@ -195,7 +196,8 @@ class MediaPlayerService : public BnMediaPlayerService
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
                 audio_format_t format, int bufferCount = 1,
                 AudioCallback cb = NULL, void *cookie = NULL,
-                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);
 
         virtual void            start();
         virtual ssize_t         write(const void* buffer, size_t size);
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 92efae8..61d6746 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -34,7 +34,7 @@ namespace android {
 
 AudioPlayer::AudioPlayer(
         const sp<MediaPlayerBase::AudioSink> &audioSink,
-        bool allowDeepBuffering,
+        uint32_t flags,
         AwesomePlayer *observer)
     : mInputBuffer(NULL),
       mSampleRate(0),
@@ -52,7 +52,7 @@ AudioPlayer::AudioPlayer(
       mFirstBufferResult(OK),
       mFirstBuffer(NULL),
       mAudioSink(audioSink),
-      mAllowDeepBuffering(allowDeepBuffering),
+      mAllowDeepBuffering((flags & ALLOW_DEEP_BUFFERING) != 0),
       mObserver(observer),
       mPinnedTimeUs(-1ll) {
 }
@@ -304,7 +304,8 @@ status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) {
 // static
 size_t AudioPlayer::AudioSinkCallback(
         MediaPlayerBase::AudioSink *audioSink,
-        void *buffer, size_t size, void *cookie) {
+        void *buffer, size_t size, void *cookie,
+        MediaPlayerBase::AudioSink::cb_event_t event) {
     AudioPlayer *me = (AudioPlayer *)cookie;
 
     return me->fillBuffer(buffer, size);
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index b0df379..e9789d3 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -471,5 +471,24 @@ AString MakeUserAgent() {
     return ua;
 }
 
+status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink,
+                           const sp<MetaData>& meta)
+{
+    // stub
+    return OK;
+}
+
+status_t mapMimeToAudioFormat(audio_format_t& format, const char* mime)
+{
+    // stub
+    return BAD_VALUE;
+}
+
+bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, bool isStreaming)
+{
+    // stub
+    return false;
+}
+
 }  // namespace android
 
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 2306f31..0d17d65 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -25,6 +25,7 @@
 #include <media/stagefright/DataSource.h>
 #include <media/stagefright/OMXClient.h>
 #include <media/stagefright/TimeSource.h>
+#include <media/stagefright/MetaData.h>
 #include <utils/threads.h>
 #include <drm/DrmManagerClient.h>
 
@@ -100,7 +101,7 @@ struct AwesomePlayer {
 
     void postAudioEOS(int64_t delayUs = 0ll);
     void postAudioSeekComplete();
-
+    void postAudioTearDown();
     status_t dump(int fd, const Vector<String16> &args) const;
 
 private:
@@ -171,6 +172,7 @@ private:
 
     ssize_t mActiveAudioTrackIndex;
     sp<MediaSource> mAudioTrack;
+    sp<MediaSource> mOmxSource;
     sp<MediaSource> mAudioSource;
     AudioPlayer *mAudioPlayer;
     int64_t mDurationUs;
@@ -211,7 +213,8 @@ private:
     bool mAudioStatusEventPending;
     sp<TimedEventQueue::Event> mVideoLagEvent;
     bool mVideoLagEventPending;
-
+    sp<TimedEventQueue::Event> mAudioTearDownEvent;
+    bool mAudioTearDownEventPending;
     sp<TimedEventQueue::Event> mAsyncPrepareEvent;
     Condition mPreparedCondition;
     bool mIsAsyncPrepare;
@@ -223,6 +226,8 @@ private:
     void postStreamDoneEvent_l(status_t status);
     void postCheckAudioStatusEvent(int64_t delayUs);
     void postVideoLagEvent_l();
+    void postAudioTearDownEvent();
+
     status_t play_l();
 
     MediaBuffer *mVideoBuffer;
@@ -257,6 +262,7 @@ private:
     void setAudioSource(sp<MediaSource> source);
     status_t initAudioDecoder();
 
+
     void setVideoSource(sp<MediaSource> source);
     status_t initVideoDecoder(uint32_t flags = 0);
 
@@ -273,6 +279,9 @@ private:
     void abortPrepare(status_t err);
     void finishAsyncPrepare_l();
     void onVideoLagUpdate();
+    void onAudioTearDownEvent();
+
+    void beginPrepareAsync_l();
 
     bool getCachedDuration_l(int64_t *durationUs, bool *eos);
 
@@ -285,6 +294,7 @@ private:
     void finishSeekIfNecessary(int64_t videoTimeUs);
     void ensureCacheIsFetching_l();
 
+    void createAudioPlayer_l();
     status_t startAudioPlayer_l(bool sendErrorNotification = true);
 
     void shutdownVideoDecoder_l();
@@ -327,6 +337,9 @@ private:
         Vector<TrackStat> mTracks;
     } mStats;
 
+    bool    mOffloadAudio;
+    bool    mAudioTearDown;
+
     status_t setVideoScalingMode(int32_t mode);
     status_t setVideoScalingMode_l(int32_t mode);
     status_t getTrackInfo(Parcel* reply) const;
diff --git a/media/libstagefright/include/ESDS.h b/media/libstagefright/include/ESDS.h
index 3a79951..2f40dae 100644
--- a/media/libstagefright/include/ESDS.h
+++ b/media/libstagefright/include/ESDS.h
@@ -33,6 +33,9 @@ public:
 
     status_t getObjectTypeIndication(uint8_t *objectTypeIndication) const;
     status_t getCodecSpecificInfo(const void **data, size_t *size) const;
+    status_t getCodecSpecificOffset(size_t *offset, size_t *size) const;
+    status_t getBitRate(uint32_t *brateMax, uint32_t *brateAvg) const;
+    status_t getStreamType(uint8_t *streamType) const;
 
 private:
     enum {
@@ -49,6 +52,9 @@ private:
     size_t mDecoderSpecificOffset;
     size_t mDecoderSpecificLength;
     uint8_t mObjectTypeIndication;
+    uint8_t mStreamType;
+    uint32_t mBitRateMax;
+    uint32_t mBitRateAvg;
 
     status_t skipDescriptorHeader(
             size_t offset, size_t size,