diff options
29 files changed, 281 insertions, 65 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 diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp index 3fa8b87..dc360a5 100755 --- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp +++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp @@ -575,10 +575,15 @@ void VideoEditorAudioPlayer::reset() { size_t VideoEditorAudioPlayer::AudioSinkCallback( MediaPlayerBase::AudioSink *audioSink, - void *buffer, size_t size, void *cookie) { + void *buffer, size_t size, void *cookie, + MediaPlayerBase::AudioSink::cb_event_t event) { VideoEditorAudioPlayer *me = (VideoEditorAudioPlayer *)cookie; - return me->fillBuffer(buffer, size); + if (event == MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER ) { + return me->fillBuffer(buffer, size); + } else { + return 0; + } } diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.h b/libvideoeditor/lvpp/VideoEditorAudioPlayer.h index a5616c1..d2e652d 100755 --- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.h +++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.h @@ -124,7 +124,8 @@ private: size_t fillBuffer(void *data, size_t size); 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); void reset(); void clear(); diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp index 4a14b40..3384e34 100755 --- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp +++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp @@ -391,7 +391,8 @@ status_t VideoEditorPlayer::VeAudioOutput::getFramesWritten(uint32_t *written) c status_t VideoEditorPlayer::VeAudioOutput::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) { mCallback = cb; mCallbackCookie = cookie; @@ -545,7 +546,8 @@ void VideoEditorPlayer::VeAudioOutput::CallbackWrapper( AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info; size_t actualSize = (*me->mCallback)( - me, buffer->raw, buffer->size, me->mCallbackCookie); + me, buffer->raw, buffer->size, me->mCallbackCookie, + MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER); buffer->size = actualSize; diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h index defc90d..69323c3 100755 --- a/libvideoeditor/lvpp/VideoEditorPlayer.h +++ b/libvideoeditor/lvpp/VideoEditorPlayer.h @@ -52,7 +52,8 @@ class VideoEditorPlayer : public MediaPlayerInterface { virtual status_t 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); virtual void start(); virtual ssize_t write(const void* buffer, size_t size); 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, diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index a6edb77..c8e8aba 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -981,11 +981,12 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t form AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; - struct audio_config config = { - sample_rate: sampleRate, - channel_mask: channelMask, - format: format, - }; + struct audio_config config; + memset(&config, 0, sizeof(config)); + config.sample_rate = sampleRate; + config.channel_mask = channelMask; + config.format = format; + audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); size_t size = dev->get_input_buffer_size(dev, &config); mHardwareStatus = AUDIO_HW_IDLE; @@ -1388,15 +1389,19 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 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) { status_t status; PlaybackThread *thread = NULL; - struct audio_config config = { - sample_rate: pSamplingRate ? *pSamplingRate : 0, - channel_mask: pChannelMask ? *pChannelMask : 0, - format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, - }; + struct audio_config config; + config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; + config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; + config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; + if (offloadInfo) { + config.offload_info = *offloadInfo; + } + audio_stream_out_t *outStream = NULL; AudioHwDevice *outHwDev; @@ -1591,11 +1596,11 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, { status_t status; RecordThread *thread = NULL; - struct audio_config config = { - sample_rate: pSamplingRate ? *pSamplingRate : 0, - channel_mask: pChannelMask ? *pChannelMask : 0, - format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, - }; + struct audio_config config; + config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; + config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; + config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; + uint32_t reqSamplingRate = config.sample_rate; audio_format_t reqFormat = config.format; audio_channel_mask_t reqChannels = config.channel_mask; diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 05dbab1..b640b31 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -160,7 +160,8 @@ 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); virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2); @@ -406,6 +407,8 @@ private: int target); virtual status_t onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); + + virtual status_t setParameters(const String8& keyValuePairs); private: const sp<PlaybackThread::Track> mTrack; }; diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index 2706880..fd4431c 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -222,15 +222,16 @@ audio_io_handle_t AudioPolicyService::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) { if (mpAudioPolicy == NULL) { return 0; } ALOGV("getOutput()"); Mutex::Autolock _l(mLock); - return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask, - flags); + return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, + format, channelMask, flags, offloadInfo); } status_t AudioPolicyService::startOutput(audio_io_handle_t output, @@ -1055,6 +1056,11 @@ int AudioPolicyService::setVoiceVolume(float volume, int delayMs) return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs); } +bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info) +{ + return false; // stub function +} + // ---------------------------------------------------------------------------- // Audio pre-processing configuration // ---------------------------------------------------------------------------- @@ -1387,7 +1393,8 @@ static audio_io_handle_t aps_open_output_on_module(void *service, 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) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); if (af == 0) { diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h index 53238fa..e723c47 100644 --- a/services/audioflinger/AudioPolicyService.h +++ b/services/audioflinger/AudioPolicyService.h @@ -67,7 +67,8 @@ public: audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = 0, audio_output_flags_t flags = - AUDIO_OUTPUT_FLAG_NONE); + AUDIO_OUTPUT_FLAG_NONE, + const audio_offload_info_t *offloadInfo = NULL); virtual status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, int session = 0); @@ -136,6 +137,7 @@ public: virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream); virtual status_t stopTone(); virtual status_t setVoiceVolume(float volume, int delayMs = 0); + virtual bool isOffloadSupported(const audio_offload_info_t &config); private: AudioPolicyService() ANDROID_API; diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index bfc197c..f0dbee3 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -250,6 +250,10 @@ void AudioFlinger::TrackHandle::pause() { mTrack->pause(); } +status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { + return INVALID_OPERATION; // stub function +} + status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) { return mTrack->attachAuxEffect(EffectId); |