diff options
111 files changed, 5082 insertions, 1728 deletions
diff --git a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp index 9b786c5..851ad2c 100644 --- a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp +++ b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp @@ -56,7 +56,7 @@ namespace android { return true; } - status_t MockDrmFactory::createDrmPlugin(const uint8_t uuid[16], DrmPlugin **plugin) + status_t MockDrmFactory::createDrmPlugin(const uint8_t /* uuid */[16], DrmPlugin **plugin) { *plugin = new MockDrmPlugin(); return OK; @@ -68,8 +68,9 @@ namespace android { return (!memcmp(uuid, mock_uuid, sizeof(mock_uuid))); } - status_t MockCryptoFactory::createPlugin(const uint8_t uuid[16], const void *data, - size_t size, CryptoPlugin **plugin) + status_t MockCryptoFactory::createPlugin(const uint8_t /* uuid */[16], + const void * /* data */, + size_t /* size */, CryptoPlugin **plugin) { *plugin = new MockCryptoPlugin(); return OK; @@ -150,7 +151,7 @@ namespace android { // Properties used in mock test, set by cts test app returned from mock plugin // byte[] mock-request -> request // string mock-default-url -> defaultUrl - // string mock-key-request-type -> keyRequestType + // string mock-keyRequestType -> keyRequestType index = mByteArrayProperties.indexOfKey(String8("mock-request")); if (index < 0) { @@ -266,8 +267,8 @@ namespace android { return OK; } - status_t MockDrmPlugin::getProvisionRequest(String8 const &certType, - String8 const &certAuthority, + status_t MockDrmPlugin::getProvisionRequest(String8 const & /* certType */, + String8 const & /* certAuthority */, Vector<uint8_t> &request, String8 &defaultUrl) { @@ -297,8 +298,8 @@ namespace android { } status_t MockDrmPlugin::provideProvisionResponse(Vector<uint8_t> const &response, - Vector<uint8_t> &certificate, - Vector<uint8_t> &wrappedKey) + Vector<uint8_t> & /* certificate */, + Vector<uint8_t> & /* wrappedKey */) { Mutex::Autolock lock(mLock); ALOGD("MockDrmPlugin::provideProvisionResponse(%s)", @@ -317,7 +318,8 @@ namespace android { return OK; } - status_t MockDrmPlugin::getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop) + status_t MockDrmPlugin::getSecureStop(Vector<uint8_t> const & /* ssid */, + Vector<uint8_t> & secureStop) { Mutex::Autolock lock(mLock); ALOGD("MockDrmPlugin::getSecureStop()"); @@ -439,6 +441,63 @@ namespace android { pData ? vectorToString(*pData) : "{}"); sendEvent(eventType, extra, pSessionId, pData); + } else if (name == "mock-send-expiration-update") { + int64_t expiryTimeMS; + sscanf(value.string(), "%jd", &expiryTimeMS); + + Vector<uint8_t> const *pSessionId = NULL; + ssize_t index = mByteArrayProperties.indexOfKey(String8("mock-event-session-id")); + if (index >= 0) { + pSessionId = &mByteArrayProperties[index]; + } + + ALOGD("sending expiration-update from mock drm plugin: %jd %s", + expiryTimeMS, pSessionId ? vectorToString(*pSessionId) : "{}"); + + sendExpirationUpdate(pSessionId, expiryTimeMS); + } else if (name == "mock-send-keys-change") { + Vector<uint8_t> const *pSessionId = NULL; + ssize_t index = mByteArrayProperties.indexOfKey(String8("mock-event-session-id")); + if (index >= 0) { + pSessionId = &mByteArrayProperties[index]; + } + + ALOGD("sending keys-change from mock drm plugin: %s", + pSessionId ? vectorToString(*pSessionId) : "{}"); + + Vector<DrmPlugin::KeyStatus> keyStatusList; + DrmPlugin::KeyStatus keyStatus; + uint8_t keyId1[] = {'k', 'e', 'y', '1'}; + keyStatus.mKeyId.clear(); + keyStatus.mKeyId.appendArray(keyId1, sizeof(keyId1)); + keyStatus.mType = DrmPlugin::kKeyStatusType_Usable; + keyStatusList.add(keyStatus); + + uint8_t keyId2[] = {'k', 'e', 'y', '2'}; + keyStatus.mKeyId.clear(); + keyStatus.mKeyId.appendArray(keyId2, sizeof(keyId2)); + keyStatus.mType = DrmPlugin::kKeyStatusType_Expired; + keyStatusList.add(keyStatus); + + uint8_t keyId3[] = {'k', 'e', 'y', '3'}; + keyStatus.mKeyId.clear(); + keyStatus.mKeyId.appendArray(keyId3, sizeof(keyId3)); + keyStatus.mType = DrmPlugin::kKeyStatusType_OutputNotAllowed; + keyStatusList.add(keyStatus); + + uint8_t keyId4[] = {'k', 'e', 'y', '4'}; + keyStatus.mKeyId.clear(); + keyStatus.mKeyId.appendArray(keyId4, sizeof(keyId4)); + keyStatus.mType = DrmPlugin::kKeyStatusType_StatusPending; + keyStatusList.add(keyStatus); + + uint8_t keyId5[] = {'k', 'e', 'y', '5'}; + keyStatus.mKeyId.clear(); + keyStatus.mKeyId.appendArray(keyId5, sizeof(keyId5)); + keyStatus.mType = DrmPlugin::kKeyStatusType_InternalError; + keyStatusList.add(keyStatus); + + sendKeysChange(pSessionId, &keyStatusList, true); } else { mStringProperties.add(name, value); } @@ -740,7 +799,7 @@ namespace android { ssize_t MockCryptoPlugin::decrypt(bool secure, const uint8_t key[16], const uint8_t iv[16], Mode mode, const void *srcPtr, const SubSample *subSamples, - size_t numSubSamples, void *dstPtr, AString *errorDetailMsg) + size_t numSubSamples, void *dstPtr, AString * /* errorDetailMsg */) { ALOGD("MockCryptoPlugin::decrypt(secure=%d, key=%s, iv=%s, mode=%d, src=%p, " "subSamples=%s, dst=%p)", @@ -769,7 +828,7 @@ namespace android { { String8 result; for (size_t i = 0; i < numSubSamples; i++) { - result.appendFormat("[%zu] {clear:%zu, encrypted:%zu} ", i, + result.appendFormat("[%zu] {clear:%u, encrypted:%u} ", i, subSamples[i].mNumBytesOfClearData, subSamples[i].mNumBytesOfEncryptedData); } diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h index b705efa..07d946d 100644 --- a/include/media/AudioResamplerPublic.h +++ b/include/media/AudioResamplerPublic.h @@ -17,6 +17,8 @@ #ifndef ANDROID_AUDIO_RESAMPLER_PUBLIC_H #define ANDROID_AUDIO_RESAMPLER_PUBLIC_H +#include <stdint.h> + // AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original // audio sample rate and the target rate when downsampling, // as permitted in the audio framework, e.g. AudioTrack and AudioFlinger. @@ -26,6 +28,22 @@ // TODO: replace with an API #define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256 +// AUDIO_RESAMPLER_UP_RATIO_MAX is the maximum suggested ratio between the original +// audio sample rate and the target rate when upsampling. It is loosely enforced by +// the system. One issue with large upsampling ratios is the approximation by +// an int32_t of the phase increments, making the resulting sample rate inexact. +#define AUDIO_RESAMPLER_UP_RATIO_MAX 65536 + +#define AUDIO_TIMESTRETCH_SPEED_MIN 0.5f +#define AUDIO_TIMESTRETCH_SPEED_MAX 2.0f +#define AUDIO_TIMESTRETCH_SPEED_NORMAL 1.0f + +#define AUDIO_TIMESTRETCH_PITCH_MIN 0.5f +#define AUDIO_TIMESTRETCH_PITCH_MAX 2.0f +#define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f + +// TODO: Consider putting these inlines into a class scope + // Returns the source frames needed to resample to destination frames. This is not a precise // value and depends on the resampler (and possibly how it handles rounding internally). // Nevertheless, this should be an upper bound on the requirements of the resampler. @@ -39,4 +57,24 @@ static inline size_t sourceFramesNeeded( size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1); } +// An upper bound for the number of destination frames possible from srcFrames +// after sample rate conversion. This may be used for buffer sizing. +static inline size_t destinationFramesPossible(size_t srcFrames, uint32_t srcSampleRate, + uint32_t dstSampleRate) { + if (srcSampleRate == dstSampleRate) { + return srcFrames; + } + uint64_t dstFrames = (uint64_t)srcFrames * dstSampleRate / srcSampleRate; + return dstFrames > 2 ? dstFrames - 2 : 0; +} + +static inline size_t sourceFramesNeededWithTimestretch( + uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate, + float speed) { + // required is the number of input frames the resampler needs + size_t required = sourceFramesNeeded(srcSampleRate, dstFramesRequired, dstSampleRate); + // to deliver this, the time stretcher requires: + return required * (double)speed + 1 + 1; // accounting for rounding dependencies +} + #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index f5db1bb..3b6db8c 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -221,14 +221,15 @@ public: audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, const audio_offload_info_t *offloadInfo = NULL); static status_t getOutputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *output, - audio_session_t session, - audio_stream_type_t *stream, - 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, - const audio_offload_info_t *offloadInfo = NULL); + audio_io_handle_t *output, + audio_session_t session, + audio_stream_type_t *stream, + 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_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, + const audio_offload_info_t *offloadInfo = NULL); static status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index d9b7057..a06197f 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -359,6 +359,21 @@ public: /* Return current source sample rate in Hz */ uint32_t getSampleRate() const; + /* Set source playback rate for timestretch + * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster + * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch + * + * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX + * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX + * + * Speed increases the playback rate of media, but does not alter pitch. + * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. + */ + status_t setPlaybackRate(float speed, float pitch); + + /* Return current playback rate */ + void getPlaybackRate(float *speed, float *pitch) const; + /* Enables looping and sets the start and end points of looping. * Only supported for static buffer mode. * @@ -477,6 +492,26 @@ private: audio_io_handle_t getOutput() const; public: + /* Selects the audio device to use for output of this AudioTrack. A value of + * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. + * + * Parameters: + * The device ID of the selected device (as returned by the AudioDevicesManager API). + * + * Returned value: + * - NO_ERROR: successful operation + * TODO: what else can happen here? + */ + status_t setOutputDevice(audio_port_handle_t deviceId); + + /* Returns the ID of the audio device used for output of this AudioTrack. + * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. + * + * Parameters: + * none. + */ + audio_port_handle_t getOutputDevice(); + /* Returns the unique session ID associated with this track. * * Parameters: @@ -699,6 +734,9 @@ protected: // increment mPosition by the delta of mServer, and return new value of mPosition uint32_t updateAndGetPosition_l(); + // check sample rate and speed is compatible with AudioTrack + bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; + // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 sp<IAudioTrack> mAudioTrack; sp<IMemory> mCblkMemory; @@ -710,6 +748,8 @@ protected: float mVolume[2]; float mSendLevel; mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it + float mSpeed; // timestretch: 1.0f for normal speed. + float mPitch; // timestretch: 1.0f for normal pitch. size_t mFrameCount; // corresponds to current IAudioTrack, value is // reported back by AudioFlinger to the client size_t mReqFrameCount; // frame count to request the first or next time @@ -817,6 +857,10 @@ protected: bool mInUnderrun; // whether track is currently in underrun state uint32_t mPausedPosition; + // For Device Selection API + // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. + int mSelectedDeviceId; + private: class DeathNotifier : public IBinder::DeathRecipient { public: diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h index fecc6f1..7506153 100644 --- a/include/media/IAudioPolicyService.h +++ b/include/media/IAudioPolicyService.h @@ -66,6 +66,7 @@ public: audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = 0, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, + audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, const audio_offload_info_t *offloadInfo = NULL) = 0; virtual status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, diff --git a/include/media/ICrypto.h b/include/media/ICrypto.h index 07742ca..aa04dbe 100644 --- a/include/media/ICrypto.h +++ b/include/media/ICrypto.h @@ -25,6 +25,7 @@ namespace android { struct AString; +struct IMemory; struct ICrypto : public IInterface { DECLARE_META_INTERFACE(Crypto); @@ -43,12 +44,14 @@ struct ICrypto : public IInterface { virtual void notifyResolution(uint32_t width, uint32_t height) = 0; + virtual status_t setMediaDrmSession(const Vector<uint8_t> &sessionId) = 0; + virtual ssize_t decrypt( bool secure, const uint8_t key[16], const uint8_t iv[16], CryptoPlugin::Mode mode, - const void *srcPtr, + const sp<IMemory> &sharedBuffer, size_t offset, const CryptoPlugin::SubSample *subSamples, size_t numSubSamples, void *dstPtr, AString *errorDetailMsg) = 0; @@ -61,6 +64,9 @@ struct BnCrypto : public BnInterface<ICrypto> { virtual status_t onTransact( uint32_t code, const Parcel &data, Parcel *reply, uint32_t flags = 0); +private: + void readVector(const Parcel &data, Vector<uint8_t> &vector) const; + void writeVector(Parcel *reply, Vector<uint8_t> const &vector) const; }; } // namespace android diff --git a/include/media/IMediaCodecList.h b/include/media/IMediaCodecList.h index e93ea8b..12b52d7 100644 --- a/include/media/IMediaCodecList.h +++ b/include/media/IMediaCodecList.h @@ -21,6 +21,8 @@ #include <binder/IInterface.h> #include <binder/Parcel.h> +#include <media/stagefright/foundation/AMessage.h> + namespace android { struct MediaCodecInfo; @@ -33,6 +35,8 @@ public: virtual size_t countCodecs() const = 0; virtual sp<MediaCodecInfo> getCodecInfo(size_t index) const = 0; + virtual const sp<AMessage> getGlobalSettings() const = 0; + virtual ssize_t findCodecByType( const char *type, bool encoder, size_t startIndex = 0) const = 0; diff --git a/include/media/MediaCodecInfo.h b/include/media/MediaCodecInfo.h index cd56adb..895a13a 100644 --- a/include/media/MediaCodecInfo.h +++ b/include/media/MediaCodecInfo.h @@ -35,6 +35,8 @@ struct AMessage; struct Parcel; struct CodecCapabilities; +typedef KeyedVector<AString, AString> CodecSettings; + struct MediaCodecInfo : public RefBase { struct ProfileLevel { uint32_t mProfile; @@ -104,6 +106,7 @@ private: MediaCodecInfo(AString name, bool encoder, const char *mime); void addQuirk(const char *name); status_t addMime(const char *mime); + status_t updateMime(const char *mime); status_t initializeCapabilities(const CodecCapabilities &caps); void addDetail(const AString &key, const AString &value); void addFeature(const AString &key, int32_t value); @@ -114,6 +117,7 @@ private: DISALLOW_EVIL_CONSTRUCTORS(MediaCodecInfo); friend class MediaCodecList; + friend class MediaCodecListOverridesTest; }; } // namespace android diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h index c1483f3..a8d0fcb 100644 --- a/include/media/stagefright/ACodec.h +++ b/include/media/stagefright/ACodec.h @@ -300,6 +300,7 @@ private: OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels); status_t setPriority(int32_t priority); + status_t setOperatingRate(float rateFloat, bool isVideo); status_t setMinBufferSize(OMX_U32 portIndex, size_t size); diff --git a/include/media/stagefright/MediaClock.h b/include/media/stagefright/MediaClock.h index e9c09a1..dd1a809 100644 --- a/include/media/stagefright/MediaClock.h +++ b/include/media/stagefright/MediaClock.h @@ -42,6 +42,7 @@ struct MediaClock : public RefBase { void updateMaxTimeMedia(int64_t maxTimeMediaUs); void setPlaybackRate(float rate); + float getPlaybackRate() const; // query media time corresponding to real time |realUs|, and save the // result in |outMediaUs|. diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h index 8241e19..3e3c276 100644 --- a/include/media/stagefright/MediaCodec.h +++ b/include/media/stagefright/MediaCodec.h @@ -30,8 +30,10 @@ struct AMessage; struct AReplyToken; struct AString; struct CodecBase; -struct ICrypto; struct IBatteryStats; +struct ICrypto; +struct IMemory; +struct MemoryDealer; struct SoftwareRenderer; struct Surface; @@ -51,6 +53,13 @@ struct MediaCodec : public AHandler { CB_OUTPUT_AVAILABLE = 2, CB_ERROR = 3, CB_OUTPUT_FORMAT_CHANGED = 4, + CB_CODEC_RELEASED = 5, + }; + + // used by CB_CODEC_RELEASED to tell the upper layer the cause of the release. + enum ReleaseReason { + REASON_UNKNOWN = 0, + REASON_RECLAIMED, // resources reclaimed by resource manager }; struct BatteryNotifier; @@ -126,6 +135,8 @@ struct MediaCodec : public AHandler { status_t getOutputFormat(sp<AMessage> *format) const; status_t getInputFormat(sp<AMessage> *format) const; + status_t getWidevineLegacyBuffers(Vector<sp<ABuffer> > *buffers) const; + status_t getInputBuffers(Vector<sp<ABuffer> > *buffers) const; status_t getOutputBuffers(Vector<sp<ABuffer> > *buffers) const; @@ -213,6 +224,7 @@ private: uint32_t mBufferID; sp<ABuffer> mData; sp<ABuffer> mEncryptedData; + sp<IMemory> mSharedEncryptedBuffer; sp<AMessage> mNotify; sp<AMessage> mFormat; bool mOwnedByClient; @@ -231,6 +243,7 @@ private: sp<AMessage> mOutputFormat; sp<AMessage> mInputFormat; sp<AMessage> mCallback; + sp<MemoryDealer> mDealer; bool mBatteryStatNotified; bool mIsVideo; diff --git a/include/media/stagefright/MediaCodecList.h b/include/media/stagefright/MediaCodecList.h index c2bbe4d..9d1d675 100644 --- a/include/media/stagefright/MediaCodecList.h +++ b/include/media/stagefright/MediaCodecList.h @@ -48,9 +48,14 @@ struct MediaCodecList : public BnMediaCodecList { return mCodecInfos.itemAt(index); } + virtual const sp<AMessage> getGlobalSettings() const; + // to be used by MediaPlayerService alone static sp<IMediaCodecList> getLocalInstance(); + // only to be used in getLocalInstance + void updateDetailsForMultipleCodecs(const KeyedVector<AString, CodecSettings>& updates); + private: class BinderDeathObserver : public IBinder::DeathRecipient { void binderDied(const wp<IBinder> &the_late_who __unused); @@ -60,6 +65,7 @@ private: enum Section { SECTION_TOPLEVEL, + SECTION_SETTINGS, SECTION_DECODERS, SECTION_DECODER, SECTION_DECODER_TYPE, @@ -74,10 +80,14 @@ private: status_t mInitCheck; Section mCurrentSection; + bool mUpdate; Vector<Section> mPastSections; int32_t mDepth; AString mHrefBase; + sp<AMessage> mGlobalSettings; + KeyedVector<AString, CodecSettings> mOverrides; + Vector<sp<MediaCodecInfo> > mCodecInfos; sp<MediaCodecInfo> mCurrentInfo; sp<IOMX> mOMX; @@ -87,7 +97,7 @@ private: status_t initCheck() const; void parseXMLFile(const char *path); - void parseTopLevelXMLFile(const char *path); + void parseTopLevelXMLFile(const char *path, bool ignore_errors = false); static void StartElementHandlerWrapper( void *me, const char *name, const char **attrs); @@ -98,9 +108,12 @@ private: void endElementHandler(const char *name); status_t includeXMLFile(const char **attrs); + status_t addSettingFromAttributes(const char **attrs); status_t addMediaCodecFromAttributes(bool encoder, const char **attrs); void addMediaCodec(bool encoder, const char *name, const char *type = NULL); + void setCurrentCodecInfo(bool encoder, const char *name, const char *type); + status_t addQuirk(const char **attrs); status_t addTypeFromAttributes(const char **attrs); status_t addLimit(const char **attrs); diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h index 5644428..6cc2e2b 100644 --- a/include/private/media/AudioTrackShared.h +++ b/include/private/media/AudioTrackShared.h @@ -25,6 +25,7 @@ #include <utils/Log.h> #include <utils/RefBase.h> #include <audio_utils/roundup.h> +#include <media/AudioResamplerPublic.h> #include <media/SingleStateQueue.h> namespace android { @@ -113,6 +114,14 @@ struct AudioTrackSharedStatic { mPosLoopQueue; }; + +struct AudioTrackPlaybackRate { + float mSpeed; + float mPitch; +}; + +typedef SingleStateQueue<AudioTrackPlaybackRate> AudioTrackPlaybackRateQueue; + // ---------------------------------------------------------------------------- // Important: do not add any virtual methods, including ~ @@ -159,6 +168,8 @@ private: uint32_t mSampleRate; // AudioTrack only: client's requested sample rate in Hz // or 0 == default. Write-only client, read-only server. + AudioTrackPlaybackRateQueue::Shared mPlaybackRateQueue; + // client write-only, server read-only uint16_t mSendLevel; // Fixed point U4.12 so 0x1000 means 1.0 @@ -313,7 +324,8 @@ public: AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool clientInServer = false) : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, - clientInServer) { } + clientInServer), + mPlaybackRateMutator(&cblk->mPlaybackRateQueue) { } virtual ~AudioTrackClientProxy() { } // No barriers on the following operations, so the ordering of loads/stores @@ -333,6 +345,13 @@ public: mCblk->mSampleRate = sampleRate; } + void setPlaybackRate(float speed, float pitch) { + AudioTrackPlaybackRate playbackRate; + playbackRate.mSpeed = speed; + playbackRate.mPitch = pitch; + mPlaybackRateMutator.push(playbackRate); + } + virtual void flush(); virtual uint32_t getUnderrunFrames() const { @@ -344,6 +363,9 @@ public: bool getStreamEndDone() const; status_t waitStreamEndDone(const struct timespec *requested); + +private: + AudioTrackPlaybackRateQueue::Mutator mPlaybackRateMutator; }; class StaticAudioTrackClientProxy : public AudioTrackClientProxy { @@ -458,8 +480,11 @@ class AudioTrackServerProxy : public ServerProxy { public: AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool clientInServer = false, uint32_t sampleRate = 0) - : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer) { + : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer), + mPlaybackRateObserver(&cblk->mPlaybackRateQueue) { mCblk->mSampleRate = sampleRate; + mPlaybackRate.mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL; + mPlaybackRate.mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL; } protected: virtual ~AudioTrackServerProxy() { } @@ -493,6 +518,13 @@ public: // Return the total number of frames that AudioFlinger has obtained and released virtual size_t framesReleased() const { return mCblk->mServer; } + + // Return the playback speed and pitch read atomically. Not multi-thread safe on server side. + void getPlaybackRate(float *speed, float *pitch); + +private: + AudioTrackPlaybackRate mPlaybackRate; // last observed playback rate + AudioTrackPlaybackRateQueue::Observer mPlaybackRateObserver; }; class StaticAudioTrackServerProxy : public AudioTrackServerProxy { diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 7fc1a78..5bbe786 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -189,13 +189,9 @@ status_t AudioRecord::set( } // validate parameters - if (!audio_is_valid_format(format)) { - ALOGE("Invalid format %#x", format); - return BAD_VALUE; - } - // Temporary restriction: AudioFlinger currently supports 16-bit PCM only - if (format != AUDIO_FORMAT_PCM_16_BIT) { - ALOGE("Format %#x is not supported", format); + // AudioFlinger capture only supports linear PCM + if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { + ALOGE("Format %#x is not linear pcm", format); return BAD_VALUE; } mFormat = format; diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 9150a94..8db72ee 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -658,13 +658,14 @@ status_t AudioSystem::getOutputForAttr(const audio_attributes_t *attr, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + audio_port_handle_t selectedDeviceId, const audio_offload_info_t *offloadInfo) { const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return NO_INIT; return aps->getOutputForAttr(attr, output, session, stream, samplingRate, format, channelMask, - flags, offloadInfo); + flags, selectedDeviceId, offloadInfo); } status_t AudioSystem::startOutput(audio_io_handle_t output, diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index cfdb19c..d32db7c 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -56,6 +56,24 @@ static int64_t getNowUs() return convertTimespecToUs(tv); } +// Must match similar computation in createTrack_l in Threads.cpp. +// TODO: Move to a common library +static size_t calculateMinFrameCount( + uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate, + uint32_t sampleRate, float speed) +{ + // Ensure that buffer depth covers at least audio hardware latency + uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate); + if (minBufCount < 2) { + minBufCount = 2; + } + ALOGV("calculateMinFrameCount afLatency %u afFrameCount %u afSampleRate %u " + "sampleRate %u speed %f minBufCount: %u", + afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount); + return minBufCount * sourceFramesNeededWithTimestretch( + sampleRate, afFrameCount, afSampleRate, speed); +} + // static status_t AudioTrack::getMinFrameCount( size_t* frameCount, @@ -94,13 +112,10 @@ status_t AudioTrack::getMinFrameCount( return status; } - // Ensure that buffer depth covers at least audio hardware latency - uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); - if (minBufCount < 2) { - minBufCount = 2; - } + // When called from createTrack, speed is 1.0f (normal speed). + // This is rechecked again on setting playback rate (TODO: on setting sample rate, too). + *frameCount = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f); - *frameCount = minBufCount * sourceFramesNeeded(sampleRate, afFrameCount, afSampleRate); // The formula above should always produce a non-zero value under normal circumstances: // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX. // Return error in the unlikely event that it does not, as that's part of the API contract. @@ -109,8 +124,8 @@ status_t AudioTrack::getMinFrameCount( streamType, sampleRate); return BAD_VALUE; } - ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%u, afSampleRate=%u, afLatency=%u", - *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); + ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u", + *frameCount, afFrameCount, afSampleRate, afLatency); return NO_ERROR; } @@ -121,7 +136,8 @@ AudioTrack::AudioTrack() mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT), - mPausedPosition(0) + mPausedPosition(0), + mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) { mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; mAttributes.usage = AUDIO_USAGE_UNKNOWN; @@ -149,7 +165,8 @@ AudioTrack::AudioTrack( mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT), - mPausedPosition(0) + mPausedPosition(0), + mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) { mStatus = set(streamType, sampleRate, format, channelMask, frameCount, flags, cbf, user, notificationFrames, @@ -177,7 +194,8 @@ AudioTrack::AudioTrack( mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT), - mPausedPosition(0) + mPausedPosition(0), + mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) { mStatus = set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags, cbf, user, notificationFrames, @@ -357,6 +375,8 @@ status_t AudioTrack::set( return BAD_VALUE; } mSampleRate = sampleRate; + mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL; + mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL; // Make copy of input parameter offloadInfo so that in the future: // (a) createTrack_l doesn't need it as an input parameter @@ -686,6 +706,7 @@ status_t AudioTrack::setSampleRate(uint32_t rate) if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { return BAD_VALUE; } + // TODO: Should we also check if the buffer size is compatible? mSampleRate = rate; mProxy->setSampleRate(rate); @@ -716,6 +737,42 @@ uint32_t AudioTrack::getSampleRate() const return mSampleRate; } +status_t AudioTrack::setPlaybackRate(float speed, float pitch) +{ + if (speed < AUDIO_TIMESTRETCH_SPEED_MIN + || speed > AUDIO_TIMESTRETCH_SPEED_MAX + || pitch < AUDIO_TIMESTRETCH_PITCH_MIN + || pitch > AUDIO_TIMESTRETCH_PITCH_MAX) { + return BAD_VALUE; + } + AutoMutex lock(mLock); + if (speed == mSpeed && pitch == mPitch) { + return NO_ERROR; + } + if (mIsTimed || isOffloadedOrDirect_l()) { + return INVALID_OPERATION; + } + if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { + return INVALID_OPERATION; + } + // Check if the buffer size is compatible. + if (!isSampleRateSpeedAllowed_l(mSampleRate, speed)) { + ALOGV("setPlaybackRate(%f, %f) failed", speed, pitch); + return BAD_VALUE; + } + mSpeed = speed; + mPitch = pitch; + mProxy->setPlaybackRate(speed, pitch); + return NO_ERROR; +} + +void AudioTrack::getPlaybackRate(float *speed, float *pitch) const +{ + AutoMutex lock(mLock); + *speed = mSpeed; + *pitch = mPitch; +} + status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) { if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { @@ -928,6 +985,21 @@ audio_io_handle_t AudioTrack::getOutput() const return mOutput; } +status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) { + AutoMutex lock(mLock); + if (mSelectedDeviceId != deviceId) { + mSelectedDeviceId = deviceId; + return restoreTrack_l("setOutputDevice() restart"); + } else { + return NO_ERROR; + } +} + +audio_port_handle_t AudioTrack::getOutputDevice() { + AutoMutex lock(mLock); + return mSelectedDeviceId; +} + status_t AudioTrack::attachAuxEffect(int effectId) { AutoMutex lock(mLock); @@ -960,11 +1032,12 @@ status_t AudioTrack::createTrack_l() audio_io_handle_t output; audio_stream_type_t streamType = mStreamType; audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; - status_t status = AudioSystem::getOutputForAttr(attr, &output, - (audio_session_t)mSessionId, &streamType, - mSampleRate, mFormat, mChannelMask, - mFlags, mOffloadInfo); + status_t status; + status = AudioSystem::getOutputForAttr(attr, &output, + (audio_session_t)mSessionId, &streamType, + mSampleRate, mFormat, mChannelMask, + mFlags, mSelectedDeviceId, mOffloadInfo); if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x," @@ -1067,8 +1140,16 @@ status_t AudioTrack::createTrack_l() // there _is_ a frameCount parameter. We silently ignore it. frameCount = mSharedBuffer->size() / mFrameSize; } else { - // For fast and normal streaming tracks, - // the frame count calculations and checks are done by server + // For fast tracks the frame count calculations and checks are done by server + + if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) { + // for normal tracks precompute the frame count based on speed. + const size_t minFrameCount = calculateMinFrameCount( + afLatency, afFrameCount, afSampleRate, mSampleRate, mSpeed); + if (frameCount < minFrameCount) { + frameCount = minFrameCount; + } + } } IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; @@ -1211,6 +1292,7 @@ status_t AudioTrack::createTrack_l() } mAudioTrack->attachAuxEffect(mAuxEffectId); + // FIXME doesn't take into account speed or future sample rate changes (until restoreTrack) // FIXME don't believe this lie mLatency = afLatency + (1000*frameCount) / mSampleRate; @@ -1236,6 +1318,7 @@ status_t AudioTrack::createTrack_l() mProxy->setSendLevel(mSendLevel); mProxy->setSampleRate(mSampleRate); + mProxy->setPlaybackRate(mSpeed, mPitch); mProxy->setMinimum(mNotificationFramesAct); mDeathNotifier = new DeathNotifier(this); @@ -1604,6 +1687,7 @@ nsecs_t AudioTrack::processAudioBuffer() // Cache other fields that will be needed soon uint32_t sampleRate = mSampleRate; + float speed = mSpeed; uint32_t notificationFrames = mNotificationFramesAct; if (mRefreshRemaining) { mRefreshRemaining = false; @@ -1732,7 +1816,7 @@ nsecs_t AudioTrack::processAudioBuffer() if (minFrames != (uint32_t) ~0) { // This "fudge factor" avoids soaking CPU, and compensates for late progress by server static const nsecs_t kFudgeNs = 10000000LL; // 10 ms - ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; + ns = ((double)minFrames * 1000000000) / ((double)sampleRate * speed) + kFudgeNs; } // If not supplying data by EVENT_MORE_DATA, then we're done @@ -1773,7 +1857,8 @@ nsecs_t AudioTrack::processAudioBuffer() if (mRetryOnPartialBuffer && !isOffloaded()) { mRetryOnPartialBuffer = false; if (avail < mRemainingFrames) { - int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; + int64_t myns = ((double)(mRemainingFrames - avail) * 1100000000) + / ((double)sampleRate * speed); if (ns < 0 || myns < ns) { ns = myns; } @@ -1828,7 +1913,7 @@ nsecs_t AudioTrack::processAudioBuffer() // that total to a sum == notificationFrames. if (0 < misalignment && misalignment <= mRemainingFrames) { mRemainingFrames = misalignment; - return (mRemainingFrames * 1100000000LL) / sampleRate; + return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed); } #endif @@ -1923,6 +2008,41 @@ uint32_t AudioTrack::updateAndGetPosition_l() return mPosition += (uint32_t) delta; } +bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const +{ + // applicable for mixing tracks only (not offloaded or direct) + if (mStaticProxy != 0) { + return true; // static tracks do not have issues with buffer sizing. + } + status_t status; + uint32_t afLatency; + status = AudioSystem::getLatency(mOutput, &afLatency); + if (status != NO_ERROR) { + ALOGE("getLatency(%d) failed status %d", mOutput, status); + return false; + } + + size_t afFrameCount; + status = AudioSystem::getFrameCount(mOutput, &afFrameCount); + if (status != NO_ERROR) { + ALOGE("getFrameCount(output=%d) status %d", mOutput, status); + return false; + } + + uint32_t afSampleRate; + status = AudioSystem::getSamplingRate(mOutput, &afSampleRate); + if (status != NO_ERROR) { + ALOGE("getSamplingRate(output=%d) status %d", mOutput, status); + return false; + } + + const size_t minFrameCount = + calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, speed); + ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu minFrameCount %zu", + mFrameCount, minFrameCount); + return mFrameCount >= minFrameCount; +} + status_t AudioTrack::setParameters(const String8& keyValuePairs) { AutoMutex lock(mLock); @@ -1988,7 +2108,8 @@ status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) return WOULD_BLOCK; // stale timestamp time, occurs before start. } const int64_t deltaTimeUs = timestampTimeUs - mStartUs; - const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; + const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000 + / ((double)mSampleRate * mSpeed); if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { // Verify that the counter can't count faster than the sample rate @@ -2075,7 +2196,8 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, mChannelCount, mFrameCount); result.append(buffer); - snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); + snprintf(buffer, 255, " sample rate(%u), speed(%f), status(%d)\n", + mSampleRate, mSpeed, mStatus); result.append(buffer); snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); result.append(buffer); diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp index 6d5f1af..ba67b40 100644 --- a/media/libmedia/AudioTrackShared.cpp +++ b/media/libmedia/AudioTrackShared.cpp @@ -793,6 +793,16 @@ void AudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount) (void) android_atomic_or(CBLK_UNDERRUN, &cblk->mFlags); } +void AudioTrackServerProxy::getPlaybackRate(float *speed, float *pitch) +{ // do not call from multiple threads without holding lock + AudioTrackPlaybackRate playbackRate; + if (mPlaybackRateObserver.poll(playbackRate)) { + mPlaybackRate = playbackRate; + } + *speed = mPlaybackRate.mSpeed; + *pitch = mPlaybackRate.mPitch; +} + // --------------------------------------------------------------------------- StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp index 39374d8..4b86532 100644 --- a/media/libmedia/IAudioPolicyService.cpp +++ b/media/libmedia/IAudioPolicyService.cpp @@ -173,6 +173,7 @@ public: audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + audio_port_handle_t selectedDeviceId, const audio_offload_info_t *offloadInfo) { Parcel data, reply; @@ -208,6 +209,7 @@ public: data.writeInt32(static_cast <uint32_t>(format)); data.writeInt32(channelMask); data.writeInt32(static_cast <uint32_t>(flags)); + data.writeInt32(selectedDeviceId); // hasOffloadInfo if (offloadInfo == NULL) { data.writeInt32(0); @@ -815,6 +817,7 @@ status_t BnAudioPolicyService::onTransact( audio_channel_mask_t channelMask = data.readInt32(); audio_output_flags_t flags = static_cast <audio_output_flags_t>(data.readInt32()); + audio_port_handle_t selectedDeviceId = data.readInt32(); bool hasOffloadInfo = data.readInt32() != 0; audio_offload_info_t offloadInfo; if (hasOffloadInfo) { @@ -824,7 +827,7 @@ status_t BnAudioPolicyService::onTransact( status_t status = getOutputForAttr(hasAttributes ? &attr : NULL, &output, session, &stream, samplingRate, format, channelMask, - flags, hasOffloadInfo ? &offloadInfo : NULL); + flags, selectedDeviceId, hasOffloadInfo ? &offloadInfo : NULL); reply->writeInt32(status); reply->writeInt32(output); reply->writeInt32(stream); diff --git a/media/libmedia/ICrypto.cpp b/media/libmedia/ICrypto.cpp index c26c5bf..9246a7c 100644 --- a/media/libmedia/ICrypto.cpp +++ b/media/libmedia/ICrypto.cpp @@ -19,6 +19,7 @@ #include <utils/Log.h> #include <binder/Parcel.h> +#include <binder/IMemory.h> #include <media/ICrypto.h> #include <media/stagefright/MediaErrors.h> #include <media/stagefright/foundation/ADebug.h> @@ -34,6 +35,7 @@ enum { REQUIRES_SECURE_COMPONENT, DECRYPT, NOTIFY_RESOLUTION, + SET_MEDIADRM_SESSION, }; struct BpCrypto : public BpInterface<ICrypto> { @@ -97,7 +99,7 @@ struct BpCrypto : public BpInterface<ICrypto> { const uint8_t key[16], const uint8_t iv[16], CryptoPlugin::Mode mode, - const void *srcPtr, + const sp<IMemory> &sharedBuffer, size_t offset, const CryptoPlugin::SubSample *subSamples, size_t numSubSamples, void *dstPtr, AString *errorDetailMsg) { @@ -126,7 +128,8 @@ struct BpCrypto : public BpInterface<ICrypto> { } data.writeInt32(totalSize); - data.write(srcPtr, totalSize); + data.writeStrongBinder(IInterface::asBinder(sharedBuffer)); + data.writeInt32(offset); data.writeInt32(numSubSamples); data.write(subSamples, sizeof(CryptoPlugin::SubSample) * numSubSamples); @@ -159,7 +162,28 @@ struct BpCrypto : public BpInterface<ICrypto> { remote()->transact(NOTIFY_RESOLUTION, data, &reply); } + virtual status_t setMediaDrmSession(const Vector<uint8_t> &sessionId) { + Parcel data, reply; + data.writeInterfaceToken(ICrypto::getInterfaceDescriptor()); + + writeVector(data, sessionId); + remote()->transact(SET_MEDIADRM_SESSION, data, &reply); + + return reply.readInt32(); + } + private: + void readVector(Parcel &reply, Vector<uint8_t> &vector) const { + uint32_t size = reply.readInt32(); + vector.insertAt((size_t)0, size); + reply.read(vector.editArray(), size); + } + + void writeVector(Parcel &data, Vector<uint8_t> const &vector) const { + data.writeInt32(vector.size()); + data.write(vector.array(), vector.size()); + } + DISALLOW_EVIL_CONSTRUCTORS(BpCrypto); }; @@ -167,6 +191,17 @@ IMPLEMENT_META_INTERFACE(Crypto, "android.hardware.ICrypto"); //////////////////////////////////////////////////////////////////////////////// +void BnCrypto::readVector(const Parcel &data, Vector<uint8_t> &vector) const { + uint32_t size = data.readInt32(); + vector.insertAt((size_t)0, size); + data.read(vector.editArray(), size); +} + +void BnCrypto::writeVector(Parcel *reply, Vector<uint8_t> const &vector) const { + reply->writeInt32(vector.size()); + reply->write(vector.array(), vector.size()); +} + status_t BnCrypto::onTransact( uint32_t code, const Parcel &data, Parcel *reply, uint32_t flags) { switch (code) { @@ -245,8 +280,9 @@ status_t BnCrypto::onTransact( data.read(iv, sizeof(iv)); size_t totalSize = data.readInt32(); - void *srcData = malloc(totalSize); - data.read(srcData, totalSize); + sp<IMemory> sharedBuffer = + interface_cast<IMemory>(data.readStrongBinder()); + int32_t offset = data.readInt32(); int32_t numSubSamples = data.readInt32(); @@ -265,15 +301,21 @@ status_t BnCrypto::onTransact( } AString errorDetailMsg; - ssize_t result = decrypt( + ssize_t result; + + if (offset + totalSize > sharedBuffer->size()) { + result = -EINVAL; + } else { + result = decrypt( secure, key, iv, mode, - srcData, + sharedBuffer, offset, subSamples, numSubSamples, dstPtr, &errorDetailMsg); + } reply->writeInt32(result); @@ -294,9 +336,6 @@ status_t BnCrypto::onTransact( delete[] subSamples; subSamples = NULL; - free(srcData); - srcData = NULL; - return OK; } @@ -311,6 +350,15 @@ status_t BnCrypto::onTransact( return OK; } + case SET_MEDIADRM_SESSION: + { + CHECK_INTERFACE(IDrm, data, reply); + Vector<uint8_t> sessionId; + readVector(data, sessionId); + reply->writeInt32(setMediaDrmSession(sessionId)); + return OK; + } + default: return BBinder::onTransact(code, data, reply, flags); } diff --git a/media/libmedia/IMediaCodecList.cpp b/media/libmedia/IMediaCodecList.cpp index 80020db..e2df104 100644 --- a/media/libmedia/IMediaCodecList.cpp +++ b/media/libmedia/IMediaCodecList.cpp @@ -30,6 +30,7 @@ enum { CREATE = IBinder::FIRST_CALL_TRANSACTION, COUNT_CODECS, GET_CODEC_INFO, + GET_GLOBAL_SETTINGS, FIND_CODEC_BY_TYPE, FIND_CODEC_BY_NAME, }; @@ -64,6 +65,19 @@ public: } } + virtual const sp<AMessage> getGlobalSettings() const + { + Parcel data, reply; + data.writeInterfaceToken(IMediaCodecList::getInterfaceDescriptor()); + remote()->transact(GET_GLOBAL_SETTINGS, data, &reply); + status_t err = reply.readInt32(); + if (err == OK) { + return AMessage::FromParcel(reply); + } else { + return NULL; + } + } + virtual ssize_t findCodecByType( const char *type, bool encoder, size_t startIndex = 0) const { @@ -125,6 +139,20 @@ status_t BnMediaCodecList::onTransact( } break; + case GET_GLOBAL_SETTINGS: + { + CHECK_INTERFACE(IMediaCodecList, data, reply); + const sp<AMessage> info = getGlobalSettings(); + if (info != NULL) { + reply->writeInt32(OK); + info->writeToParcel(reply); + } else { + reply->writeInt32(-ERANGE); + } + return NO_ERROR; + } + break; + case FIND_CODEC_BY_TYPE: { CHECK_INTERFACE(IMediaCodecList, data, reply); diff --git a/media/libmedia/MediaCodecInfo.cpp b/media/libmedia/MediaCodecInfo.cpp index 7b4c4e2..8d3fa7b 100644 --- a/media/libmedia/MediaCodecInfo.cpp +++ b/media/libmedia/MediaCodecInfo.cpp @@ -206,6 +206,17 @@ status_t MediaCodecInfo::addMime(const char *mime) { return OK; } +status_t MediaCodecInfo::updateMime(const char *mime) { + ssize_t ix = getCapabilityIndex(mime); + if (ix < 0) { + ALOGE("updateMime mime not found %s", mime); + return -EINVAL; + } + + mCurrentCaps = mCaps.valueAt(ix); + return OK; +} + void MediaCodecInfo::removeMime(const char *mime) { ssize_t ix = getCapabilityIndex(mime); if (ix >= 0) { diff --git a/media/libmediaplayerservice/Crypto.cpp b/media/libmediaplayerservice/Crypto.cpp index 8ee7c0b..f639193 100644 --- a/media/libmediaplayerservice/Crypto.cpp +++ b/media/libmediaplayerservice/Crypto.cpp @@ -22,6 +22,7 @@ #include "Crypto.h" +#include <binder/IMemory.h> #include <media/hardware/CryptoAPI.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/foundation/AString.h> @@ -238,7 +239,7 @@ ssize_t Crypto::decrypt( const uint8_t key[16], const uint8_t iv[16], CryptoPlugin::Mode mode, - const void *srcPtr, + const sp<IMemory> &sharedBuffer, size_t offset, const CryptoPlugin::SubSample *subSamples, size_t numSubSamples, void *dstPtr, AString *errorDetailMsg) { @@ -252,6 +253,8 @@ ssize_t Crypto::decrypt( return -EINVAL; } + const void *srcPtr = static_cast<uint8_t *>(sharedBuffer->pointer()) + offset; + return mPlugin->decrypt( secure, key, iv, mode, srcPtr, subSamples, numSubSamples, dstPtr, errorDetailMsg); @@ -265,4 +268,14 @@ void Crypto::notifyResolution(uint32_t width, uint32_t height) { } } +status_t Crypto::setMediaDrmSession(const Vector<uint8_t> &sessionId) { + Mutex::Autolock autoLock(mLock); + + status_t result = NO_INIT; + if (mInitCheck == OK && mPlugin != NULL) { + result = mPlugin->setMediaDrmSession(sessionId); + } + return result; +} + } // namespace android diff --git a/media/libmediaplayerservice/Crypto.h b/media/libmediaplayerservice/Crypto.h index 0037c2e..99ea95d 100644 --- a/media/libmediaplayerservice/Crypto.h +++ b/media/libmediaplayerservice/Crypto.h @@ -47,12 +47,14 @@ struct Crypto : public BnCrypto { virtual void notifyResolution(uint32_t width, uint32_t height); + virtual status_t setMediaDrmSession(const Vector<uint8_t> &sessionId); + virtual ssize_t decrypt( bool secure, const uint8_t key[16], const uint8_t iv[16], CryptoPlugin::Mode mode, - const void *srcPtr, + const sp<IMemory> &sharedBuffer, size_t offset, const CryptoPlugin::SubSample *subSamples, size_t numSubSamples, void *dstPtr, AString *errorDetailMsg); diff --git a/media/libmediaplayerservice/Drm.cpp b/media/libmediaplayerservice/Drm.cpp index 49e01d1..62cf3e5 100644 --- a/media/libmediaplayerservice/Drm.cpp +++ b/media/libmediaplayerservice/Drm.cpp @@ -136,22 +136,54 @@ void Drm::sendEvent(DrmPlugin::EventType eventType, int extra, if (listener != NULL) { Parcel obj; - if (sessionId && sessionId->size()) { - obj.writeInt32(sessionId->size()); - obj.write(sessionId->array(), sessionId->size()); - } else { - obj.writeInt32(0); - } + writeByteArray(obj, sessionId); + writeByteArray(obj, data); - if (data && data->size()) { - obj.writeInt32(data->size()); - obj.write(data->array(), data->size()); - } else { - obj.writeInt32(0); + Mutex::Autolock lock(mNotifyLock); + listener->notify(eventType, extra, &obj); + } +} + +void Drm::sendExpirationUpdate(Vector<uint8_t> const *sessionId, + int64_t expiryTimeInMS) +{ + mEventLock.lock(); + sp<IDrmClient> listener = mListener; + mEventLock.unlock(); + + if (listener != NULL) { + Parcel obj; + writeByteArray(obj, sessionId); + obj.writeInt64(expiryTimeInMS); + + Mutex::Autolock lock(mNotifyLock); + listener->notify(DrmPlugin::kDrmPluginEventExpirationUpdate, 0, &obj); + } +} + +void Drm::sendKeysChange(Vector<uint8_t> const *sessionId, + Vector<DrmPlugin::KeyStatus> const *keyStatusList, + bool hasNewUsableKey) +{ + mEventLock.lock(); + sp<IDrmClient> listener = mListener; + mEventLock.unlock(); + + if (listener != NULL) { + Parcel obj; + writeByteArray(obj, sessionId); + + size_t nkeys = keyStatusList->size(); + obj.writeInt32(keyStatusList->size()); + for (size_t i = 0; i < nkeys; ++i) { + const DrmPlugin::KeyStatus *keyStatus = &keyStatusList->itemAt(i); + writeByteArray(obj, &keyStatus->mKeyId); + obj.writeInt32(keyStatus->mType); } + obj.writeInt32(hasNewUsableKey); Mutex::Autolock lock(mNotifyLock); - listener->notify(eventType, extra, &obj); + listener->notify(DrmPlugin::kDrmPluginEventKeysChange, 0, &obj); } } @@ -756,4 +788,14 @@ void Drm::binderDied(const wp<IBinder> &the_late_who) closeFactory(); } +void Drm::writeByteArray(Parcel &obj, Vector<uint8_t> const *array) +{ + if (array && array->size()) { + obj.writeInt32(array->size()); + obj.write(array->array(), array->size()); + } else { + obj.writeInt32(0); + } +} + } // namespace android diff --git a/media/libmediaplayerservice/Drm.h b/media/libmediaplayerservice/Drm.h index 7e8f246..1591738 100644 --- a/media/libmediaplayerservice/Drm.h +++ b/media/libmediaplayerservice/Drm.h @@ -133,6 +133,13 @@ struct Drm : public BnDrm, Vector<uint8_t> const *sessionId, Vector<uint8_t> const *data); + virtual void sendExpirationUpdate(Vector<uint8_t> const *sessionId, + int64_t expiryTimeInMS); + + virtual void sendKeysChange(Vector<uint8_t> const *sessionId, + Vector<DrmPlugin::KeyStatus> const *keyStatusList, + bool hasNewUsableKey); + virtual void binderDied(const wp<IBinder> &the_late_who); private: @@ -157,7 +164,7 @@ private: void findFactoryForScheme(const uint8_t uuid[16]); bool loadLibraryForScheme(const String8 &path, const uint8_t uuid[16]); void closeFactory(); - + void writeByteArray(Parcel &obj, Vector<uint8_t> const *array); DISALLOW_EVIL_CONSTRUCTORS(Drm); }; diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp index 04ac699..3fff1e6 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp @@ -82,25 +82,69 @@ void NuPlayer::Decoder::onMessageReceived(const sp<AMessage> &msg) { switch (msg->what()) { case kWhatCodecNotify: { - if (!isStaleReply(msg)) { - int32_t numInput, numOutput; + if (mPaused) { + break; + } + + int32_t cbID; + CHECK(msg->findInt32("callbackID", &cbID)); + + ALOGV("kWhatCodecNotify: cbID = %d", cbID); + switch (cbID) { + case MediaCodec::CB_INPUT_AVAILABLE: + { + int32_t index; + CHECK(msg->findInt32("index", &index)); - if (!msg->findInt32("input-buffers", &numInput)) { - numInput = INT32_MAX; + handleAnInputBuffer(index); + break; + } + + case MediaCodec::CB_OUTPUT_AVAILABLE: + { + int32_t index; + size_t offset; + size_t size; + int64_t timeUs; + int32_t flags; + + CHECK(msg->findInt32("index", &index)); + CHECK(msg->findSize("offset", &offset)); + CHECK(msg->findSize("size", &size)); + CHECK(msg->findInt64("timeUs", &timeUs)); + CHECK(msg->findInt32("flags", &flags)); + + handleAnOutputBuffer(index, offset, size, timeUs, flags); + break; } - if (!msg->findInt32("output-buffers", &numOutput)) { - numOutput = INT32_MAX; + case MediaCodec::CB_OUTPUT_FORMAT_CHANGED: + { + sp<AMessage> format; + CHECK(msg->findMessage("format", &format)); + + handleOutputFormatChange(format); + break; } - if (!mPaused) { - while (numInput-- > 0 && handleAnInputBuffer()) {} + case MediaCodec::CB_ERROR: + { + status_t err; + CHECK(msg->findInt32("err", &err)); + ALOGE("Decoder (%s) reported error : 0x%x", + mIsAudio ? "audio" : "video", err); + + handleError(err); + break; } - while (numOutput-- > 0 && handleAnOutputBuffer()) {} + default: + { + TRESPASS(); + break; + } } - requestCodecNotification(); break; } @@ -188,6 +232,9 @@ void NuPlayer::Decoder::onConfigure(const sp<AMessage> &format) { CHECK_EQ((status_t)OK, mCodec->getOutputFormat(&mOutputFormat)); CHECK_EQ((status_t)OK, mCodec->getInputFormat(&mInputFormat)); + sp<AMessage> reply = new AMessage(kWhatCodecNotify, this); + mCodec->setCallback(reply); + err = mCodec->start(); if (err != OK) { ALOGE("Failed to start %s decoder (err=%d)", mComponentName.c_str(), err); @@ -197,18 +244,8 @@ void NuPlayer::Decoder::onConfigure(const sp<AMessage> &format) { return; } - // the following should work after start - CHECK_EQ((status_t)OK, mCodec->getInputBuffers(&mInputBuffers)); releaseAndResetMediaBuffers(); - CHECK_EQ((status_t)OK, mCodec->getOutputBuffers(&mOutputBuffers)); - ALOGV("[%s] got %zu input and %zu output buffers", - mComponentName.c_str(), - mInputBuffers.size(), - mOutputBuffers.size()); - if (mRenderer != NULL) { - requestCodecNotification(); - } mPaused = false; mResumePending = false; } @@ -217,16 +254,14 @@ void NuPlayer::Decoder::onSetRenderer(const sp<Renderer> &renderer) { bool hadNoRenderer = (mRenderer == NULL); mRenderer = renderer; if (hadNoRenderer && mRenderer != NULL) { - requestCodecNotification(); + // this means that the widevine legacy source is ready + onRequestInputBuffers(); } } void NuPlayer::Decoder::onGetInputBuffers( Vector<sp<ABuffer> > *dstBuffers) { - dstBuffers->clear(); - for (size_t i = 0; i < mInputBuffers.size(); i++) { - dstBuffers->push(mInputBuffers[i]); - } + CHECK_EQ((status_t)OK, mCodec->getWidevineLegacyBuffers(dstBuffers)); } void NuPlayer::Decoder::onResume(bool notifyComplete) { @@ -235,6 +270,7 @@ void NuPlayer::Decoder::onResume(bool notifyComplete) { if (notifyComplete) { mResumePending = true; } + mCodec->start(); } void NuPlayer::Decoder::doFlush(bool notifyComplete) { @@ -261,8 +297,10 @@ void NuPlayer::Decoder::doFlush(bool notifyComplete) { // we attempt to release the buffers even if flush fails. } releaseAndResetMediaBuffers(); + mPaused = true; } + void NuPlayer::Decoder::onFlush() { doFlush(true); @@ -276,7 +314,6 @@ void NuPlayer::Decoder::onFlush() { sp<AMessage> notify = mNotify->dup(); notify->setInt32("what", kWhatFlushCompleted); notify->post(); - mPaused = true; } void NuPlayer::Decoder::onShutdown(bool notifyComplete) { @@ -320,7 +357,9 @@ void NuPlayer::Decoder::onShutdown(bool notifyComplete) { } void NuPlayer::Decoder::doRequestBuffers() { - if (isDiscontinuityPending()) { + // mRenderer is only NULL if we have a legacy widevine source that + // is not yet ready. In this case we must not fetch input. + if (isDiscontinuityPending() || mRenderer == NULL) { return; } status_t err = OK; @@ -347,34 +386,50 @@ void NuPlayer::Decoder::doRequestBuffers() { } } -bool NuPlayer::Decoder::handleAnInputBuffer() { +void NuPlayer::Decoder::handleError(int32_t err) +{ + // We cannot immediately release the codec due to buffers still outstanding + // in the renderer. We signal to the player the error so it can shutdown/release the + // decoder after flushing and increment the generation to discard unnecessary messages. + + ++mBufferGeneration; + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatError); + notify->setInt32("err", err); + notify->post(); +} + +bool NuPlayer::Decoder::handleAnInputBuffer(size_t index) { if (isDiscontinuityPending()) { return false; } - size_t bufferIx = -1; - status_t res = mCodec->dequeueInputBuffer(&bufferIx); - ALOGV("[%s] dequeued input: %d", - mComponentName.c_str(), res == OK ? (int)bufferIx : res); - if (res != OK) { - if (res != -EAGAIN) { - ALOGE("Failed to dequeue input buffer for %s (err=%d)", - mComponentName.c_str(), res); - handleError(res); + + sp<ABuffer> buffer; + mCodec->getInputBuffer(index, &buffer); + + if (index >= mInputBuffers.size()) { + for (size_t i = mInputBuffers.size(); i <= index; ++i) { + mInputBuffers.add(); + mMediaBuffers.add(); + mInputBufferIsDequeued.add(); + mMediaBuffers.editItemAt(i) = NULL; + mInputBufferIsDequeued.editItemAt(i) = false; } - return false; } + mInputBuffers.editItemAt(index) = buffer; - CHECK_LT(bufferIx, mInputBuffers.size()); + //CHECK_LT(bufferIx, mInputBuffers.size()); - if (mMediaBuffers[bufferIx] != NULL) { - mMediaBuffers[bufferIx]->release(); - mMediaBuffers.editItemAt(bufferIx) = NULL; + if (mMediaBuffers[index] != NULL) { + mMediaBuffers[index]->release(); + mMediaBuffers.editItemAt(index) = NULL; } - mInputBufferIsDequeued.editItemAt(bufferIx) = true; + mInputBufferIsDequeued.editItemAt(index) = true; if (!mCSDsToSubmit.isEmpty()) { sp<AMessage> msg = new AMessage(); - msg->setSize("buffer-ix", bufferIx); + msg->setSize("buffer-ix", index); sp<ABuffer> buffer = mCSDsToSubmit.itemAt(0); ALOGI("[%s] resubmitting CSD", mComponentName.c_str()); @@ -392,94 +447,38 @@ bool NuPlayer::Decoder::handleAnInputBuffer() { mPendingInputMessages.erase(mPendingInputMessages.begin()); } - if (!mInputBufferIsDequeued.editItemAt(bufferIx)) { + if (!mInputBufferIsDequeued.editItemAt(index)) { return true; } - mDequeuedInputBuffers.push_back(bufferIx); + mDequeuedInputBuffers.push_back(index); onRequestInputBuffers(); return true; } -bool NuPlayer::Decoder::handleAnOutputBuffer() { - size_t bufferIx = -1; - size_t offset; - size_t size; - int64_t timeUs; - uint32_t flags; - status_t res = mCodec->dequeueOutputBuffer( - &bufferIx, &offset, &size, &timeUs, &flags); - - if (res != OK) { - ALOGV("[%s] dequeued output: %d", mComponentName.c_str(), res); - } else { - ALOGV("[%s] dequeued output: %d (time=%lld flags=%" PRIu32 ")", - mComponentName.c_str(), (int)bufferIx, timeUs, flags); +bool NuPlayer::Decoder::handleAnOutputBuffer( + size_t index, + size_t offset, + size_t size, + int64_t timeUs, + int32_t flags) { + if (mFormatChangePending) { + return false; } - if (res == INFO_OUTPUT_BUFFERS_CHANGED) { - res = mCodec->getOutputBuffers(&mOutputBuffers); - if (res != OK) { - ALOGE("Failed to get output buffers for %s after INFO event (err=%d)", - mComponentName.c_str(), res); - handleError(res); - return false; - } - // NuPlayer ignores this - return true; - } else if (res == INFO_FORMAT_CHANGED) { - sp<AMessage> format = new AMessage(); - res = mCodec->getOutputFormat(&format); - if (res != OK) { - ALOGE("Failed to get output format for %s after INFO event (err=%d)", - mComponentName.c_str(), res); - handleError(res); - return false; - } - - if (!mIsAudio) { - sp<AMessage> notify = mNotify->dup(); - notify->setInt32("what", kWhatVideoSizeChanged); - notify->setMessage("format", format); - notify->post(); - } else if (mRenderer != NULL) { - uint32_t flags; - int64_t durationUs; - bool hasVideo = (mSource->getFormat(false /* audio */) != NULL); - if (!hasVideo && - mSource->getDuration(&durationUs) == OK && - durationUs - > AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US) { - flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; - } else { - flags = AUDIO_OUTPUT_FLAG_NONE; - } +// CHECK_LT(bufferIx, mOutputBuffers.size()); + sp<ABuffer> buffer; + mCodec->getOutputBuffer(index, &buffer); - res = mRenderer->openAudioSink( - format, false /* offloadOnly */, hasVideo, flags, NULL /* isOffloaded */); - if (res != OK) { - ALOGE("Failed to open AudioSink on format change for %s (err=%d)", - mComponentName.c_str(), res); - handleError(res); - return false; - } - } - return true; - } else if (res == INFO_DISCONTINUITY) { - // nothing to do - return true; - } else if (res != OK) { - if (res != -EAGAIN) { - ALOGE("Failed to dequeue output buffer for %s (err=%d)", - mComponentName.c_str(), res); - handleError(res); + if (index >= mOutputBuffers.size()) { + for (size_t i = mOutputBuffers.size(); i <= index; ++i) { + mOutputBuffers.add(); } - return false; } - CHECK_LT(bufferIx, mOutputBuffers.size()); - sp<ABuffer> buffer = mOutputBuffers[bufferIx]; + mOutputBuffers.editItemAt(index) = buffer; + buffer->setRange(offset, size); buffer->meta()->clear(); buffer->meta()->setInt64("timeUs", timeUs); @@ -488,7 +487,7 @@ bool NuPlayer::Decoder::handleAnOutputBuffer() { // we do not expect CODECCONFIG or SYNCFRAME for decoder sp<AMessage> reply = new AMessage(kWhatRenderBuffer, this); - reply->setSize("buffer-ix", bufferIx); + reply->setSize("buffer-ix", index); reply->setInt32("generation", mBufferGeneration); if (eos) { @@ -522,6 +521,29 @@ bool NuPlayer::Decoder::handleAnOutputBuffer() { return true; } +void NuPlayer::Decoder::handleOutputFormatChange(const sp<AMessage> &format) { + if (!mIsAudio) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatVideoSizeChanged); + notify->setMessage("format", format); + notify->post(); + } else if (mRenderer != NULL) { + uint32_t flags; + int64_t durationUs; + bool hasVideo = (mSource->getFormat(false /* audio */) != NULL); + if (!hasVideo && + mSource->getDuration(&durationUs) == OK && + durationUs > AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US) { + flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; + } else { + flags = AUDIO_OUTPUT_FLAG_NONE; + } + + mRenderer->openAudioSink( + format, false /* offloadOnly */, hasVideo, flags, NULL /* isOffloaed */); + } +} + void NuPlayer::Decoder::releaseAndResetMediaBuffers() { for (size_t i = 0; i < mMediaBuffers.size(); i++) { if (mMediaBuffers[i] != NULL) { @@ -825,7 +847,8 @@ void NuPlayer::Decoder::finishHandleDiscontinuity(bool flushOnTimeChange) { mPaused = true; } else if (mTimeChangePending) { if (flushOnTimeChange) { - doFlush(false /*notifyComplete*/); + doFlush(false /* notifyComplete */); + signalResume(false /* notifyComplete */); } // restart fetching input diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h index 4aab2c6..0c0e90c 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h @@ -87,8 +87,15 @@ private: bool mResumePending; AString mComponentName; - bool handleAnInputBuffer(); - bool handleAnOutputBuffer(); + void handleError(int32_t err); + bool handleAnInputBuffer(size_t index); + bool handleAnOutputBuffer( + size_t index, + size_t offset, + size_t size, + int64_t timeUs, + int32_t flags); + void handleOutputFormatChange(const sp<AMessage> &format); void releaseAndResetMediaBuffers(); void requestCodecNotification(); diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp index a2ec51c..827bdc1 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp @@ -312,6 +312,9 @@ void NuPlayer::Renderer::onMessageReceived(const sp<AMessage> &msg) { int64_t delayUs = mAudioSink->msecsPerFrame() * numFramesPendingPlayout * 1000ll; + if (mPlaybackRate > 1.0f) { + delayUs /= mPlaybackRate; + } // Let's give it more data after about half that time // has elapsed. diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index 97f3e20..45f6339 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -1685,6 +1685,16 @@ status_t ACodec::configureCodec( err = setPriority(priority); } + int32_t rateInt = -1; + float rateFloat = -1; + if (!msg->findFloat("operating-rate", &rateFloat)) { + msg->findInt32("operating-rate", &rateInt); + rateFloat = (float)rateInt; // 16MHz (FLINTMAX) is OK for upper bound. + } + if (rateFloat > 0) { + err = setOperatingRate(rateFloat, video); + } + mBaseOutputFormat = outputFormat; CHECK_EQ(getPortFormat(kPortIndexInput, inputFormat), (status_t)OK); @@ -1711,6 +1721,34 @@ status_t ACodec::setPriority(int32_t priority) { return OK; } +status_t ACodec::setOperatingRate(float rateFloat, bool isVideo) { + if (rateFloat < 0) { + return BAD_VALUE; + } + OMX_U32 rate; + if (isVideo) { + if (rateFloat > 65535) { + return BAD_VALUE; + } + rate = (OMX_U32)(rateFloat * 65536.0f + 0.5f); + } else { + if (rateFloat > UINT_MAX) { + return BAD_VALUE; + } + rate = (OMX_U32)(rateFloat); + } + OMX_PARAM_U32TYPE config; + InitOMXParams(&config); + config.nU32 = rate; + status_t err = mOMX->setConfig( + mNode, (OMX_INDEXTYPE)OMX_IndexConfigOperatingRate, + &config, sizeof(config)); + if (err != OK) { + ALOGI("codec does not support config operating rate (err %d)", err); + } + return OK; +} + status_t ACodec::setMinBufferSize(OMX_U32 portIndex, size_t size) { OMX_PARAM_PORTDEFINITIONTYPE def; InitOMXParams(&def); @@ -4902,6 +4940,7 @@ bool ACodec::UninitializedState::onAllocateComponent(const sp<AMessage> &msg) { sp<CodecObserver> observer = new CodecObserver; IOMX::node_id node = NULL; + status_t err = OMX_ErrorComponentNotFound; for (size_t matchIndex = 0; matchIndex < matchingCodecs.size(); ++matchIndex) { componentName = matchingCodecs.itemAt(matchIndex).mName.string(); @@ -4910,7 +4949,7 @@ bool ACodec::UninitializedState::onAllocateComponent(const sp<AMessage> &msg) { pid_t tid = gettid(); int prevPriority = androidGetThreadPriority(tid); androidSetThreadPriority(tid, ANDROID_PRIORITY_FOREGROUND); - status_t err = omx->allocateNode(componentName.c_str(), observer, &node); + err = omx->allocateNode(componentName.c_str(), observer, &node); androidSetThreadPriority(tid, prevPriority); if (err == OK) { @@ -4924,13 +4963,13 @@ bool ACodec::UninitializedState::onAllocateComponent(const sp<AMessage> &msg) { if (node == NULL) { if (!mime.empty()) { - ALOGE("Unable to instantiate a %scoder for type '%s'.", - encoder ? "en" : "de", mime.c_str()); + ALOGE("Unable to instantiate a %scoder for type '%s' with err %#x.", + encoder ? "en" : "de", mime.c_str(), err); } else { - ALOGE("Unable to instantiate codec '%s'.", componentName.c_str()); + ALOGE("Unable to instantiate codec '%s' with err %#x.", componentName.c_str(), err); } - mCodec->signalError(OMX_ErrorComponentNotFound); + mCodec->signalError((OMX_ERRORTYPE)err, makeNoSideEffectStatus(err)); return false; } diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk index a2cbdaf..b0eeb7f 100644 --- a/media/libstagefright/Android.mk +++ b/media/libstagefright/Android.mk @@ -34,6 +34,7 @@ LOCAL_SRC_FILES:= \ MediaClock.cpp \ MediaCodec.cpp \ MediaCodecList.cpp \ + MediaCodecListOverrides.cpp \ MediaCodecSource.cpp \ MediaDefs.cpp \ MediaExtractor.cpp \ diff --git a/media/libstagefright/ESDS.cpp b/media/libstagefright/ESDS.cpp index 427bf7b..8fbb57c 100644 --- a/media/libstagefright/ESDS.cpp +++ b/media/libstagefright/ESDS.cpp @@ -136,6 +136,8 @@ status_t ESDS::parseESDescriptor(size_t offset, size_t size) { --size; if (streamDependenceFlag) { + if (size < 2) + return ERROR_MALFORMED; offset += 2; size -= 2; } @@ -145,11 +147,15 @@ status_t ESDS::parseESDescriptor(size_t offset, size_t size) { return ERROR_MALFORMED; } unsigned URLlength = mData[offset]; + if (URLlength >= size) + return ERROR_MALFORMED; offset += URLlength + 1; size -= URLlength + 1; } if (OCRstreamFlag) { + if (size < 2) + return ERROR_MALFORMED; offset += 2; size -= 2; diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp index d0f42cc..f7fa2b6 100644 --- a/media/libstagefright/MPEG4Extractor.cpp +++ b/media/libstagefright/MPEG4Extractor.cpp @@ -874,6 +874,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { } } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->sampleTable = new SampleTable(mDataSource); } @@ -1028,6 +1031,10 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { } original_fourcc = ntohl(original_fourcc); ALOGV("read original format: %d", original_fourcc); + + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setCString(kKeyMIMEType, FourCC2MIME(original_fourcc)); uint32_t num_channels = 0; uint32_t sample_rate = 0; @@ -1083,6 +1090,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return ERROR_IO; } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setInt32(kKeyCryptoMode, defaultAlgorithmId); mLastTrack->meta->setInt32(kKeyCryptoDefaultIVSize, defaultIVSize); mLastTrack->meta->setData(kKeyCryptoKey, 'tenc', defaultKeyId, 16); @@ -1168,6 +1178,11 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return ERROR_IO; } + if (!timescale) { + ALOGE("timescale should not be ZERO."); + return ERROR_MALFORMED; + } + mLastTrack->timescale = ntohl(timescale); // 14496-12 says all ones means indeterminate, but some files seem to use @@ -1193,7 +1208,7 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { duration = ntohl(duration32); } } - if (duration != 0) { + if (duration != 0 && mLastTrack->timescale != 0) { mLastTrack->meta->setInt64( kKeyDuration, (duration * 1000000) / mLastTrack->timescale); } @@ -1257,6 +1272,10 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { // display the timed text. // For encrypted files, there may also be more than one entry. const char *mime; + + if (mLastTrack == NULL) + return ERROR_MALFORMED; + CHECK(mLastTrack->meta->findCString(kKeyMIMEType, &mime)); if (strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) && strcasecmp(mime, "application/octet-stream")) { @@ -1303,6 +1322,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { uint16_t sample_size = U16_AT(&buffer[18]); uint32_t sample_rate = U32_AT(&buffer[24]) >> 16; + if (mLastTrack == NULL) + return ERROR_MALFORMED; + if (chunk_type != FOURCC('e', 'n', 'c', 'a')) { // if the chunk type is enca, we'll get the type from the sinf/frma box later mLastTrack->meta->setCString(kKeyMIMEType, FourCC2MIME(chunk_type)); @@ -1364,6 +1386,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { // printf("*** coding='%s' width=%d height=%d\n", // chunk, width, height); + if (mLastTrack == NULL) + return ERROR_MALFORMED; + if (chunk_type != FOURCC('e', 'n', 'c', 'v')) { // if the chunk type is encv, we'll get the type from the sinf/frma box later mLastTrack->meta->setCString(kKeyMIMEType, FourCC2MIME(chunk_type)); @@ -1389,6 +1414,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('s', 't', 'c', 'o'): case FOURCC('c', 'o', '6', '4'): { + if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) + return ERROR_MALFORMED; + status_t err = mLastTrack->sampleTable->setChunkOffsetParams( chunk_type, data_offset, chunk_data_size); @@ -1404,6 +1432,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('s', 't', 's', 'c'): { + if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) + return ERROR_MALFORMED; + status_t err = mLastTrack->sampleTable->setSampleToChunkParams( data_offset, chunk_data_size); @@ -1420,6 +1451,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('s', 't', 's', 'z'): case FOURCC('s', 't', 'z', '2'): { + if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) + return ERROR_MALFORMED; + status_t err = mLastTrack->sampleTable->setSampleSizeParams( chunk_type, data_offset, chunk_data_size); @@ -1489,6 +1523,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('s', 't', 't', 's'): { + if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) + return ERROR_MALFORMED; + *offset += chunk_size; status_t err = @@ -1504,6 +1541,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('c', 't', 't', 's'): { + if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) + return ERROR_MALFORMED; + *offset += chunk_size; status_t err = @@ -1519,6 +1559,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('s', 't', 's', 's'): { + if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) + return ERROR_MALFORMED; + *offset += chunk_size; status_t err = @@ -1591,6 +1634,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return ERROR_MALFORMED; } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setData( kKeyESDS, kTypeESDS, &buffer[4], chunk_data_size - 4); @@ -1623,6 +1669,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return ERROR_IO; } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setData( kKeyAVCC, kTypeAVCC, buffer->data(), chunk_data_size); @@ -1637,6 +1686,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return ERROR_IO; } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setData( kKeyHVCC, kTypeHVCC, buffer->data(), chunk_data_size); @@ -1670,6 +1722,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return ERROR_IO; } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setData(kKeyD263, kTypeD263, buffer, chunk_data_size); break; @@ -1767,7 +1822,7 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { } duration = d32; } - if (duration != 0) { + if (duration != 0 && mHeaderTimescale != 0) { mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale); } @@ -1816,7 +1871,7 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return ERROR_MALFORMED; } - if (duration != 0) { + if (duration != 0 && mHeaderTimescale != 0) { mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale); } @@ -1851,6 +1906,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return ERROR_IO; } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + uint32_t type = ntohl(buffer); // For the 3GPP file format, the handler-type within the 'hdlr' box // shall be 'text'. We also want to support 'sbtl' handler type @@ -1883,6 +1941,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('t', 'x', '3', 'g'): { + if (mLastTrack == NULL) + return ERROR_MALFORMED; + uint32_t type; const void *data; size_t size = 0; @@ -2024,6 +2085,8 @@ status_t MPEG4Extractor::parseSegmentIndex(off64_t offset, size_t size) { return ERROR_MALFORMED; } ALOGV("sidx refid/timescale: %d/%d", referenceId, timeScale); + if (timeScale == 0) + return ERROR_MALFORMED; uint64_t earliestPresentationTime; uint64_t firstOffset; @@ -2107,6 +2170,9 @@ status_t MPEG4Extractor::parseSegmentIndex(off64_t offset, size_t size) { uint64_t sidxDuration = total_duration * 1000000 / timeScale; + if (mLastTrack == NULL) + return ERROR_MALFORMED; + int64_t metaDuration; if (!mLastTrack->meta->findInt64(kKeyDuration, &metaDuration) || metaDuration == 0) { mLastTrack->meta->setInt64(kKeyDuration, sidxDuration); @@ -2157,6 +2223,9 @@ status_t MPEG4Extractor::parseTrackHeader( return ERROR_UNSUPPORTED; } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setInt32(kKeyTrackID, id); size_t matrixOffset = dynSize + 16; @@ -2339,6 +2408,9 @@ status_t MPEG4Extractor::parseITunesMetaData(off64_t offset, size_t size) { int32_t delay, padding; if (sscanf(mLastCommentData, " %*x %x %x %*x", &delay, &padding) == 2) { + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setInt32(kKeyEncoderDelay, delay); mLastTrack->meta->setInt32(kKeyEncoderPadding, padding); } @@ -2635,6 +2707,11 @@ status_t MPEG4Extractor::verifyTrack(Track *track) { return ERROR_MALFORMED; } + if (track->timescale == 0) { + ALOGE("timescale invalid."); + return ERROR_MALFORMED; + } + return OK; } @@ -2701,6 +2778,9 @@ status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio( if (objectTypeIndication == 0xe1) { // This isn't MPEG4 audio at all, it's QCELP 14k... + if (mLastTrack == NULL) + return ERROR_MALFORMED; + mLastTrack->meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_QCELP); return OK; } @@ -2749,6 +2829,9 @@ status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio( objectType = 32 + br.getBits(6); } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + //keep AOT type mLastTrack->meta->setInt32(kKeyAACAOT, objectType); @@ -2919,6 +3002,9 @@ status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio( return ERROR_UNSUPPORTED; } + if (mLastTrack == NULL) + return ERROR_MALFORMED; + int32_t prevSampleRate; CHECK(mLastTrack->meta->findInt32(kKeySampleRate, &prevSampleRate)); diff --git a/media/libstagefright/MediaClock.cpp b/media/libstagefright/MediaClock.cpp index 433f555..2641e4e 100644 --- a/media/libstagefright/MediaClock.cpp +++ b/media/libstagefright/MediaClock.cpp @@ -92,6 +92,11 @@ void MediaClock::setPlaybackRate(float rate) { mPlaybackRate = rate; } +float MediaClock::getPlaybackRate() const { + Mutex::Autolock autoLock(mLock); + return mPlaybackRate; +} + status_t MediaClock::getMediaTime( int64_t realUs, int64_t *outMediaUs, bool allowPastMaxTime) const { if (outMediaUs == NULL) { diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp index 0597f1d..5538cb0 100644 --- a/media/libstagefright/MediaCodec.cpp +++ b/media/libstagefright/MediaCodec.cpp @@ -22,7 +22,9 @@ #include "include/SoftwareRenderer.h" #include <binder/IBatteryStats.h> +#include <binder/IMemory.h> #include <binder/IServiceManager.h> +#include <binder/MemoryDealer.h> #include <gui/Surface.h> #include <media/ICrypto.h> #include <media/stagefright/foundation/ABuffer.h> @@ -47,16 +49,31 @@ namespace android { struct MediaCodec::BatteryNotifier : public Singleton<BatteryNotifier> { BatteryNotifier(); + virtual ~BatteryNotifier(); void noteStartVideo(); void noteStopVideo(); void noteStartAudio(); void noteStopAudio(); + void onBatteryStatServiceDied(); private: + struct DeathNotifier : public IBinder::DeathRecipient { + DeathNotifier() {} + virtual void binderDied(const wp<IBinder>& /*who*/) { + BatteryNotifier::getInstance().onBatteryStatServiceDied(); + } + }; + + Mutex mLock; int32_t mVideoRefCount; int32_t mAudioRefCount; sp<IBatteryStats> mBatteryStatService; + sp<DeathNotifier> mDeathNotifier; + + sp<IBatteryStats> getBatteryService_l(); + + DISALLOW_EVIL_CONSTRUCTORS(BatteryNotifier); }; ANDROID_SINGLETON_STATIC_INSTANCE(MediaCodec::BatteryNotifier) @@ -64,54 +81,103 @@ ANDROID_SINGLETON_STATIC_INSTANCE(MediaCodec::BatteryNotifier) MediaCodec::BatteryNotifier::BatteryNotifier() : mVideoRefCount(0), mAudioRefCount(0) { - // get battery service +} + +sp<IBatteryStats> MediaCodec::BatteryNotifier::getBatteryService_l() { + if (mBatteryStatService != NULL) { + return mBatteryStatService; + } + // get battery service from service manager const sp<IServiceManager> sm(defaultServiceManager()); if (sm != NULL) { const String16 name("batterystats"); - mBatteryStatService = interface_cast<IBatteryStats>(sm->getService(name)); + mBatteryStatService = + interface_cast<IBatteryStats>(sm->getService(name)); if (mBatteryStatService == NULL) { ALOGE("batterystats service unavailable!"); + return NULL; + } + mDeathNotifier = new DeathNotifier(); + if (IInterface::asBinder(mBatteryStatService)-> + linkToDeath(mDeathNotifier) != OK) { + mBatteryStatService.clear(); + mDeathNotifier.clear(); + return NULL; } + // notify start now if media already started + if (mVideoRefCount > 0) { + mBatteryStatService->noteStartVideo(AID_MEDIA); + } + if (mAudioRefCount > 0) { + mBatteryStatService->noteStartAudio(AID_MEDIA); + } + } + return mBatteryStatService; +} + +MediaCodec::BatteryNotifier::~BatteryNotifier() { + if (mDeathNotifier != NULL) { + IInterface::asBinder(mBatteryStatService)-> + unlinkToDeath(mDeathNotifier); } } void MediaCodec::BatteryNotifier::noteStartVideo() { - if (mVideoRefCount == 0 && mBatteryStatService != NULL) { - mBatteryStatService->noteStartVideo(AID_MEDIA); + Mutex::Autolock _l(mLock); + sp<IBatteryStats> batteryService = getBatteryService_l(); + if (mVideoRefCount == 0 && batteryService != NULL) { + batteryService->noteStartVideo(AID_MEDIA); } mVideoRefCount++; } void MediaCodec::BatteryNotifier::noteStopVideo() { + Mutex::Autolock _l(mLock); if (mVideoRefCount == 0) { ALOGW("BatteryNotifier::noteStop(): video refcount is broken!"); return; } mVideoRefCount--; - if (mVideoRefCount == 0 && mBatteryStatService != NULL) { - mBatteryStatService->noteStopVideo(AID_MEDIA); + sp<IBatteryStats> batteryService = getBatteryService_l(); + if (mVideoRefCount == 0 && batteryService != NULL) { + batteryService->noteStopVideo(AID_MEDIA); } } void MediaCodec::BatteryNotifier::noteStartAudio() { - if (mAudioRefCount == 0 && mBatteryStatService != NULL) { - mBatteryStatService->noteStartAudio(AID_MEDIA); + Mutex::Autolock _l(mLock); + sp<IBatteryStats> batteryService = getBatteryService_l(); + if (mAudioRefCount == 0 && batteryService != NULL) { + batteryService->noteStartAudio(AID_MEDIA); } mAudioRefCount++; } void MediaCodec::BatteryNotifier::noteStopAudio() { + Mutex::Autolock _l(mLock); if (mAudioRefCount == 0) { ALOGW("BatteryNotifier::noteStop(): audio refcount is broken!"); return; } mAudioRefCount--; - if (mAudioRefCount == 0 && mBatteryStatService != NULL) { - mBatteryStatService->noteStopAudio(AID_MEDIA); + sp<IBatteryStats> batteryService = getBatteryService_l(); + if (mAudioRefCount == 0 && batteryService != NULL) { + batteryService->noteStopAudio(AID_MEDIA); } } + +void MediaCodec::BatteryNotifier::onBatteryStatServiceDied() { + Mutex::Autolock _l(mLock); + mBatteryStatService.clear(); + mDeathNotifier.clear(); + // Do not reset mVideoRefCount and mAudioRefCount here. The ref + // counting is independent of the battery service availability. + // We need this if battery service becomes available after media + // started. +} + // static sp<MediaCodec> MediaCodec::CreateByType( const sp<ALooper> &looper, const char *mime, bool encoder, status_t *err) { @@ -544,6 +610,16 @@ status_t MediaCodec::getName(AString *name) const { return OK; } +status_t MediaCodec::getWidevineLegacyBuffers(Vector<sp<ABuffer> > *buffers) const { + sp<AMessage> msg = new AMessage(kWhatGetBuffers, this); + msg->setInt32("portIndex", kPortIndexInput); + msg->setPointer("buffers", buffers); + msg->setInt32("widevine", true); + + sp<AMessage> response; + return PostAndAwaitResponse(msg, &response); +} + status_t MediaCodec::getInputBuffers(Vector<sp<ABuffer> > *buffers) const { sp<AMessage> msg = new AMessage(kWhatGetBuffers, this); msg->setInt32("portIndex", kPortIndexInput); @@ -969,6 +1045,17 @@ void MediaCodec::onMessageReceived(const sp<AMessage> &msg) { size_t numBuffers = portDesc->countBuffers(); + size_t totalSize = 0; + for (size_t i = 0; i < numBuffers; ++i) { + if (portIndex == kPortIndexInput && mCrypto != NULL) { + totalSize += portDesc->bufferAt(i)->capacity(); + } + } + + if (totalSize) { + mDealer = new MemoryDealer(totalSize, "MediaCodec"); + } + for (size_t i = 0; i < numBuffers; ++i) { BufferInfo info; info.mBufferID = portDesc->bufferIDAt(i); @@ -976,8 +1063,10 @@ void MediaCodec::onMessageReceived(const sp<AMessage> &msg) { info.mData = portDesc->bufferAt(i); if (portIndex == kPortIndexInput && mCrypto != NULL) { + sp<IMemory> mem = mDealer->allocate(info.mData->capacity()); info.mEncryptedData = - new ABuffer(info.mData->capacity()); + new ABuffer(mem->pointer(), info.mData->capacity()); + info.mSharedEncryptedBuffer = mem; } buffers->push_back(info); @@ -1587,8 +1676,12 @@ void MediaCodec::onMessageReceived(const sp<AMessage> &msg) { { sp<AReplyToken> replyID; CHECK(msg->senderAwaitsResponse(&replyID)); + // Unfortunately widevine legacy source requires knowing all of the + // codec input buffers, so we have to provide them even in async mode. + int32_t widevine = 0; + msg->findInt32("widevine", &widevine); - if (!isExecuting() || (mFlags & kFlagIsAsync)) { + if (!isExecuting() || ((mFlags & kFlagIsAsync) && !widevine)) { PostReplyWithError(replyID, INVALID_OPERATION); break; } else if (mFlags & kFlagStickyError) { @@ -1953,7 +2046,8 @@ status_t MediaCodec::onQueueInputBuffer(const sp<AMessage> &msg) { key, iv, mode, - info->mEncryptedData->base() + offset, + info->mSharedEncryptedBuffer, + offset, subSamples, numSubSamples, info->mData->base(), diff --git a/media/libstagefright/MediaCodecList.cpp b/media/libstagefright/MediaCodecList.cpp index cf6e937..26798ae 100644 --- a/media/libstagefright/MediaCodecList.cpp +++ b/media/libstagefright/MediaCodecList.cpp @@ -18,6 +18,8 @@ #define LOG_TAG "MediaCodecList" #include <utils/Log.h> +#include "MediaCodecListOverrides.h" + #include <binder/IServiceManager.h> #include <media/IMediaCodecList.h> @@ -31,6 +33,7 @@ #include <media/stagefright/OMXClient.h> #include <media/stagefright/OMXCodec.h> +#include <sys/stat.h> #include <utils/threads.h> #include <libexpat/expat.h> @@ -41,21 +44,58 @@ static Mutex sInitMutex; static MediaCodecList *gCodecList = NULL; +static const char *kProfilingResults = "/data/misc/media/media_codecs_profiling_results.xml"; + +static bool parseBoolean(const char *s) { + if (!strcasecmp(s, "true") || !strcasecmp(s, "yes") || !strcasecmp(s, "y")) { + return true; + } + char *end; + unsigned long res = strtoul(s, &end, 10); + return *s != '\0' && *end == '\0' && res > 0; +} + // static sp<IMediaCodecList> MediaCodecList::sCodecList; // static sp<IMediaCodecList> MediaCodecList::getLocalInstance() { - Mutex::Autolock autoLock(sInitMutex); - - if (gCodecList == NULL) { - gCodecList = new MediaCodecList; - if (gCodecList->initCheck() == OK) { - sCodecList = gCodecList; + bool profilingNeeded = false; + KeyedVector<AString, CodecSettings> updates; + Vector<sp<MediaCodecInfo>> infos; + + { + Mutex::Autolock autoLock(sInitMutex); + + if (gCodecList == NULL) { + gCodecList = new MediaCodecList; + if (gCodecList->initCheck() == OK) { + sCodecList = gCodecList; + + struct stat s; + if (stat(kProfilingResults, &s) == -1) { + // profiling results doesn't existed + profilingNeeded = true; + for (size_t i = 0; i < gCodecList->countCodecs(); ++i) { + infos.push_back(gCodecList->getCodecInfo(i)); + } + } + } } } - return sCodecList; + if (profilingNeeded) { + profileCodecs(infos, &updates); + } + + { + Mutex::Autolock autoLock(sInitMutex); + if (updates.size() > 0) { + gCodecList->updateDetailsForMultipleCodecs(updates); + } + + return sCodecList; + } } static Mutex sRemoteInitMutex; @@ -94,11 +134,27 @@ sp<IMediaCodecList> MediaCodecList::getInstance() { } MediaCodecList::MediaCodecList() - : mInitCheck(NO_INIT) { + : mInitCheck(NO_INIT), + mUpdate(false), + mGlobalSettings(new AMessage()) { parseTopLevelXMLFile("/etc/media_codecs.xml"); + parseTopLevelXMLFile(kProfilingResults, true/* ignore_errors */); +} + +void MediaCodecList::updateDetailsForMultipleCodecs( + const KeyedVector<AString, CodecSettings>& updates) { + if (updates.size() == 0) { + return; + } + + exportResultsToXML(kProfilingResults, updates); + + for (size_t i = 0; i < updates.size(); ++i) { + applyCodecSettings(updates.keyAt(i), updates.valueAt(i), &mCodecInfos); + } } -void MediaCodecList::parseTopLevelXMLFile(const char *codecs_xml) { +void MediaCodecList::parseTopLevelXMLFile(const char *codecs_xml, bool ignore_errors) { // get href_base char *href_base_end = strrchr(codecs_xml, '/'); if (href_base_end != NULL) { @@ -119,13 +175,16 @@ void MediaCodecList::parseTopLevelXMLFile(const char *codecs_xml) { mOMX.clear(); if (mInitCheck != OK) { + if (ignore_errors) { + mInitCheck = OK; + return; + } mCodecInfos.clear(); return; } for (size_t i = mCodecInfos.size(); i-- > 0;) { const MediaCodecInfo &info = *mCodecInfos.itemAt(i).get(); - if (info.mCaps.size() == 0) { // No types supported by this component??? ALOGW("Component %s does not support any type of media?", @@ -169,6 +228,16 @@ void MediaCodecList::parseTopLevelXMLFile(const char *codecs_xml) { } ALOGV(" levels=[%s]", nice.c_str()); } + { + AString quirks; + for (size_t ix = 0; ix < info.mQuirks.size(); ix++) { + if (ix > 0) { + quirks.append(", "); + } + quirks.append(info.mQuirks[ix]); + } + ALOGV(" quirks=[%s]", quirks.c_str()); + } } #endif } @@ -328,6 +397,16 @@ void MediaCodecList::startElementHandler( mCurrentSection = SECTION_DECODERS; } else if (!strcmp(name, "Encoders")) { mCurrentSection = SECTION_ENCODERS; + } else if (!strcmp(name, "Settings")) { + mCurrentSection = SECTION_SETTINGS; + } + break; + } + + case SECTION_SETTINGS: + { + if (!strcmp(name, "Setting")) { + mInitCheck = addSettingFromAttributes(attrs); } break; } @@ -397,6 +476,14 @@ void MediaCodecList::endElementHandler(const char *name) { } switch (mCurrentSection) { + case SECTION_SETTINGS: + { + if (!strcmp(name, "Settings")) { + mCurrentSection = SECTION_TOPLEVEL; + } + break; + } + case SECTION_DECODERS: { if (!strcmp(name, "Decoders")) { @@ -462,10 +549,10 @@ void MediaCodecList::endElementHandler(const char *name) { --mDepth; } -status_t MediaCodecList::addMediaCodecFromAttributes( - bool encoder, const char **attrs) { +status_t MediaCodecList::addSettingFromAttributes(const char **attrs) { const char *name = NULL; - const char *type = NULL; + const char *value = NULL; + const char *update = NULL; size_t i = 0; while (attrs[i] != NULL) { @@ -475,11 +562,17 @@ status_t MediaCodecList::addMediaCodecFromAttributes( } name = attrs[i + 1]; ++i; - } else if (!strcmp(attrs[i], "type")) { + } else if (!strcmp(attrs[i], "value")) { if (attrs[i + 1] == NULL) { return -EINVAL; } - type = attrs[i + 1]; + value = attrs[i + 1]; + ++i; + } else if (!strcmp(attrs[i], "update")) { + if (attrs[i + 1] == NULL) { + return -EINVAL; + } + update = attrs[i + 1]; ++i; } else { return -EINVAL; @@ -488,10 +581,34 @@ status_t MediaCodecList::addMediaCodecFromAttributes( ++i; } - if (name == NULL) { + if (name == NULL || value == NULL) { return -EINVAL; } + mUpdate = (update != NULL) && parseBoolean(update); + if (mUpdate != mGlobalSettings->contains(name)) { + return -EINVAL; + } + + mGlobalSettings->setString(name, value); + return OK; +} + +void MediaCodecList::setCurrentCodecInfo(bool encoder, const char *name, const char *type) { + for (size_t i = 0; i < mCodecInfos.size(); ++i) { + if (AString(name) == mCodecInfos[i]->getCodecName()) { + if (mCodecInfos[i]->getCapabilitiesFor(type) == NULL) { + ALOGW("Overrides with an unexpected mime %s", type); + // Create a new MediaCodecInfo (but don't add it to mCodecInfos) to hold the + // overrides we don't want. + mCurrentInfo = new MediaCodecInfo(name, encoder, type); + } else { + mCurrentInfo = mCodecInfos.editItemAt(i); + mCurrentInfo->updateMime(type); // to set the current cap + } + return; + } + } mCurrentInfo = new MediaCodecInfo(name, encoder, type); // The next step involves trying to load the codec, which may // fail. Only list the codec if this succeeds. @@ -500,6 +617,78 @@ status_t MediaCodecList::addMediaCodecFromAttributes( if (initializeCapabilities(type) == OK) { mCodecInfos.push_back(mCurrentInfo); } +} + +status_t MediaCodecList::addMediaCodecFromAttributes( + bool encoder, const char **attrs) { + const char *name = NULL; + const char *type = NULL; + const char *update = NULL; + + size_t i = 0; + while (attrs[i] != NULL) { + if (!strcmp(attrs[i], "name")) { + if (attrs[i + 1] == NULL) { + return -EINVAL; + } + name = attrs[i + 1]; + ++i; + } else if (!strcmp(attrs[i], "type")) { + if (attrs[i + 1] == NULL) { + return -EINVAL; + } + type = attrs[i + 1]; + ++i; + } else if (!strcmp(attrs[i], "update")) { + if (attrs[i + 1] == NULL) { + return -EINVAL; + } + update = attrs[i + 1]; + ++i; + } else { + return -EINVAL; + } + + ++i; + } + + if (name == NULL) { + return -EINVAL; + } + + mUpdate = (update != NULL) && parseBoolean(update); + ssize_t index = -1; + for (size_t i = 0; i < mCodecInfos.size(); ++i) { + if (AString(name) == mCodecInfos[i]->getCodecName()) { + index = i; + } + } + if (mUpdate != (index >= 0)) { + return -EINVAL; + } + + if (index >= 0) { + // existing codec + mCurrentInfo = mCodecInfos.editItemAt(index); + if (type != NULL) { + // existing type + if (mCodecInfos[index]->getCapabilitiesFor(type) == NULL) { + return -EINVAL; + } + mCurrentInfo->updateMime(type); + } + } else { + // new codec + mCurrentInfo = new MediaCodecInfo(name, encoder, type); + // The next step involves trying to load the codec, which may + // fail. Only list the codec if this succeeds. + // However, keep mCurrentInfo object around until parsing + // of full codec info is completed. + if (initializeCapabilities(type) == OK) { + mCodecInfos.push_back(mCurrentInfo); + } + } + return OK; } @@ -553,6 +742,7 @@ status_t MediaCodecList::addQuirk(const char **attrs) { status_t MediaCodecList::addTypeFromAttributes(const char **attrs) { const char *name = NULL; + const char *update = NULL; size_t i = 0; while (attrs[i] != NULL) { @@ -562,6 +752,12 @@ status_t MediaCodecList::addTypeFromAttributes(const char **attrs) { } name = attrs[i + 1]; ++i; + } else if (!strcmp(attrs[i], "update")) { + if (attrs[i + 1] == NULL) { + return -EINVAL; + } + update = attrs[i + 1]; + ++i; } else { return -EINVAL; } @@ -573,14 +769,25 @@ status_t MediaCodecList::addTypeFromAttributes(const char **attrs) { return -EINVAL; } - status_t ret = mCurrentInfo->addMime(name); + bool isExistingType = (mCurrentInfo->getCapabilitiesFor(name) != NULL); + if (mUpdate != isExistingType) { + return -EINVAL; + } + + status_t ret; + if (mUpdate) { + ret = mCurrentInfo->updateMime(name); + } else { + ret = mCurrentInfo->addMime(name); + } + if (ret != OK) { return ret; } // The next step involves trying to load the codec, which may // fail. Handle this gracefully (by not reporting such mime). - if (initializeCapabilities(name) != OK) { + if (!mUpdate && initializeCapabilities(name) != OK) { mCurrentInfo->removeMime(name); } return OK; @@ -758,7 +965,8 @@ status_t MediaCodecList::addLimit(const char **attrs) { return limitFoundMissingAttr(name, "ranges", found); } else if (msg->contains("scale")) { return limitFoundMissingAttr(name, "scale"); - } else if ((name == "alignment" || name == "block-size") ^ + } else if ((name == "alignment" || name == "block-size" + || name == "max-supported-instances") ^ (found = msg->findString("value", &value))) { return limitFoundMissingAttr(name, "value", found); } @@ -780,15 +988,6 @@ status_t MediaCodecList::addLimit(const char **attrs) { return OK; } -static bool parseBoolean(const char *s) { - if (!strcasecmp(s, "true") || !strcasecmp(s, "yes") || !strcasecmp(s, "y")) { - return true; - } - char *end; - unsigned long res = strtoul(s, &end, 10); - return *s != '\0' && *end == '\0' && res > 0; -} - status_t MediaCodecList::addFeature(const char **attrs) { size_t i = 0; const char *name = NULL; @@ -860,4 +1059,8 @@ size_t MediaCodecList::countCodecs() const { return mCodecInfos.size(); } +const sp<AMessage> MediaCodecList::getGlobalSettings() const { + return mGlobalSettings; +} + } // namespace android diff --git a/media/libstagefright/MediaCodecListOverrides.cpp b/media/libstagefright/MediaCodecListOverrides.cpp new file mode 100644 index 0000000..3c54f34 --- /dev/null +++ b/media/libstagefright/MediaCodecListOverrides.cpp @@ -0,0 +1,404 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "MediaCodecListOverrides" +#include <utils/Log.h> + +#include "MediaCodecListOverrides.h" + +#include <gui/Surface.h> +#include <media/ICrypto.h> +#include <media/IMediaCodecList.h> +#include <media/MediaCodecInfo.h> + +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/MediaCodec.h> + +namespace android { + +// a limit to avoid allocating unreasonable number of codec instances in the measurement. +// this should be in sync with the MAX_SUPPORTED_INSTANCES defined in MediaCodecInfo.java. +static const int kMaxInstances = 32; + +// TODO: move MediaCodecInfo to C++. Until then, some temp methods to parse out info. +static bool getMeasureSize(sp<MediaCodecInfo::Capabilities> caps, int32_t *width, int32_t *height) { + AString sizeRange; + if (!caps->getDetails()->findString("size-range", &sizeRange)) { + return false; + } + AString minSize; + AString maxSize; + if (!splitString(sizeRange, "-", &minSize, &maxSize)) { + return false; + } + AString sWidth; + AString sHeight; + if (!splitString(minSize, "x", &sWidth, &sHeight)) { + if (!splitString(minSize, "*", &sWidth, &sHeight)) { + return false; + } + } + + *width = strtol(sWidth.c_str(), NULL, 10); + *height = strtol(sHeight.c_str(), NULL, 10); + return (*width > 0) && (*height > 0); +} + +static void getMeasureBitrate(sp<MediaCodecInfo::Capabilities> caps, int32_t *bitrate) { + // Until have native MediaCodecInfo, we cannot get bitrates based on profile/levels. + // We use 200000 as default value for our measurement. + *bitrate = 200000; + AString bitrateRange; + if (!caps->getDetails()->findString("bitrate-range", &bitrateRange)) { + return; + } + AString minBitrate; + AString maxBitrate; + if (!splitString(bitrateRange, "-", &minBitrate, &maxBitrate)) { + return; + } + + *bitrate = strtol(minBitrate.c_str(), NULL, 10); +} + +static sp<AMessage> getMeasureFormat( + bool isEncoder, AString mime, sp<MediaCodecInfo::Capabilities> caps) { + sp<AMessage> format = new AMessage(); + format->setString("mime", mime); + + if (isEncoder) { + int32_t bitrate = 0; + getMeasureBitrate(caps, &bitrate); + format->setInt32("bitrate", bitrate); + } + + if (mime.startsWith("video/")) { + int32_t width = 0; + int32_t height = 0; + if (!getMeasureSize(caps, &width, &height)) { + return NULL; + } + format->setInt32("width", width); + format->setInt32("height", height); + + Vector<uint32_t> colorFormats; + caps->getSupportedColorFormats(&colorFormats); + if (colorFormats.size() == 0) { + return NULL; + } + format->setInt32("color-format", colorFormats[0]); + + format->setFloat("frame-rate", 10.0); + format->setInt32("i-frame-interval", 10); + } else { + // TODO: profile hw audio + return NULL; + } + + return format; +} + +static size_t doProfileCodecs( + bool isEncoder, AString name, AString mime, sp<MediaCodecInfo::Capabilities> caps) { + sp<AMessage> format = getMeasureFormat(isEncoder, mime, caps); + if (format == NULL) { + return 0; + } + if (isEncoder) { + format->setInt32("encoder", 1); + } + ALOGV("doProfileCodecs %s %s %s %s", + name.c_str(), mime.c_str(), isEncoder ? "encoder" : "decoder", + format->debugString().c_str()); + + status_t err = OK; + Vector<sp<MediaCodec>> codecs; + while (err == OK && codecs.size() < kMaxInstances) { + sp<ALooper> looper = new ALooper; + looper->setName("MediaCodec_looper"); + ALOGV("doProfileCodecs for codec #%u", codecs.size()); + ALOGV("doProfileCodecs start looper"); + looper->start( + false /* runOnCallingThread */, false /* canCallJava */, ANDROID_PRIORITY_AUDIO); + ALOGV("doProfileCodecs CreateByComponentName"); + sp<MediaCodec> codec = MediaCodec::CreateByComponentName(looper, name.c_str(), &err); + if (err != OK) { + ALOGV("Failed to create codec: %s", name.c_str()); + break; + } + const sp<Surface> nativeWindow; + const sp<ICrypto> crypto; + uint32_t flags = 0; + ALOGV("doProfileCodecs configure"); + err = codec->configure(format, nativeWindow, crypto, flags); + if (err != OK) { + ALOGV("Failed to configure codec: %s with mime: %s", name.c_str(), mime.c_str()); + codec->release(); + break; + } + ALOGV("doProfileCodecs start"); + err = codec->start(); + if (err != OK) { + ALOGV("Failed to start codec: %s with mime: %s", name.c_str(), mime.c_str()); + codec->release(); + break; + } + codecs.push_back(codec); + } + + for (size_t i = 0; i < codecs.size(); ++i) { + ALOGV("doProfileCodecs release %s", name.c_str()); + err = codecs[i]->release(); + if (err != OK) { + ALOGE("Failed to release codec: %s with mime: %s", name.c_str(), mime.c_str()); + } + } + + return codecs.size(); +} + +static void printLongString(const char *buf, size_t size) { + AString print; + const char *start = buf; + size_t len; + size_t totalLen = size; + while (totalLen > 0) { + len = (totalLen > 500) ? 500 : totalLen; + print.setTo(start, len); + ALOGV("%s", print.c_str()); + totalLen -= len; + start += len; + } +} + +bool splitString(const AString &s, const AString &delimiter, AString *s1, AString *s2) { + ssize_t pos = s.find(delimiter.c_str()); + if (pos < 0) { + return false; + } + *s1 = AString(s, 0, pos); + *s2 = AString(s, pos + 1, s.size() - pos - 1); + return true; +} + +bool splitString( + const AString &s, const AString &delimiter, AString *s1, AString *s2, AString *s3) { + AString temp; + if (!splitString(s, delimiter, s1, &temp)) { + return false; + } + if (!splitString(temp, delimiter, s2, s3)) { + return false; + } + return true; +} + +void profileCodecs( + const Vector<sp<MediaCodecInfo>> &infos, + KeyedVector<AString, CodecSettings> *results, + bool forceToMeasure) { + KeyedVector<AString, sp<MediaCodecInfo::Capabilities>> codecsNeedMeasure; + for (size_t i = 0; i < infos.size(); ++i) { + const sp<MediaCodecInfo> info = infos[i]; + AString name = info->getCodecName(); + if (name.startsWith("OMX.google.") || + // TODO: reenable below codecs once fixed + name == "OMX.Intel.VideoDecoder.VP9.hybrid") { + continue; + } + + Vector<AString> mimes; + info->getSupportedMimes(&mimes); + for (size_t i = 0; i < mimes.size(); ++i) { + const sp<MediaCodecInfo::Capabilities> &caps = + info->getCapabilitiesFor(mimes[i].c_str()); + if (!forceToMeasure && caps->getDetails()->contains("max-supported-instances")) { + continue; + } + + size_t max = doProfileCodecs(info->isEncoder(), name, mimes[i], caps); + if (max > 0) { + CodecSettings settings; + char maxStr[32]; + sprintf(maxStr, "%u", max); + settings.add("max-supported-instances", maxStr); + + AString key = name; + key.append(" "); + key.append(mimes[i]); + key.append(" "); + key.append(info->isEncoder() ? "encoder" : "decoder"); + results->add(key, settings); + } + } + } +} + +void applyCodecSettings( + const AString& codecInfo, + const CodecSettings &settings, + Vector<sp<MediaCodecInfo>> *infos) { + AString name; + AString mime; + AString type; + if (!splitString(codecInfo, " ", &name, &mime, &type)) { + return; + } + + for (size_t i = 0; i < infos->size(); ++i) { + const sp<MediaCodecInfo> &info = infos->itemAt(i); + if (name != info->getCodecName()) { + continue; + } + + Vector<AString> mimes; + info->getSupportedMimes(&mimes); + for (size_t j = 0; j < mimes.size(); ++j) { + if (mimes[j] != mime) { + continue; + } + const sp<MediaCodecInfo::Capabilities> &caps = info->getCapabilitiesFor(mime.c_str()); + for (size_t k = 0; k < settings.size(); ++k) { + caps->getDetails()->setString( + settings.keyAt(k).c_str(), settings.valueAt(k).c_str()); + } + } + } +} + +void exportResultsToXML(const char *fileName, const KeyedVector<AString, CodecSettings>& results) { +#if LOG_NDEBUG == 0 + ALOGE("measurement results"); + for (size_t i = 0; i < results.size(); ++i) { + ALOGE("key %s", results.keyAt(i).c_str()); + const CodecSettings &settings = results.valueAt(i); + for (size_t j = 0; j < settings.size(); ++j) { + ALOGE("name %s value %s", settings.keyAt(j).c_str(), settings.valueAt(j).c_str()); + } + } +#endif + + AString overrides; + FILE *f = fopen(fileName, "rb"); + if (f != NULL) { + fseek(f, 0, SEEK_END); + long size = ftell(f); + rewind(f); + + char *buf = (char *)malloc(size); + if (fread(buf, size, 1, f) == 1) { + overrides.setTo(buf, size); +#if LOG_NDEBUG == 0 + ALOGV("Existing overrides:"); + printLongString(buf, size); +#endif + } else { + ALOGE("Failed to read %s", fileName); + } + fclose(f); + free(buf); + } + + for (size_t i = 0; i < results.size(); ++i) { + AString name; + AString mime; + AString type; + if (!splitString(results.keyAt(i), " ", &name, &mime, &type)) { + continue; + } + name = AStringPrintf("\"%s\"", name.c_str()); + mime = AStringPrintf("\"%s\"", mime.c_str()); + ALOGV("name(%s) mime(%s) type(%s)", name.c_str(), mime.c_str(), type.c_str()); + ssize_t posCodec = overrides.find(name.c_str()); + size_t posInsert = 0; + if (posCodec < 0) { + AString encodersDecoders = (type == "encoder") ? "<Encoders>" : "<Decoders>"; + AString encodersDecodersEnd = (type == "encoder") ? "</Encoders>" : "</Decoders>"; + ssize_t posEncodersDecoders = overrides.find(encodersDecoders.c_str()); + if (posEncodersDecoders < 0) { + AString mediaCodecs = "<MediaCodecs>"; + ssize_t posMediaCodec = overrides.find(mediaCodecs.c_str()); + if (posMediaCodec < 0) { + posMediaCodec = overrides.size(); + overrides.insert("\n<MediaCodecs>\n</MediaCodecs>\n", posMediaCodec); + posMediaCodec = overrides.find(mediaCodecs.c_str(), posMediaCodec); + } + posEncodersDecoders = posMediaCodec + mediaCodecs.size(); + AString codecs = AStringPrintf( + "\n %s\n %s", encodersDecoders.c_str(), encodersDecodersEnd.c_str()); + overrides.insert(codecs.c_str(), posEncodersDecoders); + posEncodersDecoders = overrides.find(encodersDecoders.c_str(), posEncodersDecoders); + } + posCodec = posEncodersDecoders + encodersDecoders.size(); + AString codec = AStringPrintf( + "\n <MediaCodec name=%s type=%s update=\"true\" >\n </MediaCodec>", + name.c_str(), + mime.c_str()); + overrides.insert(codec.c_str(), posCodec); + posCodec = overrides.find(name.c_str()); + } + + // insert to existing entry + ssize_t posMime = overrides.find(mime.c_str(), posCodec); + ssize_t posEnd = overrides.find(">", posCodec); + if (posEnd < 0) { + ALOGE("Format error in overrides file."); + return; + } + if (posMime < 0 || posMime > posEnd) { + // new mime for an existing component + AString codecEnd = "</MediaCodec>"; + posInsert = overrides.find(codecEnd.c_str(), posCodec) + codecEnd.size(); + AString codec = AStringPrintf( + "\n <MediaCodec name=%s type=%s update=\"true\" >\n </MediaCodec>", + name.c_str(), + mime.c_str()); + overrides.insert(codec.c_str(), posInsert); + posInsert = overrides.find(">", posInsert) + 1; + } else { + posInsert = posEnd + 1; + } + + CodecSettings settings = results.valueAt(i); + for (size_t i = 0; i < settings.size(); ++i) { + // WARNING: we assume all the settings are "Limit". Currently we have only one type + // of setting in this case, which is "max-supported-instances". + AString strInsert = AStringPrintf( + "\n <Limit name=\"%s\" value=\"%s\" />", + settings.keyAt(i).c_str(), + settings.valueAt(i).c_str()); + overrides.insert(strInsert, posInsert); + } + } + +#if LOG_NDEBUG == 0 + ALOGV("New overrides:"); + printLongString(overrides.c_str(), overrides.size()); +#endif + + f = fopen(fileName, "wb"); + if (f == NULL) { + ALOGE("Failed to open %s for writing.", fileName); + return; + } + if (fwrite(overrides.c_str(), 1, overrides.size(), f) != overrides.size()) { + ALOGE("Failed to write to %s.", fileName); + } + fclose(f); +} + +} // namespace android diff --git a/media/libstagefright/MediaCodecListOverrides.h b/media/libstagefright/MediaCodecListOverrides.h new file mode 100644 index 0000000..f97ce63 --- /dev/null +++ b/media/libstagefright/MediaCodecListOverrides.h @@ -0,0 +1,50 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef MEDIA_CODEC_LIST_OVERRIDES_H_ + +#define MEDIA_CODEC_LIST_OVERRIDES_H_ + +#include <media/MediaCodecInfo.h> +#include <media/stagefright/foundation/AString.h> + +#include <utils/StrongPointer.h> +#include <utils/KeyedVector.h> + +namespace android { + +class MediaCodecInfo; + +bool splitString(const AString &s, const AString &delimiter, AString *s1, AString *s2); + +bool splitString( + const AString &s, const AString &delimiter, AString *s1, AString *s2, AString *s3); + +void profileCodecs( + const Vector<sp<MediaCodecInfo>> &infos, + KeyedVector<AString, CodecSettings> *results, + bool forceToMeasure = false); // forceToMeasure is mainly for testing + +void applyCodecSettings( + const AString& codecInfo, + const CodecSettings &settings, + Vector<sp<MediaCodecInfo>> *infos); + +void exportResultsToXML(const char *fileName, const KeyedVector<AString, CodecSettings>& results); + +} // namespace android + +#endif // MEDIA_CODEC_LIST_OVERRIDES_H_ diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp index b6fa810..6568d25 100644 --- a/media/libstagefright/MediaCodecSource.cpp +++ b/media/libstagefright/MediaCodecSource.cpp @@ -399,6 +399,9 @@ status_t MediaCodecSource::initEncoder() { ALOGV("output format is '%s'", mOutputFormat->debugString(0).c_str()); + mEncoderActivityNotify = new AMessage(kWhatEncoderActivity, mReflector); + mEncoder->setCallback(mEncoderActivityNotify); + status_t err = mEncoder->configure( mOutputFormat, NULL /* nativeWindow */, @@ -422,9 +425,6 @@ status_t MediaCodecSource::initEncoder() { } } - mEncoderActivityNotify = new AMessage(kWhatEncoderActivity, mReflector); - mEncoder->setCallback(mEncoderActivityNotify); - err = mEncoder->start(); if (err != OK) { diff --git a/media/libstagefright/SampleTable.cpp b/media/libstagefright/SampleTable.cpp index bdd6d56..aba64d5 100644 --- a/media/libstagefright/SampleTable.cpp +++ b/media/libstagefright/SampleTable.cpp @@ -230,8 +230,13 @@ status_t SampleTable::setSampleToChunkParams( return ERROR_MALFORMED; } + if (SIZE_MAX / sizeof(SampleToChunkEntry) <= mNumSampleToChunkOffsets) + return ERROR_OUT_OF_RANGE; + mSampleToChunkEntries = - new SampleToChunkEntry[mNumSampleToChunkOffsets]; + new (std::nothrow) SampleToChunkEntry[mNumSampleToChunkOffsets]; + if (!mSampleToChunkEntries) + return ERROR_OUT_OF_RANGE; for (uint32_t i = 0; i < mNumSampleToChunkOffsets; ++i) { uint8_t buffer[12]; @@ -330,11 +335,13 @@ status_t SampleTable::setTimeToSampleParams( } mTimeToSampleCount = U32_AT(&header[4]); - uint64_t allocSize = mTimeToSampleCount * 2 * sizeof(uint32_t); + uint64_t allocSize = mTimeToSampleCount * 2 * (uint64_t)sizeof(uint32_t); if (allocSize > SIZE_MAX) { return ERROR_OUT_OF_RANGE; } - mTimeToSample = new uint32_t[mTimeToSampleCount * 2]; + mTimeToSample = new (std::nothrow) uint32_t[mTimeToSampleCount * 2]; + if (!mTimeToSample) + return ERROR_OUT_OF_RANGE; size_t size = sizeof(uint32_t) * mTimeToSampleCount * 2; if (mDataSource->readAt( @@ -376,12 +383,14 @@ status_t SampleTable::setCompositionTimeToSampleParams( } mNumCompositionTimeDeltaEntries = numEntries; - uint64_t allocSize = numEntries * 2 * sizeof(uint32_t); + uint64_t allocSize = numEntries * 2 * (uint64_t)sizeof(uint32_t); if (allocSize > SIZE_MAX) { return ERROR_OUT_OF_RANGE; } - mCompositionTimeDeltaEntries = new uint32_t[2 * numEntries]; + mCompositionTimeDeltaEntries = new (std::nothrow) uint32_t[2 * numEntries]; + if (!mCompositionTimeDeltaEntries) + return ERROR_OUT_OF_RANGE; if (mDataSource->readAt( data_offset + 8, mCompositionTimeDeltaEntries, numEntries * 8) @@ -426,12 +435,15 @@ status_t SampleTable::setSyncSampleParams(off64_t data_offset, size_t data_size) ALOGV("Table of sync samples is empty or has only a single entry!"); } - uint64_t allocSize = mNumSyncSamples * sizeof(uint32_t); + uint64_t allocSize = mNumSyncSamples * (uint64_t)sizeof(uint32_t); if (allocSize > SIZE_MAX) { return ERROR_OUT_OF_RANGE; } - mSyncSamples = new uint32_t[mNumSyncSamples]; + mSyncSamples = new (std::nothrow) uint32_t[mNumSyncSamples]; + if (!mSyncSamples) + return ERROR_OUT_OF_RANGE; + size_t size = mNumSyncSamples * sizeof(uint32_t); if (mDataSource->readAt(mSyncSampleOffset + 8, mSyncSamples, size) != (ssize_t)size) { @@ -499,7 +511,9 @@ void SampleTable::buildSampleEntriesTable() { return; } - mSampleTimeEntries = new SampleTimeEntry[mNumSampleSizes]; + mSampleTimeEntries = new (std::nothrow) SampleTimeEntry[mNumSampleSizes]; + if (!mSampleTimeEntries) + return; uint32_t sampleIndex = 0; uint32_t sampleTime = 0; diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp index 6e6a78a..a35909e 100644 --- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp +++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp @@ -139,7 +139,7 @@ bool SoftVPX::outputBuffers(bool flushDecoder, bool display, bool eos, bool *por uint32_t height = mImg->d_h; outInfo = *outQueue.begin(); outHeader = outInfo->mHeader; - CHECK_EQ(mImg->fmt, IMG_FMT_I420); + CHECK_EQ(mImg->fmt, VPX_IMG_FMT_I420); handlePortSettingsChange(portWillReset, width, height); if (*portWillReset) { return true; @@ -151,12 +151,12 @@ bool SoftVPX::outputBuffers(bool flushDecoder, bool display, bool eos, bool *por outHeader->nTimeStamp = *(OMX_TICKS *)mImg->user_priv; uint8_t *dst = outHeader->pBuffer; - const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y]; - const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U]; - const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V]; - size_t srcYStride = mImg->stride[PLANE_Y]; - size_t srcUStride = mImg->stride[PLANE_U]; - size_t srcVStride = mImg->stride[PLANE_V]; + const uint8_t *srcY = (const uint8_t *)mImg->planes[VPX_PLANE_Y]; + const uint8_t *srcU = (const uint8_t *)mImg->planes[VPX_PLANE_U]; + const uint8_t *srcV = (const uint8_t *)mImg->planes[VPX_PLANE_V]; + size_t srcYStride = mImg->stride[VPX_PLANE_Y]; + size_t srcUStride = mImg->stride[VPX_PLANE_U]; + size_t srcVStride = mImg->stride[VPX_PLANE_V]; copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride); mImg = NULL; diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index 2d93152..74f58e9 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -141,6 +141,21 @@ const char *LiveSession::getKeyForStream(StreamType type) { return NULL; } +//static +const char *LiveSession::getNameForStream(StreamType type) { + switch (type) { + case STREAMTYPE_VIDEO: + return "video"; + case STREAMTYPE_AUDIO: + return "audio"; + case STREAMTYPE_SUBTITLES: + return "subs"; + default: + break; + } + return "unknown"; +} + LiveSession::LiveSession( const sp<AMessage> ¬ify, uint32_t flags, const sp<IMediaHTTPService> &httpService) @@ -192,7 +207,11 @@ status_t LiveSession::dequeueAccessUnit( status_t finalResult = OK; sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream); - ssize_t idx = typeToIndex(stream); + ssize_t streamIdx = typeToIndex(stream); + if (streamIdx < 0) { + return INVALID_VALUE; + } + const char *streamStr = getNameForStream(stream); // Do not let client pull data if we don't have data packets yet. // We might only have a format discontinuity queued without data. // When NuPlayerDecoder dequeues the format discontinuity, it will @@ -200,6 +219,9 @@ status_t LiveSession::dequeueAccessUnit( // thinks it can do seamless change, so will not shutdown decoder. // When the actual format arrives, it can't handle it and get stuck. if (!packetSource->hasDataBufferAvailable(&finalResult)) { + ALOGV("[%s] dequeueAccessUnit: no buffer available (finalResult=%d)", + streamStr, finalResult); + if (finalResult == OK) { return -EAGAIN; } else { @@ -212,25 +234,6 @@ status_t LiveSession::dequeueAccessUnit( status_t err = packetSource->dequeueAccessUnit(accessUnit); - size_t streamIdx; - const char *streamStr; - switch (stream) { - case STREAMTYPE_AUDIO: - streamIdx = kAudioIndex; - streamStr = "audio"; - break; - case STREAMTYPE_VIDEO: - streamIdx = kVideoIndex; - streamStr = "video"; - break; - case STREAMTYPE_SUBTITLES: - streamIdx = kSubtitleIndex; - streamStr = "subs"; - break; - default: - TRESPASS(); - } - StreamItem& strm = mStreams[streamIdx]; if (err == INFO_DISCONTINUITY) { // adaptive streaming, discontinuities in the playlist @@ -249,9 +252,10 @@ status_t LiveSession::dequeueAccessUnit( } else if (err == OK) { if (stream == STREAMTYPE_AUDIO || stream == STREAMTYPE_VIDEO) { - int64_t timeUs; + int64_t timeUs, originalTimeUs; int32_t discontinuitySeq = 0; CHECK((*accessUnit)->meta()->findInt64("timeUs", &timeUs)); + originalTimeUs = timeUs; (*accessUnit)->meta()->findInt32("discontinuitySeq", &discontinuitySeq); if (discontinuitySeq > (int32_t) strm.mCurDiscontinuitySeq) { int64_t offsetTimeUs; @@ -303,7 +307,8 @@ status_t LiveSession::dequeueAccessUnit( timeUs += mDiscontinuityOffsetTimesUs.valueFor(discontinuitySeq); } - ALOGV("[%s] read buffer at time %" PRId64 " us", streamStr, timeUs); + ALOGV("[%s] dequeueAccessUnit: time %lld us, original %lld us", + streamStr, (long long)timeUs, (long long)originalTimeUs); (*accessUnit)->meta()->setInt64("timeUs", timeUs); mLastDequeuedTimeUs = timeUs; mRealTimeBaseUs = ALooper::GetNowUs() - timeUs; @@ -409,7 +414,7 @@ bool LiveSession::checkSwitchProgress( if (lastDequeueMeta == NULL) { // this means we don't have enough cushion, try again later ALOGV("[%s] up switching failed due to insufficient buffer", - stream == STREAMTYPE_AUDIO ? "audio" : "video"); + getNameForStream(stream)); return false; } } else { @@ -428,7 +433,7 @@ bool LiveSession::checkSwitchProgress( if (firstNewMeta[i] == NULL) { HLSTime dequeueTime(lastDequeueMeta); ALOGV("[%s] dequeue time (%d, %lld) past start time", - stream == STREAMTYPE_AUDIO ? "audio" : "video", + getNameForStream(stream), dequeueTime.mSeq, (long long) dequeueTime.mTimeUs); return false; } @@ -493,16 +498,15 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { case kWhatSeek: { - sp<AReplyToken> seekReplyID; - CHECK(msg->senderAwaitsResponse(&seekReplyID)); - mSeekReplyID = seekReplyID; - mSeekReply = new AMessage; - - status_t err = onSeek(msg); - - if (err != OK) { + if (mReconfigurationInProgress) { msg->post(50000); + break; } + + CHECK(msg->senderAwaitsResponse(&mSeekReplyID)); + mSeekReply = new AMessage; + + onSeek(msg); break; } @@ -525,6 +529,11 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { break; } + ALOGV("fetcher-%d %s", + mFetcherInfos[index].mFetcher->getFetcherID(), + what == PlaylistFetcher::kWhatPaused ? + "paused" : "stopped"); + if (what == PlaylistFetcher::kWhatStopped) { mFetcherLooper->unregisterHandler( mFetcherInfos[index].mFetcher->id()); @@ -544,6 +553,7 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { if (--mContinuationCounter == 0) { mContinuation->post(); } + ALOGV("%zu fetcher(s) left", mContinuationCounter); } break; } @@ -636,6 +646,9 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { int32_t switchGeneration; CHECK(msg->findInt32("switchGeneration", &switchGeneration)); + ALOGV("kWhatStartedAt: switchGen=%d, mSwitchGen=%d", + switchGeneration, mSwitchGeneration); + if (switchGeneration != mSwitchGeneration) { break; } @@ -667,6 +680,7 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { if (checkSwitchProgress(stopParams, delayUs, &needResumeUntil)) { // playback time hasn't passed startAt time if (!needResumeUntil) { + ALOGV("finish switch"); for (size_t i = 0; i < kMaxStreams; ++i) { if ((mSwapMask & indexToType(i)) && uri == mStreams[i].mNewUri) { @@ -682,6 +696,7 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { // Resume fetcher for the original variant; the resumed fetcher should // continue until the timestamps found in msg, which is stored by the // new fetcher to indicate where the new variant has started buffering. + ALOGV("finish switch with resumeUntilAsync"); for (size_t i = 0; i < mFetcherInfos.size(); i++) { const FetcherInfo &info = mFetcherInfos.valueAt(i); if (info.mToBeRemoved) { @@ -693,8 +708,10 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { // playback time passed startAt time if (switchUp) { // if switching up, cancel and retry if condition satisfies again + ALOGV("cancel up switch because we're too late"); cancelBandwidthSwitch(true /* resume */); } else { + ALOGV("retry down switch at next sample"); resumeFetcher(uri, mSwapMask, -1, true /* newUri */); } } @@ -933,7 +950,8 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) { notify->setInt32("switchGeneration", mSwitchGeneration); FetcherInfo info; - info.mFetcher = new PlaylistFetcher(notify, this, uri, mSubtitleGeneration); + info.mFetcher = new PlaylistFetcher( + notify, this, uri, mCurBandwidthIndex, mSubtitleGeneration); info.mDurationUs = -1ll; info.mToBeRemoved = false; info.mToBeResumed = false; @@ -1167,9 +1185,13 @@ bool LiveSession::resumeFetcher( } if (resume) { - ALOGV("resuming fetcher %s, timeUs %lld", uri.c_str(), (long long)timeUs); + sp<PlaylistFetcher> &fetcher = mFetcherInfos.editValueAt(index).mFetcher; SeekMode seekMode = newUri ? kSeekModeNextSample : kSeekModeExactPosition; - mFetcherInfos.editValueAt(index).mFetcher->startAsync( + + ALOGV("resuming fetcher-%d, timeUs=%lld, seekMode=%d", + fetcher->getFetcherID(), (long long)timeUs, seekMode); + + fetcher->startAsync( sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex], @@ -1349,16 +1371,10 @@ HLSTime LiveSession::latestMediaSegmentStartTime() const { return audioTime < videoTime ? videoTime : audioTime; } -status_t LiveSession::onSeek(const sp<AMessage> &msg) { +void LiveSession::onSeek(const sp<AMessage> &msg) { int64_t timeUs; CHECK(msg->findInt64("timeUs", &timeUs)); - - if (!mReconfigurationInProgress) { - changeConfiguration(timeUs); - return OK; - } else { - return -EWOULDBLOCK; - } + changeConfiguration(timeUs); } status_t LiveSession::getDuration(int64_t *durationUs) const { @@ -1406,6 +1422,9 @@ status_t LiveSession::selectTrack(size_t index, bool select) { return INVALID_OPERATION; } + ALOGV("selectTrack: index=%zu, select=%d, mSubtitleGen=%d++", + index, select, mSubtitleGeneration); + ++mSubtitleGeneration; status_t err = mPlaylist->selectTrack(index, select); if (err == OK) { @@ -1426,6 +1445,9 @@ ssize_t LiveSession::getSelectedTrack(media_track_type type) const { void LiveSession::changeConfiguration( int64_t timeUs, ssize_t bandwidthIndex, bool pickTrack) { + ALOGV("changeConfiguration: timeUs=%lld us, bwIndex=%zd, pickTrack=%d", + (long long)timeUs, bandwidthIndex, pickTrack); + cancelBandwidthSwitch(); CHECK(!mReconfigurationInProgress); @@ -1433,6 +1455,10 @@ void LiveSession::changeConfiguration( if (bandwidthIndex >= 0) { mOrigBandwidthIndex = mCurBandwidthIndex; mCurBandwidthIndex = bandwidthIndex; + if (mOrigBandwidthIndex != mCurBandwidthIndex) { + ALOGI("#### Starting Bandwidth Switch: %zd => %zd", + mOrigBandwidthIndex, mCurBandwidthIndex); + } } CHECK_LT(mCurBandwidthIndex, mBandwidthItems.size()); const BandwidthItem &item = mBandwidthItems.itemAt(mCurBandwidthIndex); @@ -1478,6 +1504,7 @@ void LiveSession::changeConfiguration( } if (discardFetcher) { + ALOGV("discarding fetcher-%d", fetcher->getFetcherID()); fetcher->stopAsync(); } else { float threshold = -1.0f; // always finish fetching by default @@ -1490,8 +1517,8 @@ void LiveSession::changeConfiguration( mOrigBandwidthIndex, mCurBandwidthIndex); } - ALOGV("Pausing with threshold %.3f", threshold); - + ALOGV("pausing fetcher-%d, threshold=%.2f", + fetcher->getFetcherID(), threshold); fetcher->pauseAsync(threshold); } } @@ -1526,6 +1553,8 @@ void LiveSession::changeConfiguration( } void LiveSession::onChangeConfiguration(const sp<AMessage> &msg) { + ALOGV("onChangeConfiguration"); + if (!mReconfigurationInProgress) { int32_t pickTrack = 0; msg->findInt32("pickTrack", &pickTrack); @@ -1536,6 +1565,8 @@ void LiveSession::onChangeConfiguration(const sp<AMessage> &msg) { } void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) { + ALOGV("onChangeConfiguration2"); + mContinuation.clear(); // All fetchers are either suspended or have been removed now. @@ -1547,6 +1578,7 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) { if (timeUs >= 0) { mLastSeekTimeUs = timeUs; + mLastDequeuedTimeUs = timeUs; for (size_t i = 0; i < mPacketSources.size(); i++) { mPacketSources.editValueAt(i)->clear(); @@ -1599,8 +1631,10 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) { ALOGV("stream %zu changed: oldURI %s, newURI %s", i, mStreams[i].mUri.c_str(), URIs[i].c_str()); sp<AnotherPacketSource> source = mPacketSources.valueFor(indexToType(i)); - source->queueDiscontinuity( - ATSParser::DISCONTINUITY_FORMATCHANGE, NULL, true); + if (source->getLatestDequeuedMeta() != NULL) { + source->queueDiscontinuity( + ATSParser::DISCONTINUITY_FORMATCHANGE, NULL, true); + } } // Determine which decoders to shutdown on the player side, // a decoder has to be shutdown if its streamtype was active @@ -1660,16 +1694,17 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { // and resume audio. mSwapMask = mNewStreamMask & mStreamMask & ~resumeMask; switching = (mSwapMask != 0); - if (!switching) { - ALOGV("#### Finishing Bandwidth Switch Early: %zd => %zd", - mOrigBandwidthIndex, mCurBandwidthIndex); - } } mRealTimeBaseUs = ALooper::GetNowUs() - mLastDequeuedTimeUs; } else { mRealTimeBaseUs = ALooper::GetNowUs() - timeUs; } + ALOGV("onChangeConfiguration3: timeUs=%lld, switching=%d, pickTrack=%d, " + "mStreamMask=0x%x, mNewStreamMask=0x%x, mSwapMask=0x%x", + (long long)timeUs, switching, pickTrack, + mStreamMask, mNewStreamMask, mSwapMask); + for (size_t i = 0; i < kMaxStreams; ++i) { if (streamMask & indexToType(i)) { if (switching) { @@ -1687,6 +1722,9 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { for (size_t i = 0; i < mFetcherInfos.size(); ++i) { const AString &uri = mFetcherInfos.keyAt(i); if (!resumeFetcher(uri, resumeMask, timeUs)) { + ALOGV("marking fetcher-%d to be removed", + mFetcherInfos[i].mFetcher->getFetcherID()); + mFetcherInfos.editValueAt(i).mToBeRemoved = true; } } @@ -1776,6 +1814,14 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { } } + ALOGV("[fetcher-%d] startAsync: startTimeUs %lld mLastSeekTimeUs %lld " + "segmentStartTimeUs %lld seekMode %d", + fetcher->getFetcherID(), + (long long)startTime.mTimeUs, + (long long)mLastSeekTimeUs, + (long long)startTime.getSegmentTimeUs(true /* midpoint */), + seekMode); + // Set the target segment start time to the middle point of the // segment where the last sample was. // This gives a better guess if segments of the two variants are not @@ -1795,22 +1841,28 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { // All fetchers have now been started, the configuration change // has completed. - ALOGV("XXX configuration change completed."); mReconfigurationInProgress = false; if (switching) { mSwitchInProgress = true; } else { mStreamMask = mNewStreamMask; - mOrigBandwidthIndex = mCurBandwidthIndex; + if (mOrigBandwidthIndex != mCurBandwidthIndex) { + ALOGV("#### Finished Bandwidth Switch Early: %zd => %zd", + mOrigBandwidthIndex, mCurBandwidthIndex); + mOrigBandwidthIndex = mCurBandwidthIndex; + } } + ALOGV("onChangeConfiguration3: mSwitchInProgress %d, mStreamMask 0x%x", + mSwitchInProgress, mStreamMask); + if (mDisconnectReplyID != NULL) { finishDisconnect(); } } void LiveSession::swapPacketSource(StreamType stream) { - ALOGV("swapPacketSource: stream = %d", stream); + ALOGV("[%s] swapPacketSource", getNameForStream(stream)); // transfer packets from source2 to source sp<AnotherPacketSource> &aps = mPacketSources.editValueFor(stream); @@ -1858,7 +1910,7 @@ void LiveSession::tryToFinishBandwidthSwitch(const AString &oldUri) { mFetcherInfos.editValueAt(index).mFetcher->stopAsync(false /* clear */); - ALOGV("tryToFinishBandwidthSwitch: mSwapMask=%x", mSwapMask); + ALOGV("tryToFinishBandwidthSwitch: mSwapMask=0x%x", mSwapMask); if (mSwapMask != 0) { return; } @@ -1925,11 +1977,19 @@ void LiveSession::onPollBuffering() { bool underflow, ready, down, up; if (checkBuffering(underflow, ready, down, up)) { - if (mInPreparationPhase && ready) { - postPrepared(OK); + if (mInPreparationPhase) { + // Allow down switch even if we're still preparing. + // + // Some streams have a high bandwidth index as default, + // when bandwidth is low, it takes a long time to buffer + // to ready mark, then it immediately pauses after start + // as we have to do a down switch. It's better experience + // to restart from a lower index, if we detect low bw. + if (!switchBandwidthIfNeeded(false /* up */, down) && ready) { + postPrepared(OK); + } } - // don't switch before we report prepared if (!mInPreparationPhase) { if (ready) { stopBufferingIfNecessary(); @@ -1937,8 +1997,7 @@ void LiveSession::onPollBuffering() { startBufferingIfNecessary(); } switchBandwidthIfNeeded(up, down); - } - + } } schedulePollBuffering(); @@ -1983,7 +2042,7 @@ void LiveSession::cancelBandwidthSwitch(bool resume) { } ALOGI("#### Canceled Bandwidth Switch: %zd => %zd", - mCurBandwidthIndex, mOrigBandwidthIndex); + mOrigBandwidthIndex, mCurBandwidthIndex); mSwitchGeneration++; mSwitchInProgress = false; @@ -2022,13 +2081,16 @@ bool LiveSession::checkBuffering( int64_t bufferedDurationUs = mPacketSources[i]->getEstimatedDurationUs(); - ALOGV("source[%zu]: buffered %lld us", i, (long long)bufferedDurationUs); + ALOGV("[%s] buffered %lld us", + getNameForStream(mPacketSources.keyAt(i)), + (long long)bufferedDurationUs); if (durationUs >= 0) { int32_t percent; if (mPacketSources[i]->isFinished(0 /* duration */)) { percent = 100; } else { - percent = (int32_t)(100.0 * (mLastDequeuedTimeUs + bufferedDurationUs) / durationUs); + percent = (int32_t)(100.0 * + (mLastDequeuedTimeUs + bufferedDurationUs) / durationUs); } if (minBufferPercent < 0 || percent < minBufferPercent) { minBufferPercent = percent; @@ -2111,10 +2173,14 @@ void LiveSession::notifyBufferingUpdate(int32_t percentage) { notify->post(); } -void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) { +/* + * returns true if a bandwidth switch is actually needed (and started), + * returns false otherwise + */ +bool LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) { // no need to check bandwidth if we only have 1 bandwidth settings if (mSwitchInProgress || mBandwidthItems.size() < 2) { - return; + return false; } int32_t bandwidthBps; @@ -2123,7 +2189,7 @@ void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) { mLastBandwidthBps = bandwidthBps; } else { ALOGV("no bandwidth estimate."); - return; + return false; } int32_t curBandwidth = mBandwidthItems.itemAt(mCurBandwidthIndex).mBandwidth; @@ -2142,16 +2208,16 @@ void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) { // bandwidthIndex is < mCurBandwidthIndex, as getBandwidthIndex() only uses 70% // of measured bw. In that case we don't want to do anything, since we have // both enough buffer and enough bw. - if (bandwidthIndex == mCurBandwidthIndex - || (canSwitchUp && bandwidthIndex < mCurBandwidthIndex) - || (canSwithDown && bandwidthIndex > mCurBandwidthIndex)) { - return; + if ((canSwitchUp && bandwidthIndex > mCurBandwidthIndex) + || (canSwithDown && bandwidthIndex < mCurBandwidthIndex)) { + // if not yet prepared, just restart again with new bw index. + // this is faster and playback experience is cleaner. + changeConfiguration( + mInPreparationPhase ? 0 : -1ll, bandwidthIndex); + return true; } - - ALOGI("#### Starting Bandwidth Switch: %zd => %zd", - mCurBandwidthIndex, bandwidthIndex); - changeConfiguration(-1, bandwidthIndex, false); } + return false; } void LiveSession::postError(status_t err) { diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index b5e31c9..9117bb1 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -91,6 +91,7 @@ struct LiveSession : public AHandler { bool hasDynamicDuration() const; static const char *getKeyForStream(StreamType type); + static const char *getNameForStream(StreamType type); enum { kWhatStreamsChanged, @@ -236,7 +237,7 @@ private: sp<PlaylistFetcher> addFetcher(const char *uri); void onConnect(const sp<AMessage> &msg); - status_t onSeek(const sp<AMessage> &msg); + void onSeek(const sp<AMessage> &msg); void onFinishDisconnect2(); // If given a non-zero block_size (default 0), it is used to cap the number of @@ -291,7 +292,7 @@ private: bool checkSwitchProgress( sp<AMessage> &msg, int64_t delayUs, bool *needResumeUntil); - void switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow); + bool switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow); void schedulePollBuffering(); void cancelPollBuffering(); diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 368612d..ce79cc2 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -45,6 +45,10 @@ #include <openssl/aes.h> #include <openssl/md5.h> +#define FLOGV(fmt, ...) ALOGV("[fetcher-%d] " fmt, mFetcherID, ##__VA_ARGS__) +#define FSLOGV(stream, fmt, ...) ALOGV("[fetcher-%d] [%s] " fmt, mFetcherID, \ + LiveSession::getNameForStream(stream), ##__VA_ARGS__) + namespace android { // static @@ -143,10 +147,12 @@ PlaylistFetcher::PlaylistFetcher( const sp<AMessage> ¬ify, const sp<LiveSession> &session, const char *uri, + int32_t id, int32_t subtitleGeneration) : mNotify(notify), mSession(session), mURI(uri), + mFetcherID(id), mStreamTypeMask(0), mStartTimeUs(-1ll), mSegmentStartTimeUs(-1ll), @@ -176,6 +182,10 @@ PlaylistFetcher::PlaylistFetcher( PlaylistFetcher::~PlaylistFetcher() { } +int32_t PlaylistFetcher::getFetcherID() const { + return mFetcherID; +} + int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const { CHECK(mPlaylist != NULL); @@ -436,7 +446,7 @@ void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) { maxDelayUs = minDelayUs; } if (delayUs > maxDelayUs) { - ALOGV("Need to refresh playlist in %" PRId64 , maxDelayUs); + FLOGV("Need to refresh playlist in %lld", (long long)maxDelayUs); delayUs = maxDelayUs; } sp<AMessage> msg = new AMessage(kWhatMonitorQueue, this); @@ -507,6 +517,8 @@ void PlaylistFetcher::stopAsync(bool clear) { } void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> ¶ms) { + FLOGV("resumeUntilAsync: params=%s", params->debugString().c_str()); + AMessage* msg = new AMessage(kWhatResumeUntil, this); msg->setMessage("params", params); msg->post(); @@ -763,8 +775,9 @@ void PlaylistFetcher::onMonitorQueue() { int64_t bufferedStreamDurationUs = mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); - ALOGV("buffered %" PRId64 " for stream %d", - bufferedStreamDurationUs, mPacketSources.keyAt(i)); + + FSLOGV(mPacketSources.keyAt(i), "buffered %lld", (long long)bufferedStreamDurationUs); + if (bufferedDurationUs == -1ll || bufferedStreamDurationUs < bufferedDurationUs) { bufferedDurationUs = bufferedStreamDurationUs; @@ -776,8 +789,9 @@ void PlaylistFetcher::onMonitorQueue() { } if (finalResult == OK && bufferedDurationUs < kMinBufferedDurationUs) { - ALOGV("monitoring, buffered=%" PRId64 " < %" PRId64 "", - bufferedDurationUs, kMinBufferedDurationUs); + FLOGV("monitoring, buffered=%lld < %lld", + (long long)bufferedDurationUs, (long long)kMinBufferedDurationUs); + // delay the next download slightly; hopefully this gives other concurrent fetchers // a better chance to run. // onDownloadNext(); @@ -792,8 +806,12 @@ void PlaylistFetcher::onMonitorQueue() { if (delayUs > targetDurationUs / 2) { delayUs = targetDurationUs / 2; } - ALOGV("pausing for %" PRId64 ", buffered=%" PRId64 " > %" PRId64 "", - delayUs, bufferedDurationUs, kMinBufferedDurationUs); + + FLOGV("pausing for %lld, buffered=%lld > %lld", + (long long)delayUs, + (long long)bufferedDurationUs, + (long long)kMinBufferedDurationUs); + postMonitorQueue(delayUs); } } @@ -891,6 +909,12 @@ bool PlaylistFetcher::shouldPauseDownload() { } } lastEnqueueUs -= mSegmentFirstPTS; + + FLOGV("%spausing now, thresholdUs %lld, remaining %lld", + targetDurationUs - lastEnqueueUs > thresholdUs ? "" : "not ", + (long long)thresholdUs, + (long long)(targetDurationUs - lastEnqueueUs)); + if (targetDurationUs - lastEnqueueUs > thresholdUs) { return true; } @@ -940,8 +964,8 @@ bool PlaylistFetcher::initDownloadState( mStartTimeUs -= getSegmentStartTimeUs(mSeqNumber); } mStartTimeUsRelative = true; - ALOGV("Initial sequence number for time %" PRId64 " is %d from (%d .. %d)", - mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist, + FLOGV("Initial sequence number for time %lld is %d from (%d .. %d)", + (long long)mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist); } else { // When adapting or track switching, mSegmentStartTimeUs (relative @@ -966,7 +990,7 @@ bool PlaylistFetcher::initDownloadState( if (mSeqNumber > lastSeqNumberInPlaylist) { mSeqNumber = lastSeqNumberInPlaylist; } - ALOGV("Initial sequence number for live event %d from (%d .. %d)", + FLOGV("Initial sequence number is %d from (%d .. %d)", mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist); } @@ -995,10 +1019,10 @@ bool PlaylistFetcher::initDownloadState( if (delayUs > kMaxMonitorDelayUs) { delayUs = kMaxMonitorDelayUs; } - ALOGV("sequence number high: %d from (%d .. %d), " - "monitor in %" PRId64 " (retry=%d)", + FLOGV("sequence number high: %d from (%d .. %d), " + "monitor in %lld (retry=%d)", mSeqNumber, firstSeqNumberInPlaylist, - lastSeqNumberInPlaylist, delayUs, mNumRetries); + lastSeqNumberInPlaylist, (long long)delayUs, mNumRetries); postMonitorQueue(delayUs); return false; } @@ -1067,9 +1091,9 @@ bool PlaylistFetcher::initDownloadState( // Seek jumped to a new discontinuity sequence. We need to signal // a format change to decoder. Decoder needs to shutdown and be // created again if seamless format change is unsupported. - ALOGV("saw discontinuity: mStartup %d, mLastDiscontinuitySeq %d, " + FLOGV("saw discontinuity: mStartup %d, mLastDiscontinuitySeq %d, " "mDiscontinuitySeq %d, mStartTimeUs %lld", - mStartup, mLastDiscontinuitySeq, mDiscontinuitySeq, (long long)mStartTimeUs); + mStartup, mLastDiscontinuitySeq, mDiscontinuitySeq, (long long)mStartTimeUs); discontinuity = true; } mLastDiscontinuitySeq = -1; @@ -1134,7 +1158,7 @@ bool PlaylistFetcher::initDownloadState( } } - ALOGV("fetching segment %d from (%d .. %d)", + FLOGV("fetching segment %d from (%d .. %d)", mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist); return true; } @@ -1157,7 +1181,7 @@ void PlaylistFetcher::onDownloadNext() { firstSeqNumberInPlaylist, lastSeqNumberInPlaylist); connectHTTP = false; - ALOGV("resuming: '%s'", uri.c_str()); + FLOGV("resuming: '%s'", uri.c_str()); } else { if (!initDownloadState( uri, @@ -1166,7 +1190,7 @@ void PlaylistFetcher::onDownloadNext() { lastSeqNumberInPlaylist)) { return; } - ALOGV("fetching: '%s'", uri.c_str()); + FLOGV("fetching: '%s'", uri.c_str()); } int64_t range_offset, range_length; @@ -1196,6 +1220,11 @@ void PlaylistFetcher::onDownloadNext() { | LiveSession::STREAMTYPE_VIDEO))) { int64_t delayUs = ALooper::GetNowUs() - startUs; mSession->addBandwidthMeasurement(bytesRead, delayUs); + + if (delayUs > 2000000ll) { + FLOGV("bytesRead %zd took %.2f seconds - abnormal bandwidth dip", + bytesRead, (double)delayUs / 1.0e6); + } } connectHTTP = false; @@ -1584,6 +1613,16 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu // (newSeqNumber), start at least 1 segment prior. int32_t newSeqNumber = getSeqNumberWithAnchorTime( timeUs, targetDiffUs); + + FLOGV("guessed wrong seq number: timeUs=%lld, mStartTimeUs=%lld, " + "targetDurationUs=%lld, mSeqNumber=%d, newSeq=%d, firstSeq=%d", + (long long)timeUs, + (long long)mStartTimeUs, + (long long)targetDurationUs, + mSeqNumber, + newSeqNumber, + firstSeqNumberInPlaylist); + if (newSeqNumber >= mSeqNumber) { --mSeqNumber; } else { @@ -1604,8 +1643,13 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu } bool startTimeReached = true; if (mStartTimeUsRelative) { + FLOGV("startTimeUsRelative, timeUs (%lld) - %lld = %lld", + (long long)timeUs, + (long long)mFirstTimeUs, + (long long)(timeUs - mFirstTimeUs)); timeUs -= mFirstTimeUs; if (timeUs < 0) { + FLOGV("clamp negative timeUs to 0"); timeUs = 0; } startTimeReached = (timeUs >= mStartTimeUs); @@ -1614,13 +1658,17 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu if (!startTimeReached || (isAvc && !mIDRFound)) { // buffer up to the closest preceding IDR frame in the next segement, // or the closest succeeding IDR frame after the exact position + FSLOGV(stream, "timeUs=%lld, mStartTimeUs=%lld, mIDRFound=%d", + (long long)timeUs, (long long)mStartTimeUs, mIDRFound); if (isAvc) { if (IsIDR(accessUnit)) { mVideoBuffer->clear(); + FSLOGV(stream, "found IDR, clear mVideoBuffer"); mIDRFound = true; } if (mIDRFound && mStartTimeUsRelative && !startTimeReached) { mVideoBuffer->queueAccessUnit(accessUnit); + FSLOGV(stream, "saving AVC video AccessUnit"); } } if (!startTimeReached || (isAvc && !mIDRFound)) { @@ -1635,15 +1683,17 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu if (!(streamMask & mPacketSources.keyAt(i))) { streamMask |= mPacketSources.keyAt(i); mStartTimeUsNotify->setInt32("streamMask", streamMask); + FSLOGV(stream, "found start point, timeUs=%lld, streamMask becomes %x", + (long long)timeUs, streamMask); if (streamMask == mStreamTypeMask) { + FLOGV("found start point for all streams"); mStartup = false; } } } if (mStopParams != NULL) { - // Queue discontinuity in original stream. int32_t discontinuitySeq; int64_t stopTimeUs; if (!mStopParams->findInt32("discontinuitySeq", &discontinuitySeq) @@ -1651,13 +1701,13 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu || !mStopParams->findInt64(key, &stopTimeUs) || (discontinuitySeq == mDiscontinuitySeq && timeUs >= stopTimeUs)) { + FSLOGV(stream, "reached stop point, timeUs=%lld", (long long)timeUs); mStreamTypeMask &= ~stream; mPacketSources.removeItemsAt(i); break; } } - // Note that we do NOT dequeue any discontinuities except for format change. if (stream == LiveSession::STREAMTYPE_VIDEO) { const bool discard = true; status_t status; @@ -1666,11 +1716,16 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu mVideoBuffer->dequeueAccessUnit(&videoBuffer); setAccessUnitProperties(videoBuffer, source, discard); packetSource->queueAccessUnit(videoBuffer); + int64_t bufferTimeUs; + CHECK(videoBuffer->meta()->findInt64("timeUs", &bufferTimeUs)); + FSLOGV(stream, "queueAccessUnit (saved), timeUs=%lld", + (long long)bufferTimeUs); } } setAccessUnitProperties(accessUnit, source); packetSource->queueAccessUnit(accessUnit); + FSLOGV(stream, "queueAccessUnit, timeUs=%lld", (long long)timeUs); } if (err != OK) { @@ -1688,7 +1743,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu if (!mStreamTypeMask) { // Signal gap is filled between original and new stream. - ALOGV("ERROR OUT OF RANGE"); + FLOGV("reached stop point for all streams"); return ERROR_OUT_OF_RANGE; } @@ -1918,7 +1973,6 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( } if (mStopParams != NULL) { - // Queue discontinuity in original stream. int32_t discontinuitySeq; int64_t stopTimeUs; if (!mStopParams->findInt32("discontinuitySeq", &discontinuitySeq) diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index dab56df..f64d160 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -55,8 +55,11 @@ struct PlaylistFetcher : public AHandler { const sp<AMessage> ¬ify, const sp<LiveSession> &session, const char *uri, + int32_t id, int32_t subtitleGeneration); + int32_t getFetcherID() const; + sp<DataSource> getDataSource(); void startAsync( @@ -113,6 +116,8 @@ private: sp<LiveSession> mSession; AString mURI; + int32_t mFetcherID; + uint32_t mStreamTypeMask; int64_t mStartTimeUs; diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp index c5bb41b..c7912c0 100644 --- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp +++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp @@ -355,10 +355,15 @@ int64_t AnotherPacketSource::getBufferedDurationUs_l(status_t *finalResult) { int64_t time2 = -1; int64_t durationUs = 0; - List<sp<ABuffer> >::iterator it = mBuffers.begin(); - while (it != mBuffers.end()) { + List<sp<ABuffer> >::iterator it; + for (it = mBuffers.begin(); it != mBuffers.end(); it++) { const sp<ABuffer> &buffer = *it; + int32_t discard; + if (buffer->meta()->findInt32("discard", &discard) && discard) { + continue; + } + int64_t timeUs; if (buffer->meta()->findInt64("timeUs", &timeUs)) { if (time1 < 0 || timeUs < time1) { @@ -373,8 +378,6 @@ int64_t AnotherPacketSource::getBufferedDurationUs_l(status_t *finalResult) { durationUs += time2 - time1; time1 = time2 = -1; } - - ++it; } return durationUs + (time2 - time1); @@ -393,11 +396,19 @@ int64_t AnotherPacketSource::getEstimatedDurationUs() { return getBufferedDurationUs_l(&finalResult); } - List<sp<ABuffer> >::iterator it = mBuffers.begin(); - sp<ABuffer> buffer = *it; + sp<ABuffer> buffer; + int32_t discard; + int64_t startTimeUs = -1ll; + List<sp<ABuffer> >::iterator it; + for (it = mBuffers.begin(); it != mBuffers.end(); it++) { + buffer = *it; + if (buffer->meta()->findInt32("discard", &discard) && discard) { + continue; + } + buffer->meta()->findInt64("timeUs", &startTimeUs); + break; + } - int64_t startTimeUs; - buffer->meta()->findInt64("timeUs", &startTimeUs); if (startTimeUs < 0) { return 0; } @@ -514,7 +525,7 @@ void AnotherPacketSource::trimBuffersAfterMeta( } HLSTime stopTime(meta); - ALOGV("trimBuffersAfterMeta: discontinuitySeq %zu, timeUs %lld", + ALOGV("trimBuffersAfterMeta: discontinuitySeq %d, timeUs %lld", stopTime.mSeq, (long long)stopTime.mTimeUs); List<sp<ABuffer> >::iterator it; @@ -554,7 +565,7 @@ void AnotherPacketSource::trimBuffersAfterMeta( sp<AMessage> AnotherPacketSource::trimBuffersBeforeMeta( const sp<AMessage> &meta) { HLSTime startTime(meta); - ALOGV("trimBuffersBeforeMeta: discontinuitySeq %zu, timeUs %lld", + ALOGV("trimBuffersBeforeMeta: discontinuitySeq %d, timeUs %lld", startTime.mSeq, (long long)startTime.mTimeUs); sp<AMessage> firstMeta; diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp index 1f43d6d..33cfd1d 100644 --- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp +++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp @@ -85,12 +85,6 @@ status_t MPEG2TSSource::read( MediaBuffer **out, const ReadOptions *options) { *out = NULL; - int64_t seekTimeUs; - ReadOptions::SeekMode seekMode; - if (mSeekable && options && options->getSeekTo(&seekTimeUs, &seekMode)) { - return ERROR_UNSUPPORTED; - } - status_t finalResult; while (!mImpl->hasBufferAvailable(&finalResult)) { if (finalResult != OK) { @@ -103,6 +97,17 @@ status_t MPEG2TSSource::read( } } + int64_t seekTimeUs; + ReadOptions::SeekMode seekMode; + if (mSeekable && options && options->getSeekTo(&seekTimeUs, &seekMode)) { + // A seek was requested, but we don't actually support seeking and so can only "seek" to + // the current position + int64_t nextBufTimeUs; + if (mImpl->nextBufferTime(&nextBufTimeUs) != OK || seekTimeUs != nextBufTimeUs) { + return ERROR_UNSUPPORTED; + } + } + return mImpl->read(out, options); } diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp index 9b6958a..3ab241a 100644 --- a/media/libstagefright/omx/SoftOMXPlugin.cpp +++ b/media/libstagefright/omx/SoftOMXPlugin.cpp @@ -85,7 +85,7 @@ OMX_ERRORTYPE SoftOMXPlugin::makeComponentInstance( void *libHandle = dlopen(libName.c_str(), RTLD_NOW); if (libHandle == NULL) { - ALOGE("unable to dlopen %s", libName.c_str()); + ALOGE("unable to dlopen %s: %s", libName.c_str(), dlerror()); return OMX_ErrorComponentNotFound; } diff --git a/media/libstagefright/tests/Android.mk b/media/libstagefright/tests/Android.mk index 8d6ff5b..51e1c78 100644 --- a/media/libstagefright/tests/Android.mk +++ b/media/libstagefright/tests/Android.mk @@ -62,6 +62,33 @@ LOCAL_C_INCLUDES := \ include $(BUILD_NATIVE_TEST) +include $(CLEAR_VARS) +LOCAL_ADDITIONAL_DEPENDENCIES := $(LOCAL_PATH)/Android.mk + +LOCAL_MODULE := MediaCodecListOverrides_test + +LOCAL_MODULE_TAGS := tests + +LOCAL_SRC_FILES := \ + MediaCodecListOverrides_test.cpp \ + +LOCAL_SHARED_LIBRARIES := \ + libmedia \ + libstagefright \ + libstagefright_foundation \ + libstagefright_omx \ + libutils \ + liblog + +LOCAL_C_INCLUDES := \ + frameworks/av/media/libstagefright \ + frameworks/av/media/libstagefright/include \ + frameworks/native/include/media/openmax \ + +LOCAL_32_BIT_ONLY := true + +include $(BUILD_NATIVE_TEST) + # Include subdirectory makefiles # ============================================================ diff --git a/media/libstagefright/tests/MediaCodecListOverrides_test.cpp b/media/libstagefright/tests/MediaCodecListOverrides_test.cpp new file mode 100644 index 0000000..cacaa84 --- /dev/null +++ b/media/libstagefright/tests/MediaCodecListOverrides_test.cpp @@ -0,0 +1,316 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +// #define LOG_NDEBUG 0 +#define LOG_TAG "MediaCodecListOverrides_test" +#include <utils/Log.h> + +#include <gtest/gtest.h> + +#include "MediaCodecListOverrides.h" + +#include <media/MediaCodecInfo.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/MediaCodecList.h> + +namespace android { + +static const char kTestOverridesStr[] = +"<MediaCodecs>\n" +" <Settings>\n" +" <Setting name=\"max-max-supported-instances\" value=\"8\" update=\"true\" />\n" +" </Settings>\n" +" <Encoders>\n" +" <MediaCodec name=\"OMX.qcom.video.encoder.mpeg4\" type=\"video/mp4v-es\" update=\"true\" >\n" +" <Quirk name=\"requires-allocate-on-input-ports\" />\n" +" <Limit name=\"bitrate\" range=\"1-20000000\" />\n" +" <Feature name=\"can-swap-width-height\" />\n" +" </MediaCodec>\n" +" </Encoders>\n" +" <Decoders>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.avc\" type=\"video/avc\" update=\"true\" >\n" +" <Quirk name=\"requires-allocate-on-input-ports\" />\n" +" <Limit name=\"size\" min=\"64x64\" max=\"1920x1088\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.mpeg2\" type=\"different_mime\" update=\"true\" >\n" +" </MediaCodec>\n" +" </Decoders>\n" +"</MediaCodecs>\n"; + +static const char kTestOverridesStrNew1[] = +"<MediaCodecs>\n" +" <Settings>\n" +" <Setting name=\"max-max-supported-instances\" value=\"8\" update=\"true\" />\n" +" </Settings>\n" +" <Encoders>\n" +" <MediaCodec name=\"OMX.qcom.video.encoder.avc\" type=\"video/avc\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"4\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.encoder.mpeg4\" type=\"video/mp4v-es\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"4\" />\n" +" <Quirk name=\"requires-allocate-on-input-ports\" />\n" +" <Limit name=\"bitrate\" range=\"1-20000000\" />\n" +" <Feature name=\"can-swap-width-height\" />\n" +" </MediaCodec>\n" +" </Encoders>\n" +" <Decoders>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.mpeg4\" type=\"video/mp4v-es\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"3\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.h263\" type=\"video/3gpp\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"4\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.avc.secure\" type=\"video/avc\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"1\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.avc\" type=\"video/avc\" update=\"true\" >\n" +" <Quirk name=\"requires-allocate-on-input-ports\" />\n" +" <Limit name=\"size\" min=\"64x64\" max=\"1920x1088\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.mpeg2\" type=\"different_mime\" update=\"true\" >\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.mpeg2\" type=\"video/mpeg2\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"3\" />\n" +" </MediaCodec>\n" +" </Decoders>\n" +"</MediaCodecs>\n"; + +static const char kTestOverridesStrNew2[] = +"\n" +"<MediaCodecs>\n" +" <Encoders>\n" +" <MediaCodec name=\"OMX.qcom.video.encoder.mpeg4\" type=\"video/mp4v-es\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"4\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.encoder.avc\" type=\"video/avc\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"4\" />\n" +" </MediaCodec>\n" +" </Encoders>\n" +" <Decoders>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.mpeg4\" type=\"video/mp4v-es\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"3\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.mpeg2\" type=\"video/mpeg2\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"3\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.h263\" type=\"video/3gpp\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"4\" />\n" +" </MediaCodec>\n" +" <MediaCodec name=\"OMX.qcom.video.decoder.avc.secure\" type=\"video/avc\" update=\"true\" >\n" +" <Limit name=\"max-supported-instances\" value=\"1\" />\n" +" </MediaCodec>\n" +" </Decoders>\n" +"</MediaCodecs>\n"; + +class MediaCodecListOverridesTest : public ::testing::Test { +public: + MediaCodecListOverridesTest() {} + + void verifyOverrides(const KeyedVector<AString, CodecSettings> &overrides) { + EXPECT_EQ(3u, overrides.size()); + + EXPECT_TRUE(overrides.keyAt(0) == "OMX.qcom.video.decoder.avc video/avc decoder"); + const CodecSettings &settings0 = overrides.valueAt(0); + EXPECT_EQ(1u, settings0.size()); + EXPECT_TRUE(settings0.keyAt(0) == "max-supported-instances"); + EXPECT_TRUE(settings0.valueAt(0) == "4"); + + EXPECT_TRUE(overrides.keyAt(1) == "OMX.qcom.video.encoder.avc video/avc encoder"); + const CodecSettings &settings1 = overrides.valueAt(1); + EXPECT_EQ(1u, settings1.size()); + EXPECT_TRUE(settings1.keyAt(0) == "max-supported-instances"); + EXPECT_TRUE(settings1.valueAt(0) == "3"); + + EXPECT_TRUE(overrides.keyAt(2) == "global"); + const CodecSettings &settings2 = overrides.valueAt(2); + EXPECT_EQ(3u, settings2.size()); + EXPECT_TRUE(settings2.keyAt(0) == "max-max-supported-instances"); + EXPECT_TRUE(settings2.valueAt(0) == "8"); + EXPECT_TRUE(settings2.keyAt(1) == "supports-multiple-secure-codecs"); + EXPECT_TRUE(settings2.valueAt(1) == "false"); + EXPECT_TRUE(settings2.keyAt(2) == "supports-secure-with-non-secure-codec"); + EXPECT_TRUE(settings2.valueAt(2) == "true"); + } + + void verifySetting(const sp<AMessage> &details, const char *name, const char *value) { + AString value1; + EXPECT_TRUE(details->findString(name, &value1)); + EXPECT_TRUE(value1 == value); + } + + void createTestInfos(Vector<sp<MediaCodecInfo>> *infos) { + const char *name = "OMX.qcom.video.decoder.avc"; + const bool encoder = false; + const char *mime = "video/avc"; + sp<MediaCodecInfo> info = new MediaCodecInfo(name, encoder, mime); + infos->push_back(info); + const sp<MediaCodecInfo::Capabilities> caps = info->getCapabilitiesFor(mime); + const sp<AMessage> details = caps->getDetails(); + details->setString("cap1", "value1"); + details->setString("max-max-supported-instances", "16"); + + info = new MediaCodecInfo("anothercodec", true, "anothermime"); + infos->push_back(info); + } + + void addMaxInstancesSetting( + const AString &key, + const AString &value, + KeyedVector<AString, CodecSettings> *results) { + CodecSettings settings; + settings.add("max-supported-instances", value); + results->add(key, settings); + } + + void exportTestResultsToXML(const char *fileName) { + KeyedVector<AString, CodecSettings> r; + addMaxInstancesSetting("OMX.qcom.video.decoder.avc.secure video/avc decoder", "1", &r); + addMaxInstancesSetting("OMX.qcom.video.decoder.h263 video/3gpp decoder", "4", &r); + addMaxInstancesSetting("OMX.qcom.video.decoder.mpeg2 video/mpeg2 decoder", "3", &r); + addMaxInstancesSetting("OMX.qcom.video.decoder.mpeg4 video/mp4v-es decoder", "3", &r); + addMaxInstancesSetting("OMX.qcom.video.encoder.avc video/avc encoder", "4", &r); + addMaxInstancesSetting("OMX.qcom.video.encoder.mpeg4 video/mp4v-es encoder", "4", &r); + + exportResultsToXML(fileName, r); + } +}; + +TEST_F(MediaCodecListOverridesTest, splitString) { + AString s = "abc123"; + AString delimiter = " "; + AString s1; + AString s2; + EXPECT_FALSE(splitString(s, delimiter, &s1, &s2)); + s = "abc 123"; + EXPECT_TRUE(splitString(s, delimiter, &s1, &s2)); + EXPECT_TRUE(s1 == "abc"); + EXPECT_TRUE(s2 == "123"); + + s = "abc123xyz"; + delimiter = ","; + AString s3; + EXPECT_FALSE(splitString(s, delimiter, &s1, &s2, &s3)); + s = "abc,123xyz"; + EXPECT_FALSE(splitString(s, delimiter, &s1, &s2, &s3)); + s = "abc,123,xyz"; + EXPECT_TRUE(splitString(s, delimiter, &s1, &s2, &s3)); + EXPECT_TRUE(s1 == "abc"); + EXPECT_TRUE(s2 == "123" ); + EXPECT_TRUE(s3 == "xyz"); +} + +// TODO: the codec component never returns OMX_EventCmdComplete in unit test. +TEST_F(MediaCodecListOverridesTest, DISABLED_profileCodecs) { + sp<IMediaCodecList> list = MediaCodecList::getInstance(); + Vector<sp<MediaCodecInfo>> infos; + for (size_t i = 0; i < list->countCodecs(); ++i) { + infos.push_back(list->getCodecInfo(i)); + } + KeyedVector<AString, CodecSettings> results; + profileCodecs(infos, &results, true /* forceToMeasure */); + EXPECT_LT(0u, results.size()); + for (size_t i = 0; i < results.size(); ++i) { + AString key = results.keyAt(i); + CodecSettings settings = results.valueAt(i); + EXPECT_EQ(1u, settings.size()); + EXPECT_TRUE(settings.keyAt(0) == "max-supported-instances"); + AString valueS = settings.valueAt(0); + int32_t value = strtol(valueS.c_str(), NULL, 10); + EXPECT_LT(0, value); + ALOGV("profileCodecs results %s %s", key.c_str(), valueS.c_str()); + } +} + +TEST_F(MediaCodecListOverridesTest, applyCodecSettings) { + AString codecInfo = "OMX.qcom.video.decoder.avc video/avc decoder"; + Vector<sp<MediaCodecInfo>> infos; + createTestInfos(&infos); + CodecSettings settings; + settings.add("max-supported-instances", "3"); + settings.add("max-max-supported-instances", "8"); + applyCodecSettings(codecInfo, settings, &infos); + + EXPECT_EQ(2u, infos.size()); + EXPECT_TRUE(AString(infos[0]->getCodecName()) == "OMX.qcom.video.decoder.avc"); + const sp<AMessage> details = infos[0]->getCapabilitiesFor("video/avc")->getDetails(); + verifySetting(details, "max-supported-instances", "3"); + verifySetting(details, "max-max-supported-instances", "8"); + + EXPECT_TRUE(AString(infos[1]->getCodecName()) == "anothercodec"); + EXPECT_EQ(0u, infos[1]->getCapabilitiesFor("anothermime")->getDetails()->countEntries()); +} + +TEST_F(MediaCodecListOverridesTest, exportResultsToExistingFile) { + const char *fileName = "/sdcard/mediacodec_list_overrides_test.xml"; + remove(fileName); + + FILE *f = fopen(fileName, "wb"); + if (f == NULL) { + ALOGW("Failed to open %s for writing.", fileName); + return; + } + EXPECT_EQ( + strlen(kTestOverridesStr), + fwrite(kTestOverridesStr, 1, strlen(kTestOverridesStr), f)); + fclose(f); + + exportTestResultsToXML(fileName); + + // verify + AString overrides; + f = fopen(fileName, "rb"); + ASSERT_TRUE(f != NULL); + fseek(f, 0, SEEK_END); + long size = ftell(f); + rewind(f); + + char *buf = (char *)malloc(size); + EXPECT_EQ(1, fread(buf, size, 1, f)); + overrides.setTo(buf, size); + fclose(f); + free(buf); + + EXPECT_TRUE(overrides == kTestOverridesStrNew1); + + remove(fileName); +} + +TEST_F(MediaCodecListOverridesTest, exportResultsToEmptyFile) { + const char *fileName = "/sdcard/mediacodec_list_overrides_test.xml"; + remove(fileName); + + exportTestResultsToXML(fileName); + + // verify + AString overrides; + FILE *f = fopen(fileName, "rb"); + ASSERT_TRUE(f != NULL); + fseek(f, 0, SEEK_END); + long size = ftell(f); + rewind(f); + + char *buf = (char *)malloc(size); + EXPECT_EQ(1, fread(buf, size, 1, f)); + overrides.setTo(buf, size); + fclose(f); + free(buf); + + EXPECT_TRUE(overrides == kTestOverridesStrNew2); + + remove(fileName); +} + +} // namespace android diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk index 0e2e48c..ba47172 100644 --- a/media/mediaserver/Android.mk +++ b/media/mediaserver/Android.mk @@ -45,7 +45,8 @@ LOCAL_C_INCLUDES := \ frameworks/av/services/mediaresourcemanager \ $(call include-path-for, audio-utils) \ frameworks/av/services/soundtrigger \ - frameworks/av/services/radio + frameworks/av/services/radio \ + external/sonic LOCAL_MODULE:= mediaserver LOCAL_32_BIT_ONLY := true diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk index fee2347..c359be5 100644 --- a/services/audioflinger/Android.mk +++ b/services/audioflinger/Android.mk @@ -44,12 +44,13 @@ LOCAL_SRC_FILES:= \ SpdifStreamOut.cpp \ Effects.cpp \ AudioMixer.cpp.arm \ - PatchPanel.cpp - -LOCAL_SRC_FILES += StateQueue.cpp + BufferProviders.cpp \ + PatchPanel.cpp \ + StateQueue.cpp LOCAL_C_INCLUDES := \ $(TOPDIR)frameworks/av/services/audiopolicy \ + $(TOPDIR)external/sonic \ $(call include-path-for, audio-effects) \ $(call include-path-for, audio-utils) @@ -68,7 +69,8 @@ LOCAL_SHARED_LIBRARIES := \ libhardware_legacy \ libeffects \ libpowermanager \ - libserviceutility + libserviceutility \ + libsonic LOCAL_STATIC_LIBRARIES := \ libscheduling_policy \ diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index f3206cb..5002099 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -45,6 +45,8 @@ #include "AudioFlinger.h" #include "ServiceUtilities.h" +#include <media/AudioResamplerPublic.h> + #include <media/EffectsFactoryApi.h> #include <audio_effects/effect_visualizer.h> #include <audio_effects/effect_ns.h> @@ -1140,19 +1142,46 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t form if (ret != NO_ERROR) { return 0; } + if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { + return 0; + } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; - audio_config_t config; - memset(&config, 0, sizeof(config)); - config.sample_rate = sampleRate; - config.channel_mask = channelMask; - config.format = format; + audio_config_t config, proposed; + memset(&proposed, 0, sizeof(proposed)); + proposed.sample_rate = sampleRate; + proposed.channel_mask = channelMask; + proposed.format = format; audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); - size_t size = dev->get_input_buffer_size(dev, &config); + size_t frames; + for (;;) { + // Note: config is currently a const parameter for get_input_buffer_size() + // but we use a copy from proposed in case config changes from the call. + config = proposed; + frames = dev->get_input_buffer_size(dev, &config); + if (frames != 0) { + break; // hal success, config is the result + } + // change one parameter of the configuration each iteration to a more "common" value + // to see if the device will support it. + if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { + proposed.format = AUDIO_FORMAT_PCM_16_BIT; + } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as + proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? + } else { + ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " + "format %#x, channelMask 0x%X", + sampleRate, format, channelMask); + break; // retries failed, break out of loop with frames == 0. + } + } mHardwareStatus = AUDIO_HW_IDLE; - return size; + if (frames > 0 && config.sample_rate != sampleRate) { + frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); + } + return frames; // may be converted to bytes at the Java level. } uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const @@ -1419,9 +1448,8 @@ sp<IAudioRecord> AudioFlinger::openRecord( goto Exit; } - // we don't yet support anything other than 16-bit PCM - if (!(audio_is_valid_format(format) && - audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { + // we don't yet support anything other than linear PCM + if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { ALOGE("openRecord() invalid format %#x", format); lStatus = BAD_VALUE; goto Exit; @@ -2002,11 +2030,11 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m status, address.string()); // If the input could not be opened with the requested parameters and we can handle the - // conversion internally, try to open again with the proposed parameters. The AudioFlinger can - // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. + // conversion internally, try to open again with the proposed parameters. if (status == BAD_VALUE && - config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT && - (halconfig.sample_rate <= 2 * config->sample_rate) && + audio_is_linear_pcm(config->format) && + audio_is_linear_pcm(halconfig.format) && + (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { // FIXME describe the change proposed by HAL (save old values so we can log them here) diff --git a/services/audioflinger/AudioHwDevice.cpp b/services/audioflinger/AudioHwDevice.cpp index 09d86ea..3191598 100644 --- a/services/audioflinger/AudioHwDevice.cpp +++ b/services/audioflinger/AudioHwDevice.cpp @@ -44,7 +44,7 @@ status_t AudioHwDevice::openOutputStream( AudioStreamOut *outputStream = new AudioStreamOut(this, flags); // Try to open the HAL first using the current format. - ALOGV("AudioHwDevice::openOutputStream(), try " + ALOGV("openOutputStream(), try " " sampleRate %d, Format %#x, " "channelMask %#x", config->sample_rate, @@ -59,7 +59,7 @@ status_t AudioHwDevice::openOutputStream( // FIXME Look at any modification to the config. // The HAL might modify the config to suggest a wrapped format. // Log this so we can see what the HALs are doing. - ALOGI("AudioHwDevice::openOutputStream(), HAL returned" + ALOGI("openOutputStream(), HAL returned" " sampleRate %d, Format %#x, " "channelMask %#x, status %d", config->sample_rate, @@ -72,16 +72,19 @@ status_t AudioHwDevice::openOutputStream( && ((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0); - // FIXME - Add isEncodingSupported() query to SPDIF wrapper then - // call it from here. if (wrapperNeeded) { - outputStream = new SpdifStreamOut(this, flags); - status = outputStream->open(handle, devices, &originalConfig, address); - if (status != NO_ERROR) { - ALOGE("ERROR - AudioHwDevice::openOutputStream(), SPDIF open returned %d", - status); - delete outputStream; - outputStream = NULL; + if (SPDIFEncoder::isFormatSupported(originalConfig.format)) { + outputStream = new SpdifStreamOut(this, flags, originalConfig.format); + status = outputStream->open(handle, devices, &originalConfig, address); + if (status != NO_ERROR) { + ALOGE("ERROR - openOutputStream(), SPDIF open returned %d", + status); + delete outputStream; + outputStream = NULL; + } + } else { + ALOGE("ERROR - openOutputStream(), SPDIFEncoder does not support format 0x%08x", + originalConfig.format); } } } diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index dddca02..c2c791f 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -38,9 +38,7 @@ #include <audio_utils/format.h> #include <common_time/local_clock.h> #include <common_time/cc_helper.h> - -#include <media/EffectsFactoryApi.h> -#include <audio_effects/effect_downmix.h> +#include <media/AudioResamplerPublic.h> #include "AudioMixerOps.h" #include "AudioMixer.h" @@ -91,323 +89,6 @@ T min(const T& a, const T& b) return a < b ? a : b; } -AudioMixer::CopyBufferProvider::CopyBufferProvider(size_t inputFrameSize, - size_t outputFrameSize, size_t bufferFrameCount) : - mInputFrameSize(inputFrameSize), - mOutputFrameSize(outputFrameSize), - mLocalBufferFrameCount(bufferFrameCount), - mLocalBufferData(NULL), - mConsumed(0) -{ - ALOGV("CopyBufferProvider(%p)(%zu, %zu, %zu)", this, - inputFrameSize, outputFrameSize, bufferFrameCount); - LOG_ALWAYS_FATAL_IF(inputFrameSize < outputFrameSize && bufferFrameCount == 0, - "Requires local buffer if inputFrameSize(%zu) < outputFrameSize(%zu)", - inputFrameSize, outputFrameSize); - if (mLocalBufferFrameCount) { - (void)posix_memalign(&mLocalBufferData, 32, mLocalBufferFrameCount * mOutputFrameSize); - } - mBuffer.frameCount = 0; -} - -AudioMixer::CopyBufferProvider::~CopyBufferProvider() -{ - ALOGV("~CopyBufferProvider(%p)", this); - if (mBuffer.frameCount != 0) { - mTrackBufferProvider->releaseBuffer(&mBuffer); - } - free(mLocalBufferData); -} - -status_t AudioMixer::CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer, - int64_t pts) -{ - //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)", - // this, pBuffer, pBuffer->frameCount, pts); - if (mLocalBufferFrameCount == 0) { - status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts); - if (res == OK) { - copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount); - } - return res; - } - if (mBuffer.frameCount == 0) { - mBuffer.frameCount = pBuffer->frameCount; - status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts); - // At one time an upstream buffer provider had - // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014. - // - // By API spec, if res != OK, then mBuffer.frameCount == 0. - // but there may be improper implementations. - ALOG_ASSERT(res == OK || mBuffer.frameCount == 0); - if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe. - pBuffer->raw = NULL; - pBuffer->frameCount = 0; - return res; - } - mConsumed = 0; - } - ALOG_ASSERT(mConsumed < mBuffer.frameCount); - size_t count = min(mLocalBufferFrameCount, mBuffer.frameCount - mConsumed); - count = min(count, pBuffer->frameCount); - pBuffer->raw = mLocalBufferData; - pBuffer->frameCount = count; - copyFrames(pBuffer->raw, (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize, - pBuffer->frameCount); - return OK; -} - -void AudioMixer::CopyBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) -{ - //ALOGV("CopyBufferProvider(%p)::releaseBuffer(%p(%zu))", - // this, pBuffer, pBuffer->frameCount); - if (mLocalBufferFrameCount == 0) { - mTrackBufferProvider->releaseBuffer(pBuffer); - return; - } - // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount"); - mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content - if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) { - mTrackBufferProvider->releaseBuffer(&mBuffer); - ALOG_ASSERT(mBuffer.frameCount == 0); - } - pBuffer->raw = NULL; - pBuffer->frameCount = 0; -} - -void AudioMixer::CopyBufferProvider::reset() -{ - if (mBuffer.frameCount != 0) { - mTrackBufferProvider->releaseBuffer(&mBuffer); - } - mConsumed = 0; -} - -AudioMixer::DownmixerBufferProvider::DownmixerBufferProvider( - audio_channel_mask_t inputChannelMask, - audio_channel_mask_t outputChannelMask, audio_format_t format, - uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) : - CopyBufferProvider( - audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask), - audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask), - bufferFrameCount) // set bufferFrameCount to 0 to do in-place -{ - ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d)", - this, inputChannelMask, outputChannelMask, format, - sampleRate, sessionId); - if (!sIsMultichannelCapable - || EffectCreate(&sDwnmFxDesc.uuid, - sessionId, - SESSION_ID_INVALID_AND_IGNORED, - &mDownmixHandle) != 0) { - ALOGE("DownmixerBufferProvider() error creating downmixer effect"); - mDownmixHandle = NULL; - return; - } - // channel input configuration will be overridden per-track - mDownmixConfig.inputCfg.channels = inputChannelMask; // FIXME: Should be bits - mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits - mDownmixConfig.inputCfg.format = format; - mDownmixConfig.outputCfg.format = format; - mDownmixConfig.inputCfg.samplingRate = sampleRate; - mDownmixConfig.outputCfg.samplingRate = sampleRate; - mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; - mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; - // input and output buffer provider, and frame count will not be used as the downmix effect - // process() function is called directly (see DownmixerBufferProvider::getNextBuffer()) - mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | - EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE; - mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask; - - int cmdStatus; - uint32_t replySize = sizeof(int); - - // Configure downmixer - status_t status = (*mDownmixHandle)->command(mDownmixHandle, - EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/, - &mDownmixConfig /*pCmdData*/, - &replySize, &cmdStatus /*pReplyData*/); - if (status != 0 || cmdStatus != 0) { - ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer", - status, cmdStatus); - EffectRelease(mDownmixHandle); - mDownmixHandle = NULL; - return; - } - - // Enable downmixer - replySize = sizeof(int); - status = (*mDownmixHandle)->command(mDownmixHandle, - EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/, - &replySize, &cmdStatus /*pReplyData*/); - if (status != 0 || cmdStatus != 0) { - ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer", - status, cmdStatus); - EffectRelease(mDownmixHandle); - mDownmixHandle = NULL; - return; - } - - // Set downmix type - // parameter size rounded for padding on 32bit boundary - const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int); - const int downmixParamSize = - sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t); - effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize); - param->psize = sizeof(downmix_params_t); - const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE; - memcpy(param->data, &downmixParam, param->psize); - const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD; - param->vsize = sizeof(downmix_type_t); - memcpy(param->data + psizePadded, &downmixType, param->vsize); - replySize = sizeof(int); - status = (*mDownmixHandle)->command(mDownmixHandle, - EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */, - param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/); - free(param); - if (status != 0 || cmdStatus != 0) { - ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type", - status, cmdStatus); - EffectRelease(mDownmixHandle); - mDownmixHandle = NULL; - return; - } - ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType); -} - -AudioMixer::DownmixerBufferProvider::~DownmixerBufferProvider() -{ - ALOGV("~DownmixerBufferProvider (%p)", this); - EffectRelease(mDownmixHandle); - mDownmixHandle = NULL; -} - -void AudioMixer::DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames) -{ - mDownmixConfig.inputCfg.buffer.frameCount = frames; - mDownmixConfig.inputCfg.buffer.raw = const_cast<void *>(src); - mDownmixConfig.outputCfg.buffer.frameCount = frames; - mDownmixConfig.outputCfg.buffer.raw = dst; - // may be in-place if src == dst. - status_t res = (*mDownmixHandle)->process(mDownmixHandle, - &mDownmixConfig.inputCfg.buffer, &mDownmixConfig.outputCfg.buffer); - ALOGE_IF(res != OK, "DownmixBufferProvider error %d", res); -} - -/* call once in a pthread_once handler. */ -/*static*/ status_t AudioMixer::DownmixerBufferProvider::init() -{ - // find multichannel downmix effect if we have to play multichannel content - uint32_t numEffects = 0; - int ret = EffectQueryNumberEffects(&numEffects); - if (ret != 0) { - ALOGE("AudioMixer() error %d querying number of effects", ret); - return NO_INIT; - } - ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects); - - for (uint32_t i = 0 ; i < numEffects ; i++) { - if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) { - ALOGV("effect %d is called %s", i, sDwnmFxDesc.name); - if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) { - ALOGI("found effect \"%s\" from %s", - sDwnmFxDesc.name, sDwnmFxDesc.implementor); - sIsMultichannelCapable = true; - break; - } - } - } - ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect"); - return NO_INIT; -} - -/*static*/ bool AudioMixer::DownmixerBufferProvider::sIsMultichannelCapable = false; -/*static*/ effect_descriptor_t AudioMixer::DownmixerBufferProvider::sDwnmFxDesc; - -AudioMixer::RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask, - audio_channel_mask_t outputChannelMask, audio_format_t format, - size_t bufferFrameCount) : - CopyBufferProvider( - audio_bytes_per_sample(format) - * audio_channel_count_from_out_mask(inputChannelMask), - audio_bytes_per_sample(format) - * audio_channel_count_from_out_mask(outputChannelMask), - bufferFrameCount), - mFormat(format), - mSampleSize(audio_bytes_per_sample(format)), - mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)), - mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask)) -{ - ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu", - this, format, inputChannelMask, outputChannelMask, - mInputChannels, mOutputChannels); - - const audio_channel_representation_t inputRepresentation = - audio_channel_mask_get_representation(inputChannelMask); - const audio_channel_representation_t outputRepresentation = - audio_channel_mask_get_representation(outputChannelMask); - const uint32_t inputBits = audio_channel_mask_get_bits(inputChannelMask); - const uint32_t outputBits = audio_channel_mask_get_bits(outputChannelMask); - - switch (inputRepresentation) { - case AUDIO_CHANNEL_REPRESENTATION_POSITION: - switch (outputRepresentation) { - case AUDIO_CHANNEL_REPRESENTATION_POSITION: - memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry), - outputBits, inputBits); - return; - case AUDIO_CHANNEL_REPRESENTATION_INDEX: - // TODO: output channel index mask not currently allowed - // fall through - default: - break; - } - break; - case AUDIO_CHANNEL_REPRESENTATION_INDEX: - switch (outputRepresentation) { - case AUDIO_CHANNEL_REPRESENTATION_POSITION: - memcpy_by_index_array_initialization_src_index(mIdxAry, ARRAY_SIZE(mIdxAry), - outputBits, inputBits); - return; - case AUDIO_CHANNEL_REPRESENTATION_INDEX: - // TODO: output channel index mask not currently allowed - // fall through - default: - break; - } - break; - default: - break; - } - LOG_ALWAYS_FATAL("invalid channel mask conversion from %#x to %#x", - inputChannelMask, outputChannelMask); -} - -void AudioMixer::RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames) -{ - memcpy_by_index_array(dst, mOutputChannels, - src, mInputChannels, mIdxAry, mSampleSize, frames); -} - -AudioMixer::ReformatBufferProvider::ReformatBufferProvider(int32_t channels, - audio_format_t inputFormat, audio_format_t outputFormat, - size_t bufferFrameCount) : - CopyBufferProvider( - channels * audio_bytes_per_sample(inputFormat), - channels * audio_bytes_per_sample(outputFormat), - bufferFrameCount), - mChannels(channels), - mInputFormat(inputFormat), - mOutputFormat(outputFormat) -{ - ALOGV("ReformatBufferProvider(%p)(%d, %#x, %#x)", this, channels, inputFormat, outputFormat); -} - -void AudioMixer::ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames) -{ - memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannels); -} - // ---------------------------------------------------------------------------- // Ensure mConfiguredNames bitmask is initialized properly on all architectures. @@ -442,6 +123,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr t->resampler = NULL; t->downmixerBufferProvider = NULL; t->mReformatBufferProvider = NULL; + t->mTimestretchBufferProvider = NULL; t++; } @@ -454,6 +136,7 @@ AudioMixer::~AudioMixer() delete t->resampler; delete t->downmixerBufferProvider; delete t->mReformatBufferProvider; + delete t->mTimestretchBufferProvider; t++; } delete [] mState.outputTemp; @@ -532,6 +215,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, t->mReformatBufferProvider = NULL; t->downmixerBufferProvider = NULL; t->mPostDownmixReformatBufferProvider = NULL; + t->mTimestretchBufferProvider = NULL; t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT; t->mFormat = format; t->mMixerInFormat = selectMixerInFormat(format); @@ -539,6 +223,8 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits( AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO); t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask); + t->mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL; + t->mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL; // Check the downmixing (or upmixing) requirements. status_t status = t->prepareForDownmix(); if (status != OK) { @@ -731,6 +417,10 @@ void AudioMixer::track_t::reconfigureBufferProviders() mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider); bufferProvider = mPostDownmixReformatBufferProvider; } + if (mTimestretchBufferProvider) { + mTimestretchBufferProvider->setBufferProvider(bufferProvider); + bufferProvider = mTimestretchBufferProvider; + } } void AudioMixer::deleteTrackName(int name) @@ -751,7 +441,9 @@ void AudioMixer::deleteTrackName(int name) mState.tracks[name].unprepareForDownmix(); // delete the reformatter mState.tracks[name].unprepareForReformat(); - + // delete the timestretch provider + delete track.mTimestretchBufferProvider; + track.mTimestretchBufferProvider = NULL; mTrackNames &= ~(1<<name); } @@ -973,6 +665,26 @@ void AudioMixer::setParameter(int name, int target, int param, void *value) } } break; + case TIMESTRETCH: + switch (param) { + case PLAYBACK_RATE: { + const float speed = reinterpret_cast<float*>(value)[0]; + const float pitch = reinterpret_cast<float*>(value)[1]; + ALOG_ASSERT(AUDIO_TIMESTRETCH_SPEED_MIN <= speed + && speed <= AUDIO_TIMESTRETCH_SPEED_MAX, + "bad speed %f", speed); + ALOG_ASSERT(AUDIO_TIMESTRETCH_PITCH_MIN <= pitch + && pitch <= AUDIO_TIMESTRETCH_PITCH_MAX, + "bad pitch %f", pitch); + if (track.setPlaybackRate(speed, pitch)) { + ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, %f %f", speed, pitch); + // invalidateState(1 << name); + } + } break; + default: + LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param); + } + break; default: LOG_ALWAYS_FATAL("setParameter: bad target %d", target); @@ -1018,6 +730,28 @@ bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSam return false; } +bool AudioMixer::track_t::setPlaybackRate(float speed, float pitch) +{ + if (speed == mSpeed && pitch == mPitch) { + return false; + } + mSpeed = speed; + mPitch = pitch; + if (mTimestretchBufferProvider == NULL) { + // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer + // but if none exists, it is the channel count (1 for mono). + const int timestretchChannelCount = downmixerBufferProvider != NULL + ? mMixerChannelCount : channelCount; + mTimestretchBufferProvider = new TimestretchBufferProvider(timestretchChannelCount, + mMixerInFormat, sampleRate, speed, pitch); + reconfigureBufferProviders(); + } else { + reinterpret_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider) + ->setPlaybackRate(speed, pitch); + } + return true; +} + /* Checks to see if the volume ramp has completed and clears the increment * variables appropriately. * @@ -1096,6 +830,8 @@ void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider mState.tracks[name].downmixerBufferProvider->reset(); } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) { mState.tracks[name].mPostDownmixReformatBufferProvider->reset(); + } else if (mState.tracks[name].mTimestretchBufferProvider != NULL) { + mState.tracks[name].mTimestretchBufferProvider->reset(); } mState.tracks[name].mInputBufferProvider = bufferProvider; diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h index 381036b..e27a0d1 100644 --- a/services/audioflinger/AudioMixer.h +++ b/services/audioflinger/AudioMixer.h @@ -29,6 +29,7 @@ #include <utils/threads.h> #include "AudioResampler.h" +#include "BufferProviders.h" // FIXME This is actually unity gain, which might not be max in future, expressed in U.12 #define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT @@ -72,6 +73,7 @@ public: RESAMPLE = 0x3001, RAMP_VOLUME = 0x3002, // ramp to new volume VOLUME = 0x3003, // don't ramp + TIMESTRETCH = 0x3004, // set Parameter names // for target TRACK @@ -99,6 +101,9 @@ public: VOLUME0 = 0x4200, VOLUME1 = 0x4201, AUXLEVEL = 0x4210, + // for target TIMESTRETCH + PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name; + // parameter 'value' is a pointer to the new playback rate. }; @@ -159,7 +164,6 @@ private: struct state_t; struct track_t; - class CopyBufferProvider; typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux); @@ -214,6 +218,9 @@ private: /* Buffer providers are constructed to translate the track input data as needed. * + * TODO: perhaps make a single PlaybackConverterProvider class to move + * all pre-mixer track buffer conversions outside the AudioMixer class. + * * 1) mInputBufferProvider: The AudioTrack buffer provider. * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to * match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer @@ -223,13 +230,14 @@ private: * the number of channels required by the mixer sink. * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from * the downmixer requirements to the mixer engine input requirements. + * 5) mTimestretchBufferProvider: Adds timestretching for playback rate */ AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider. - CopyBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting. - CopyBufferProvider* downmixerBufferProvider; // wrapper for channel conversion. - CopyBufferProvider* mPostDownmixReformatBufferProvider; + PassthruBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting. + PassthruBufferProvider* downmixerBufferProvider; // wrapper for channel conversion. + PassthruBufferProvider* mPostDownmixReformatBufferProvider; + PassthruBufferProvider* mTimestretchBufferProvider; - // 16-byte boundary int32_t sessionId; audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT) @@ -251,6 +259,9 @@ private: audio_channel_mask_t mMixerChannelMask; uint32_t mMixerChannelCount; + float mSpeed; + float mPitch; + bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; } bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate); bool doesResample() const { return resampler != NULL; } @@ -263,6 +274,7 @@ private: void unprepareForDownmix(); status_t prepareForReformat(); void unprepareForReformat(); + bool setPlaybackRate(float speed, float pitch); void reconfigureBufferProviders(); }; @@ -282,112 +294,6 @@ private: track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32))); }; - // Base AudioBufferProvider class used for DownMixerBufferProvider, RemixBufferProvider, - // and ReformatBufferProvider. - // It handles a private buffer for use in converting format or channel masks from the - // input data to a form acceptable by the mixer. - // TODO: Make a ResamplerBufferProvider when integers are entirely removed from the - // processing pipeline. - class CopyBufferProvider : public AudioBufferProvider { - public: - // Use a private buffer of bufferFrameCount frames (each frame is outputFrameSize bytes). - // If bufferFrameCount is 0, no private buffer is created and in-place modification of - // the upstream buffer provider's buffers is performed by copyFrames(). - CopyBufferProvider(size_t inputFrameSize, size_t outputFrameSize, - size_t bufferFrameCount); - virtual ~CopyBufferProvider(); - - // Overrides AudioBufferProvider methods - virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); - virtual void releaseBuffer(Buffer* buffer); - - // Other public methods - - // call this to release the buffer to the upstream provider. - // treat it as an audio discontinuity for future samples. - virtual void reset(); - - // this function should be supplied by the derived class. It converts - // #frames in the *src pointer to the *dst pointer. It is public because - // some providers will allow this to work on arbitrary buffers outside - // of the internal buffers. - virtual void copyFrames(void *dst, const void *src, size_t frames) = 0; - - // set the upstream buffer provider. Consider calling "reset" before this function. - void setBufferProvider(AudioBufferProvider *p) { - mTrackBufferProvider = p; - } - - protected: - AudioBufferProvider* mTrackBufferProvider; - const size_t mInputFrameSize; - const size_t mOutputFrameSize; - private: - AudioBufferProvider::Buffer mBuffer; - const size_t mLocalBufferFrameCount; - void* mLocalBufferData; - size_t mConsumed; - }; - - // DownmixerBufferProvider wraps a track AudioBufferProvider to provide - // position dependent downmixing by an Audio Effect. - class DownmixerBufferProvider : public CopyBufferProvider { - public: - DownmixerBufferProvider(audio_channel_mask_t inputChannelMask, - audio_channel_mask_t outputChannelMask, audio_format_t format, - uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount); - virtual ~DownmixerBufferProvider(); - virtual void copyFrames(void *dst, const void *src, size_t frames); - bool isValid() const { return mDownmixHandle != NULL; } - - static status_t init(); - static bool isMultichannelCapable() { return sIsMultichannelCapable; } - - protected: - effect_handle_t mDownmixHandle; - effect_config_t mDownmixConfig; - - // effect descriptor for the downmixer used by the mixer - static effect_descriptor_t sDwnmFxDesc; - // indicates whether a downmix effect has been found and is usable by this mixer - static bool sIsMultichannelCapable; - // FIXME: should we allow effects outside of the framework? - // We need to here. A special ioId that must be <= -2 so it does not map to a session. - static const int32_t SESSION_ID_INVALID_AND_IGNORED = -2; - }; - - // RemixBufferProvider wraps a track AudioBufferProvider to perform an - // upmix or downmix to the proper channel count and mask. - class RemixBufferProvider : public CopyBufferProvider { - public: - RemixBufferProvider(audio_channel_mask_t inputChannelMask, - audio_channel_mask_t outputChannelMask, audio_format_t format, - size_t bufferFrameCount); - virtual void copyFrames(void *dst, const void *src, size_t frames); - - protected: - const audio_format_t mFormat; - const size_t mSampleSize; - const size_t mInputChannels; - const size_t mOutputChannels; - int8_t mIdxAry[sizeof(uint32_t)*8]; // 32 bits => channel indices - }; - - // ReformatBufferProvider wraps a track AudioBufferProvider to convert the input data - // to an acceptable mixer input format type. - class ReformatBufferProvider : public CopyBufferProvider { - public: - ReformatBufferProvider(int32_t channels, - audio_format_t inputFormat, audio_format_t outputFormat, - size_t bufferFrameCount); - virtual void copyFrames(void *dst, const void *src, size_t frames); - - protected: - const int32_t mChannels; - const audio_format_t mInputFormat; - const audio_format_t mOutputFormat; - }; - // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. uint32_t mTrackNames; diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index 46e3d6c..e49b7b1 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -41,7 +41,7 @@ public: AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) : AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { } - virtual void resample(int32_t* out, size_t outFrameCount, + virtual size_t resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); private: // number of bits used in interpolation multiply - 15 bits avoids overflow @@ -51,9 +51,9 @@ private: static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits; void init() {} - void resampleMono16(int32_t* out, size_t outFrameCount, + size_t resampleMono16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); - void resampleStereo16(int32_t* out, size_t outFrameCount, + size_t resampleStereo16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); #ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1 void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx, @@ -329,7 +329,7 @@ void AudioResampler::reset() { // ---------------------------------------------------------------------------- -void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount, +size_t AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { // should never happen, but we overflow if it does @@ -338,15 +338,16 @@ void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount, // select the appropriate resampler switch (mChannelCount) { case 1: - resampleMono16(out, outFrameCount, provider); - break; + return resampleMono16(out, outFrameCount, provider); case 2: - resampleStereo16(out, outFrameCount, provider); - break; + return resampleStereo16(out, outFrameCount, provider); + default: + LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount); + return 0; } } -void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, +size_t AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { int32_t vl = mVolume[0]; @@ -442,9 +443,10 @@ resampleStereo16_exit: // save state mInputIndex = inputIndex; mPhaseFraction = phaseFraction; + return outputIndex / 2 /* channels for stereo */; } -void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, +size_t AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { int32_t vl = mVolume[0]; @@ -538,6 +540,7 @@ resampleMono16_exit: // save state mInputIndex = inputIndex; mPhaseFraction = phaseFraction; + return outputIndex; } #ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1 diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h index 863614a..a8e3e6f 100644 --- a/services/audioflinger/AudioResampler.h +++ b/services/audioflinger/AudioResampler.h @@ -67,12 +67,18 @@ public: // Resample int16_t samples from provider and accumulate into 'out'. // A mono provider delivers a sequence of samples. // A stereo provider delivers a sequence of interleaved pairs of samples. - // Multi-channel providers are not supported. + // // In either case, 'out' holds interleaved pairs of fixed-point Q4.27. // That is, for a mono provider, there is an implicit up-channeling. // Since this method accumulates, the caller is responsible for clearing 'out' initially. - // FIXME assumes provider is always successful; it should return the actual frame count. - virtual void resample(int32_t* out, size_t outFrameCount, + // + // For a float resampler, 'out' holds interleaved pairs of float samples. + // + // Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY, + // DYN_MED_QUALITY, and DYN_HIGH_QUALITY. + // + // Returns the number of frames resampled into the out buffer. + virtual size_t resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) = 0; virtual void reset(); diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp index d3cbd1c..172c2a5 100644 --- a/services/audioflinger/AudioResamplerCubic.cpp +++ b/services/audioflinger/AudioResamplerCubic.cpp @@ -14,7 +14,7 @@ * limitations under the License. */ -#define LOG_TAG "AudioSRC" +#define LOG_TAG "AudioResamplerCubic" #include <stdint.h> #include <string.h> @@ -32,7 +32,7 @@ void AudioResamplerCubic::init() { memset(&right, 0, sizeof(state)); } -void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, +size_t AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { // should never happen, but we overflow if it does @@ -41,15 +41,16 @@ void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, // select the appropriate resampler switch (mChannelCount) { case 1: - resampleMono16(out, outFrameCount, provider); - break; + return resampleMono16(out, outFrameCount, provider); case 2: - resampleStereo16(out, outFrameCount, provider); - break; + return resampleStereo16(out, outFrameCount, provider); + default: + LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount); + return 0; } } -void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, +size_t AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { int32_t vl = mVolume[0]; @@ -67,7 +68,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, mPTS); if (mBuffer.raw == NULL) { - return; + return 0; } // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); } @@ -115,9 +116,10 @@ save_state: // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); mInputIndex = inputIndex; mPhaseFraction = phaseFraction; + return outputIndex / 2 /* channels for stereo */; } -void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, +size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { int32_t vl = mVolume[0]; @@ -135,7 +137,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, mPTS); if (mBuffer.raw == NULL) { - return; + return 0; } // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); } @@ -182,6 +184,7 @@ save_state: // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); mInputIndex = inputIndex; mPhaseFraction = phaseFraction; + return outputIndex; } // ---------------------------------------------------------------------------- diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h index 1ddc5f9..4b45b0b 100644 --- a/services/audioflinger/AudioResamplerCubic.h +++ b/services/audioflinger/AudioResamplerCubic.h @@ -31,7 +31,7 @@ public: AudioResamplerCubic(int inChannelCount, int32_t sampleRate) : AudioResampler(inChannelCount, sampleRate, MED_QUALITY) { } - virtual void resample(int32_t* out, size_t outFrameCount, + virtual size_t resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); private: // number of bits used in interpolation multiply - 14 bits avoids overflow @@ -43,9 +43,9 @@ private: int32_t a, b, c, y0, y1, y2, y3; } state; void init(); - void resampleMono16(int32_t* out, size_t outFrameCount, + size_t resampleMono16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); - void resampleStereo16(int32_t* out, size_t outFrameCount, + size_t resampleStereo16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); static inline int32_t interp(state* p, int32_t x) { return (((((p->a * x >> 14) + p->b) * x >> 14) + p->c) * x >> 14) + p->y1; diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp index c21d4ca..6481b85 100644 --- a/services/audioflinger/AudioResamplerDyn.cpp +++ b/services/audioflinger/AudioResamplerDyn.cpp @@ -477,15 +477,15 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) } template<typename TC, typename TI, typename TO> -void AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount, +size_t AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { - (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider); + return (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider); } template<typename TC, typename TI, typename TO> template<int CHANNELS, bool LOCKED, int STRIDE> -void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, +size_t AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider) { // TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out. @@ -610,6 +610,7 @@ resample_exit: ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer mInBuffer.setImpulse(impulse); mPhaseFraction = phaseFraction; + return outputIndex / OUTPUT_CHANNELS; } /* instantiate templates used by AudioResampler::create */ diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h index 238b163..3b1c381 100644 --- a/services/audioflinger/AudioResamplerDyn.h +++ b/services/audioflinger/AudioResamplerDyn.h @@ -52,7 +52,7 @@ public: virtual void setVolume(float left, float right); - virtual void resample(int32_t* out, size_t outFrameCount, + virtual size_t resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); private: @@ -111,10 +111,10 @@ private: int inSampleRate, int outSampleRate, double tbwCheat); template<int CHANNELS, bool LOCKED, int STRIDE> - void resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider); + size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider); // define a pointer to member function type for resample - typedef void (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out, + typedef size_t (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out, size_t outFrameCount, AudioBufferProvider* provider); // data - the contiguous storage and layout of these is important. diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp index ba9a356..41730ee 100644 --- a/services/audioflinger/AudioResamplerSinc.cpp +++ b/services/audioflinger/AudioResamplerSinc.cpp @@ -256,7 +256,7 @@ void AudioResamplerSinc::setVolume(float left, float right) { mVolumeSIMD[1] = u4_28_from_float(clampFloatVol(right)); } -void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, +size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { // FIXME store current state (up or down sample) and only load the coefs when the state @@ -272,17 +272,18 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, // select the appropriate resampler switch (mChannelCount) { case 1: - resample<1>(out, outFrameCount, provider); - break; + return resample<1>(out, outFrameCount, provider); case 2: - resample<2>(out, outFrameCount, provider); - break; + return resample<2>(out, outFrameCount, provider); + default: + LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount); + return 0; } } template<int CHANNELS> -void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, +size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { const Constants& c(*mConstants); @@ -357,6 +358,7 @@ resample_exit: mImpulse = impulse; mInputIndex = inputIndex; mPhaseFraction = phaseFraction; + return outputIndex / CHANNELS; } template<int CHANNELS> diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h index 6d8e85d..0fbeac8 100644 --- a/services/audioflinger/AudioResamplerSinc.h +++ b/services/audioflinger/AudioResamplerSinc.h @@ -39,7 +39,7 @@ public: virtual ~AudioResamplerSinc(); - virtual void resample(int32_t* out, size_t outFrameCount, + virtual size_t resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); private: void init(); @@ -47,7 +47,7 @@ private: virtual void setVolume(float left, float right); template<int CHANNELS> - void resample(int32_t* out, size_t outFrameCount, + size_t resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); template<int CHANNELS> diff --git a/services/audioflinger/BufferProviders.cpp b/services/audioflinger/BufferProviders.cpp new file mode 100644 index 0000000..dcae5e7 --- /dev/null +++ b/services/audioflinger/BufferProviders.cpp @@ -0,0 +1,540 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "BufferProvider" +//#define LOG_NDEBUG 0 + +#include <audio_effects/effect_downmix.h> +#include <audio_utils/primitives.h> +#include <audio_utils/format.h> +#include <media/AudioResamplerPublic.h> +#include <media/EffectsFactoryApi.h> + +#include <utils/Log.h> + +#include "Configuration.h" +#include "BufferProviders.h" + +#ifndef ARRAY_SIZE +#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0])) +#endif + +namespace android { + +// ---------------------------------------------------------------------------- + +template <typename T> +static inline T min(const T& a, const T& b) +{ + return a < b ? a : b; +} + +CopyBufferProvider::CopyBufferProvider(size_t inputFrameSize, + size_t outputFrameSize, size_t bufferFrameCount) : + mInputFrameSize(inputFrameSize), + mOutputFrameSize(outputFrameSize), + mLocalBufferFrameCount(bufferFrameCount), + mLocalBufferData(NULL), + mConsumed(0) +{ + ALOGV("CopyBufferProvider(%p)(%zu, %zu, %zu)", this, + inputFrameSize, outputFrameSize, bufferFrameCount); + LOG_ALWAYS_FATAL_IF(inputFrameSize < outputFrameSize && bufferFrameCount == 0, + "Requires local buffer if inputFrameSize(%zu) < outputFrameSize(%zu)", + inputFrameSize, outputFrameSize); + if (mLocalBufferFrameCount) { + (void)posix_memalign(&mLocalBufferData, 32, mLocalBufferFrameCount * mOutputFrameSize); + } + mBuffer.frameCount = 0; +} + +CopyBufferProvider::~CopyBufferProvider() +{ + ALOGV("~CopyBufferProvider(%p)", this); + if (mBuffer.frameCount != 0) { + mTrackBufferProvider->releaseBuffer(&mBuffer); + } + free(mLocalBufferData); +} + +status_t CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer, + int64_t pts) +{ + //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)", + // this, pBuffer, pBuffer->frameCount, pts); + if (mLocalBufferFrameCount == 0) { + status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts); + if (res == OK) { + copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount); + } + return res; + } + if (mBuffer.frameCount == 0) { + mBuffer.frameCount = pBuffer->frameCount; + status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts); + // At one time an upstream buffer provider had + // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014. + // + // By API spec, if res != OK, then mBuffer.frameCount == 0. + // but there may be improper implementations. + ALOG_ASSERT(res == OK || mBuffer.frameCount == 0); + if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe. + pBuffer->raw = NULL; + pBuffer->frameCount = 0; + return res; + } + mConsumed = 0; + } + ALOG_ASSERT(mConsumed < mBuffer.frameCount); + size_t count = min(mLocalBufferFrameCount, mBuffer.frameCount - mConsumed); + count = min(count, pBuffer->frameCount); + pBuffer->raw = mLocalBufferData; + pBuffer->frameCount = count; + copyFrames(pBuffer->raw, (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize, + pBuffer->frameCount); + return OK; +} + +void CopyBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) +{ + //ALOGV("CopyBufferProvider(%p)::releaseBuffer(%p(%zu))", + // this, pBuffer, pBuffer->frameCount); + if (mLocalBufferFrameCount == 0) { + mTrackBufferProvider->releaseBuffer(pBuffer); + return; + } + // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount"); + mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content + if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) { + mTrackBufferProvider->releaseBuffer(&mBuffer); + ALOG_ASSERT(mBuffer.frameCount == 0); + } + pBuffer->raw = NULL; + pBuffer->frameCount = 0; +} + +void CopyBufferProvider::reset() +{ + if (mBuffer.frameCount != 0) { + mTrackBufferProvider->releaseBuffer(&mBuffer); + } + mConsumed = 0; +} + +DownmixerBufferProvider::DownmixerBufferProvider( + audio_channel_mask_t inputChannelMask, + audio_channel_mask_t outputChannelMask, audio_format_t format, + uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) : + CopyBufferProvider( + audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask), + audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask), + bufferFrameCount) // set bufferFrameCount to 0 to do in-place +{ + ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d)", + this, inputChannelMask, outputChannelMask, format, + sampleRate, sessionId); + if (!sIsMultichannelCapable + || EffectCreate(&sDwnmFxDesc.uuid, + sessionId, + SESSION_ID_INVALID_AND_IGNORED, + &mDownmixHandle) != 0) { + ALOGE("DownmixerBufferProvider() error creating downmixer effect"); + mDownmixHandle = NULL; + return; + } + // channel input configuration will be overridden per-track + mDownmixConfig.inputCfg.channels = inputChannelMask; // FIXME: Should be bits + mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits + mDownmixConfig.inputCfg.format = format; + mDownmixConfig.outputCfg.format = format; + mDownmixConfig.inputCfg.samplingRate = sampleRate; + mDownmixConfig.outputCfg.samplingRate = sampleRate; + mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; + mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; + // input and output buffer provider, and frame count will not be used as the downmix effect + // process() function is called directly (see DownmixerBufferProvider::getNextBuffer()) + mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | + EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE; + mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask; + + int cmdStatus; + uint32_t replySize = sizeof(int); + + // Configure downmixer + status_t status = (*mDownmixHandle)->command(mDownmixHandle, + EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/, + &mDownmixConfig /*pCmdData*/, + &replySize, &cmdStatus /*pReplyData*/); + if (status != 0 || cmdStatus != 0) { + ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer", + status, cmdStatus); + EffectRelease(mDownmixHandle); + mDownmixHandle = NULL; + return; + } + + // Enable downmixer + replySize = sizeof(int); + status = (*mDownmixHandle)->command(mDownmixHandle, + EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/, + &replySize, &cmdStatus /*pReplyData*/); + if (status != 0 || cmdStatus != 0) { + ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer", + status, cmdStatus); + EffectRelease(mDownmixHandle); + mDownmixHandle = NULL; + return; + } + + // Set downmix type + // parameter size rounded for padding on 32bit boundary + const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int); + const int downmixParamSize = + sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t); + effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize); + param->psize = sizeof(downmix_params_t); + const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE; + memcpy(param->data, &downmixParam, param->psize); + const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD; + param->vsize = sizeof(downmix_type_t); + memcpy(param->data + psizePadded, &downmixType, param->vsize); + replySize = sizeof(int); + status = (*mDownmixHandle)->command(mDownmixHandle, + EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */, + param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/); + free(param); + if (status != 0 || cmdStatus != 0) { + ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type", + status, cmdStatus); + EffectRelease(mDownmixHandle); + mDownmixHandle = NULL; + return; + } + ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType); +} + +DownmixerBufferProvider::~DownmixerBufferProvider() +{ + ALOGV("~DownmixerBufferProvider (%p)", this); + EffectRelease(mDownmixHandle); + mDownmixHandle = NULL; +} + +void DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames) +{ + mDownmixConfig.inputCfg.buffer.frameCount = frames; + mDownmixConfig.inputCfg.buffer.raw = const_cast<void *>(src); + mDownmixConfig.outputCfg.buffer.frameCount = frames; + mDownmixConfig.outputCfg.buffer.raw = dst; + // may be in-place if src == dst. + status_t res = (*mDownmixHandle)->process(mDownmixHandle, + &mDownmixConfig.inputCfg.buffer, &mDownmixConfig.outputCfg.buffer); + ALOGE_IF(res != OK, "DownmixBufferProvider error %d", res); +} + +/* call once in a pthread_once handler. */ +/*static*/ status_t DownmixerBufferProvider::init() +{ + // find multichannel downmix effect if we have to play multichannel content + uint32_t numEffects = 0; + int ret = EffectQueryNumberEffects(&numEffects); + if (ret != 0) { + ALOGE("AudioMixer() error %d querying number of effects", ret); + return NO_INIT; + } + ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects); + + for (uint32_t i = 0 ; i < numEffects ; i++) { + if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) { + ALOGV("effect %d is called %s", i, sDwnmFxDesc.name); + if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) { + ALOGI("found effect \"%s\" from %s", + sDwnmFxDesc.name, sDwnmFxDesc.implementor); + sIsMultichannelCapable = true; + break; + } + } + } + ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect"); + return NO_INIT; +} + +/*static*/ bool DownmixerBufferProvider::sIsMultichannelCapable = false; +/*static*/ effect_descriptor_t DownmixerBufferProvider::sDwnmFxDesc; + +RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask, + audio_channel_mask_t outputChannelMask, audio_format_t format, + size_t bufferFrameCount) : + CopyBufferProvider( + audio_bytes_per_sample(format) + * audio_channel_count_from_out_mask(inputChannelMask), + audio_bytes_per_sample(format) + * audio_channel_count_from_out_mask(outputChannelMask), + bufferFrameCount), + mFormat(format), + mSampleSize(audio_bytes_per_sample(format)), + mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)), + mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask)) +{ + ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu", + this, format, inputChannelMask, outputChannelMask, + mInputChannels, mOutputChannels); + + const audio_channel_representation_t inputRepresentation = + audio_channel_mask_get_representation(inputChannelMask); + const audio_channel_representation_t outputRepresentation = + audio_channel_mask_get_representation(outputChannelMask); + const uint32_t inputBits = audio_channel_mask_get_bits(inputChannelMask); + const uint32_t outputBits = audio_channel_mask_get_bits(outputChannelMask); + + switch (inputRepresentation) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + switch (outputRepresentation) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry), + outputBits, inputBits); + return; + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + // TODO: output channel index mask not currently allowed + // fall through + default: + break; + } + break; + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + switch (outputRepresentation) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + memcpy_by_index_array_initialization_src_index(mIdxAry, ARRAY_SIZE(mIdxAry), + outputBits, inputBits); + return; + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + // TODO: output channel index mask not currently allowed + // fall through + default: + break; + } + break; + default: + break; + } + LOG_ALWAYS_FATAL("invalid channel mask conversion from %#x to %#x", + inputChannelMask, outputChannelMask); +} + +void RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames) +{ + memcpy_by_index_array(dst, mOutputChannels, + src, mInputChannels, mIdxAry, mSampleSize, frames); +} + +ReformatBufferProvider::ReformatBufferProvider(int32_t channelCount, + audio_format_t inputFormat, audio_format_t outputFormat, + size_t bufferFrameCount) : + CopyBufferProvider( + channelCount * audio_bytes_per_sample(inputFormat), + channelCount * audio_bytes_per_sample(outputFormat), + bufferFrameCount), + mChannelCount(channelCount), + mInputFormat(inputFormat), + mOutputFormat(outputFormat) +{ + ALOGV("ReformatBufferProvider(%p)(%u, %#x, %#x)", + this, channelCount, inputFormat, outputFormat); +} + +void ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames) +{ + memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount); +} + +TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount, + audio_format_t format, uint32_t sampleRate, float speed, float pitch) : + mChannelCount(channelCount), + mFormat(format), + mSampleRate(sampleRate), + mFrameSize(channelCount * audio_bytes_per_sample(format)), + mSpeed(speed), + mPitch(pitch), + mLocalBufferFrameCount(0), + mLocalBufferData(NULL), + mRemaining(0), + mSonicStream(sonicCreateStream(sampleRate, mChannelCount)) +{ + ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f)", + this, channelCount, format, sampleRate, speed, pitch); + mBuffer.frameCount = 0; + + LOG_ALWAYS_FATAL_IF(mSonicStream == NULL, + "TimestretchBufferProvider can't allocate Sonic stream"); + sonicSetSpeed(mSonicStream, speed); +} + +TimestretchBufferProvider::~TimestretchBufferProvider() +{ + ALOGV("~TimestretchBufferProvider(%p)", this); + sonicDestroyStream(mSonicStream); + if (mBuffer.frameCount != 0) { + mTrackBufferProvider->releaseBuffer(&mBuffer); + } + free(mLocalBufferData); +} + +status_t TimestretchBufferProvider::getNextBuffer( + AudioBufferProvider::Buffer *pBuffer, int64_t pts) +{ + ALOGV("TimestretchBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)", + this, pBuffer, pBuffer->frameCount, pts); + + // BYPASS + //return mTrackBufferProvider->getNextBuffer(pBuffer, pts); + + // check if previously processed data is sufficient. + if (pBuffer->frameCount <= mRemaining) { + ALOGV("previous sufficient"); + pBuffer->raw = mLocalBufferData; + return OK; + } + + // do we need to resize our buffer? + if (pBuffer->frameCount > mLocalBufferFrameCount) { + void *newmem; + if (posix_memalign(&newmem, 32, pBuffer->frameCount * mFrameSize) == OK) { + if (mRemaining != 0) { + memcpy(newmem, mLocalBufferData, mRemaining * mFrameSize); + } + free(mLocalBufferData); + mLocalBufferData = newmem; + mLocalBufferFrameCount = pBuffer->frameCount; + } + } + + // need to fetch more data + const size_t outputDesired = pBuffer->frameCount - mRemaining; + mBuffer.frameCount = mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL + ? outputDesired : outputDesired * mSpeed + 1; + + status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts); + + ALOG_ASSERT(res == OK || mBuffer.frameCount == 0); + if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe. + ALOGD("buffer error"); + if (mRemaining == 0) { + pBuffer->raw = NULL; + pBuffer->frameCount = 0; + return res; + } else { // return partial count + pBuffer->raw = mLocalBufferData; + pBuffer->frameCount = mRemaining; + return OK; + } + } + + // time-stretch the data + size_t dstAvailable = min(mLocalBufferFrameCount - mRemaining, outputDesired); + size_t srcAvailable = mBuffer.frameCount; + processFrames((uint8_t*)mLocalBufferData + mRemaining * mFrameSize, &dstAvailable, + mBuffer.raw, &srcAvailable); + + // release all data consumed + mBuffer.frameCount = srcAvailable; + mTrackBufferProvider->releaseBuffer(&mBuffer); + + // update buffer vars with the actual data processed and return with buffer + mRemaining += dstAvailable; + + pBuffer->raw = mLocalBufferData; + pBuffer->frameCount = mRemaining; + + return OK; +} + +void TimestretchBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) +{ + ALOGV("TimestretchBufferProvider(%p)::releaseBuffer(%p (%zu))", + this, pBuffer, pBuffer->frameCount); + + // BYPASS + //return mTrackBufferProvider->releaseBuffer(pBuffer); + + // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount"); + if (pBuffer->frameCount < mRemaining) { + memcpy(mLocalBufferData, + (uint8_t*)mLocalBufferData + pBuffer->frameCount * mFrameSize, + (mRemaining - pBuffer->frameCount) * mFrameSize); + mRemaining -= pBuffer->frameCount; + } else if (pBuffer->frameCount == mRemaining) { + mRemaining = 0; + } else { + LOG_ALWAYS_FATAL("Releasing more frames(%zu) than available(%zu)", + pBuffer->frameCount, mRemaining); + } + + pBuffer->raw = NULL; + pBuffer->frameCount = 0; +} + +void TimestretchBufferProvider::reset() +{ + mRemaining = 0; +} + +status_t TimestretchBufferProvider::setPlaybackRate(float speed, float pitch) +{ + mSpeed = speed; + mPitch = pitch; + + sonicSetSpeed(mSonicStream, speed); + //TODO: pitch is ignored for now + return OK; +} + +void TimestretchBufferProvider::processFrames(void *dstBuffer, size_t *dstFrames, + const void *srcBuffer, size_t *srcFrames) +{ + ALOGV("processFrames(%zu %zu) remaining(%zu)", *dstFrames, *srcFrames, mRemaining); + // Note dstFrames is the required number of frames. + + // Ensure consumption from src is as expected. + const size_t targetSrc = *dstFrames * mSpeed; + if (*srcFrames < targetSrc) { // limit dst frames to that possible + *dstFrames = *srcFrames / mSpeed; + } else if (*srcFrames > targetSrc + 1) { + *srcFrames = targetSrc + 1; + } + + switch (mFormat) { + case AUDIO_FORMAT_PCM_FLOAT: + if (sonicWriteFloatToStream(mSonicStream, (float*)srcBuffer, *srcFrames) != 1) { + ALOGE("sonicWriteFloatToStream cannot realloc"); + *srcFrames = 0; // cannot consume all of srcBuffer + } + *dstFrames = sonicReadFloatFromStream(mSonicStream, (float*)dstBuffer, *dstFrames); + break; + case AUDIO_FORMAT_PCM_16_BIT: + if (sonicWriteShortToStream(mSonicStream, (short*)srcBuffer, *srcFrames) != 1) { + ALOGE("sonicWriteShortToStream cannot realloc"); + *srcFrames = 0; // cannot consume all of srcBuffer + } + *dstFrames = sonicReadShortFromStream(mSonicStream, (short*)dstBuffer, *dstFrames); + break; + default: + // could also be caught on construction + LOG_ALWAYS_FATAL("invalid format %#x for TimestretchBufferProvider", mFormat); + } +} + +// ---------------------------------------------------------------------------- +} // namespace android diff --git a/services/audioflinger/BufferProviders.h b/services/audioflinger/BufferProviders.h new file mode 100644 index 0000000..42030c0 --- /dev/null +++ b/services/audioflinger/BufferProviders.h @@ -0,0 +1,193 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef ANDROID_BUFFER_PROVIDERS_H +#define ANDROID_BUFFER_PROVIDERS_H + +#include <stdint.h> +#include <sys/types.h> + +#include <hardware/audio_effect.h> +#include <media/AudioBufferProvider.h> +#include <system/audio.h> +#include <sonic.h> + +namespace android { + +// ---------------------------------------------------------------------------- + +class PassthruBufferProvider : public AudioBufferProvider { +public: + PassthruBufferProvider() : mTrackBufferProvider(NULL) { } + + virtual ~PassthruBufferProvider() { } + + // call this to release the buffer to the upstream provider. + // treat it as an audio discontinuity for future samples. + virtual void reset() { } + + // set the upstream buffer provider. Consider calling "reset" before this function. + virtual void setBufferProvider(AudioBufferProvider *p) { + mTrackBufferProvider = p; + } + +protected: + AudioBufferProvider *mTrackBufferProvider; +}; + +// Base AudioBufferProvider class used for DownMixerBufferProvider, RemixBufferProvider, +// and ReformatBufferProvider. +// It handles a private buffer for use in converting format or channel masks from the +// input data to a form acceptable by the mixer. +// TODO: Make a ResamplerBufferProvider when integers are entirely removed from the +// processing pipeline. +class CopyBufferProvider : public PassthruBufferProvider { +public: + // Use a private buffer of bufferFrameCount frames (each frame is outputFrameSize bytes). + // If bufferFrameCount is 0, no private buffer is created and in-place modification of + // the upstream buffer provider's buffers is performed by copyFrames(). + CopyBufferProvider(size_t inputFrameSize, size_t outputFrameSize, + size_t bufferFrameCount); + virtual ~CopyBufferProvider(); + + // Overrides AudioBufferProvider methods + virtual status_t getNextBuffer(Buffer *buffer, int64_t pts); + virtual void releaseBuffer(Buffer *buffer); + + // Overrides PassthruBufferProvider + virtual void reset(); + + // this function should be supplied by the derived class. It converts + // #frames in the *src pointer to the *dst pointer. It is public because + // some providers will allow this to work on arbitrary buffers outside + // of the internal buffers. + virtual void copyFrames(void *dst, const void *src, size_t frames) = 0; + +protected: + const size_t mInputFrameSize; + const size_t mOutputFrameSize; +private: + AudioBufferProvider::Buffer mBuffer; + const size_t mLocalBufferFrameCount; + void *mLocalBufferData; + size_t mConsumed; +}; + +// DownmixerBufferProvider derives from CopyBufferProvider to provide +// position dependent downmixing by an Audio Effect. +class DownmixerBufferProvider : public CopyBufferProvider { +public: + DownmixerBufferProvider(audio_channel_mask_t inputChannelMask, + audio_channel_mask_t outputChannelMask, audio_format_t format, + uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount); + virtual ~DownmixerBufferProvider(); + //Overrides + virtual void copyFrames(void *dst, const void *src, size_t frames); + + bool isValid() const { return mDownmixHandle != NULL; } + static status_t init(); + static bool isMultichannelCapable() { return sIsMultichannelCapable; } + +protected: + effect_handle_t mDownmixHandle; + effect_config_t mDownmixConfig; + + // effect descriptor for the downmixer used by the mixer + static effect_descriptor_t sDwnmFxDesc; + // indicates whether a downmix effect has been found and is usable by this mixer + static bool sIsMultichannelCapable; + // FIXME: should we allow effects outside of the framework? + // We need to here. A special ioId that must be <= -2 so it does not map to a session. + static const int32_t SESSION_ID_INVALID_AND_IGNORED = -2; +}; + +// RemixBufferProvider derives from CopyBufferProvider to perform an +// upmix or downmix to the proper channel count and mask. +class RemixBufferProvider : public CopyBufferProvider { +public: + RemixBufferProvider(audio_channel_mask_t inputChannelMask, + audio_channel_mask_t outputChannelMask, audio_format_t format, + size_t bufferFrameCount); + //Overrides + virtual void copyFrames(void *dst, const void *src, size_t frames); + +protected: + const audio_format_t mFormat; + const size_t mSampleSize; + const size_t mInputChannels; + const size_t mOutputChannels; + int8_t mIdxAry[sizeof(uint32_t) * 8]; // 32 bits => channel indices +}; + +// ReformatBufferProvider derives from CopyBufferProvider to convert the input data +// to an acceptable mixer input format type. +class ReformatBufferProvider : public CopyBufferProvider { +public: + ReformatBufferProvider(int32_t channelCount, + audio_format_t inputFormat, audio_format_t outputFormat, + size_t bufferFrameCount); + virtual void copyFrames(void *dst, const void *src, size_t frames); + +protected: + const uint32_t mChannelCount; + const audio_format_t mInputFormat; + const audio_format_t mOutputFormat; +}; + +// TimestretchBufferProvider derives from PassthruBufferProvider for time stretching +class TimestretchBufferProvider : public PassthruBufferProvider { +public: + TimestretchBufferProvider(int32_t channelCount, + audio_format_t format, uint32_t sampleRate, float speed, float pitch); + virtual ~TimestretchBufferProvider(); + + // Overrides AudioBufferProvider methods + virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); + virtual void releaseBuffer(Buffer* buffer); + + // Overrides PassthruBufferProvider + virtual void reset(); + + virtual status_t setPlaybackRate(float speed, float pitch); + + // processes frames + // dstBuffer is where to place the data + // dstFrames [in/out] is the desired frames (return with actual placed in buffer) + // srcBuffer is the source data + // srcFrames [in/out] is the available source frames (return with consumed) + virtual void processFrames(void *dstBuffer, size_t *dstFrames, + const void *srcBuffer, size_t *srcFrames); + +protected: + const uint32_t mChannelCount; + const audio_format_t mFormat; + const uint32_t mSampleRate; // const for now (TODO change this) + const size_t mFrameSize; + float mSpeed; + float mPitch; + +private: + AudioBufferProvider::Buffer mBuffer; + size_t mLocalBufferFrameCount; + void *mLocalBufferData; + size_t mRemaining; + sonicStream mSonicStream; +}; + +// ---------------------------------------------------------------------------- +} // namespace android + +#endif // ANDROID_BUFFER_PROVIDERS_H diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp index efbdcff..834947f 100644 --- a/services/audioflinger/PatchPanel.cpp +++ b/services/audioflinger/PatchPanel.cpp @@ -200,26 +200,17 @@ status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *pa status = BAD_VALUE; goto exit; } - // limit to connections between devices and input streams for HAL before 3.0 - if (patch->sinks[i].ext.mix.hw_module == srcModule && - (audioHwDevice->version() < AUDIO_DEVICE_API_VERSION_3_0) && - (patch->sinks[i].type != AUDIO_PORT_TYPE_MIX)) { - ALOGW("createAudioPatch() invalid sink type %d for device source", - patch->sinks[i].type); - status = BAD_VALUE; - goto exit; - } } - if (patch->sinks[0].ext.device.hw_module != srcModule) { - // limit to device to device connection if not on same hw module - if (patch->sinks[0].type != AUDIO_PORT_TYPE_DEVICE) { - ALOGW("createAudioPatch() invalid sink type for cross hw module"); - status = INVALID_OPERATION; - goto exit; - } - // special case num sources == 2 -=> reuse an exiting output mix to connect to the - // sink + // manage patches requiring a software bridge + // - Device to device AND + // - source HW module != destination HW module OR + // - audio HAL version < 3.0 + // - special patch request with 2 sources (reuse one existing output mix) + if ((patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) && + ((patch->sinks[0].ext.device.hw_module != srcModule) || + (audioHwDevice->version() < AUDIO_DEVICE_API_VERSION_3_0) || + (patch->num_sources == 2))) { if (patch->num_sources == 2) { if (patch->sources[1].type != AUDIO_PORT_TYPE_MIX || patch->sinks[0].ext.device.hw_module != @@ -304,6 +295,11 @@ status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *pa &halHandle); } } else { + if (patch->sinks[0].type != AUDIO_PORT_TYPE_MIX) { + status = INVALID_OPERATION; + goto exit; + } + sp<ThreadBase> thread = audioflinger->checkRecordThread_l( patch->sinks[0].ext.mix.handle); if (thread == 0) { @@ -472,6 +468,7 @@ status_t AudioFlinger::PatchPanel::createPatchConnections(Patch *patch, // this track is given the same buffer as the PatchRecord buffer patch->mPatchTrack = new PlaybackThread::PatchTrack( patch->mPlaybackThread.get(), + audioPatch->sources[1].ext.mix.usecase.stream, sampleRate, outChannelMask, format, @@ -578,8 +575,8 @@ status_t AudioFlinger::PatchPanel::releaseAudioPatch(audio_patch_handle_t handle break; } - if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE && - patch->sinks[0].ext.device.hw_module != srcModule) { + if (removedPatch->mRecordPatchHandle != AUDIO_PATCH_HANDLE_NONE || + removedPatch->mPlaybackPatchHandle != AUDIO_PATCH_HANDLE_NONE) { clearPatchConnections(removedPatch); break; } @@ -693,5 +690,4 @@ status_t AudioFlinger::PatchPanel::setAudioPortConfig(const struct audio_port_co return NO_ERROR; } - } // namespace android diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h index 45df6a9..c51021b 100644 --- a/services/audioflinger/PlaybackTracks.h +++ b/services/audioflinger/PlaybackTracks.h @@ -298,6 +298,7 @@ class PatchTrack : public Track, public PatchProxyBufferProvider { public: PatchTrack(PlaybackThread *playbackThread, + audio_stream_type_t streamType, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h index 204a9d6..25d6d95 100644 --- a/services/audioflinger/RecordTracks.h +++ b/services/audioflinger/RecordTracks.h @@ -34,6 +34,7 @@ public: IAudioFlinger::track_flags_t flags, track_type type); virtual ~RecordTrack(); + virtual status_t initCheck() const; virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); virtual void stop(); @@ -66,21 +67,6 @@ private: bool mOverflow; // overflow on most recent attempt to fill client buffer - // updated by RecordThread::readInputParameters_l() - AudioResampler *mResampler; - - // interleaved stereo pairs of fixed-point Q4.27 - int32_t *mRsmpOutBuffer; - // current allocated frame count for the above, which may be larger than needed - size_t mRsmpOutFrameCount; - - size_t mRsmpInUnrel; // unreleased frames remaining from - // most recent getNextBuffer - // for debug only - - // rolling counter that is never cleared - int32_t mRsmpInFront; // next available frame - AudioBufferProvider::Buffer mSink; // references client's buffer sink in shared memory // sync event triggering actual audio capture. Frames read before this event will @@ -93,7 +79,10 @@ private: ssize_t mFramesToDrop; // used by resampler to find source frames - ResamplerBufferProvider *mResamplerBufferProvider; + ResamplerBufferProvider *mResamplerBufferProvider; + + // used by the record thread to convert frames to proper destination format + RecordBufferConverter *mRecordBufferConverter; }; // playback track, used by PatchPanel diff --git a/services/audioflinger/ServiceUtilities.cpp b/services/audioflinger/ServiceUtilities.cpp index fae19a1..8246fef 100644 --- a/services/audioflinger/ServiceUtilities.cpp +++ b/services/audioflinger/ServiceUtilities.cpp @@ -50,13 +50,6 @@ bool captureHotwordAllowed() { return ok; } -bool captureFmTunerAllowed() { - static const String16 sCaptureFmTunerAllowed("android.permission.ACCESS_FM_RADIO"); - bool ok = checkCallingPermission(sCaptureFmTunerAllowed); - if (!ok) ALOGE("android.permission.ACCESS_FM_RADIO"); - return ok; -} - bool settingsAllowed() { if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true; static const String16 sAudioSettings("android.permission.MODIFY_AUDIO_SETTINGS"); diff --git a/services/audioflinger/ServiceUtilities.h b/services/audioflinger/ServiceUtilities.h index ce18a90..df6f6f4 100644 --- a/services/audioflinger/ServiceUtilities.h +++ b/services/audioflinger/ServiceUtilities.h @@ -23,7 +23,6 @@ extern pid_t getpid_cached; bool recordingAllowed(); bool captureAudioOutputAllowed(); bool captureHotwordAllowed(); -bool captureFmTunerAllowed(); bool settingsAllowed(); bool modifyAudioRoutingAllowed(); bool dumpAllowed(); diff --git a/services/audioflinger/SpdifStreamOut.cpp b/services/audioflinger/SpdifStreamOut.cpp index d23588e..45b541a 100644 --- a/services/audioflinger/SpdifStreamOut.cpp +++ b/services/audioflinger/SpdifStreamOut.cpp @@ -32,10 +32,12 @@ namespace android { * If the AudioFlinger is processing encoded data and the HAL expects * PCM then we need to wrap the data in an SPDIF wrapper. */ -SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags) +SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev, + audio_output_flags_t flags, + audio_format_t format) : AudioStreamOut(dev,flags) , mRateMultiplier(1) - , mSpdifEncoder(this) + , mSpdifEncoder(this, format) , mRenderPositionHal(0) , mPreviousHalPosition32(0) { @@ -49,15 +51,15 @@ status_t SpdifStreamOut::open( { struct audio_config customConfig = *config; - customConfig.format = AUDIO_FORMAT_PCM_16_BIT; - customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO; - // Some data bursts run at a higher sample rate. + // TODO Move this into the audio_utils as a static method. switch(config->format) { case AUDIO_FORMAT_E_AC3: mRateMultiplier = 4; break; case AUDIO_FORMAT_AC3: + case AUDIO_FORMAT_DTS: + case AUDIO_FORMAT_DTS_HD: mRateMultiplier = 1; break; default: @@ -67,6 +69,9 @@ status_t SpdifStreamOut::open( } customConfig.sample_rate = config->sample_rate * mRateMultiplier; + customConfig.format = AUDIO_FORMAT_PCM_16_BIT; + customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO; + // Always print this because otherwise it could be very confusing if the // HAL and AudioFlinger are using different formats. // Print before open() because HAL may modify customConfig. diff --git a/services/audioflinger/SpdifStreamOut.h b/services/audioflinger/SpdifStreamOut.h index cb82ac7..d81c064 100644 --- a/services/audioflinger/SpdifStreamOut.h +++ b/services/audioflinger/SpdifStreamOut.h @@ -38,7 +38,8 @@ namespace android { class SpdifStreamOut : public AudioStreamOut { public: - SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags); + SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags, + audio_format_t format); virtual ~SpdifStreamOut() { } @@ -77,8 +78,9 @@ private: class MySPDIFEncoder : public SPDIFEncoder { public: - MySPDIFEncoder(SpdifStreamOut *spdifStreamOut) - : mSpdifStreamOut(spdifStreamOut) + MySPDIFEncoder(SpdifStreamOut *spdifStreamOut, audio_format_t format) + : SPDIFEncoder(format) + , mSpdifStreamOut(spdifStreamOut) { } diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 4efb3d7..b30fd20 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -86,7 +86,13 @@ #define ALOGVV(a...) do { } while(0) #endif +// TODO: Move these macro/inlines to a header file. #define max(a, b) ((a) > (b) ? (a) : (b)) +template <typename T> +static inline T min(const T& a, const T& b) +{ + return a < b ? a : b; +} namespace android { @@ -1602,13 +1608,19 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac // If you change this calculation, also review the start threshold which is related. if (!(*flags & IAudioFlinger::TRACK_FAST) && audio_is_linear_pcm(format) && sharedBuffer == 0) { + // this must match AudioTrack.cpp calculateMinFrameCount(). + // TODO: Move to a common library uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); if (minBufCount < 2) { minBufCount = 2; } + // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack + // or the client should compute and pass in a larger buffer request. size_t minFrameCount = - minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate); + minBufCount * sourceFramesNeededWithTimestretch( + sampleRate, mNormalFrameCount, + mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); if (frameCount < minFrameCount) { // including frameCount == 0 frameCount = minFrameCount; } @@ -3586,21 +3598,17 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed // during last round size_t desiredFrames; - uint32_t sr = track->sampleRate(); - if (sr == mSampleRate) { - desiredFrames = mNormalFrameCount; - } else { - desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); - // add frames already consumed but not yet released by the resampler - // because mAudioTrackServerProxy->framesReady() will include these frames - desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); -#if 0 - // the minimum track buffer size is normally twice the number of frames necessary - // to fill one buffer and the resampler should not leave more than one buffer worth - // of unreleased frames after each pass, but just in case... - ALOG_ASSERT(desiredFrames <= cblk->frameCount_); -#endif - } + const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); + float speed, pitch; + track->mAudioTrackServerProxy->getPlaybackRate(&speed, &pitch); + + desiredFrames = sourceFramesNeededWithTimestretch( + sampleRate, mNormalFrameCount, mSampleRate, speed); + // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. + // add frames already consumed but not yet released by the resampler + // because mAudioTrackServerProxy->framesReady() will include these frames + desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); + uint32_t minFrames = 1; if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { @@ -3763,6 +3771,17 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac AudioMixer::RESAMPLE, AudioMixer::SAMPLE_RATE, (void *)(uintptr_t)reqSampleRate); + + // set the playback rate as an float array {speed, pitch} + float playbackRate[2]; + track->mAudioTrackServerProxy->getPlaybackRate( + &playbackRate[0] /*speed*/, &playbackRate[1] /*pitch*/); + mAudioMixer->setParameter( + name, + AudioMixer::TIMESTRETCH, + AudioMixer::PLAYBACK_RATE, + playbackRate); + /* * Select the appropriate output buffer for the track. * @@ -5290,7 +5309,6 @@ failed: ; // FIXME mNormalSource } - AudioFlinger::RecordThread::~RecordThread() { if (mFastCapture != 0) { @@ -5594,6 +5612,9 @@ reacquire_wakelock: continue; } + // TODO: This code probably should be moved to RecordTrack. + // TODO: Update the activeTrack buffer converter in case of reconfigure. + enum { OVERRUN_UNKNOWN, OVERRUN_TRUE, @@ -5608,131 +5629,28 @@ reacquire_wakelock: size_t framesOut = activeTrack->mSink.frameCount; LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); - int32_t front = activeTrack->mRsmpInFront; - ssize_t filled = rear - front; + // check available frames and handle overrun conditions + // if the record track isn't draining fast enough. + bool hasOverrun; size_t framesIn; - - if (filled < 0) { - // should not happen, but treat like a massive overrun and re-sync - framesIn = 0; - activeTrack->mRsmpInFront = rear; - overrun = OVERRUN_TRUE; - } else if ((size_t) filled <= mRsmpInFrames) { - framesIn = (size_t) filled; - } else { - // client is not keeping up with server, but give it latest data - framesIn = mRsmpInFrames; - activeTrack->mRsmpInFront = front = rear - framesIn; + activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); + if (hasOverrun) { overrun = OVERRUN_TRUE; } - if (framesOut == 0 || framesIn == 0) { break; } - if (activeTrack->mResampler == NULL) { - // no resampling - if (framesIn > framesOut) { - framesIn = framesOut; - } else { - framesOut = framesIn; - } - int8_t *dst = activeTrack->mSink.i8; - while (framesIn > 0) { - front &= mRsmpInFramesP2 - 1; - size_t part1 = mRsmpInFramesP2 - front; - if (part1 > framesIn) { - part1 = framesIn; - } - int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); - if (mChannelCount == activeTrack->mChannelCount) { - memcpy(dst, src, part1 * mFrameSize); - } else if (mChannelCount == 1) { - upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, - part1); - } else { - downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, - (const int16_t *)src, part1); - } - dst += part1 * activeTrack->mFrameSize; - front += part1; - framesIn -= part1; - } - activeTrack->mRsmpInFront += framesOut; - - } else { - // resampling - // FIXME framesInNeeded should really be part of resampler API, and should - // depend on the SRC ratio - // to keep mRsmpInBuffer full so resampler always has sufficient input - size_t framesInNeeded; - // FIXME only re-calculate when it changes, and optimize for common ratios - // Do not precompute in/out because floating point is not associative - // e.g. a*b/c != a*(b/c). - const double in(mSampleRate); - const double out(activeTrack->mSampleRate); - framesInNeeded = ceil(framesOut * in / out) + 1; - ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", - framesInNeeded, framesOut, in / out); - // Although we theoretically have framesIn in circular buffer, some of those are - // unreleased frames, and thus must be discounted for purpose of budgeting. - size_t unreleased = activeTrack->mRsmpInUnrel; - framesIn = framesIn > unreleased ? framesIn - unreleased : 0; - if (framesIn < framesInNeeded) { - ALOGV("not enough to resample: have %u frames in but need %u in to " - "produce %u out given in/out ratio of %.4g", - framesIn, framesInNeeded, framesOut, in / out); - size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; - LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); - if (newFramesOut == 0) { - break; - } - framesInNeeded = ceil(newFramesOut * in / out) + 1; - ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", - framesInNeeded, newFramesOut, out / in); - LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); - ALOGV("success 2: have %u frames in and need %u in to produce %u out " - "given in/out ratio of %.4g", - framesIn, framesInNeeded, newFramesOut, in / out); - framesOut = newFramesOut; - } else { - ALOGV("success 1: have %u in and need %u in to produce %u out " - "given in/out ratio of %.4g", - framesIn, framesInNeeded, framesOut, in / out); - } - - // reallocate mRsmpOutBuffer as needed; we will grow but never shrink - if (activeTrack->mRsmpOutFrameCount < framesOut) { - // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? - delete[] activeTrack->mRsmpOutBuffer; - // resampler always outputs stereo - activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; - activeTrack->mRsmpOutFrameCount = framesOut; - } - - // resampler accumulates, but we only have one source track - memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); - activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, - // FIXME how about having activeTrack implement this interface itself? - activeTrack->mResamplerBufferProvider - /*this*/ /* AudioBufferProvider* */); - // ditherAndClamp() works as long as all buffers returned by - // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. - if (activeTrack->mChannelCount == 1) { - // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t - ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, - framesOut); - // the resampler always outputs stereo samples: - // do post stereo to mono conversion - downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, - (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); - } else { - ditherAndClamp((int32_t *)activeTrack->mSink.raw, - activeTrack->mRsmpOutBuffer, framesOut); - } - // now done with mRsmpOutBuffer - - } + // Don't allow framesOut to be larger than what is possible with resampling + // from framesIn. + // This isn't strictly necessary but helps limit buffer resizing in + // RecordBufferConverter. TODO: remove when no longer needed. + framesOut = min(framesOut, + destinationFramesPossible( + framesIn, mSampleRate, activeTrack->mSampleRate)); + // process frames from the RecordThread buffer provider to the RecordTrack buffer + framesOut = activeTrack->mRecordBufferConverter->convert( + activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { overrun = OVERRUN_FALSE; @@ -6041,12 +5959,9 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac // was initialized to some value closer to the thread's mRsmpInFront, then the track could // see previously buffered data before it called start(), but with greater risk of overrun. - recordTrack->mRsmpInFront = mRsmpInRear; - recordTrack->mRsmpInUnrel = 0; - // FIXME why reset? - if (recordTrack->mResampler != NULL) { - recordTrack->mResampler->reset(); - } + recordTrack->mResamplerBufferProvider->reset(); + // clear any converter state as new data will be discontinuous + recordTrack->mRecordBufferConverter->reset(); recordTrack->mState = TrackBase::STARTING_2; // signal thread to start mWaitWorkCV.broadcast(); @@ -6222,12 +6137,52 @@ void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args write(fd, result.string(), result.size()); } + +void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() +{ + sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); + RecordThread *recordThread = (RecordThread *) threadBase.get(); + mRsmpInFront = recordThread->mRsmpInRear; + mRsmpInUnrel = 0; +} + +void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( + size_t *framesAvailable, bool *hasOverrun) +{ + sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); + RecordThread *recordThread = (RecordThread *) threadBase.get(); + const int32_t rear = recordThread->mRsmpInRear; + const int32_t front = mRsmpInFront; + const ssize_t filled = rear - front; + + size_t framesIn; + bool overrun = false; + if (filled < 0) { + // should not happen, but treat like a massive overrun and re-sync + framesIn = 0; + mRsmpInFront = rear; + overrun = true; + } else if ((size_t) filled <= recordThread->mRsmpInFrames) { + framesIn = (size_t) filled; + } else { + // client is not keeping up with server, but give it latest data + framesIn = recordThread->mRsmpInFrames; + mRsmpInFront = /* front = */ rear - framesIn; + overrun = true; + } + if (framesAvailable != NULL) { + *framesAvailable = framesIn; + } + if (hasOverrun != NULL) { + *hasOverrun = overrun; + } +} + // AudioBufferProvider interface status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( AudioBufferProvider::Buffer* buffer, int64_t pts __unused) { - RecordTrack *activeTrack = mRecordTrack; - sp<ThreadBase> threadBase = activeTrack->mThread.promote(); + sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); if (threadBase == 0) { buffer->frameCount = 0; buffer->raw = NULL; @@ -6235,7 +6190,7 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( } RecordThread *recordThread = (RecordThread *) threadBase.get(); int32_t rear = recordThread->mRsmpInRear; - int32_t front = activeTrack->mRsmpInFront; + int32_t front = mRsmpInFront; ssize_t filled = rear - front; // FIXME should not be P2 (don't want to increase latency) // FIXME if client not keeping up, discard @@ -6252,17 +6207,16 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( part1 = ask; } if (part1 == 0) { - // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty - LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); + // out of data is fine since the resampler will return a short-count. buffer->raw = NULL; buffer->frameCount = 0; - activeTrack->mRsmpInUnrel = 0; + mRsmpInUnrel = 0; return NOT_ENOUGH_DATA; } buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; buffer->frameCount = part1; - activeTrack->mRsmpInUnrel = part1; + mRsmpInUnrel = part1; return NO_ERROR; } @@ -6270,18 +6224,197 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( AudioBufferProvider::Buffer* buffer) { - RecordTrack *activeTrack = mRecordTrack; size_t stepCount = buffer->frameCount; if (stepCount == 0) { return; } - ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); - activeTrack->mRsmpInUnrel -= stepCount; - activeTrack->mRsmpInFront += stepCount; + ALOG_ASSERT(stepCount <= mRsmpInUnrel); + mRsmpInUnrel -= stepCount; + mRsmpInFront += stepCount; buffer->raw = NULL; buffer->frameCount = 0; } +AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( + audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, + uint32_t srcSampleRate, + audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, + uint32_t dstSampleRate) : + mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars + // mSrcFormat + // mSrcSampleRate + // mDstChannelMask + // mDstFormat + // mDstSampleRate + // mSrcChannelCount + // mDstChannelCount + // mDstFrameSize + mBuf(NULL), mBufFrames(0), mBufFrameSize(0), + mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0) +{ + (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, + dstChannelMask, dstFormat, dstSampleRate); +} + +AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { + free(mBuf); + delete mResampler; + free(mRsmpOutBuffer); +} + +size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, + AudioBufferProvider *provider, size_t frames) +{ + if (mSrcSampleRate == mDstSampleRate) { + ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", + mSrcSampleRate, mSrcFormat, mDstFormat); + + AudioBufferProvider::Buffer buffer; + for (size_t i = frames; i > 0; ) { + buffer.frameCount = i; + status_t status = provider->getNextBuffer(&buffer, 0); + if (status != OK || buffer.frameCount == 0) { + frames -= i; // cannot fill request. + break; + } + // convert to destination buffer + convert(dst, buffer.raw, buffer.frameCount); + + dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; + i -= buffer.frameCount; + provider->releaseBuffer(&buffer); + } + } else { + ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", + mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); + + // reallocate mRsmpOutBuffer as needed; we will grow but never shrink + if (mRsmpOutFrameCount < frames) { + // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? + free(mRsmpOutBuffer); + // resampler always outputs stereo (FOR NOW) + (void)posix_memalign(&mRsmpOutBuffer, 32, frames * FCC_2 * sizeof(int32_t) /*Q4.27*/); + mRsmpOutFrameCount = frames; + } + // resampler accumulates, but we only have one source track + memset(mRsmpOutBuffer, 0, frames * FCC_2 * sizeof(int32_t)); + frames = mResampler->resample((int32_t*)mRsmpOutBuffer, frames, provider); + + // convert to destination buffer + convert(dst, mRsmpOutBuffer, frames); + } + return frames; +} + +status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( + audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, + uint32_t srcSampleRate, + audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, + uint32_t dstSampleRate) +{ + // quick evaluation if there is any change. + if (mSrcFormat == srcFormat + && mSrcChannelMask == srcChannelMask + && mSrcSampleRate == srcSampleRate + && mDstFormat == dstFormat + && mDstChannelMask == dstChannelMask + && mDstSampleRate == dstSampleRate) { + return NO_ERROR; + } + + const bool valid = + audio_is_input_channel(srcChannelMask) + && audio_is_input_channel(dstChannelMask) + && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) + && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) + && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) + ; // no upsampling checks for now + if (!valid) { + return BAD_VALUE; + } + + mSrcFormat = srcFormat; + mSrcChannelMask = srcChannelMask; + mSrcSampleRate = srcSampleRate; + mDstFormat = dstFormat; + mDstChannelMask = dstChannelMask; + mDstSampleRate = dstSampleRate; + + // compute derived parameters + mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); + mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); + mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); + + // do we need a format buffer? + if (mSrcFormat != mDstFormat && mDstChannelCount != mSrcChannelCount) { + mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); + } else { + mBufFrameSize = 0; + } + mBufFrames = 0; // force the buffer to be resized. + + // do we need to resample? + if (mSrcSampleRate != mDstSampleRate) { + if (mResampler != NULL) { + delete mResampler; + } + mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, + mSrcChannelCount, mDstSampleRate); // may seem confusing... + mResampler->setSampleRate(mSrcSampleRate); + mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); + } + return NO_ERROR; +} + +void AudioFlinger::RecordThread::RecordBufferConverter::convert( + void *dst, /*const*/ void *src, size_t frames) +{ + // check if a memcpy will do + if (mResampler == NULL + && mSrcChannelCount == mDstChannelCount + && mSrcFormat == mDstFormat) { + memcpy(dst, src, + frames * mDstChannelCount * audio_bytes_per_sample(mDstFormat)); + return; + } + // reallocate buffer if needed + if (mBufFrameSize != 0 && mBufFrames < frames) { + free(mBuf); + mBufFrames = frames; + (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); + } + // do processing + if (mResampler != NULL) { + // src channel count is always >= 2. + void *dstBuf = mBuf != NULL ? mBuf : dst; + // ditherAndClamp() works as long as all buffers returned by + // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. + if (mDstChannelCount == 1) { + // the resampler always outputs stereo samples. + // FIXME: this rewrites back into src + ditherAndClamp((int32_t *)src, (const int32_t *)src, frames); + downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf, + (const int16_t *)src, frames); + } else { + ditherAndClamp((int32_t *)dstBuf, (const int32_t *)src, frames); + } + } else if (mSrcChannelCount != mDstChannelCount) { + void *dstBuf = mBuf != NULL ? mBuf : dst; + if (mSrcChannelCount == 1) { + upmix_to_stereo_i16_from_mono_i16((int16_t *)dstBuf, (const int16_t *)src, + frames); + } else { + downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf, + (const int16_t *)src, frames); + } + } + if (mSrcFormat != mDstFormat) { + void *srcBuf = mBuf != NULL ? mBuf : src; + memcpy_by_audio_format(dst, mDstFormat, srcBuf, mSrcFormat, + frames * mDstChannelCount); + } +} + bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, status_t& status) { @@ -6303,7 +6436,7 @@ bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValueP reconfig = true; } if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { - if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { + if (!audio_is_linear_pcm((audio_format_t) value)) { status = BAD_VALUE; } else { reqFormat = (audio_format_t) value; @@ -6377,10 +6510,10 @@ bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValueP } if (reconfig) { if (status == BAD_VALUE && - reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && - reqFormat == AUDIO_FORMAT_PCM_16_BIT && + audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && + audio_is_linear_pcm(reqFormat) && (mInput->stream->common.get_sample_rate(&mInput->stream->common) - <= (2 * samplingRate)) && + <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && audio_channel_count_from_in_mask( mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && (channelMask == AUDIO_CHANNEL_IN_MONO || @@ -6451,6 +6584,8 @@ void AudioFlinger::RecordThread::readInputParameters_l() // The value is somewhat arbitrary, and could probably be even larger. // A larger value should allow more old data to be read after a track calls start(), // without increasing latency. + // + // Note this is independent of the maximum downsampling ratio permitted for capture. mRsmpInFrames = mFrameCount * 7; mRsmpInFramesP2 = roundup(mRsmpInFrames); delete[] mRsmpInBuffer; diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index d600ea9..27bc56b 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -1036,17 +1036,127 @@ class RecordThread : public ThreadBase public: class RecordTrack; + + /* The ResamplerBufferProvider is used to retrieve recorded input data from the + * RecordThread. It maintains local state on the relative position of the read + * position of the RecordTrack compared with the RecordThread. + */ class ResamplerBufferProvider : public AudioBufferProvider - // derives from AudioBufferProvider interface for use by resampler { public: - ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { } + ResamplerBufferProvider(RecordTrack* recordTrack) : + mRecordTrack(recordTrack), + mRsmpInUnrel(0), mRsmpInFront(0) { } virtual ~ResamplerBufferProvider() { } + + // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, + // skipping any previous data read from the hal. + virtual void reset(); + + /* Synchronizes RecordTrack position with the RecordThread. + * Calculates available frames and handle overruns if the RecordThread + * has advanced faster than the ResamplerBufferProvider has retrieved data. + * TODO: why not do this for every getNextBuffer? + * + * Parameters + * framesAvailable: pointer to optional output size_t to store record track + * frames available. + * hasOverrun: pointer to optional boolean, returns true if track has overrun. + */ + + virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); + // AudioBufferProvider interface virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); private: RecordTrack * const mRecordTrack; + size_t mRsmpInUnrel; // unreleased frames remaining from + // most recent getNextBuffer + // for debug only + int32_t mRsmpInFront; // next available frame + // rolling counter that is never cleared + }; + + /* The RecordBufferConverter is used for format, channel, and sample rate + * conversion for a RecordTrack. + * + * TODO: Self contained, so move to a separate file later. + * + * RecordBufferConverter uses the convert() method rather than exposing a + * buffer provider interface; this is to save a memory copy. + */ + class RecordBufferConverter + { + public: + RecordBufferConverter( + audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, + uint32_t srcSampleRate, + audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, + uint32_t dstSampleRate); + + ~RecordBufferConverter(); + + /* Converts input data from an AudioBufferProvider by format, channelMask, + * and sampleRate to a destination buffer. + * + * Parameters + * dst: buffer to place the converted data. + * provider: buffer provider to obtain source data. + * frames: number of frames to convert + * + * Returns the number of frames converted. + */ + size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); + + // returns NO_ERROR if constructor was successful + status_t initCheck() const { + // mSrcChannelMask set on successful updateParameters + return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; + } + + // allows dynamic reconfigure of all parameters + status_t updateParameters( + audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, + uint32_t srcSampleRate, + audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, + uint32_t dstSampleRate); + + // called to reset resampler buffers on record track discontinuity + void reset() { + if (mResampler != NULL) { + mResampler->reset(); + } + } + + private: + // internal convert function for format and channel mask. + void convert(void *dst, /*const*/ void *src, size_t frames); + + // user provided information + audio_channel_mask_t mSrcChannelMask; + audio_format_t mSrcFormat; + uint32_t mSrcSampleRate; + audio_channel_mask_t mDstChannelMask; + audio_format_t mDstFormat; + uint32_t mDstSampleRate; + + // derived information + uint32_t mSrcChannelCount; + uint32_t mDstChannelCount; + size_t mDstFrameSize; + + // format conversion buffer + void *mBuf; + size_t mBufFrames; + size_t mBufFrameSize; + + // resampler info + AudioResampler *mResampler; + // interleaved stereo pairs of fixed-point Q4.27 or float depending on resampler + void *mRsmpOutBuffer; + // current allocated frame count for the above, which may be larger than needed + size_t mRsmpOutFrameCount; }; #include "RecordTracks.h" diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index dc9f249..da2d634 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -903,9 +903,14 @@ status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& times mPreviousTimestampValid = false; return INVALID_OPERATION; } + // FIXME Not accurate under dynamic changes of sample rate and speed. + // Do not use track's mSampleRate as it is not current for mixer tracks. + uint32_t sampleRate = mAudioTrackServerProxy->getSampleRate(); + float speed, pitch; + mAudioTrackServerProxy->getPlaybackRate(&speed, &pitch); uint32_t unpresentedFrames = - ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / - playbackThread->mSampleRate; + ((double) playbackThread->mLatchQ.mUnpresentedFrames * sampleRate * speed) + / playbackThread->mSampleRate; // FIXME Since we're using a raw pointer as the key, it is theoretically possible // for a brand new track to share the same address as a recently destroyed // track, and thus for us to get the frames released of the wrong track. @@ -1861,13 +1866,14 @@ void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread, + audio_stream_type_t streamType, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, IAudioFlinger::track_flags_t flags) - : Track(playbackThread, NULL, AUDIO_STREAM_PATCH, + : Track(playbackThread, NULL, streamType, sampleRate, format, channelMask, frameCount, buffer, 0, 0, getuid(), flags, TYPE_PATCH), mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)) @@ -1989,29 +1995,30 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) : ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE), type), - mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0), - // See real initialization of mRsmpInFront at RecordThread::start() - mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) + mOverflow(false), + mFramesToDrop(0) { if (mCblk == NULL) { return; } + mRecordBufferConverter = new RecordBufferConverter( + thread->mChannelMask, thread->mFormat, thread->mSampleRate, + channelMask, format, sampleRate); + // Check if the RecordBufferConverter construction was successful. + // If not, don't continue with construction. + // + // NOTE: It would be extremely rare that the record track cannot be created + // for the current device, but a pending or future device change would make + // the record track configuration valid. + if (mRecordBufferConverter->initCheck() != NO_ERROR) { + ALOGE("RecordTrack unable to create record buffer converter"); + return; + } + mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize, !isExternalTrack()); - - uint32_t channelCount = audio_channel_count_from_in_mask(channelMask); - // FIXME I don't understand either of the channel count checks - if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && - channelCount <= FCC_2) { - // sink SR - mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, - thread->mChannelCount, sampleRate); - // source SR - mResampler->setSampleRate(thread->mSampleRate); - mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); - mResamplerBufferProvider = new ResamplerBufferProvider(this); - } + mResamplerBufferProvider = new ResamplerBufferProvider(this); if (flags & IAudioFlinger::TRACK_FAST) { ALOG_ASSERT(thread->mFastTrackAvail); @@ -2022,11 +2029,19 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( AudioFlinger::RecordThread::RecordTrack::~RecordTrack() { ALOGV("%s", __func__); - delete mResampler; - delete[] mRsmpOutBuffer; + delete mRecordBufferConverter; delete mResamplerBufferProvider; } +status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const +{ + status_t status = TrackBase::initCheck(); + if (status == NO_ERROR && mServerProxy == 0) { + status = BAD_VALUE; + } + return status; +} + // AudioBufferProvider interface status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts __unused) diff --git a/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk index 8604ef5..536eb93 100644 --- a/services/audioflinger/tests/Android.mk +++ b/services/audioflinger/tests/Android.mk @@ -39,11 +39,13 @@ endif LOCAL_SRC_FILES:= \ test-mixer.cpp \ ../AudioMixer.cpp.arm \ + ../BufferProviders.cpp LOCAL_C_INCLUDES := \ $(call include-path-for, audio-effects) \ $(call include-path-for, audio-utils) \ - frameworks/av/services/audioflinger + frameworks/av/services/audioflinger \ + external/sonic LOCAL_STATIC_LIBRARIES := \ libsndfile @@ -57,7 +59,8 @@ LOCAL_SHARED_LIBRARIES := \ libdl \ libcutils \ libutils \ - liblog + liblog \ + libsonic LOCAL_MODULE:= test-mixer diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp index d6217ba..9e375db 100644 --- a/services/audioflinger/tests/resampler_tests.cpp +++ b/services/audioflinger/tests/resampler_tests.cpp @@ -48,7 +48,10 @@ void resample(int channels, void *output, if (thisFrames == 0 || thisFrames > outputFrames - i) { thisFrames = outputFrames - i; } - resampler->resample((int32_t*) output + channels*i, thisFrames, provider); + size_t framesResampled = resampler->resample( + (int32_t*) output + channels*i, thisFrames, provider); + // we should have enough buffer space, so there is no short count. + ASSERT_EQ(thisFrames, framesResampled); i += thisFrames; } } diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h index 116d0d6..48d0e29 100644 --- a/services/audiopolicy/AudioPolicyInterface.h +++ b/services/audiopolicy/AudioPolicyInterface.h @@ -110,6 +110,7 @@ public: audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + int selectedDeviceId, const audio_offload_info_t *offloadInfo) = 0; // indicates to the audio policy manager that the output starts being used by corresponding stream. virtual status_t startOutput(audio_io_handle_t output, diff --git a/services/audiopolicy/common/include/Volume.h b/services/audiopolicy/common/include/Volume.h index a4cc759..4205589 100755 --- a/services/audiopolicy/common/include/Volume.h +++ b/services/audiopolicy/common/include/Volume.h @@ -18,6 +18,10 @@ #include <system/audio.h> #include <utils/Log.h> +#include <math.h> + +// Absolute min volume in dB (can be represented in single precision normal float value) +#define VOLUME_MIN_DB (-758) class VolumeCurvePoint { @@ -32,7 +36,7 @@ public: /** * 4 points to define the volume attenuation curve, each characterized by the volume * index (from 0 to 100) at which they apply, and the attenuation in dB at that index. - * we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl() + * we use 100 steps to avoid rounding errors when computing the volume in volIndexToDb() * * @todo shall become configurable */ @@ -134,4 +138,20 @@ public: } } + static inline float DbToAmpl(float decibels) + { + if (decibels <= VOLUME_MIN_DB) { + return 0.0f; + } + return exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) + } + + static inline float AmplToDb(float amplification) + { + if (amplification == 0) { + return VOLUME_MIN_DB; + } + return 20 * log10(amplification); + } + }; diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk index 71ba1cb..7c265aa 100644 --- a/services/audiopolicy/common/managerdefinitions/Android.mk +++ b/services/audiopolicy/common/managerdefinitions/Android.mk @@ -25,6 +25,7 @@ LOCAL_SHARED_LIBRARIES := \ LOCAL_C_INCLUDES += \ $(LOCAL_PATH)/include \ $(TOPDIR)frameworks/av/services/audiopolicy/common/include \ + $(TOPDIR)frameworks/av/services/audiopolicy LOCAL_EXPORT_C_INCLUDE_DIRS := \ $(LOCAL_PATH)/include diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h index 7536a37..18bcfdb 100644 --- a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h +++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h @@ -34,12 +34,11 @@ class AudioInputDescriptor: public AudioPortConfig public: AudioInputDescriptor(const sp<IOProfile>& profile); void setIoHandle(audio_io_handle_t ioHandle); - + audio_port_handle_t getId() const; audio_module_handle_t getModuleHandle() const; status_t dump(int fd); - audio_port_handle_t mId; audio_io_handle_t mIoHandle; // input handle audio_devices_t mDevice; // current device this input is routed to AudioMix *mPolicyMix; // non NULL when used by a dynamic policy @@ -57,6 +56,9 @@ public: const struct audio_port_config *srcConfig = NULL) const; virtual sp<AudioPort> getAudioPort() const { return mProfile; } void toAudioPort(struct audio_port *port) const; + +private: + audio_port_handle_t mId; }; class AudioInputCollection : diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h index 43ee691..f1aee46 100644 --- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h +++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h @@ -27,24 +27,36 @@ namespace android { class IOProfile; class AudioMix; +class AudioPolicyClientInterface; // descriptor for audio outputs. Used to maintain current configuration of each opened audio output // and keep track of the usage of this output by each audio stream type. class AudioOutputDescriptor: public AudioPortConfig { public: - AudioOutputDescriptor(const sp<IOProfile>& profile); + AudioOutputDescriptor(const sp<AudioPort>& port, + AudioPolicyClientInterface *clientInterface); + virtual ~AudioOutputDescriptor() {} status_t dump(int fd); + void log(const char* indent); + + audio_port_handle_t getId() const; + virtual audio_devices_t device() const; + virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc); + virtual audio_devices_t supportedDevices(); + virtual bool isDuplicated() const { return false; } + virtual uint32_t latency() { return 0; } + virtual bool isFixedVolume(audio_devices_t device); + virtual sp<AudioOutputDescriptor> subOutput1() { return 0; } + virtual sp<AudioOutputDescriptor> subOutput2() { return 0; } + virtual bool setVolume(float volume, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t delayMs, + bool force); + virtual void changeRefCount(audio_stream_type_t stream, int delta); - audio_devices_t device() const; - void changeRefCount(audio_stream_type_t stream, int delta); - - void setIoHandle(audio_io_handle_t ioHandle); - bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } - audio_devices_t supportedDevices(); - uint32_t latency(); - bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc); bool isActive(uint32_t inPastMs = 0) const; bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0, @@ -52,32 +64,69 @@ public: virtual void toAudioPortConfig(struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig = NULL) const; - virtual sp<AudioPort> getAudioPort() const { return mProfile; } - void toAudioPort(struct audio_port *port) const; + virtual sp<AudioPort> getAudioPort() const { return mPort; } + virtual void toAudioPort(struct audio_port *port) const; audio_module_handle_t getModuleHandle() const; - audio_port_handle_t mId; - audio_io_handle_t mIoHandle; // output handle - uint32_t mLatency; // - audio_output_flags_t mFlags; // + sp<AudioPort> mPort; audio_devices_t mDevice; // current device this output is routed to - AudioMix *mPolicyMix; // non NULL when used by a dynamic policy audio_patch_handle_t mPatchHandle; uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output nsecs_t mStopTime[AUDIO_STREAM_CNT]; - sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output - sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output - float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume + float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume in dB int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter - const sp<IOProfile> mProfile; // I/O profile this output derives from bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible // device selection. See checkDeviceMuteStrategies() + AudioPolicyClientInterface *mClientInterface; + +protected: + audio_port_handle_t mId; +}; + +// Audio output driven by a software mixer in audio flinger. +class SwAudioOutputDescriptor: public AudioOutputDescriptor +{ +public: + SwAudioOutputDescriptor(const sp<IOProfile>& profile, + AudioPolicyClientInterface *clientInterface); + virtual ~SwAudioOutputDescriptor() {} + + status_t dump(int fd); + + void setIoHandle(audio_io_handle_t ioHandle); + + virtual audio_devices_t device() const; + virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc); + virtual audio_devices_t supportedDevices(); + virtual uint32_t latency(); + virtual bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } + virtual bool isFixedVolume(audio_devices_t device); + virtual sp<AudioOutputDescriptor> subOutput1() { return mOutput1; } + virtual sp<AudioOutputDescriptor> subOutput2() { return mOutput2; } + virtual void changeRefCount(audio_stream_type_t stream, int delta); + virtual bool setVolume(float volume, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t delayMs, + bool force); + + virtual void toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig = NULL) const; + virtual void toAudioPort(struct audio_port *port) const; + + const sp<IOProfile> mProfile; // I/O profile this output derives from + audio_io_handle_t mIoHandle; // output handle + uint32_t mLatency; // + audio_output_flags_t mFlags; // + AudioMix *mPolicyMix; // non NULL when used by a dynamic policy + sp<SwAudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output + sp<SwAudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) }; -class AudioOutputCollection : - public DefaultKeyedVector< audio_io_handle_t, sp<AudioOutputDescriptor> > +class SwAudioOutputCollection : + public DefaultKeyedVector< audio_io_handle_t, sp<SwAudioOutputDescriptor> > { public: bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; @@ -96,9 +145,9 @@ public: */ audio_io_handle_t getA2dpOutput() const; - sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const; + sp<SwAudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const; - sp<AudioOutputDescriptor> getPrimaryOutput() const; + sp<SwAudioOutputDescriptor> getPrimaryOutput() const; /** * return true if any output is playing anything besides the stream to ignore diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h index 988aed6..d51f4e1 100644 --- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h +++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h @@ -24,7 +24,7 @@ namespace android { -class AudioOutputDescriptor; +class SwAudioOutputDescriptor; /** * custom mix entry in mPolicyMixes @@ -33,19 +33,19 @@ class AudioPolicyMix : public RefBase { public: AudioPolicyMix() {} - const sp<AudioOutputDescriptor> &getOutput() const; + const sp<SwAudioOutputDescriptor> &getOutput() const; - void setOutput(sp<AudioOutputDescriptor> &output); + void setOutput(sp<SwAudioOutputDescriptor> &output); void clearOutput(); - android::AudioMix &getMix(); + android::AudioMix *getMix(); void setMix(AudioMix &mix); private: AudioMix mMix; // Audio policy mix descriptor - sp<AudioOutputDescriptor> mOutput; // Corresponding output stream + sp<SwAudioOutputDescriptor> mOutput; // Corresponding output stream }; @@ -58,24 +58,24 @@ public: status_t unregisterMix(String8 address); - void closeOutput(sp<AudioOutputDescriptor> &desc); + void closeOutput(sp<SwAudioOutputDescriptor> &desc); /** * Try to find an output descriptor for the given attributes. * - * @param[in] attributes to consider for the research of output descriptor. + * @param[in] attributes to consider fowr the research of output descriptor. * @param[out] desc to return if an output could be found. * * @return NO_ERROR if an output was found for the given attribute (in this case, the * descriptor output param is initialized), error code otherwise. */ - status_t getOutputForAttr(audio_attributes_t attributes, sp<AudioOutputDescriptor> &desc); + status_t getOutputForAttr(audio_attributes_t attributes, sp<SwAudioOutputDescriptor> &desc); audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource, audio_devices_t availableDeviceTypes, AudioMix **policyMix); - status_t getInputMixForAttr(audio_attributes_t attr, AudioMix *&policyMix); + status_t getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix); }; }; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h index 4f7f2bc..1c2c27e 100644 --- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h +++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h @@ -32,13 +32,11 @@ class AudioPort : public virtual RefBase { public: AudioPort(const String8& name, audio_port_type_t type, - audio_port_role_t role, const sp<HwModule>& module); + audio_port_role_t role); virtual ~AudioPort() {} - audio_port_handle_t getHandle() { return mId; } - - void attach(const sp<HwModule>& module); - bool isAttached() { return mId != 0; } + virtual void attach(const sp<HwModule>& module); + bool isAttached() { return mModule != 0; } static audio_port_handle_t getNextUniqueId(); @@ -64,8 +62,12 @@ public: // searches for an exact match status_t checkExactChannelMask(audio_channel_mask_t channelMask) const; // searches for a compatible match, currently implemented for input channel masks only - status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const; - status_t checkFormat(audio_format_t format) const; + status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask, + audio_channel_mask_t *updatedChannelMask) const; + + status_t checkExactFormat(audio_format_t format) const; + // searches for a compatible match, currently implemented for input formats only + status_t checkCompatibleFormat(audio_format_t format, audio_format_t *updatedFormat) const; status_t checkGain(const struct audio_gain_config *gainConfig, int index) const; uint32_t pickSamplingRate() const; @@ -73,11 +75,19 @@ public: audio_format_t pickFormat() const; static const audio_format_t sPcmFormatCompareTable[]; + static int compareFormatsGoodToBad( + const audio_format_t *format1, const audio_format_t *format2) { + // compareFormats sorts from bad to good, we reverse it here + return compareFormats(*format2, *format1); + } static int compareFormats(audio_format_t format1, audio_format_t format2); audio_module_handle_t getModuleHandle() const; + uint32_t getModuleVersion() const; + const char *getModuleName() const; void dump(int fd, int spaces) const; + void log(const char* indent) const; String8 mName; audio_port_type_t mType; @@ -94,13 +104,6 @@ public: uint32_t mFlags; // attribute flags (e.g primary output, // direct output...). - -protected: - //TODO - clarify the role of mId in this case, both an "attached" indicator - // and a unique ID for identifying a port to the (upcoming) selection API, - // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor. - audio_port_handle_t mId; - private: static volatile int32_t mNextUniqueId; }; diff --git a/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h index 14a7d36..f8c4d08 100644 --- a/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h +++ b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h @@ -39,11 +39,12 @@ struct StringToEnum { }; #define STRING_TO_ENUM(string) { #string, string } +#define NAME_TO_ENUM(name, value) { name, value } #ifndef ARRAY_SIZE #define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) #endif -const StringToEnum sDeviceNameToEnumTable[] = { +const StringToEnum sDeviceTypeToEnumTable[] = { STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE), STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER), STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE), @@ -94,6 +95,57 @@ const StringToEnum sDeviceNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK), }; +const StringToEnum sDeviceNameToEnumTable[] = { + NAME_TO_ENUM("Earpiece", AUDIO_DEVICE_OUT_EARPIECE), + NAME_TO_ENUM("Speaker", AUDIO_DEVICE_OUT_SPEAKER), + NAME_TO_ENUM("Speaker Protected", AUDIO_DEVICE_OUT_SPEAKER_SAFE), + NAME_TO_ENUM("Wired Headset", AUDIO_DEVICE_OUT_WIRED_HEADSET), + NAME_TO_ENUM("Wired Headphones", AUDIO_DEVICE_OUT_WIRED_HEADPHONE), + NAME_TO_ENUM("BT SCO", AUDIO_DEVICE_OUT_BLUETOOTH_SCO), + NAME_TO_ENUM("BT SCO Headset", AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET), + NAME_TO_ENUM("BT SCO Car Kit", AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT), + NAME_TO_ENUM("", AUDIO_DEVICE_OUT_ALL_SCO), + NAME_TO_ENUM("BT A2DP Out", AUDIO_DEVICE_OUT_BLUETOOTH_A2DP), + NAME_TO_ENUM("BT A2DP Headphones", AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES), + NAME_TO_ENUM("BT A2DP Speaker", AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), + NAME_TO_ENUM("", AUDIO_DEVICE_OUT_ALL_A2DP), + NAME_TO_ENUM("HDMI Out", AUDIO_DEVICE_OUT_AUX_DIGITAL), + NAME_TO_ENUM("HDMI Out", AUDIO_DEVICE_OUT_HDMI), + NAME_TO_ENUM("Analog Dock Out", AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET), + NAME_TO_ENUM("Digital Dock Out", AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET), + NAME_TO_ENUM("USB Host Out", AUDIO_DEVICE_OUT_USB_ACCESSORY), + NAME_TO_ENUM("USB Device Out", AUDIO_DEVICE_OUT_USB_DEVICE), + NAME_TO_ENUM("", AUDIO_DEVICE_OUT_ALL_USB), + NAME_TO_ENUM("Reroute Submix Out", AUDIO_DEVICE_OUT_REMOTE_SUBMIX), + NAME_TO_ENUM("Telephony Tx", AUDIO_DEVICE_OUT_TELEPHONY_TX), + NAME_TO_ENUM("Line Out", AUDIO_DEVICE_OUT_LINE), + NAME_TO_ENUM("HDMI ARC Out", AUDIO_DEVICE_OUT_HDMI_ARC), + NAME_TO_ENUM("S/PDIF Out", AUDIO_DEVICE_OUT_SPDIF), + NAME_TO_ENUM("FM transceiver Out", AUDIO_DEVICE_OUT_FM), + NAME_TO_ENUM("Aux Line Out", AUDIO_DEVICE_OUT_AUX_LINE), + NAME_TO_ENUM("Ambient Mic", AUDIO_DEVICE_IN_AMBIENT), + NAME_TO_ENUM("Built-In Mic", AUDIO_DEVICE_IN_BUILTIN_MIC), + NAME_TO_ENUM("BT SCO Headset Mic", AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET), + NAME_TO_ENUM("", AUDIO_DEVICE_IN_ALL_SCO), + NAME_TO_ENUM("Wired Headset Mic", AUDIO_DEVICE_IN_WIRED_HEADSET), + NAME_TO_ENUM("HDMI In", AUDIO_DEVICE_IN_AUX_DIGITAL), + NAME_TO_ENUM("HDMI In", AUDIO_DEVICE_IN_HDMI), + NAME_TO_ENUM("Telephony Rx", AUDIO_DEVICE_IN_TELEPHONY_RX), + NAME_TO_ENUM("Telephony Rx", AUDIO_DEVICE_IN_VOICE_CALL), + NAME_TO_ENUM("Built-In Back Mic", AUDIO_DEVICE_IN_BACK_MIC), + NAME_TO_ENUM("Reroute Submix In", AUDIO_DEVICE_IN_REMOTE_SUBMIX), + NAME_TO_ENUM("Analog Dock In", AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET), + NAME_TO_ENUM("Digital Dock In", AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET), + NAME_TO_ENUM("USB Host In", AUDIO_DEVICE_IN_USB_ACCESSORY), + NAME_TO_ENUM("USB Device In", AUDIO_DEVICE_IN_USB_DEVICE), + NAME_TO_ENUM("FM Tuner In", AUDIO_DEVICE_IN_FM_TUNER), + NAME_TO_ENUM("TV Tuner In", AUDIO_DEVICE_IN_TV_TUNER), + NAME_TO_ENUM("Line In", AUDIO_DEVICE_IN_LINE), + NAME_TO_ENUM("S/PDIF In", AUDIO_DEVICE_IN_SPDIF), + NAME_TO_ENUM("BT A2DP In", AUDIO_DEVICE_IN_BLUETOOTH_A2DP), + NAME_TO_ENUM("Loopback In", AUDIO_DEVICE_IN_LOOPBACK), +}; + const StringToEnum sOutputFlagNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT), STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY), diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h index d15f6b4..aa37eec 100644 --- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h +++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h @@ -41,19 +41,22 @@ public: const struct audio_port_config *srcConfig = NULL) const; // AudioPort + virtual void attach(const sp<HwModule>& module); virtual void loadGains(cnode *root); virtual void toAudioPort(struct audio_port *port) const; + audio_port_handle_t getId() const; audio_devices_t type() const { return mDeviceType; } status_t dump(int fd, int spaces, int index) const; + void log() const; String8 mAddress; - audio_port_handle_t mId; static String8 emptyNameStr; private: - audio_devices_t mDeviceType; + audio_devices_t mDeviceType; + audio_port_handle_t mId; friend class DeviceVector; }; diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h index 095e759..ab6fcc1 100644 --- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h +++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h @@ -33,7 +33,7 @@ class HwModule; class IOProfile : public AudioPort { public: - IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module); + IOProfile(const String8& name, audio_port_role_t role); virtual ~IOProfile(); // This method is used for both output and input. @@ -45,7 +45,9 @@ public: uint32_t samplingRate, uint32_t *updatedSamplingRate, audio_format_t format, + audio_format_t *updatedFormat, audio_channel_mask_t channelMask, + audio_channel_mask_t *updatedChannelMask, uint32_t flags) const; void dump(int fd); diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp index fa66728..937160b 100644 --- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp @@ -27,9 +27,9 @@ namespace android { AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile) - : mId(0), mIoHandle(0), + : mIoHandle(0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0), - mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false) + mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false), mId(0) { if (profile != NULL) { mSamplingRate = profile->pickSamplingRate(); @@ -49,9 +49,17 @@ void AudioInputDescriptor::setIoHandle(audio_io_handle_t ioHandle) audio_module_handle_t AudioInputDescriptor::getModuleHandle() const { + if (mProfile == 0) { + return 0; + } return mProfile->getModuleHandle(); } +audio_port_handle_t AudioInputDescriptor::getId() const +{ + return mId; +} + void AudioInputDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig) const { @@ -68,7 +76,7 @@ void AudioInputDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig dstConfig->id = mId; dstConfig->role = AUDIO_PORT_ROLE_SINK; dstConfig->type = AUDIO_PORT_TYPE_MIX; - dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle; + dstConfig->ext.mix.hw_module = getModuleHandle(); dstConfig->ext.mix.handle = mIoHandle; dstConfig->ext.mix.usecase.source = mInputSource; } @@ -80,7 +88,7 @@ void AudioInputDescriptor::toAudioPort(struct audio_port *port) const mProfile->toAudioPort(port); port->id = mId; toAudioPortConfig(&port->active_config); - port->ext.mix.hw_module = mProfile->mModule->mHandle; + port->ext.mix.hw_module = getModuleHandle(); port->ext.mix.handle = mIoHandle; port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL; } @@ -91,7 +99,7 @@ status_t AudioInputDescriptor::dump(int fd) char buffer[SIZE]; String8 result; - snprintf(buffer, SIZE, " ID: %d\n", mId); + snprintf(buffer, SIZE, " ID: %d\n", getId()); result.append(buffer); snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); result.append(buffer); @@ -130,7 +138,7 @@ sp<AudioInputDescriptor> AudioInputCollection::getInputFromId(audio_port_handle_ sp<AudioInputDescriptor> inputDesc = NULL; for (size_t i = 0; i < size(); i++) { inputDesc = valueAt(i); - if (inputDesc->mId == id) { + if (inputDesc->getId() == id) { break; } } diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp index cdb5b51..596aa1d 100644 --- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp @@ -17,9 +17,11 @@ #define LOG_TAG "APM::AudioOutputDescriptor" //#define LOG_NDEBUG 0 +#include <AudioPolicyInterface.h> #include "AudioOutputDescriptor.h" #include "IOProfile.h" #include "AudioGain.h" +#include "Volume.h" #include "HwModule.h" #include <media/AudioPolicy.h> @@ -29,11 +31,10 @@ namespace android { -AudioOutputDescriptor::AudioOutputDescriptor(const sp<IOProfile>& profile) - : mId(0), mIoHandle(0), mLatency(0), - mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), - mPatchHandle(0), - mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0) +AudioOutputDescriptor::AudioOutputDescriptor(const sp<AudioPort>& port, + AudioPolicyClientInterface *clientInterface) + : mPort(port), mDevice(AUDIO_DEVICE_NONE), + mPatchHandle(0), mClientInterface(clientInterface), mId(0) { // clear usage count for all stream types for (int i = 0; i < AUDIO_STREAM_CNT; i++) { @@ -45,66 +46,50 @@ AudioOutputDescriptor::AudioOutputDescriptor(const sp<IOProfile>& profile) for (int i = 0; i < NUM_STRATEGIES; i++) { mStrategyMutedByDevice[i] = false; } - if (profile != NULL) { - mFlags = (audio_output_flags_t)profile->mFlags; - mSamplingRate = profile->pickSamplingRate(); - mFormat = profile->pickFormat(); - mChannelMask = profile->pickChannelMask(); - if (profile->mGains.size() > 0) { - profile->mGains[0]->getDefaultConfig(&mGain); + if (port != NULL) { + mSamplingRate = port->pickSamplingRate(); + mFormat = port->pickFormat(); + mChannelMask = port->pickChannelMask(); + if (port->mGains.size() > 0) { + port->mGains[0]->getDefaultConfig(&mGain); } } } audio_module_handle_t AudioOutputDescriptor::getModuleHandle() const { - return mProfile->getModuleHandle(); + return mPort->getModuleHandle(); } -audio_devices_t AudioOutputDescriptor::device() const +audio_port_handle_t AudioOutputDescriptor::getId() const { - if (isDuplicated()) { - return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); - } else { - return mDevice; - } + return mId; } -void AudioOutputDescriptor::setIoHandle(audio_io_handle_t ioHandle) +audio_devices_t AudioOutputDescriptor::device() const { - mId = AudioPort::getNextUniqueId(); - mIoHandle = ioHandle; + return mDevice; } -uint32_t AudioOutputDescriptor::latency() +audio_devices_t AudioOutputDescriptor::supportedDevices() { - if (isDuplicated()) { - return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; - } else { - return mLatency; - } + return mDevice; } bool AudioOutputDescriptor::sharesHwModuleWith( const sp<AudioOutputDescriptor> outputDesc) { - if (isDuplicated()) { - return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); - } else if (outputDesc->isDuplicated()){ - return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2); + if (outputDesc->isDuplicated()) { + return sharesHwModuleWith(outputDesc->subOutput1()) || + sharesHwModuleWith(outputDesc->subOutput2()); } else { - return (mProfile->mModule == outputDesc->mProfile->mModule); + return (getModuleHandle() == outputDesc->getModuleHandle()); } } void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream, int delta) { - // forward usage count change to attached outputs - if (isDuplicated()) { - mOutput1->changeRefCount(stream, delta); - mOutput2->changeRefCount(stream, delta); - } if ((delta + (int)mRefCount[stream]) < 0) { ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]); @@ -115,15 +100,6 @@ void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream, ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); } -audio_devices_t AudioOutputDescriptor::supportedDevices() -{ - if (isDuplicated()) { - return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); - } else { - return mProfile->mSupportedDevices.types() ; - } -} - bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const { nsecs_t sysTime = 0; @@ -160,12 +136,33 @@ bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream, return false; } + +bool AudioOutputDescriptor::isFixedVolume(audio_devices_t device __unused) +{ + return false; +} + +bool AudioOutputDescriptor::setVolume(float volume, + audio_stream_type_t stream, + audio_devices_t device __unused, + uint32_t delayMs, + bool force) +{ + // We actually change the volume if: + // - the float value returned by computeVolume() changed + // - the force flag is set + if (volume != mCurVolume[stream] || force) { + ALOGV("setVolume() for stream %d, volume %f, delay %d", stream, volume, delayMs); + mCurVolume[stream] = volume; + return true; + } + return false; +} + void AudioOutputDescriptor::toAudioPortConfig( struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig) const { - ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle); - dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN; if (srcConfig != NULL) { @@ -176,22 +173,16 @@ void AudioOutputDescriptor::toAudioPortConfig( dstConfig->id = mId; dstConfig->role = AUDIO_PORT_ROLE_SOURCE; dstConfig->type = AUDIO_PORT_TYPE_MIX; - dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle; - dstConfig->ext.mix.handle = mIoHandle; + dstConfig->ext.mix.hw_module = getModuleHandle(); dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; } void AudioOutputDescriptor::toAudioPort( struct audio_port *port) const { - ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle); - mProfile->toAudioPort(port); + mPort->toAudioPort(port); port->id = mId; - toAudioPortConfig(&port->active_config); - port->ext.mix.hw_module = mProfile->mModule->mHandle; - port->ext.mix.handle = mIoHandle; - port->ext.mix.latency_class = - mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL; + port->ext.mix.hw_module = getModuleHandle(); } status_t AudioOutputDescriptor::dump(int fd) @@ -208,10 +199,6 @@ status_t AudioOutputDescriptor::dump(int fd) result.append(buffer); snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); result.append(buffer); - snprintf(buffer, SIZE, " Latency: %d\n", mLatency); - result.append(buffer); - snprintf(buffer, SIZE, " Flags %08x\n", mFlags); - result.append(buffer); snprintf(buffer, SIZE, " Devices %08x\n", device()); result.append(buffer); snprintf(buffer, SIZE, " Stream volume refCount muteCount\n"); @@ -226,11 +213,165 @@ status_t AudioOutputDescriptor::dump(int fd) return NO_ERROR; } -bool AudioOutputCollection::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const +void AudioOutputDescriptor::log(const char* indent) +{ + ALOGI("%sID: %d,0x%X, [rt:%d fmt:0x%X ch:0x%X]", + indent, mId, mId, mSamplingRate, mFormat, mChannelMask); +} + +// SwAudioOutputDescriptor implementation +SwAudioOutputDescriptor::SwAudioOutputDescriptor( + const sp<IOProfile>& profile, AudioPolicyClientInterface *clientInterface) + : AudioOutputDescriptor(profile, clientInterface), + mProfile(profile), mIoHandle(0), mLatency(0), + mFlags((audio_output_flags_t)0), mPolicyMix(NULL), + mOutput1(0), mOutput2(0), mDirectOpenCount(0) +{ + if (profile != NULL) { + mFlags = (audio_output_flags_t)profile->mFlags; + } +} + +void SwAudioOutputDescriptor::setIoHandle(audio_io_handle_t ioHandle) +{ + mId = AudioPort::getNextUniqueId(); + mIoHandle = ioHandle; +} + + +status_t SwAudioOutputDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " Latency: %d\n", mLatency); + result.append(buffer); + snprintf(buffer, SIZE, " Flags %08x\n", mFlags); + result.append(buffer); + write(fd, result.string(), result.size()); + + AudioOutputDescriptor::dump(fd); + + return NO_ERROR; +} + +audio_devices_t SwAudioOutputDescriptor::device() const +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); + } else { + return mDevice; + } +} + +bool SwAudioOutputDescriptor::sharesHwModuleWith( + const sp<AudioOutputDescriptor> outputDesc) +{ + if (isDuplicated()) { + return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); + } else if (outputDesc->isDuplicated()){ + return sharesHwModuleWith(outputDesc->subOutput1()) || + sharesHwModuleWith(outputDesc->subOutput2()); + } else { + return AudioOutputDescriptor::sharesHwModuleWith(outputDesc); + } +} + +audio_devices_t SwAudioOutputDescriptor::supportedDevices() +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); + } else { + return mProfile->mSupportedDevices.types() ; + } +} + +uint32_t SwAudioOutputDescriptor::latency() +{ + if (isDuplicated()) { + return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; + } else { + return mLatency; + } +} + +void SwAudioOutputDescriptor::changeRefCount(audio_stream_type_t stream, + int delta) +{ + // forward usage count change to attached outputs + if (isDuplicated()) { + mOutput1->changeRefCount(stream, delta); + mOutput2->changeRefCount(stream, delta); + } + AudioOutputDescriptor::changeRefCount(stream, delta); +} + + +bool SwAudioOutputDescriptor::isFixedVolume(audio_devices_t device) +{ + // unit gain if rerouting to external policy + if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { + if (mPolicyMix != NULL) { + ALOGV("max gain when rerouting for output=%d", mIoHandle); + return true; + } + } + return false; +} + +void SwAudioOutputDescriptor::toAudioPortConfig( + struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig) const +{ + + ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle); + AudioOutputDescriptor::toAudioPortConfig(dstConfig, srcConfig); + + dstConfig->ext.mix.handle = mIoHandle; +} + +void SwAudioOutputDescriptor::toAudioPort( + struct audio_port *port) const +{ + ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle); + + AudioOutputDescriptor::toAudioPort(port); + + toAudioPortConfig(&port->active_config); + port->ext.mix.handle = mIoHandle; + port->ext.mix.latency_class = + mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL; +} + +bool SwAudioOutputDescriptor::setVolume(float volume, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t delayMs, + bool force) +{ + bool changed = AudioOutputDescriptor::setVolume(volume, stream, device, delayMs, force); + + if (changed) { + // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is + // enabled + float volume = Volume::DbToAmpl(mCurVolume[stream]); + if (stream == AUDIO_STREAM_BLUETOOTH_SCO) { + mClientInterface->setStreamVolume( + AUDIO_STREAM_VOICE_CALL, volume, mIoHandle, delayMs); + } + mClientInterface->setStreamVolume(stream, volume, mIoHandle, delayMs); + } + return changed; +} + +// SwAudioOutputCollection implementation + +bool SwAudioOutputCollection::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const { nsecs_t sysTime = systemTime(); for (size_t i = 0; i < this->size(); i++) { - const sp<AudioOutputDescriptor> outputDesc = this->valueAt(i); + const sp<SwAudioOutputDescriptor> outputDesc = this->valueAt(i); if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) { return true; } @@ -238,12 +379,12 @@ bool AudioOutputCollection::isStreamActive(audio_stream_type_t stream, uint32_t return false; } -bool AudioOutputCollection::isStreamActiveRemotely(audio_stream_type_t stream, +bool SwAudioOutputCollection::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const { nsecs_t sysTime = systemTime(); for (size_t i = 0; i < size(); i++) { - const sp<AudioOutputDescriptor> outputDesc = valueAt(i); + const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i); if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && outputDesc->isStreamActive(stream, inPastMs, sysTime)) { // do not consider re routing (when the output is going to a dynamic policy) @@ -256,10 +397,10 @@ bool AudioOutputCollection::isStreamActiveRemotely(audio_stream_type_t stream, return false; } -audio_io_handle_t AudioOutputCollection::getA2dpOutput() const +audio_io_handle_t SwAudioOutputCollection::getA2dpOutput() const { for (size_t i = 0; i < size(); i++) { - sp<AudioOutputDescriptor> outputDesc = valueAt(i); + sp<SwAudioOutputDescriptor> outputDesc = valueAt(i); if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { return this->keyAt(i); } @@ -267,10 +408,10 @@ audio_io_handle_t AudioOutputCollection::getA2dpOutput() const return 0; } -sp<AudioOutputDescriptor> AudioOutputCollection::getPrimaryOutput() const +sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getPrimaryOutput() const { for (size_t i = 0; i < size(); i++) { - const sp<AudioOutputDescriptor> outputDesc = valueAt(i); + const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i); if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { return outputDesc; } @@ -278,26 +419,26 @@ sp<AudioOutputDescriptor> AudioOutputCollection::getPrimaryOutput() const return NULL; } -sp<AudioOutputDescriptor> AudioOutputCollection::getOutputFromId(audio_port_handle_t id) const +sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getOutputFromId(audio_port_handle_t id) const { - sp<AudioOutputDescriptor> outputDesc = NULL; + sp<SwAudioOutputDescriptor> outputDesc = NULL; for (size_t i = 0; i < size(); i++) { outputDesc = valueAt(i); - if (outputDesc->mId == id) { + if (outputDesc->getId() == id) { break; } } return outputDesc; } -bool AudioOutputCollection::isAnyOutputActive(audio_stream_type_t streamToIgnore) const +bool SwAudioOutputCollection::isAnyOutputActive(audio_stream_type_t streamToIgnore) const { for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) { if (s == (size_t) streamToIgnore) { continue; } for (size_t i = 0; i < size(); i++) { - const sp<AudioOutputDescriptor> outputDesc = valueAt(i); + const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i); if (outputDesc->mRefCount[s] != 0) { return true; } @@ -306,15 +447,15 @@ bool AudioOutputCollection::isAnyOutputActive(audio_stream_type_t streamToIgnore return false; } -audio_devices_t AudioOutputCollection::getSupportedDevices(audio_io_handle_t handle) const +audio_devices_t SwAudioOutputCollection::getSupportedDevices(audio_io_handle_t handle) const { - sp<AudioOutputDescriptor> outputDesc = valueFor(handle); + sp<SwAudioOutputDescriptor> outputDesc = valueFor(handle); audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types(); return devices; } -status_t AudioOutputCollection::dump(int fd) const +status_t SwAudioOutputCollection::dump(int fd) const { const size_t SIZE = 256; char buffer[SIZE]; diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp index 3a317fa..a06d867 100644 --- a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp @@ -54,8 +54,8 @@ status_t AudioPatch::dump(int fd, int spaces, int index) const for (size_t i = 0; i < mPatch.num_sources; i++) { if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) { snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "", - mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), + mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), mPatch.sources[i].ext.device.type)); } else { snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "", @@ -68,8 +68,8 @@ status_t AudioPatch::dump(int fd, int spaces, int index) const for (size_t i = 0; i < mPatch.num_sinks; i++) { if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) { snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "", - mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), + mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), mPatch.sinks[i].ext.device.type)); } else { snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "", diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp index 84a53eb..77fc0b9 100644 --- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp @@ -26,12 +26,12 @@ namespace android { -void AudioPolicyMix::setOutput(sp<AudioOutputDescriptor> &output) +void AudioPolicyMix::setOutput(sp<SwAudioOutputDescriptor> &output) { mOutput = output; } -const sp<AudioOutputDescriptor> &AudioPolicyMix::getOutput() const +const sp<SwAudioOutputDescriptor> &AudioPolicyMix::getOutput() const { return mOutput; } @@ -46,9 +46,9 @@ void AudioPolicyMix::setMix(AudioMix &mix) mMix = mix; } -android::AudioMix &AudioPolicyMix::getMix() +android::AudioMix *AudioPolicyMix::getMix() { - return mMix; + return &mMix; } status_t AudioPolicyMixCollection::registerMix(String8 address, AudioMix mix) @@ -88,7 +88,7 @@ status_t AudioPolicyMixCollection::getAudioPolicyMix(String8 address, return NO_ERROR; } -void AudioPolicyMixCollection::closeOutput(sp<AudioOutputDescriptor> &desc) +void AudioPolicyMixCollection::closeOutput(sp<SwAudioOutputDescriptor> &desc) { for (size_t i = 0; i < size(); i++) { sp<AudioPolicyMix> policyMix = valueAt(i); @@ -99,40 +99,40 @@ void AudioPolicyMixCollection::closeOutput(sp<AudioOutputDescriptor> &desc) } status_t AudioPolicyMixCollection::getOutputForAttr(audio_attributes_t attributes, - sp<AudioOutputDescriptor> &desc) + sp<SwAudioOutputDescriptor> &desc) { for (size_t i = 0; i < size(); i++) { sp<AudioPolicyMix> policyMix = valueAt(i); - AudioMix mix = policyMix->getMix(); - - if (mix.mMixType == MIX_TYPE_PLAYERS) { - for (size_t j = 0; j < mix.mCriteria.size(); j++) { - if ((RULE_MATCH_ATTRIBUTE_USAGE == mix.mCriteria[j].mRule && - mix.mCriteria[j].mAttr.mUsage == attributes.usage) || - (RULE_EXCLUDE_ATTRIBUTE_USAGE == mix.mCriteria[j].mRule && - mix.mCriteria[j].mAttr.mUsage != attributes.usage)) { + AudioMix *mix = policyMix->getMix(); + + if (mix->mMixType == MIX_TYPE_PLAYERS) { + for (size_t j = 0; j < mix->mCriteria.size(); j++) { + if ((RULE_MATCH_ATTRIBUTE_USAGE == mix->mCriteria[j].mRule && + mix->mCriteria[j].mAttr.mUsage == attributes.usage) || + (RULE_EXCLUDE_ATTRIBUTE_USAGE == mix->mCriteria[j].mRule && + mix->mCriteria[j].mAttr.mUsage != attributes.usage)) { desc = policyMix->getOutput(); break; } if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && strncmp(attributes.tags + strlen("addr="), - mix.mRegistrationId.string(), + mix->mRegistrationId.string(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { desc = policyMix->getOutput(); break; } } - } else if (mix.mMixType == MIX_TYPE_RECORDERS) { + } else if (mix->mMixType == MIX_TYPE_RECORDERS) { if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE && strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && strncmp(attributes.tags + strlen("addr="), - mix.mRegistrationId.string(), + mix->mRegistrationId.string(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { desc = policyMix->getOutput(); } } if (desc != 0) { - desc->mPolicyMix = &mix; + desc->mPolicyMix = mix; return NO_ERROR; } } @@ -144,19 +144,19 @@ audio_devices_t AudioPolicyMixCollection::getDeviceAndMixForInputSource(audio_so AudioMix **policyMix) { for (size_t i = 0; i < size(); i++) { - AudioMix mix = valueAt(i)->getMix(); + AudioMix *mix = valueAt(i)->getMix(); - if (mix.mMixType != MIX_TYPE_RECORDERS) { + if (mix->mMixType != MIX_TYPE_RECORDERS) { continue; } - for (size_t j = 0; j < mix.mCriteria.size(); j++) { - if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mix.mCriteria[j].mRule && - mix.mCriteria[j].mAttr.mSource == inputSource) || - (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mix.mCriteria[j].mRule && - mix.mCriteria[j].mAttr.mSource != inputSource)) { + for (size_t j = 0; j < mix->mCriteria.size(); j++) { + if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mix->mCriteria[j].mRule && + mix->mCriteria[j].mAttr.mSource == inputSource) || + (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mix->mCriteria[j].mRule && + mix->mCriteria[j].mAttr.mSource != inputSource)) { if (availDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { if (policyMix != NULL) { - *policyMix = &mix; + *policyMix = mix; } return AUDIO_DEVICE_IN_REMOTE_SUBMIX; } @@ -167,7 +167,7 @@ audio_devices_t AudioPolicyMixCollection::getDeviceAndMixForInputSource(audio_so return AUDIO_DEVICE_NONE; } -status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, AudioMix *&policyMix) +status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix) { if (strncmp(attr.tags, "addr=", strlen("addr=")) != 0) { return BAD_VALUE; @@ -180,13 +180,13 @@ status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, A return BAD_VALUE; } sp<AudioPolicyMix> audioPolicyMix = valueAt(index); - AudioMix mix = audioPolicyMix->getMix(); + AudioMix *mix = audioPolicyMix->getMix(); - if (mix.mMixType != MIX_TYPE_PLAYERS) { + if (mix->mMixType != MIX_TYPE_PLAYERS) { ALOGW("getInputForAttr() bad policy mix type for address %s", address.string()); return BAD_VALUE; } - policyMix = &mix; + *policyMix = mix; return NO_ERROR; } diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp index 46a119e..f3978ec 100644 --- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp @@ -16,7 +16,7 @@ #define LOG_TAG "APM::AudioPort" //#define LOG_NDEBUG 0 - +#include <media/AudioResamplerPublic.h> #include "AudioPort.h" #include "HwModule.h" #include "AudioGain.h" @@ -31,8 +31,8 @@ int32_t volatile AudioPort::mNextUniqueId = 1; // --- AudioPort class implementation AudioPort::AudioPort(const String8& name, audio_port_type_t type, - audio_port_role_t role, const sp<HwModule>& module) : - mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0) + audio_port_role_t role) : + mName(name), mType(type), mRole(role), mFlags(0) { mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) || ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK)); @@ -40,7 +40,6 @@ AudioPort::AudioPort(const String8& name, audio_port_type_t type, void AudioPort::attach(const sp<HwModule>& module) { - mId = getNextUniqueId(); mModule = module; } @@ -51,9 +50,28 @@ audio_port_handle_t AudioPort::getNextUniqueId() audio_module_handle_t AudioPort::getModuleHandle() const { + if (mModule == 0) { + return 0; + } return mModule->mHandle; } +uint32_t AudioPort::getModuleVersion() const +{ + if (mModule == 0) { + return 0; + } + return mModule->mHalVersion; +} + +const char *AudioPort::getModuleName() const +{ + if (mModule == 0) { + return ""; + } + return mModule->mName; +} + void AudioPort::toAudioPort(struct audio_port *port) const { port->role = mRole; @@ -198,6 +216,7 @@ void AudioPort::loadFormats(char *name) } str = strtok(NULL, "|"); } + mFormats.sort(compareFormatsGoodToBad); } void AudioPort::loadInChannels(char *name) @@ -340,6 +359,9 @@ status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate, uint32_t *updatedSamplingRate) const { if (mSamplingRates.isEmpty()) { + if (updatedSamplingRate != NULL) { + *updatedSamplingRate = samplingRate; + } return NO_ERROR; } @@ -369,16 +391,11 @@ status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate, } } } - // This uses hard-coded knowledge about AudioFlinger resampling ratios. - // TODO Move these assumptions out. - static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs - static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur - // due to approximation by an int32_t of the - // phase increments + // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum. if (minAbove >= 0) { candidate = mSamplingRates[minAbove]; - if (candidate / kMaxDownSampleRatio <= samplingRate) { + if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) { if (updatedSamplingRate != NULL) { *updatedSamplingRate = candidate; } @@ -388,7 +405,7 @@ status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate, // But if we have to up-sample from a lower sampling rate, that's OK. if (maxBelow >= 0) { candidate = mSamplingRates[maxBelow]; - if (candidate * kMaxUpSampleRatio >= samplingRate) { + if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) { if (updatedSamplingRate != NULL) { *updatedSamplingRate = candidate; } @@ -413,10 +430,13 @@ status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) cons return BAD_VALUE; } -status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask) - const +status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask, + audio_channel_mask_t *updatedChannelMask) const { if (mChannelMasks.isEmpty()) { + if (updatedChannelMask != NULL) { + *updatedChannelMask = channelMask; + } return NO_ERROR; } @@ -425,6 +445,9 @@ status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask) // FIXME Does not handle multi-channel automatic conversions yet audio_channel_mask_t supported = mChannelMasks[i]; if (supported == channelMask) { + if (updatedChannelMask != NULL) { + *updatedChannelMask = channelMask; + } return NO_ERROR; } if (isRecordThread) { @@ -434,6 +457,9 @@ status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask) && channelMask == AUDIO_CHANNEL_IN_MONO) || (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK || channelMask == AUDIO_CHANNEL_IN_STEREO))) { + if (updatedChannelMask != NULL) { + *updatedChannelMask = supported; + } return NO_ERROR; } } @@ -441,7 +467,7 @@ status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask) return BAD_VALUE; } -status_t AudioPort::checkFormat(audio_format_t format) const +status_t AudioPort::checkExactFormat(audio_format_t format) const { if (mFormats.isEmpty()) { return NO_ERROR; @@ -455,6 +481,33 @@ status_t AudioPort::checkFormat(audio_format_t format) const return BAD_VALUE; } +status_t AudioPort::checkCompatibleFormat(audio_format_t format, audio_format_t *updatedFormat) + const +{ + if (mFormats.isEmpty()) { + if (updatedFormat != NULL) { + *updatedFormat = format; + } + return NO_ERROR; + } + + const bool checkInexact = // when port is input and format is linear pcm + mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK + && audio_is_linear_pcm(format); + + for (size_t i = 0; i < mFormats.size(); ++i) { + if (mFormats[i] == format || + (checkInexact && audio_is_linear_pcm(mFormats[i]))) { + // for inexact checks we take the first linear pcm format since + // mFormats is sorted from best PCM format to worst PCM format. + if (updatedFormat != NULL) { + *updatedFormat = mFormats[i]; + } + return NO_ERROR; + } + } + return BAD_VALUE; +} uint32_t AudioPort::pickSamplingRate() const { @@ -629,7 +682,7 @@ void AudioPort::dump(int fd, int spaces) const char buffer[SIZE]; String8 result; - if (mName.size() != 0) { + if (mName.length() != 0) { snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string()); result.append(buffer); } @@ -687,13 +740,16 @@ void AudioPort::dump(int fd, int spaces) const if (mGains.size() != 0) { snprintf(buffer, SIZE, "%*s- gains:\n", spaces, ""); write(fd, buffer, strlen(buffer) + 1); - result.append(buffer); for (size_t i = 0; i < mGains.size(); i++) { mGains[i]->dump(fd, spaces + 2, i); } } } +void AudioPort::log(const char* indent) const +{ + ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.string(), mType, mRole); +} // --- AudioPortConfig class implementation @@ -735,7 +791,7 @@ status_t AudioPortConfig::applyAudioPortConfig( mChannelMask = config->channel_mask; } if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) { - status = audioport->checkFormat(config->format); + status = audioport->checkExactFormat(config->format); if (status != NO_ERROR) { goto exit; } diff --git a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp index fe5bc5f..9ab1d61 100644 --- a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp @@ -113,8 +113,8 @@ audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name) char *devName = strtok(name, "|"); while (devName != NULL) { if (strlen(devName) != 0) { - device |= stringToEnum(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), + device |= stringToEnum(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), devName); } devName = strtok(NULL, "|"); @@ -224,8 +224,8 @@ void ConfigParsingUtils::loadGlobalConfig(cnode *root, const sp<HwModule>& modul availableOutputDevices.types()); } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) { audio_devices_t device = (audio_devices_t)stringToEnum( - sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), + sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), (char *)node->value); if (device != AUDIO_DEVICE_NONE) { defaultOutputDevice = new DeviceDescriptor(String8("default-output"), device); diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp index 7df7d75..9573583 100644 --- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp @@ -29,13 +29,23 @@ String8 DeviceDescriptor::emptyNameStr = String8(""); DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) : AudioPort(name, AUDIO_PORT_TYPE_DEVICE, audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK : - AUDIO_PORT_ROLE_SOURCE, - NULL), - mAddress(""), mDeviceType(type) + AUDIO_PORT_ROLE_SOURCE), + mAddress(""), mDeviceType(type), mId(0) { } +audio_port_handle_t DeviceDescriptor::getId() const +{ + return mId; +} + +void DeviceDescriptor::attach(const sp<HwModule>& module) +{ + AudioPort::attach(module); + mId = getNextUniqueId(); +} + bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const { // Devices are considered equal if they: @@ -139,11 +149,14 @@ void DeviceVector::loadDevicesFromName(char *name, char *devName = strtok(name, "|"); while (devName != NULL) { if (strlen(devName) != 0) { - audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), + audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), devName); if (type != AUDIO_DEVICE_NONE) { - sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type); + devName = (char *)ConfigParsingUtils::enumToString(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + type); + sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(devName), type); if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX || type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) { dev->mAddress = String8("0"); @@ -183,7 +196,7 @@ sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const { sp<DeviceDescriptor> device; for (size_t i = 0; i < size(); i++) { - if (itemAt(i)->getHandle() == id) { + if (itemAt(i)->getId() == id) { device = itemAt(i); break; } @@ -303,8 +316,8 @@ status_t DeviceDescriptor::dump(int fd, int spaces, int index) const result.append(buffer); } snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "", - ConfigParsingUtils::enumToString(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), + ConfigParsingUtils::enumToString(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), mDeviceType)); result.append(buffer); if (mAddress.size() != 0) { @@ -317,4 +330,16 @@ status_t DeviceDescriptor::dump(int fd, int spaces, int index) const return NO_ERROR; } +void DeviceDescriptor::log() const +{ + ALOGI("Device id:%d type:0x%X:%s, addr:%s", + mId, + mDeviceType, + ConfigParsingUtils::enumToString( + sDeviceNameToEnumTable, ARRAY_SIZE(sDeviceNameToEnumTable), mDeviceType), + mAddress.string()); + + AudioPort::log(" "); +} + }; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp index 0097d69..e955447 100644 --- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp @@ -48,7 +48,7 @@ status_t HwModule::loadInput(cnode *root) { cnode *node = root->first_child; - sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this); + sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK); while (node) { if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { @@ -83,6 +83,7 @@ status_t HwModule::loadInput(cnode *root) ALOGV("loadInput() adding input Supported Devices %04x", profile->mSupportedDevices.types()); + profile->attach(this); mInputProfiles.add(profile); return NO_ERROR; } else { @@ -94,7 +95,7 @@ status_t HwModule::loadOutput(cnode *root) { cnode *node = root->first_child; - sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this); + sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE); while (node) { if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { @@ -128,7 +129,7 @@ status_t HwModule::loadOutput(cnode *root) ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x", profile->mSupportedDevices.types(), profile->mFlags); - + profile->attach(this); mOutputProfiles.add(profile); return NO_ERROR; } else { @@ -154,7 +155,6 @@ status_t HwModule::loadDevice(cnode *root) return BAD_VALUE; } sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type); - deviceDesc->mModule = this; node = root->first_child; while (node) { @@ -183,7 +183,7 @@ status_t HwModule::loadDevice(cnode *root) status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config, audio_devices_t device, String8 address) { - sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this); + sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE); profile->mSamplingRates.add(config->sample_rate); profile->mChannelMasks.add(config->channel_mask); @@ -193,6 +193,7 @@ status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config, devDesc->mAddress = address; profile->mSupportedDevices.add(devDesc); + profile->attach(this); mOutputProfiles.add(profile); return NO_ERROR; @@ -213,7 +214,7 @@ status_t HwModule::removeOutputProfile(String8 name) status_t HwModule::addInputProfile(String8 name, const audio_config_t *config, audio_devices_t device, String8 address) { - sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this); + sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK); profile->mSamplingRates.add(config->sample_rate); profile->mChannelMasks.add(config->channel_mask); @@ -225,6 +226,7 @@ status_t HwModule::addInputProfile(String8 name, const audio_config_t *config, ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask); + profile->attach(this); mInputProfiles.add(profile); return NO_ERROR; diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp index 376dd22..7b6d51d 100644 --- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp +++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp @@ -23,9 +23,8 @@ namespace android { -IOProfile::IOProfile(const String8& name, audio_port_role_t role, - const sp<HwModule>& module) - : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module) +IOProfile::IOProfile(const String8& name, audio_port_role_t role) + : AudioPort(name, AUDIO_PORT_TYPE_MIX, role) { } @@ -41,7 +40,9 @@ bool IOProfile::isCompatibleProfile(audio_devices_t device, uint32_t samplingRate, uint32_t *updatedSamplingRate, audio_format_t format, + audio_format_t *updatedFormat, audio_channel_mask_t channelMask, + audio_channel_mask_t *updatedChannelMask, uint32_t flags) const { const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE; @@ -72,7 +73,14 @@ bool IOProfile::isCompatibleProfile(audio_devices_t device, return false; } - if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) { + if (!audio_is_valid_format(format)) { + return false; + } + if (isPlaybackThread && checkExactFormat(format) != NO_ERROR) { + return false; + } + audio_format_t myUpdatedFormat = format; + if (isRecordThread && checkCompatibleFormat(format, &myUpdatedFormat) != NO_ERROR) { return false; } @@ -80,8 +88,9 @@ bool IOProfile::isCompatibleProfile(audio_devices_t device, checkExactChannelMask(channelMask) != NO_ERROR)) { return false; } + audio_channel_mask_t myUpdatedChannelMask = channelMask; if (isRecordThread && (!audio_is_input_channel(channelMask) || - checkCompatibleChannelMask(channelMask) != NO_ERROR)) { + checkCompatibleChannelMask(channelMask, &myUpdatedChannelMask) != NO_ERROR)) { return false; } @@ -100,6 +109,12 @@ bool IOProfile::isCompatibleProfile(audio_devices_t device, if (updatedSamplingRate != NULL) { *updatedSamplingRate = myUpdatedSamplingRate; } + if (updatedFormat != NULL) { + *updatedFormat = myUpdatedFormat; + } + if (updatedChannelMask != NULL) { + *updatedChannelMask = myUpdatedChannelMask; + } return true; } diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h b/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h index eadaa77..db0573f 100755 --- a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h +++ b/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h @@ -134,16 +134,16 @@ public: audio_policy_dev_state_t state) = 0; /** - * Translate a volume index given by the UI to an amplification value for a stream type + * Translate a volume index given by the UI to an amplification value in dB for a stream type * and a device category. * * @param[in] deviceCategory for which the conversion is requested. * @param[in] stream type for which the conversion is requested. * @param[in] indexInUi index received from the UI to be translated. * - * @return amplification value matching the UI index for this given device and stream. + * @return amplification value in dB matching the UI index for this given device and stream. */ - virtual float volIndexToAmpl(Volume::device_category deviceCategory, audio_stream_type_t stream, + virtual float volIndexToDb(Volume::device_category deviceCategory, audio_stream_type_t stream, int indexInUi) = 0; /** diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h index 4f5427e..6d43df2 100755 --- a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h +++ b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h @@ -43,7 +43,7 @@ public: virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const = 0; - virtual const AudioOutputCollection &getOutputs() const = 0; + virtual const SwAudioOutputCollection &getOutputs() const = 0; virtual const AudioInputCollection &getInputs() const = 0; diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp index 1fd3341..50f1609 100755 --- a/services/audiopolicy/enginedefault/src/Engine.cpp +++ b/services/audiopolicy/enginedefault/src/Engine.cpp @@ -63,13 +63,14 @@ status_t Engine::initCheck() return (mApmObserver != NULL) ? NO_ERROR : NO_INIT; } -float Engine::volIndexToAmpl(Volume::device_category category, audio_stream_type_t streamType, +float Engine::volIndexToDb(Volume::device_category category, audio_stream_type_t streamType, int indexInUi) { const StreamDescriptor &streamDesc = mApmObserver->getStreamDescriptors().valueAt(streamType); - return Gains::volIndexToAmpl(category, streamDesc, indexInUi); + return Gains::volIndexToDb(category, streamDesc, indexInUi); } + status_t Engine::initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) { ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); @@ -243,7 +244,7 @@ routing_strategy Engine::getStrategyForStream(audio_stream_type_t stream) routing_strategy Engine::getStrategyForUsage(audio_usage_t usage) { - const AudioOutputCollection &outputs = mApmObserver->getOutputs(); + const SwAudioOutputCollection &outputs = mApmObserver->getOutputs(); // usage to strategy mapping switch (usage) { @@ -291,7 +292,7 @@ audio_devices_t Engine::getDeviceForStrategy(routing_strategy strategy) const const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices(); const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices(); - const AudioOutputCollection &outputs = mApmObserver->getOutputs(); + const SwAudioOutputCollection &outputs = mApmObserver->getOutputs(); uint32_t device = AUDIO_DEVICE_NONE; uint32_t availableOutputDevicesType = availableOutputDevices.types(); @@ -358,7 +359,7 @@ audio_devices_t Engine::getDeviceForStrategy(routing_strategy strategy) const if (((availableInputDevices.types() & AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) || (((txDevice & availPrimaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) && - (primaryOutput->getAudioPort()->mModule->mHalVersion < + (primaryOutput->getAudioPort()->getModuleVersion() < AUDIO_DEVICE_API_VERSION_3_0))) { availableOutputDevicesType = availPrimaryOutputDevices; } @@ -582,7 +583,7 @@ audio_devices_t Engine::getDeviceForInputSource(audio_source_t inputSource) cons { const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices(); const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices(); - const AudioOutputCollection &outputs = mApmObserver->getOutputs(); + const SwAudioOutputCollection &outputs = mApmObserver->getOutputs(); audio_devices_t availableDeviceTypes = availableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; uint32_t device = AUDIO_DEVICE_NONE; diff --git a/services/audiopolicy/enginedefault/src/Engine.h b/services/audiopolicy/enginedefault/src/Engine.h index f44556c..56a4748 100755 --- a/services/audiopolicy/enginedefault/src/Engine.h +++ b/services/audiopolicy/enginedefault/src/Engine.h @@ -101,10 +101,10 @@ private: { return mPolicyEngine->initializeVolumeCurves(isSpeakerDrcEnabled); } - virtual float volIndexToAmpl(Volume::device_category deviceCategory, + virtual float volIndexToDb(Volume::device_category deviceCategory, audio_stream_type_t stream,int indexInUi) { - return mPolicyEngine->volIndexToAmpl(deviceCategory, stream, indexInUi); + return mPolicyEngine->volIndexToDb(deviceCategory, stream, indexInUi); } private: Engine *mPolicyEngine; @@ -141,7 +141,7 @@ private: audio_devices_t getDeviceForStrategy(routing_strategy strategy) const; audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const; - float volIndexToAmpl(Volume::device_category category, + float volIndexToDb(Volume::device_category category, audio_stream_type_t stream, int indexInUi); status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax); void initializeVolumeCurves(bool isSpeakerDrcEnabled); diff --git a/services/audiopolicy/enginedefault/src/Gains.cpp b/services/audiopolicy/enginedefault/src/Gains.cpp index a684fdd..78f2909 100644 --- a/services/audiopolicy/enginedefault/src/Gains.cpp +++ b/services/audiopolicy/enginedefault/src/Gains.cpp @@ -197,10 +197,10 @@ const VolumeCurvePoint *Gains::sVolumeProfiles[AUDIO_STREAM_CNT] }; //static -float Gains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, - int indexInUi) +float Gains::volIndexToDb(Volume::device_category deviceCategory, + const StreamDescriptor& streamDesc, + int indexInUi) { - Volume::device_category deviceCategory = Volume::getDeviceCategory(device); const VolumeCurvePoint *curve = streamDesc.getVolumeCurvePoint(deviceCategory); // the volume index in the UI is relative to the min and max volume indices for this stream type @@ -212,7 +212,7 @@ float Gains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& stre // find what part of the curve this index volume belongs to, or if it's out of bounds int segment = 0; if (volIdx < curve[Volume::VOLMIN].mIndex) { // out of bounds - return 0.0f; + return VOLUME_MIN_DB; } else if (volIdx < curve[Volume::VOLKNEE1].mIndex) { segment = 0; } else if (volIdx < curve[Volume::VOLKNEE2].mIndex) { @@ -220,7 +220,7 @@ float Gains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& stre } else if (volIdx <= curve[Volume::VOLMAX].mIndex) { segment = 2; } else { // out of bounds - return 1.0f; + return 0.0f; } // linear interpolation in the attenuation table in dB @@ -231,17 +231,25 @@ float Gains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& stre ((float)(curve[segment+1].mIndex - curve[segment].mIndex)) ); - float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) - - ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", + ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f]", curve[segment].mIndex, volIdx, curve[segment+1].mIndex, curve[segment].mDBAttenuation, decibels, - curve[segment+1].mDBAttenuation, - amplification); + curve[segment+1].mDBAttenuation); + + return decibels; +} - return amplification; + +//static +float Gains::volIndexToAmpl(Volume::device_category deviceCategory, + const StreamDescriptor& streamDesc, + int indexInUi) +{ + return Volume::DbToAmpl(volIndexToDb(deviceCategory, streamDesc, indexInUi)); } + + }; // namespace android diff --git a/services/audiopolicy/enginedefault/src/Gains.h b/services/audiopolicy/enginedefault/src/Gains.h index b5601ca..7620b7d 100644 --- a/services/audiopolicy/enginedefault/src/Gains.h +++ b/services/audiopolicy/enginedefault/src/Gains.h @@ -29,8 +29,13 @@ class StreamDescriptor; class Gains { public : - static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, - int indexInUi); + static float volIndexToAmpl(Volume::device_category deviceCategory, + const StreamDescriptor& streamDesc, + int indexInUi); + + static float volIndexToDb(Volume::device_category deviceCategory, + const StreamDescriptor& streamDesc, + int indexInUi); // default volume curve static const VolumeCurvePoint sDefaultVolumeCurve[Volume::VOLCNT]; diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp index 797a2b4..ba9f996 100644 --- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp +++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp @@ -157,7 +157,7 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, // outputs must be closed after checkOutputForAllStrategies() is executed if (!outputs.isEmpty()) { for (size_t i = 0; i < outputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); // close unused outputs after device disconnection or direct outputs that have been // opened by checkOutputsForDevice() to query dynamic parameters if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) || @@ -176,18 +176,17 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, updateCallRouting(newDevice); } for (size_t i = 0; i < mOutputs.size(); i++) { - audio_io_handle_t output = mOutputs.keyAt(i); - if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { - audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i), - true /*fromCache*/); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); + if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) { + audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/); // do not force device change on duplicated output because if device is 0, it will // also force a device 0 for the two outputs it is duplicated to which may override // a valid device selection on those outputs. - bool force = !mOutputs.valueAt(i)->isDuplicated() + bool force = !desc->isDuplicated() && (!device_distinguishes_on_address(device) // always force when disconnecting (a non-duplicated device) || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); - setOutputDevice(output, newDevice, force, 0); + setOutputDevice(desc, newDevice, force, 0); } } @@ -349,10 +348,11 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); if (output != AUDIO_IO_HANDLE_NONE) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); ALOG_ASSERT(!outputDesc->isDuplicated(), "updateCallRouting() RX device output is duplicated"); outputDesc->toAudioPortConfig(&patch.sources[1]); + patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; patch.num_sources = 2; } @@ -395,6 +395,7 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs ALOG_ASSERT(!outputDesc->isDuplicated(), "updateCallRouting() RX device output is duplicated"); outputDesc->toAudioPortConfig(&patch.sources[1]); + patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; patch.num_sources = 2; } @@ -448,13 +449,13 @@ void AudioPolicyManager::setPhoneState(audio_mode_t state) checkOutputForAllStrategies(); updateDevicesAndOutputs(); - sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); + sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput; int delayMs = 0; if (isStateInCall(state)) { nsecs_t sysTime = systemTime(); for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); // mute media and sonification strategies and delay device switch by the largest // latency of any output where either strategy is active. // This avoid sending the ring tone or music tail into the earpiece or headset. @@ -464,14 +465,14 @@ void AudioPolicyManager::setPhoneState(audio_mode_t state) isStrategyActive(desc, STRATEGY_SONIFICATION, SONIFICATION_HEADSET_MUSIC_DELAY, sysTime)) && - (delayMs < (int)desc->mLatency*2)) { - delayMs = desc->mLatency*2; + (delayMs < (int)desc->latency()*2)) { + delayMs = desc->latency()*2; } - setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); - setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, + setStrategyMute(STRATEGY_MEDIA, true, desc); + setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS, getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); - setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); - setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, + setStrategyMute(STRATEGY_SONIFICATION, true, desc); + setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS, getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); } } @@ -547,13 +548,13 @@ void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage, updateCallRouting(newDevice); } for (size_t i = 0; i < mOutputs.size(); i++) { - audio_io_handle_t output = mOutputs.keyAt(i); - audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/); - if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { - setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); + audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/); + if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) { + setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE)); } if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { - applyStreamVolumes(output, newDevice, 0, true); + applyStreamVolumes(outputDesc, newDevice, 0, true); } } @@ -584,8 +585,10 @@ sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput( } for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j]; - bool found = profile->isCompatibleProfile(device, String8(""), samplingRate, - NULL /*updatedSamplingRate*/, format, channelMask, + bool found = profile->isCompatibleProfile(device, String8(""), + samplingRate, NULL /*updatedSamplingRate*/, + format, NULL /*updatedFormat*/, + channelMask, NULL /*updatedChannelMask*/, flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ? AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT); if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) { @@ -621,6 +624,7 @@ status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + audio_port_handle_t selectedDeviceId, const audio_offload_info_t *offloadInfo) { audio_attributes_t attributes; @@ -639,7 +643,7 @@ status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, } stream_type_to_audio_attributes(*stream, &attributes); } - sp<AudioOutputDescriptor> desc; + sp<SwAudioOutputDescriptor> desc; if (mPolicyMixes.getOutputForAttr(attributes, desc) == NO_ERROR) { ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr"); if (!audio_is_linear_pcm(format)) { @@ -675,6 +679,17 @@ status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, if (*output == AUDIO_IO_HANDLE_NONE) { return INVALID_OPERATION; } + + // Explicit routing? + sp<DeviceDescriptor> deviceDesc; + + for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { + if (mAvailableOutputDevices[i]->getId() == selectedDeviceId) { + deviceDesc = mAvailableOutputDevices[i]; + break; + } + } + mOutputRoutes.addRoute(session, *stream, deviceDesc); return NO_ERROR; } @@ -699,7 +714,8 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( if (mTestOutputs[mCurOutput] == 0) { ALOGV("getOutput() opening test output"); - sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL); + sp<AudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(NULL, + mpClientInterface); outputDesc->mDevice = mTestDevice; outputDesc->mLatency = mTestLatencyMs; outputDesc->mFlags = @@ -775,10 +791,10 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( } if (profile != 0) { - sp<AudioOutputDescriptor> outputDesc = NULL; + sp<SwAudioOutputDescriptor> outputDesc = NULL; for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); if (!desc->isDuplicated() && (profile == desc->mProfile)) { outputDesc = desc; // reuse direct output if currently open and configured with same parameters @@ -795,7 +811,7 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( if (outputDesc != NULL) { closeOutput(outputDesc->mIoHandle); } - outputDesc = new AudioOutputDescriptor(profile); + outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface); outputDesc->mDevice = device; outputDesc->mLatency = 0; outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); @@ -806,7 +822,7 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( if (offloadInfo != NULL) { config.offload_info = *offloadInfo; } - status = mpClientInterface->openOutput(profile->mModule->mHandle, + status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &outputDesc->mDevice, @@ -856,7 +872,6 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( } non_direct_output: - // ignoring channel mask due to downmix capability in mixer // open a non direct output @@ -874,7 +889,7 @@ non_direct_output: ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); - ALOGV("getOutput() returns output %d", output); + ALOGV(" getOutputForDevice() returns output %d", output); return output; } @@ -902,7 +917,7 @@ audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_h audio_io_handle_t outputPrimary = 0; for (size_t i = 0; i < outputs.size(); i++) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]); if (!outputDesc->isDuplicated()) { // if a valid format is specified, skip output if not compatible if (format != AUDIO_FORMAT_INVALID) { @@ -941,15 +956,59 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session) { - ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); + ALOGV("startOutput() output %d, stream %d, session %d", + output, stream, session); ssize_t index = mOutputs.indexOfKey(output); if (index < 0) { ALOGW("startOutput() unknown output %d", output); return BAD_VALUE; } + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); + + audio_devices_t newDevice; + if (outputDesc->mPolicyMix != NULL) { + newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; + } else { + newDevice = AUDIO_DEVICE_NONE; + } + + uint32_t delayMs = 0; + + // Routing? + mOutputRoutes.incRouteActivity(session); + + status_t status = startSource(outputDesc, stream, newDevice, &delayMs); + + if (status != NO_ERROR) { + mOutputRoutes.decRouteActivity(session); + } + // Automatically enable the remote submix input when output is started on a re routing mix + // of type MIX_TYPE_RECORDERS + if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL && + outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + outputDesc->mPolicyMix->mRegistrationId, + "remote-submix"); + } + + if (delayMs != 0) { + usleep(delayMs * 1000); + } + + return status; +} + +status_t AudioPolicyManager::startSource(sp<AudioOutputDescriptor> outputDesc, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t *delayMs) +{ // cannot start playback of STREAM_TTS if any other output is being used uint32_t beaconMuteLatency = 0; + + *delayMs = 0; if (stream == AUDIO_STREAM_TTS) { ALOGV("\t found BEACON stream"); if (mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) { @@ -962,8 +1021,6 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); } - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); - // increment usage count for this stream on the requested output: // NOTE that the usage count is the same for duplicated output and hardware output which is // necessary for a correct control of hardware output routing by startOutput() and stopOutput() @@ -971,11 +1028,8 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, if (outputDesc->mRefCount[stream] == 1) { // starting an output being rerouted? - audio_devices_t newDevice; - if (outputDesc->mPolicyMix != NULL) { - newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; - } else { - newDevice = getNewOutputDevice(output, false /*fromCache*/); + if (device == AUDIO_DEVICE_NONE) { + device = getNewOutputDevice(outputDesc, false /*fromCache*/); } routing_strategy strategy = getStrategy(stream); bool shouldWait = (strategy == STRATEGY_SONIFICATION) || @@ -991,7 +1045,7 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, // In this case, the audio HAL must receive the new device selection so that it can // change the device currently selected by the other active output. if (outputDesc->sharesHwModuleWith(desc) && - desc->device() != newDevice) { + desc->device() != device) { force = true; } // wait for audio on other active outputs to be presented when starting @@ -1003,7 +1057,7 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, } } } - uint32_t muteWaitMs = setOutputDevice(output, newDevice, force); + uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force); // handle special case for sonification while in call if (isInCall()) { @@ -1012,32 +1066,18 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, // apply volume rules for current stream and device if necessary checkAndSetVolume(stream, - mStreams[stream].getVolumeIndex(newDevice), - output, - newDevice); + mStreams.valueFor(stream).getVolumeIndex(device), + outputDesc, + device); // update the outputs if starting an output with a stream that can affect notification // routing handleNotificationRoutingForStream(stream); - // Automatically enable the remote submix input when output is started on a re routing mix - // of type MIX_TYPE_RECORDERS - if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL && - outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { - setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - outputDesc->mPolicyMix->mRegistrationId, - "remote-submix"); - } - // force reevaluating accessibility routing when ringtone or alarm starts if (strategy == STRATEGY_SONIFICATION) { mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); } - - if (waitMs > muteWaitMs) { - usleep((waitMs - muteWaitMs) * 2 * 1000); - } } return NO_ERROR; } @@ -1054,8 +1094,32 @@ status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, return BAD_VALUE; } - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); + + if (outputDesc->mRefCount[stream] == 1) { + // Automatically disable the remote submix input when output is stopped on a + // re routing mix of type MIX_TYPE_RECORDERS + if (audio_is_remote_submix_device(outputDesc->mDevice) && + outputDesc->mPolicyMix != NULL && + outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, + outputDesc->mPolicyMix->mRegistrationId, + "remote-submix"); + } + } + + // Routing? + if (outputDesc->mRefCount[stream] > 0) { + mOutputRoutes.decRouteActivity(session); + } + + return stopSource(outputDesc, stream); +} +status_t AudioPolicyManager::stopSource(sp<AudioOutputDescriptor> outputDesc, + audio_stream_type_t stream) +{ // always handle stream stop, check which stream type is stopping handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); @@ -1067,41 +1131,31 @@ status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, if (outputDesc->mRefCount[stream] > 0) { // decrement usage count of this stream on the output outputDesc->changeRefCount(stream, -1); + // store time at which the stream was stopped - see isStreamActive() if (outputDesc->mRefCount[stream] == 0) { - // Automatically disable the remote submix input when output is stopped on a - // re routing mix of type MIX_TYPE_RECORDERS - if (audio_is_remote_submix_device(outputDesc->mDevice) && - outputDesc->mPolicyMix != NULL && - outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { - setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - outputDesc->mPolicyMix->mRegistrationId, - "remote-submix"); - } - outputDesc->mStopTime[stream] = systemTime(); - audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/); + audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/); // delay the device switch by twice the latency because stopOutput() is executed when // the track stop() command is received and at that time the audio track buffer can // still contain data that needs to be drained. The latency only covers the audio HAL // and kernel buffers. Also the latency does not always include additional delay in the // audio path (audio DSP, CODEC ...) - setOutputDevice(output, newDevice, false, outputDesc->mLatency*2); + setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2); // force restoring the device selection on other active outputs if it differs from the // one being selected for this output for (size_t i = 0; i < mOutputs.size(); i++) { audio_io_handle_t curOutput = mOutputs.keyAt(i); sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); - if (curOutput != output && + if (desc != outputDesc && desc->isActive() && outputDesc->sharesHwModuleWith(desc) && (newDevice != desc->device())) { - setOutputDevice(curOutput, - getNewOutputDevice(curOutput, false /*fromCache*/), + setOutputDevice(desc, + getNewOutputDevice(desc, false /*fromCache*/), true, - outputDesc->mLatency*2); + outputDesc->latency()*2); } } // update the outputs if stopping one with a stream that can affect notification routing @@ -1109,7 +1163,7 @@ status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, } return NO_ERROR; } else { - ALOGW("stopOutput() refcount is already 0 for output %d", output); + ALOGW("stopOutput() refcount is already 0"); return INVALID_OPERATION; } } @@ -1138,7 +1192,10 @@ void AudioPolicyManager::releaseOutput(audio_io_handle_t output, } #endif //AUDIO_POLICY_TEST - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index); + // Routing + mOutputRoutes.removeRoute(session); + + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(index); if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { if (desc->mDirectOpenCount <= 0) { ALOGW("releaseOutput() invalid open count %d for output %d", @@ -1150,8 +1207,9 @@ void AudioPolicyManager::releaseOutput(audio_io_handle_t output, // If effects where present on the output, audioflinger moved them to the primary // output by default: move them back to the appropriate output. audio_io_handle_t dstOutput = getOutputForEffect(); - if (dstOutput != mPrimaryOutput) { - mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput); + if (dstOutput != mPrimaryOutput->mIoHandle) { + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, + mPrimaryOutput->mIoHandle, dstOutput); } mpClientInterface->onAudioPortListUpdate(); } @@ -1189,7 +1247,7 @@ status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX && strncmp(attr->tags, "addr=", strlen("addr=")) == 0) { - status_t ret = mPolicyMixes.getInputMixForAttr(*attr, policyMix); + status_t ret = mPolicyMixes.getInputMixForAttr(*attr, &policyMix); if (ret != NO_ERROR) { return ret; } @@ -1247,48 +1305,54 @@ status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, } } - sp<IOProfile> profile = getInputProfile(device, address, - samplingRate, format, channelMask, - flags); - if (profile == 0) { - //retry without flags - audio_input_flags_t log_flags = flags; - flags = AUDIO_INPUT_FLAG_NONE; + // find a compatible input profile (not necessarily identical in parameters) + sp<IOProfile> profile; + // samplingRate and flags may be updated by getInputProfile + uint32_t profileSamplingRate = samplingRate; + audio_format_t profileFormat = format; + audio_channel_mask_t profileChannelMask = channelMask; + audio_input_flags_t profileFlags = flags; + for (;;) { profile = getInputProfile(device, address, - samplingRate, format, channelMask, - flags); - if (profile == 0) { + profileSamplingRate, profileFormat, profileChannelMask, + profileFlags); + if (profile != 0) { + break; // success + } else if (profileFlags != AUDIO_INPUT_FLAG_NONE) { + profileFlags = AUDIO_INPUT_FLAG_NONE; // retry + } else { // fail ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u," "format %#x, channelMask 0x%X, flags %#x", - device, samplingRate, format, channelMask, log_flags); + device, samplingRate, format, channelMask, flags); return BAD_VALUE; } } - if (profile->mModule->mHandle == 0) { - ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName); + if (profile->getModuleHandle() == 0) { + ALOGE("getInputForAttr(): HW module %s not opened", profile->getModuleName()); return NO_INIT; } audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = samplingRate; - config.channel_mask = channelMask; - config.format = format; + config.sample_rate = profileSamplingRate; + config.channel_mask = profileChannelMask; + config.format = profileFormat; - status_t status = mpClientInterface->openInput(profile->mModule->mHandle, + status_t status = mpClientInterface->openInput(profile->getModuleHandle(), input, &config, &device, address, halInputSource, - flags); + profileFlags); // only accept input with the exact requested set of parameters if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE || - (samplingRate != config.sample_rate) || - (format != config.format) || - (channelMask != config.channel_mask)) { - ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x", + (profileSamplingRate != config.sample_rate) || + (profileFormat != config.format) || + (profileChannelMask != config.channel_mask)) { + ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d," + " channelMask %x", samplingRate, format, channelMask); if (*input != AUDIO_IO_HANDLE_NONE) { mpClientInterface->closeInput(*input); @@ -1300,15 +1364,15 @@ status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, inputDesc->mInputSource = inputSource; inputDesc->mRefCount = 0; inputDesc->mOpenRefCount = 1; - inputDesc->mSamplingRate = samplingRate; - inputDesc->mFormat = format; - inputDesc->mChannelMask = channelMask; + inputDesc->mSamplingRate = profileSamplingRate; + inputDesc->mFormat = profileFormat; + inputDesc->mChannelMask = profileChannelMask; inputDesc->mDevice = device; inputDesc->mSessions.add(session); inputDesc->mIsSoundTrigger = isSoundTrigger; inputDesc->mPolicyMix = policyMix; - ALOGV("getInputForAttr() returns input type = %d", inputType); + ALOGV("getInputForAttr() returns input type = %d", *inputType); addInput(*input, inputDesc); mpClientInterface->onAudioPortListUpdate(); @@ -1505,8 +1569,8 @@ status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, audio_devices_t device) { - if ((index < mStreams[stream].getVolumeIndexMin()) || - (index > mStreams[stream].getVolumeIndexMax())) { + if ((index < mStreams.valueFor(stream).getVolumeIndexMin()) || + (index > mStreams.valueFor(stream).getVolumeIndexMax())) { return BAD_VALUE; } if (!audio_is_output_device(device)) { @@ -1514,7 +1578,7 @@ status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, } // Force max volume if stream cannot be muted - if (!mStreams.canBeMuted(stream)) index = mStreams[stream].getVolumeIndexMax(); + if (!mStreams.canBeMuted(stream)) index = mStreams.valueFor(stream).getVolumeIndexMax(); ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", stream, device, index); @@ -1543,16 +1607,17 @@ status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, } status_t status = NO_ERROR; for (size_t i = 0; i < mOutputs.size(); i++) { - audio_devices_t curDevice = Volume::getDeviceForVolume(mOutputs.valueAt(i)->device()); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); + audio_devices_t curDevice = Volume::getDeviceForVolume(desc->device()); if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) { - status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); + status_t volStatus = checkAndSetVolume(stream, index, desc, curDevice); if (volStatus != NO_ERROR) { status = volStatus; } } if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) { status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY, - index, mOutputs.keyAt(i), curDevice); + index, desc, curDevice); } } return status; @@ -1575,7 +1640,7 @@ status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream, } device = Volume::getDeviceForVolume(device); - *index = mStreams[stream].getVolumeIndex(device); + *index = mStreams.valueFor(stream).getVolumeIndex(device); ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); return NO_ERROR; } @@ -1599,7 +1664,7 @@ audio_io_handle_t AudioPolicyManager::selectOutputForEffects( audio_io_handle_t outputDeepBuffer = 0; for (size_t i = 0; i < outputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags); if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { outputOffloaded = outputs[i]; @@ -1653,6 +1718,16 @@ status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc, return mEffects.registerEffect(desc, io, strategy, session, id); } +bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const +{ + return mOutputs.isStreamActive(stream, inPastMs); +} + +bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const +{ + return mOutputs.isStreamActiveRemotely(stream, inPastMs); +} + bool AudioPolicyManager::isSourceActive(audio_source_t source) const { for (size_t i = 0; i < mInputs.size(); i++) { @@ -1803,7 +1878,7 @@ status_t AudioPolicyManager::dump(int fd) snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this); result.append(buffer); - snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput); + snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput->mIoHandle); result.append(buffer); snprintf(buffer, SIZE, " Phone state: %d\n", mEngine->getPhoneState()); result.append(buffer); @@ -2021,7 +2096,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, } if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id); return BAD_VALUE; @@ -2055,9 +2130,12 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, patch->sources[0].sample_rate, NULL, // updatedSamplingRate patch->sources[0].format, + NULL, // updatedFormat patch->sources[0].channel_mask, + NULL, // updatedChannelMask AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) { - ALOGV("createAudioPatch() profile not supported for device %08x", devDesc->type()); + ALOGV("createAudioPatch() profile not supported for device %08x", + devDesc->type()); return INVALID_OPERATION; } devices.add(devDesc); @@ -2069,7 +2147,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, // TODO: reconfigure output format and channels here ALOGV("createAudioPatch() setting device %08x on output %d", devices.types(), outputDesc->mIoHandle); - setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle); + setOutputDevice(outputDesc, devices.types(), true, 0, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { @@ -2109,7 +2187,9 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, patch->sinks[0].sample_rate, NULL, /*updatedSampleRate*/ patch->sinks[0].format, + NULL, /*updatedFormat*/ patch->sinks[0].channel_mask, + NULL, /*updatedChannelMask*/ // FIXME for the parameter type, // and the NONE (audio_output_flags_t) @@ -2163,8 +2243,12 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, } sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]); - if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) { - // only one sink supported when connected devices across HW modules + // create a software bridge in PatchPanel if: + // - source and sink devices are on differnt HW modules OR + // - audio HAL version is < 3.0 + if ((srcDeviceDesc->getModuleHandle() != sinkDeviceDesc->getModuleHandle()) || + (srcDeviceDesc->mModule->mHalVersion < AUDIO_DEVICE_API_VERSION_3_0)) { + // support only one sink device for now to simplify output selection logic if (patch->num_sinks > 1) { return INVALID_OPERATION; } @@ -2181,6 +2265,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, return INVALID_OPERATION; } outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]); + newPatch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; newPatch.num_sources = 2; } } @@ -2242,14 +2327,14 @@ status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle, struct audio_patch *patch = &patchDesc->mPatch; patchDesc->mUid = mUidCached; if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id); return BAD_VALUE; } - setOutputDevice(outputDesc->mIoHandle, - getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/), + setOutputDevice(outputDesc, + getNewOutputDevice(outputDesc, true /*fromCache*/), true, 0, NULL); @@ -2308,7 +2393,7 @@ status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config * sp<AudioPortConfig> audioPortConfig; if (config->type == AUDIO_PORT_TYPE_MIX) { if (config->role == AUDIO_PORT_ROLE_SOURCE) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id); if (outputDesc == NULL) { return BAD_VALUE; } @@ -2390,7 +2475,6 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa #ifdef AUDIO_POLICY_TEST Thread(false), #endif //AUDIO_POLICY_TEST - mPrimaryOutput((audio_io_handle_t)0), mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), mA2dpSuspended(false), mSpeakerDrcEnabled(false), @@ -2474,7 +2558,8 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa if ((profileType & outputDeviceTypes) == 0) { continue; } - sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile); + sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile, + mpClientInterface); outputDesc->mDevice = profileType; audio_config_t config = AUDIO_CONFIG_INITIALIZER; @@ -2482,7 +2567,7 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa config.channel_mask = outputDesc->mChannelMask; config.format = outputDesc->mFormat; audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle, + status_t status = mpClientInterface->openOutput(outProfile->getModuleHandle(), &output, &config, &outputDesc->mDevice, @@ -2510,10 +2595,10 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa } if (mPrimaryOutput == 0 && outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { - mPrimaryOutput = output; + mPrimaryOutput = outputDesc; } addOutput(output, outputDesc); - setOutputDevice(output, + setOutputDevice(outputDesc, outputDesc->mDevice, true); } @@ -2558,7 +2643,7 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa config.channel_mask = inputDesc->mChannelMask; config.format = inputDesc->mFormat; audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle, + status_t status = mpClientInterface->openInput(inProfile->getModuleHandle(), &input, &config, &inputDesc->mDevice, @@ -2620,7 +2705,7 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa if (mPrimaryOutput != 0) { AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); + mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, outputCmd.toString()); mTestDevice = AUDIO_DEVICE_OUT_SPEAKER; mTestSamplingRate = 44100; @@ -2760,20 +2845,21 @@ bool AudioPolicyManager::threadLoop() if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_reopen")); - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput); - mpClientInterface->closeOutput(mPrimaryOutput); + mpClientInterface->closeOutput(mpClientInterface->closeOutput(mPrimaryOutput);); - audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; + audio_module_handle_t moduleHandle = mPrimaryOutput->getModuleHandle(); - removeOutput(mPrimaryOutput); - sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL); + removeOutput(mPrimaryOutput->mIoHandle); + sp<SwAudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL, + mpClientInterface); outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = outputDesc->mSamplingRate; config.channel_mask = outputDesc->mChannelMask; config.format = outputDesc->mFormat; + audio_io_handle_t handle; status_t status = mpClientInterface->openOutput(moduleHandle, - &mPrimaryOutput, + &handle, &config, &outputDesc->mDevice, String8(""), @@ -2787,10 +2873,11 @@ bool AudioPolicyManager::threadLoop() outputDesc->mSamplingRate = config.sample_rate; outputDesc->mChannelMask = config.channel_mask; outputDesc->mFormat = config.format; + mPrimaryOutput = outputDesc; AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); - addOutput(mPrimaryOutput, outputDesc); + mpClientInterface->setParameters(handle, outputCmd.toString()); + addOutput(handle, outputDesc); } } @@ -2822,7 +2909,7 @@ int AudioPolicyManager::testOutputIndex(audio_io_handle_t output) // --- -void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc) +void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<SwAudioOutputDescriptor> outputDesc) { outputDesc->setIoHandle(output); mOutputs.add(output, outputDesc); @@ -2841,7 +2928,7 @@ void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescript nextAudioPortGeneration(); } -void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/, +void AudioPolicyManager::findIoHandlesByAddress(sp<SwAudioOutputDescriptor> desc /*in*/, const audio_devices_t device /*in*/, const String8 address /*in*/, SortedVector<audio_io_handle_t>& outputs /*out*/) { @@ -2860,7 +2947,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de const String8 address) { audio_devices_t device = devDesc->type(); - sp<AudioOutputDescriptor> desc; + sp<SwAudioOutputDescriptor> desc; // erase all current sample rates, formats and channel masks devDesc->clearCapabilities(); @@ -2868,7 +2955,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de // first list already open outputs that can be routed to this device for (size_t i = 0; i < mOutputs.size(); i++) { desc = mOutputs.valueAt(i); - if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) { + if (!desc->isDuplicated() && (desc->supportedDevices() & device)) { if (!device_distinguishes_on_address(device)) { ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); @@ -2927,7 +3014,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de ALOGV("opening output for device %08x with params %s profile %p", device, address.string(), profile.get()); - desc = new AudioOutputDescriptor(profile); + desc = new SwAudioOutputDescriptor(profile, mpClientInterface); desc->mDevice = device; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = desc->mSamplingRate; @@ -2937,7 +3024,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de config.offload_info.channel_mask = desc->mChannelMask; config.offload_info.format = desc->mFormat; audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openOutput(profile->mModule->mHandle, + status_t status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &desc->mDevice, @@ -3007,7 +3094,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de config.offload_info.sample_rate = config.sample_rate; config.offload_info.channel_mask = config.channel_mask; config.offload_info.format = config.format; - status = mpClientInterface->openOutput(profile->mModule->mHandle, + status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &desc->mDevice, @@ -3032,7 +3119,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de address.string()); } policyMix->setOutput(desc); - desc->mPolicyMix = &(policyMix->getMix()); + desc->mPolicyMix = policyMix->getMix(); } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) { // no duplicated output for direct outputs and @@ -3040,28 +3127,29 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE; // set initial stream volume for device - applyStreamVolumes(output, device, 0, true); + applyStreamVolumes(desc, device, 0, true); //TODO: configure audio effect output stage here // open a duplicating output thread for the new output and the primary output - duplicatedOutput = mpClientInterface->openDuplicateOutput(output, - mPrimaryOutput); + duplicatedOutput = + mpClientInterface->openDuplicateOutput(output, + mPrimaryOutput->mIoHandle); if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) { // add duplicated output descriptor - sp<AudioOutputDescriptor> dupOutputDesc = - new AudioOutputDescriptor(NULL); - dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); - dupOutputDesc->mOutput2 = mOutputs.valueFor(output); + sp<SwAudioOutputDescriptor> dupOutputDesc = + new SwAudioOutputDescriptor(NULL, mpClientInterface); + dupOutputDesc->mOutput1 = mPrimaryOutput; + dupOutputDesc->mOutput2 = desc; dupOutputDesc->mSamplingRate = desc->mSamplingRate; dupOutputDesc->mFormat = desc->mFormat; dupOutputDesc->mChannelMask = desc->mChannelMask; dupOutputDesc->mLatency = desc->mLatency; addOutput(duplicatedOutput, dupOutputDesc); - applyStreamVolumes(duplicatedOutput, device, 0, true); + applyStreamVolumes(dupOutputDesc, device, 0, true); } else { ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", - mPrimaryOutput, output); + mPrimaryOutput->mIoHandle, output); mpClientInterface->closeOutput(output); removeOutput(output); nextAudioPortGeneration(); @@ -3083,7 +3171,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de if (device_distinguishes_on_address(device)) { ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)", device, address.string()); - setOutputDevice(output, device, true/*force*/, 0/*delay*/, + setOutputDevice(desc, device, true/*force*/, 0/*delay*/, NULL/*patch handle*/, address.string()); } ALOGV("checkOutputsForDevice(): adding output %d", output); @@ -3101,10 +3189,9 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de if (!desc->isDuplicated()) { // exact match on device if (device_distinguishes_on_address(device) && - (desc->mProfile->mSupportedDevices.types() == device)) { + (desc->supportedDevices() == device)) { findIoHandlesByAddress(desc, device, address, outputs); - } else if (!(desc->mProfile->mSupportedDevices.types() - & mAvailableOutputDevices.types())) { + } else if (!(desc->supportedDevices() & mAvailableOutputDevices.types())) { ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); @@ -3212,7 +3299,7 @@ status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device, config.channel_mask = desc->mChannelMask; config.format = desc->mFormat; audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openInput(profile->mModule->mHandle, + status_t status = mpClientInterface->openInput(profile->getModuleHandle(), &input, &config, &desc->mDevice, @@ -3339,7 +3426,7 @@ void AudioPolicyManager::closeOutput(audio_io_handle_t output) { ALOGV("closeOutput(%d)", output); - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); if (outputDesc == NULL) { ALOGW("closeOutput() unknown output %d", output); return; @@ -3348,7 +3435,7 @@ void AudioPolicyManager::closeOutput(audio_io_handle_t output) // look for duplicated outputs connected to the output being removed. for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i); + sp<SwAudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i); if (dupOutputDesc->isDuplicated() && (dupOutputDesc->mOutput1 == outputDesc || dupOutputDesc->mOutput2 == outputDesc)) { @@ -3417,8 +3504,9 @@ void AudioPolicyManager::closeInput(audio_io_handle_t input) mInputs.removeItem(input); } -SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device, - AudioOutputCollection openOutputs) +SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice( + audio_devices_t device, + SwAudioOutputCollection openOutputs) { SortedVector<audio_io_handle_t> outputs; @@ -3459,14 +3547,14 @@ void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) // associated with policies in the "before" and "after" output vectors ALOGVV("checkOutputForStrategy(): policy related outputs"); for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) { - const sp<AudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i); + const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i); if (desc != 0 && desc->mPolicyMix != NULL) { srcOutputs.add(desc->mIoHandle); ALOGVV(" previous outputs: adding %d", desc->mIoHandle); } } for (size_t i = 0 ; i < mOutputs.size() ; i++) { - const sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); + const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); if (desc != 0 && desc->mPolicyMix != NULL) { dstOutputs.add(desc->mIoHandle); ALOGVV(" new outputs: adding %d", desc->mIoHandle); @@ -3478,10 +3566,10 @@ void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) strategy, srcOutputs[0], dstOutputs[0]); // mute strategy while moving tracks from one output to another for (size_t i = 0; i < srcOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]); if (isStrategyActive(desc, strategy)) { - setStrategyMute(strategy, true, srcOutputs[i]); - setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); + setStrategyMute(strategy, true, desc); + setStrategyMute(strategy, false, desc, MUTE_TIME_MS, newDevice); } } @@ -3578,12 +3666,11 @@ void AudioPolicyManager::checkA2dpSuspend() } } -audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache) +audio_devices_t AudioPolicyManager::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, + bool fromCache) { audio_devices_t device = AUDIO_DEVICE_NONE; - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); - ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); if (index >= 0) { sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index); @@ -3761,9 +3848,9 @@ uint32_t AudioPolicyManager::setBeaconMute(bool mute) { ALOGV("\t muting %d", mute); uint32_t maxLatency = 0; for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); setStreamMute(AUDIO_STREAM_TTS, mute/*on*/, - desc->mIoHandle, + desc, 0 /*delay*/, AUDIO_DEVICE_NONE); const uint32_t latency = desc->latency() * 2; if (latency > maxLatency) { @@ -3779,6 +3866,21 @@ uint32_t AudioPolicyManager::setBeaconMute(bool mute) { audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy, bool fromCache) { + // Routing + // see if we have an explicit route + // scan the whole RouteMap, for each entry, convert the stream type to a strategy + // (getStrategy(stream)). + // if the strategy from the stream type in the RouteMap is the same as the argument above, + // and activity count is non-zero + // the device = the device from the descriptor in the RouteMap, and exit. + for (size_t routeIndex = 0; routeIndex < mOutputRoutes.size(); routeIndex++) { + sp<SessionRoute> route = mOutputRoutes.valueAt(routeIndex); + routing_strategy strat = getStrategy(route->mStreamType); + if (strat == strategy && route->mDeviceDescriptor != 0 /*&& route->mActivityCount != 0*/) { + return route->mDeviceDescriptor->type(); + } + } + if (fromCache) { ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]); @@ -3812,7 +3914,7 @@ uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> for (size_t i = 0; i < NUM_STRATEGIES; i++) { audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); - curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types(); + curDevice = curDevice & outputDesc->supportedDevices(); bool mute = shouldMute && (curDevice & device) && (curDevice != device); bool doMute = false; @@ -3831,10 +3933,9 @@ uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> == AUDIO_DEVICE_NONE) { continue; } - audio_io_handle_t curOutput = mOutputs.keyAt(j); - ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", - mute ? "muting" : "unmuting", i, curDevice, curOutput); - setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs); + ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x)", + mute ? "muting" : "unmuting", i, curDevice); + setStrategyMute((routing_strategy)i, mute, desc, mute ? 0 : delayMs); if (isStrategyActive(desc, (routing_strategy)i)) { if (mute) { // FIXME: should not need to double latency if volume could be applied @@ -3859,9 +3960,9 @@ uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> } for (size_t i = 0; i < NUM_STRATEGIES; i++) { if (isStrategyActive(outputDesc, (routing_strategy)i)) { - setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle); + setStrategyMute((routing_strategy)i, true, outputDesc); // do tempMute unmute after twice the mute wait time - setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle, + setStrategyMute((routing_strategy)i, false, outputDesc, muteWaitMs *2, device); } } @@ -3876,36 +3977,35 @@ uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> return 0; } -uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, +uint32_t AudioPolicyManager::setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, audio_devices_t device, bool force, int delayMs, audio_patch_handle_t *patchHandle, const char* address) { - ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); + ALOGV("setOutputDevice() device %04x delayMs %d", device, delayMs); AudioParameter param; uint32_t muteWaitMs; if (outputDesc->isDuplicated()) { - muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs); - muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs); + muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs); + muteWaitMs += setOutputDevice(outputDesc->subOutput2(), device, force, delayMs); return muteWaitMs; } // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current // output profile if ((device != AUDIO_DEVICE_NONE) && - ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) { + ((device & outputDesc->supportedDevices()) == 0)) { return 0; } // filter devices according to output selected - device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types()); + device = (audio_devices_t)(device & outputDesc->supportedDevices()); audio_devices_t prevDevice = outputDesc->mDevice; - ALOGV("setOutputDevice() prevDevice %04x", prevDevice); + ALOGV("setOutputDevice() prevDevice 0x%04x", prevDevice); if (device != AUDIO_DEVICE_NONE) { outputDesc->mDevice = device; @@ -3918,10 +4018,10 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, // AND force is not specified // AND the output is connected by a valid audio patch. // Doing this check here allows the caller to call setOutputDevice() without conditions - if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force && - outputDesc->mPatchHandle != 0) { - ALOGV("setOutputDevice() setting same device %04x or null device for output %d", - device, output); + if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && + !force && + outputDesc->mPatchHandle != 0) { + ALOGV("setOutputDevice() setting same device 0x%04x or null device", device); return muteWaitMs; } @@ -3929,7 +4029,7 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, // do the routing if (device == AUDIO_DEVICE_NONE) { - resetOutputDevice(output, delayMs, NULL); + resetOutputDevice(outputDesc, delayMs, NULL); } else { DeviceVector deviceList = (address == NULL) ? mAvailableOutputDevices.getDevicesFromType(device) @@ -3996,16 +4096,15 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, } // update stream volumes according to new device - applyStreamVolumes(output, device, delayMs); + applyStreamVolumes(outputDesc, device, delayMs); return muteWaitMs; } -status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output, +status_t AudioPolicyManager::resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, int delayMs, audio_patch_handle_t *patchHandle) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); ssize_t index; if (patchHandle) { index = mAudioPatches.indexOfKey(*patchHandle); @@ -4115,12 +4214,15 @@ status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input, sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device, String8 address, uint32_t& samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, + audio_format_t& format, + audio_channel_mask_t& channelMask, audio_input_flags_t flags) { // Choose an input profile based on the requested capture parameters: select the first available // profile supporting all requested parameters. + // + // TODO: perhaps isCompatibleProfile should return a "matching" score so we can return + // the best matching profile, not the first one. for (size_t i = 0; i < mHwModules.size(); i++) { @@ -4133,7 +4235,11 @@ sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device, // profile->log(); if (profile->isCompatibleProfile(device, address, samplingRate, &samplingRate /*updatedSamplingRate*/, - format, channelMask, (audio_output_flags_t) flags)) { + format, + &format /*updatedFormat*/, + channelMask, + &channelMask /*updatedChannelMask*/, + (audio_output_flags_t) flags)) { return profile; } @@ -4162,17 +4268,10 @@ audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t input } float AudioPolicyManager::computeVolume(audio_stream_type_t stream, - int index, - audio_io_handle_t output, - audio_devices_t device) + int index, + audio_devices_t device) { - float volume = 1.0; - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); - - if (device == AUDIO_DEVICE_NONE) { - device = outputDesc->device(); - } - volume = mEngine->volIndexToAmpl(Volume::getDeviceCategory(device), stream, index); + float volumeDb = mEngine->volIndexToDb(Volume::getDeviceCategory(device), stream, index); // if a headset is connected, apply the following rules to ring tones and notifications // to avoid sound level bursts in user's ears: @@ -4190,41 +4289,39 @@ float AudioPolicyManager::computeVolume(audio_stream_type_t stream, || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) && mStreams.canBeMuted(stream)) { - volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; + volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB; // when the phone is ringing we must consider that music could have been paused just before // by the music application and behave as if music was active if the last music track was // just stopped if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || mLimitRingtoneVolume) { audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); - float musicVol = computeVolume(AUDIO_STREAM_MUSIC, - mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice), - output, + float musicVolDB = computeVolume(AUDIO_STREAM_MUSIC, + mStreams.valueFor(AUDIO_STREAM_MUSIC).getVolumeIndex(musicDevice), musicDevice); - float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? - musicVol : SONIFICATION_HEADSET_VOLUME_MIN; - if (volume > minVol) { - volume = minVol; - ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); + float minVolDB = (musicVolDB > SONIFICATION_HEADSET_VOLUME_MIN_DB) ? + musicVolDB : SONIFICATION_HEADSET_VOLUME_MIN_DB; + if (volumeDb > minVolDB) { + volumeDb = minVolDB; + ALOGV("computeVolume limiting volume to %f musicVol %f", minVolDB, musicVolDB); } } } - return volume; + return volumeDb; } status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, - int index, - audio_io_handle_t output, - audio_devices_t device, - int delayMs, - bool force) + int index, + const sp<AudioOutputDescriptor>& outputDesc, + audio_devices_t device, + int delayMs, + bool force) { - // do not change actual stream volume if the stream is muted - if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { + if (outputDesc->mMuteCount[stream] != 0) { ALOGVV("checkAndSetVolume() stream %d muted count %d", - stream, mOutputs.valueFor(output)->mMuteCount[stream]); + stream, outputDesc->mMuteCount[stream]); return NO_ERROR; } audio_policy_forced_cfg_t forceUseForComm = @@ -4237,45 +4334,28 @@ status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, return INVALID_OPERATION; } - float volume = computeVolume(stream, index, output, device); - // unit gain if rerouting to external policy - if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { - ssize_t index = mOutputs.indexOfKey(output); - if (index >= 0) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); - if (outputDesc->mPolicyMix != NULL) { - ALOGV("max gain when rerouting for output=%d", output); - volume = 1.0f; - } - } - + if (device == AUDIO_DEVICE_NONE) { + device = outputDesc->device(); } - // We actually change the volume if: - // - the float value returned by computeVolume() changed - // - the force flag is set - if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || - force) { - mOutputs.valueFor(output)->mCurVolume[stream] = volume; - ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); - // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is - // enabled - if (stream == AUDIO_STREAM_BLUETOOTH_SCO) { - mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs); - } - mpClientInterface->setStreamVolume(stream, volume, output, delayMs); + + float volumeDb = computeVolume(stream, index, device); + if (outputDesc->isFixedVolume(device)) { + volumeDb = 0.0f; } + outputDesc->setVolume(volumeDb, stream, device, delayMs, force); + if (stream == AUDIO_STREAM_VOICE_CALL || stream == AUDIO_STREAM_BLUETOOTH_SCO) { float voiceVolume; // Force voice volume to max for bluetooth SCO as volume is managed by the headset if (stream == AUDIO_STREAM_VOICE_CALL) { - voiceVolume = (float)index/(float)mStreams[stream].getVolumeIndexMax(); + voiceVolume = (float)index/(float)mStreams.valueFor(stream).getVolumeIndexMax(); } else { voiceVolume = 1.0; } - if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) { + if (voiceVolume != mLastVoiceVolume && outputDesc == mPrimaryOutput) { mpClientInterface->setVoiceVolume(voiceVolume, delayMs); mLastVoiceVolume = voiceVolume; } @@ -4284,20 +4364,20 @@ status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, return NO_ERROR; } -void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output, - audio_devices_t device, - int delayMs, - bool force) +void AudioPolicyManager::applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc, + audio_devices_t device, + int delayMs, + bool force) { - ALOGVV("applyStreamVolumes() for output %d and device %x", output, device); + ALOGVV("applyStreamVolumes() for device %08x", device); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } checkAndSetVolume((audio_stream_type_t)stream, - mStreams[stream].getVolumeIndex(device), - output, + mStreams.valueFor((audio_stream_type_t)stream).getVolumeIndex(device), + outputDesc, device, delayMs, force); @@ -4305,10 +4385,10 @@ void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output, } void AudioPolicyManager::setStrategyMute(routing_strategy strategy, - bool on, - audio_io_handle_t output, - int delayMs, - audio_devices_t device) + bool on, + const sp<AudioOutputDescriptor>& outputDesc, + int delayMs, + audio_devices_t device) { ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { @@ -4316,32 +4396,31 @@ void AudioPolicyManager::setStrategyMute(routing_strategy strategy, continue; } if (getStrategy((audio_stream_type_t)stream) == strategy) { - setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device); + setStreamMute((audio_stream_type_t)stream, on, outputDesc, delayMs, device); } } } void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, - bool on, - audio_io_handle_t output, - int delayMs, - audio_devices_t device) + bool on, + const sp<AudioOutputDescriptor>& outputDesc, + int delayMs, + audio_devices_t device) { - const StreamDescriptor &streamDesc = mStreams[stream]; - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); + const StreamDescriptor& streamDesc = mStreams.valueFor(stream); if (device == AUDIO_DEVICE_NONE) { device = outputDesc->device(); } - ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", - stream, on, output, outputDesc->mMuteCount[stream], device); + ALOGVV("setStreamMute() stream %d, mute %d, mMuteCount %d device %04x", + stream, on, outputDesc->mMuteCount[stream], device); if (on) { if (outputDesc->mMuteCount[stream] == 0) { if (streamDesc.canBeMuted() && ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) || (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) { - checkAndSetVolume(stream, 0, output, device, delayMs); + checkAndSetVolume(stream, 0, outputDesc, device, delayMs); } } // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored @@ -4354,7 +4433,7 @@ void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, if (--outputDesc->mMuteCount[stream] == 0) { checkAndSetVolume(stream, streamDesc.getVolumeIndex(device), - output, + outputDesc, device, delayMs); } @@ -4373,7 +4452,7 @@ void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, const routing_strategy stream_strategy = getStrategy(stream); if ((stream_strategy == STRATEGY_SONIFICATION) || ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput); + sp<SwAudioOutputDescriptor> outputDesc = mPrimaryOutput; ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", stream, starting, outputDesc->mDevice, stateChange); if (outputDesc->mRefCount[stream]) { @@ -4406,6 +4485,70 @@ void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, } } +// --- SessionRoute class implementation +void AudioPolicyManager::SessionRoute::log(const char* prefix) { + ALOGI("%s[SessionRoute strm:0x%X, sess:0x%X, dev:0x%X refs:%d act:%d", + prefix, mStreamType, mSession, + mDeviceDescriptor != 0 ? mDeviceDescriptor->type() : AUDIO_DEVICE_NONE, + mRefCount, mActivityCount); +} + +// --- SessionRouteMap class implementation +bool AudioPolicyManager::SessionRouteMap::hasRoute(audio_session_t session) +{ + return indexOfKey(session) >= 0 && valueFor(session)->mDeviceDescriptor != 0; +} + +void AudioPolicyManager::SessionRouteMap::addRoute(audio_session_t session, + audio_stream_type_t streamType, + sp<DeviceDescriptor> deviceDescriptor) +{ + sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0; + if (route != NULL) { + route->mRefCount++; + route->mDeviceDescriptor = deviceDescriptor; + } else { + route = new AudioPolicyManager::SessionRoute(session, streamType, deviceDescriptor); + route->mRefCount++; + add(session, route); + } +} + +void AudioPolicyManager::SessionRouteMap::removeRoute(audio_session_t session) +{ + sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0; + if (route != 0) { + ALOG_ASSERT(route->mRefCount > 0); + --route->mRefCount; + if (route->mRefCount <= 0) { + removeItem(session); + } + } +} + +int AudioPolicyManager::SessionRouteMap::incRouteActivity(audio_session_t session) +{ + sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0; + return route != 0 ? ++(route->mActivityCount) : -1; +} + +int AudioPolicyManager::SessionRouteMap::decRouteActivity(audio_session_t session) +{ + sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0; + if (route != 0 && route->mActivityCount > 0) { + return --(route->mActivityCount); + } else { + return -1; + } +} + +void AudioPolicyManager::SessionRouteMap::log(const char* caption) { + ALOGI("%s ----", caption); + for(size_t index = 0; index < size(); index++) { + valueAt(index)->log(" "); + } +} + void AudioPolicyManager::defaultAudioPolicyConfig(void) { sp<HwModule> module; @@ -4417,7 +4560,8 @@ void AudioPolicyManager::defaultAudioPolicyConfig(void) module = new HwModule("primary"); - profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module); + profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE); + profile->attach(module); profile->mSamplingRates.add(44100); profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); @@ -4425,7 +4569,8 @@ void AudioPolicyManager::defaultAudioPolicyConfig(void) profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; module->mOutputProfiles.add(profile); - profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module); + profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK); + profile->attach(module); profile->mSamplingRates.add(8000); profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h index 02b678a..fe6b986 100644 --- a/services/audiopolicy/managerdefault/AudioPolicyManager.h +++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h @@ -49,8 +49,11 @@ namespace android { // Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB #define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 +#define SONIFICATION_HEADSET_VOLUME_FACTOR_DB (-6) // Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB #define SONIFICATION_HEADSET_VOLUME_MIN 0.016 +#define SONIFICATION_HEADSET_VOLUME_MIN_DB (-36) + // Time in milliseconds during which we consider that music is still active after a music // track was stopped - see computeVolume() #define SONIFICATION_HEADSET_MUSIC_DELAY 5000 @@ -110,6 +113,7 @@ public: audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + audio_port_handle_t selectedDeviceId, const audio_offload_info_t *offloadInfo); virtual status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, @@ -172,19 +176,15 @@ public: return mEffects.setEffectEnabled(id, enabled); } - virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const - { - return mOutputs.isStreamActive(stream, inPastMs); - } + virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; // return whether a stream is playing remotely, override to change the definition of // local/remote playback, used for instance by notification manager to not make // media players lose audio focus when not playing locally // For the base implementation, "remotely" means playing during screen mirroring which // uses an output for playback with a non-empty, non "0" address. - virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const - { - return mOutputs.isStreamActiveRemotely(stream, inPastMs); - } + virtual bool isStreamActiveRemotely(audio_stream_type_t stream, + uint32_t inPastMs = 0) const; + virtual bool isSourceActive(audio_source_t source) const; virtual status_t dump(int fd); @@ -227,6 +227,46 @@ public: // return the strategy corresponding to a given stream type routing_strategy getStrategy(audio_stream_type_t stream) const; +protected: + class SessionRoute : public RefBase + { + public: + friend class SessionRouteMap; + SessionRoute(audio_session_t session, + audio_stream_type_t streamType, + sp<DeviceDescriptor> deviceDescriptor) + : mSession(session), + mStreamType(streamType), + mDeviceDescriptor(deviceDescriptor), + mRefCount(0), + mActivityCount(0) {} + + audio_session_t mSession; + audio_stream_type_t mStreamType; + + sp<DeviceDescriptor> mDeviceDescriptor; + + // "reference" counting + int mRefCount; // +/- on references + int mActivityCount; // +/- on start/stop + + void log(const char* prefix); + }; + + class SessionRouteMap: public KeyedVector<audio_session_t, sp<SessionRoute>> + { + public: + bool hasRoute(audio_session_t session); + void addRoute(audio_session_t session, audio_stream_type_t streamType, + sp<DeviceDescriptor> deviceDescriptor); + void removeRoute(audio_session_t session); + + int incRouteActivity(audio_session_t session); + int decRouteActivity(audio_session_t session); + + void log(const char* caption); + }; + // From AudioPolicyManagerObserver virtual const AudioPatchCollection &getAudioPatches() const { @@ -240,7 +280,7 @@ public: { return mPolicyMixes; } - virtual const AudioOutputCollection &getOutputs() const + virtual const SwAudioOutputCollection &getOutputs() const { return mOutputs; } @@ -265,7 +305,7 @@ public: return mDefaultOutputDevice; } protected: - void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc); + void addOutput(audio_io_handle_t output, sp<SwAudioOutputDescriptor> outputDesc); void removeOutput(audio_io_handle_t output); void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc); @@ -288,13 +328,13 @@ protected: // change the route of the specified output. Returns the number of ms we have slept to // allow new routing to take effect in certain cases. - virtual uint32_t setOutputDevice(audio_io_handle_t output, + virtual uint32_t setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, audio_devices_t device, bool force = false, int delayMs = 0, audio_patch_handle_t *patchHandle = NULL, const char* address = NULL); - status_t resetOutputDevice(audio_io_handle_t output, + status_t resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, int delayMs = 0, audio_patch_handle_t *patchHandle = NULL); status_t setInputDevice(audio_io_handle_t input, @@ -309,29 +349,31 @@ protected: // compute the actual volume for a given stream according to the requested index and a particular // device - virtual float computeVolume(audio_stream_type_t stream, int index, - audio_io_handle_t output, audio_devices_t device); + virtual float computeVolume(audio_stream_type_t stream, + int index, + audio_devices_t device); // check that volume change is permitted, compute and send new volume to audio hardware virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index, - audio_io_handle_t output, + const sp<AudioOutputDescriptor>& outputDesc, audio_devices_t device, int delayMs = 0, bool force = false); // apply all stream volumes to the specified output and device - void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); + void applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc, + audio_devices_t device, int delayMs = 0, bool force = false); // Mute or unmute all streams handled by the specified strategy on the specified output void setStrategyMute(routing_strategy strategy, bool on, - audio_io_handle_t output, + const sp<AudioOutputDescriptor>& outputDesc, int delayMs = 0, audio_devices_t device = (audio_devices_t)0); // Mute or unmute the stream on the specified output void setStreamMute(audio_stream_type_t stream, bool on, - audio_io_handle_t output, + const sp<AudioOutputDescriptor>& outputDesc, int delayMs = 0, audio_devices_t device = (audio_devices_t)0); @@ -384,7 +426,8 @@ protected: // must be called every time a condition that affects the device choice for a given output is // changed: connected device, phone state, force use, output start, output stop.. // see getDeviceForStrategy() for the use of fromCache parameter - audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache); + audio_devices_t getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, + bool fromCache); // updates cache of device used by all strategies (mDeviceForStrategy[]) // must be called every time a condition that affects the device choice for a given strategy is @@ -412,7 +455,7 @@ protected: #endif //AUDIO_POLICY_TEST SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device, - AudioOutputCollection openOutputs); + SwAudioOutputCollection openOutputs); bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, SortedVector<audio_io_handle_t>& outputs2); @@ -427,12 +470,12 @@ protected: audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs, audio_output_flags_t flags, audio_format_t format); - // samplingRate parameter is an in/out and so may be modified + // samplingRate, format, channelMask are in/out and so may be modified sp<IOProfile> getInputProfile(audio_devices_t device, String8 address, uint32_t& samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, + audio_format_t& format, + audio_channel_mask_t& channelMask, audio_input_flags_t flags); sp<IOProfile> getProfileForDirectOutput(audio_devices_t device, uint32_t samplingRate, @@ -453,28 +496,39 @@ protected: audio_devices_t availablePrimaryOutputDevices() const { - return mOutputs.getSupportedDevices(mPrimaryOutput) & mAvailableOutputDevices.types(); + return mPrimaryOutput->supportedDevices() & mAvailableOutputDevices.types(); } audio_devices_t availablePrimaryInputDevices() const { - return mAvailableInputDevices.getDevicesFromHwModule( - mOutputs.valueFor(mPrimaryOutput)->getModuleHandle()); + return mAvailableInputDevices.getDevicesFromHwModule(mPrimaryOutput->getModuleHandle()); } void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0); + status_t startSource(sp<AudioOutputDescriptor> outputDesc, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t *delayMs); + status_t stopSource(sp<AudioOutputDescriptor> outputDesc, + audio_stream_type_t stream); + uid_t mUidCached; AudioPolicyClientInterface *mpClientInterface; // audio policy client interface - audio_io_handle_t mPrimaryOutput; // primary output handle + sp<SwAudioOutputDescriptor> mPrimaryOutput; // primary output descriptor // list of descriptors for outputs currently opened - AudioOutputCollection mOutputs; + + SwAudioOutputCollection mOutputs; // copy of mOutputs before setDeviceConnectionState() opens new outputs // reset to mOutputs when updateDevicesAndOutputs() is called. - AudioOutputCollection mPreviousOutputs; + SwAudioOutputCollection mPreviousOutputs; AudioInputCollection mInputs; // list of input descriptors + DeviceVector mAvailableOutputDevices; // all available output devices DeviceVector mAvailableInputDevices; // all available input devices + SessionRouteMap mOutputRoutes; + SessionRouteMap mInputRoutes; + StreamDescriptorCollection mStreams; // stream descriptors for volume control bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected audio_devices_t mDeviceForStrategy[NUM_STRATEGIES]; @@ -539,7 +593,7 @@ private: // in mProfile->mSupportedDevices) matches the device whose address is to be matched. // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one // where addresses are used to distinguish between one connected device and another. - void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/, + void findIoHandlesByAddress(sp<SwAudioOutputDescriptor> desc /*in*/, const audio_devices_t device /*in*/, const String8 address /*in*/, SortedVector<audio_io_handle_t>& outputs /*out*/); diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp index e9ff838..9510727 100644 --- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp +++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp @@ -150,6 +150,7 @@ status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + int mSelectedDeviceId, const audio_offload_info_t *offloadInfo) { if (mAudioPolicyManager == NULL) { @@ -158,7 +159,7 @@ status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, ALOGV("getOutput()"); Mutex::Autolock _l(mLock); return mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, samplingRate, - format, channelMask, flags, offloadInfo); + format, channelMask, flags, mSelectedDeviceId, offloadInfo); } status_t AudioPolicyService::startOutput(audio_io_handle_t output, @@ -261,8 +262,7 @@ status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr, return BAD_VALUE; } - if (((attr->source == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) || - ((attr->source == AUDIO_SOURCE_FM_TUNER) && !captureFmTunerAllowed())) { + if ((attr->source == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) { return BAD_VALUE; } sp<AudioPolicyEffects>audioPolicyEffects; diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp index 5a91192..e4ca5dc 100644 --- a/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp +++ b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp @@ -255,8 +255,7 @@ status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr, inputSource = AUDIO_SOURCE_MIC; } - if (((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) || - ((inputSource == AUDIO_SOURCE_FM_TUNER) && !captureFmTunerAllowed())) { + if ((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) { return BAD_VALUE; } @@ -569,6 +568,7 @@ status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + int selectedDeviceId __unused, const audio_offload_info_t *offloadInfo) { if (attr != NULL) { diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h index 0378384..f8dabd3 100644 --- a/services/audiopolicy/service/AudioPolicyService.h +++ b/services/audiopolicy/service/AudioPolicyService.h @@ -84,6 +84,7 @@ public: audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = 0, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, + int selectedDeviceId = AUDIO_PORT_HANDLE_NONE, const audio_offload_info_t *offloadInfo = NULL); virtual status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp index e9c96c6..414d563 100644 --- a/services/camera/libcameraservice/CameraService.cpp +++ b/services/camera/libcameraservice/CameraService.cpp @@ -122,7 +122,7 @@ static void torch_mode_status_change( // should be ok for now. static CameraService *gCameraService; -CameraService::CameraService() : mEventLog(DEFAULT_EVICTION_LOG_LENGTH), +CameraService::CameraService() : mEventLog(DEFAULT_EVENT_LOG_LENGTH), mLastUserId(DEFAULT_LAST_USER_ID), mSoundRef(0), mModule(0), mFlashlight(0) { ALOGI("CameraService started (pid=%d)", getpid()); gCameraService = this; @@ -242,6 +242,8 @@ void CameraService::onDeviceStatusChanged(camera_device_status_t cameraId, } if (newStatus == CAMERA_DEVICE_STATUS_NOT_PRESENT) { + logDeviceRemoved(id, String8::format("Device status changed from %d to %d", oldStatus, + newStatus)); sp<BasicClient> clientToDisconnect; { // Don't do this in updateStatus to avoid deadlock over mServiceLock @@ -274,6 +276,10 @@ void CameraService::onDeviceStatusChanged(camera_device_status_t cameraId, } } else { + if (oldStatus == ICameraServiceListener::Status::STATUS_NOT_PRESENT) { + logDeviceAdded(id, String8::format("Device status changed from %d to %d", oldStatus, + newStatus)); + } updateStatus(static_cast<ICameraServiceListener::Status>(newStatus), id); } @@ -765,8 +771,8 @@ status_t CameraService::validateConnectLocked(const String8& cameraId, /*inout*/ } else { // We only trust our own process to forward client UIDs if (callingPid != getpid()) { - ALOGE("CameraService::connect X (PID %d) rejected (don't trust clientUid)", - callingPid); + ALOGE("CameraService::connect X (PID %d) rejected (don't trust clientUid %d)", + callingPid, clientUid); return PERMISSION_DENIED; } } @@ -796,10 +802,12 @@ status_t CameraService::validateConnectLocked(const String8& cameraId, /*inout*/ return -EACCES; } - // Only allow clients who are being used by the current foreground device user. - if (mLastUserId != clientUserId && mLastUserId != DEFAULT_LAST_USER_ID) { - ALOGE("CameraService::connect X (PID %d) rejected (cannot connect from non-foreground " - "device user)", callingPid); + // Only allow clients who are being used by the current foreground device user, unless calling + // from our own process. + if (callingPid != getpid() && + (mLastUserId != clientUserId && mLastUserId != DEFAULT_LAST_USER_ID)) { + ALOGE("CameraService::connect X (PID %d) rejected (cannot connect from previous " + "device user %d, current device user %d)", callingPid, clientUserId, mLastUserId); return PERMISSION_DENIED; } @@ -858,7 +866,7 @@ status_t CameraService::handleEvictionsLocked(const String8& cameraId, int clien std::shared_ptr<resource_policy::ClientDescriptor<String8, sp<BasicClient>>>* partial) { status_t ret = NO_ERROR; - std::vector<sp<BasicClient>> evictedClients; + std::vector<DescriptorPtr> evictedClients; DescriptorPtr clientDescriptor; { if (effectiveApiLevel == API_1) { @@ -934,7 +942,7 @@ status_t CameraService::handleEvictionsLocked(const String8& cameraId, int clien mActiveClientManager.getIncompatibleClients(clientDescriptor); String8 msg = String8::format("%s : DENIED connect device %s client for package %s " - "(PID %d, priority %d)", curTime.string(), + "(PID %d, priority %d) due to eviction policy", curTime.string(), cameraId.string(), packageName.string(), clientPid, getCameraPriorityFromProcState(priorities[priorities.size() - 1])); @@ -946,6 +954,7 @@ status_t CameraService::handleEvictionsLocked(const String8& cameraId, int clien } // Log the client's attempt + Mutex::Autolock l(mLogLock); mEventLog.add(msg); return -EBUSY; @@ -965,14 +974,12 @@ status_t CameraService::handleEvictionsLocked(const String8& cameraId, int clien ALOGE("CameraService::connect evicting conflicting client for camera ID %s", i->getKey().string()); - evictedClients.push_back(clientSp); - - String8 curTime = getFormattedCurrentTime(); + evictedClients.push_back(i); // Log the clients evicted - mEventLog.add(String8::format("%s : EVICT device %s client for package %s (PID %" - PRId32 ", priority %" PRId32 ")\n - Evicted by device %s client for " - "package %s (PID %d, priority %" PRId32 ")", curTime.string(), + logEvent(String8::format("EVICT device %s client held by package %s (PID" + " %" PRId32 ", priority %" PRId32 ")\n - Evicted by device %s client for" + " package %s (PID %d, priority %" PRId32 ")", i->getKey().string(), String8{clientSp->getPackageName()}.string(), i->getOwnerId(), i->getPriority(), cameraId.string(), packageName.string(), clientPid, @@ -994,12 +1001,31 @@ status_t CameraService::handleEvictionsLocked(const String8& cameraId, int clien // Destroy evicted clients for (auto& i : evictedClients) { // Disconnect is blocking, and should only have returned when HAL has cleaned up - i->disconnect(); // Clients will remove themselves from the active client list here + i->getValue()->disconnect(); // Clients will remove themselves from the active client list } - evictedClients.clear(); IPCThreadState::self()->restoreCallingIdentity(token); + for (const auto& i : evictedClients) { + ALOGV("%s: Waiting for disconnect to complete for client for device %s (PID %" PRId32 ")", + __FUNCTION__, i->getKey().string(), i->getOwnerId()); + ret = mActiveClientManager.waitUntilRemoved(i, DEFAULT_DISCONNECT_TIMEOUT_NS); + if (ret == TIMED_OUT) { + ALOGE("%s: Timed out waiting for client for device %s to disconnect, " + "current clients:\n%s", __FUNCTION__, i->getKey().string(), + mActiveClientManager.toString().string()); + return -EBUSY; + } + if (ret != NO_ERROR) { + ALOGE("%s: Received error waiting for client for device %s to disconnect: %s (%d), " + "current clients:\n%s", __FUNCTION__, i->getKey().string(), strerror(-ret), + ret, mActiveClientManager.toString().string()); + return ret; + } + } + + evictedClients.clear(); + // Once clients have been disconnected, relock mServiceLock.lock(); @@ -1027,6 +1053,8 @@ status_t CameraService::connect( clientPackageName, clientUid, API_1, false, false, /*out*/client); if(ret != NO_ERROR) { + logRejected(id, getCallingPid(), String8(clientPackageName), + String8::format("%s (%d)", strerror(-ret), ret)); return ret; } @@ -1042,6 +1070,7 @@ status_t CameraService::connectLegacy( /*out*/ sp<ICamera>& device) { + String8 id = String8::format("%d", cameraId); int apiVersion = mModule->getModuleApiVersion(); if (halVersion != CAMERA_HAL_API_VERSION_UNSPECIFIED && apiVersion < CAMERA_MODULE_API_VERSION_2_3) { @@ -1053,16 +1082,19 @@ status_t CameraService::connectLegacy( */ ALOGE("%s: camera HAL module version %x doesn't support connecting to legacy HAL devices!", __FUNCTION__, apiVersion); + logRejected(id, getCallingPid(), String8(clientPackageName), + String8("HAL module version doesn't support legacy HAL connections")); return INVALID_OPERATION; } status_t ret = NO_ERROR; - String8 id = String8::format("%d", cameraId); sp<Client> client = nullptr; ret = connectHelper<ICameraClient,Client>(cameraClient, id, halVersion, clientPackageName, clientUid, API_1, true, false, /*out*/client); if(ret != NO_ERROR) { + logRejected(id, getCallingPid(), String8(clientPackageName), + String8::format("%s (%d)", strerror(-ret), ret)); return ret; } @@ -1086,6 +1118,8 @@ status_t CameraService::connectDevice( /*out*/client); if(ret != NO_ERROR) { + logRejected(id, getCallingPid(), String8(clientPackageName), + String8::format("%s (%d)", strerror(-ret), ret)); return ret; } @@ -1426,6 +1460,8 @@ void CameraService::doUserSwitch(int newUserId) { newUserId = DEFAULT_LAST_USER_ID; } + logUserSwitch(mLastUserId, newUserId); + mLastUserId = newUserId; // Current user has switched, evict all current clients. @@ -1444,12 +1480,12 @@ void CameraService::doUserSwitch(int newUserId) { ALOGE("Evicting conflicting client for camera ID %s due to user change", i->getKey().string()); + // Log the clients evicted - mEventLog.add(String8::format("%s : EVICT device %s client for package %s (PID %" + logEvent(String8::format("EVICT device %s client held by package %s (PID %" PRId32 ", priority %" PRId32 ")\n - Evicted due to user switch.", - curTime.string(), i->getKey().string(), - String8{clientSp->getPackageName()}.string(), i->getOwnerId(), - i->getPriority())); + i->getKey().string(), String8{clientSp->getPackageName()}.string(), + i->getOwnerId(), i->getPriority())); } @@ -1470,22 +1506,52 @@ void CameraService::doUserSwitch(int newUserId) { mServiceLock.lock(); } -void CameraService::logDisconnected(const String8& cameraId, int clientPid, - const String8& clientPackage) { - +void CameraService::logEvent(const char* event) { String8 curTime = getFormattedCurrentTime(); - // Log the clients evicted - mEventLog.add(String8::format("%s : DISCONNECT device %s client for package %s (PID %d)", - curTime.string(), cameraId.string(), clientPackage.string(), clientPid)); + Mutex::Autolock l(mLogLock); + mEventLog.add(String8::format("%s : %s", curTime.string(), event)); } -void CameraService::logConnected(const String8& cameraId, int clientPid, - const String8& clientPackage) { +void CameraService::logDisconnected(const char* cameraId, int clientPid, + const char* clientPackage) { + // Log the clients evicted + logEvent(String8::format("DISCONNECT device %s client for package %s (PID %d)", cameraId, + clientPackage, clientPid)); +} - String8 curTime = getFormattedCurrentTime(); +void CameraService::logConnected(const char* cameraId, int clientPid, + const char* clientPackage) { // Log the clients evicted - mEventLog.add(String8::format("%s : CONNECT device %s client for package %s (PID %d)", - curTime.string(), cameraId.string(), clientPackage.string(), clientPid)); + logEvent(String8::format("CONNECT device %s client for package %s (PID %d)", cameraId, + clientPackage, clientPid)); +} + +void CameraService::logRejected(const char* cameraId, int clientPid, + const char* clientPackage, const char* reason) { + // Log the client rejected + logEvent(String8::format("REJECT device %s client for package %s (PID %d), reason: (%s)", + cameraId, clientPackage, clientPid, reason)); +} + +void CameraService::logUserSwitch(int oldUserId, int newUserId) { + // Log the new and old users + logEvent(String8::format("USER_SWITCH from old user: %d , to new user: %d", oldUserId, + newUserId)); +} + +void CameraService::logDeviceRemoved(const char* cameraId, const char* reason) { + // Log the device removal + logEvent(String8::format("REMOVE device %s, reason: (%s)", cameraId, reason)); +} + +void CameraService::logDeviceAdded(const char* cameraId, const char* reason) { + // Log the device removal + logEvent(String8::format("ADD device %s, reason: (%s)", cameraId, reason)); +} + +void CameraService::logClientDied(int clientPid, const char* reason) { + // Log the device removal + logEvent(String8::format("DIED client(s) with PID %d, reason: (%s)", clientPid, reason)); } status_t CameraService::onTransact(uint32_t code, const Parcel& data, Parcel* reply, @@ -1911,7 +1977,7 @@ static bool tryLock(Mutex& mutex) } status_t CameraService::dump(int fd, const Vector<String16>& args) { - String8 result; + String8 result("Dump of the Camera Service:\n"); if (checkCallingPermission(String16("android.permission.DUMP")) == false) { result.appendFormat("Permission Denial: " "can't dump CameraService from pid=%d, uid=%d\n", @@ -1957,12 +2023,15 @@ status_t CameraService::dump(int fd, const Vector<String16>& args) { result = String8("Prior client events (most recent at top):\n"); - for (const auto& msg : mEventLog) { - result.appendFormat("%s\n", msg.string()); - } + { + Mutex::Autolock l(mLogLock); + for (const auto& msg : mEventLog) { + result.appendFormat("%s\n", msg.string()); + } - if (mEventLog.size() == DEFAULT_EVICTION_LOG_LENGTH) { - result.append("...\n"); + if (mEventLog.size() == DEFAULT_EVENT_LOG_LENGTH) { + result.append("...\n"); + } } write(fd, result.string(), result.size()); @@ -2094,10 +2163,12 @@ void CameraService::handleTorchClientBinderDied(const wp<IBinder> &who) { /*virtual*/void CameraService::binderDied(const wp<IBinder> &who) { /** - * While tempting to promote the wp<IBinder> into a sp, - * it's actually not supported by the binder driver + * While tempting to promote the wp<IBinder> into a sp, it's actually not supported by the + * binder driver */ + logClientDied(getCallingPid(), String8("Binder died unexpectedly")); + // check torch client handleTorchClientBinderDied(who); diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h index ca1c504..91c7d59 100644 --- a/services/camera/libcameraservice/CameraService.h +++ b/services/camera/libcameraservice/CameraService.h @@ -90,8 +90,11 @@ public: // 3 second busy timeout when other clients are connecting static const nsecs_t DEFAULT_CONNECT_TIMEOUT_NS = 3000000000; + // 1 second busy timeout when other clients are disconnecting + static const nsecs_t DEFAULT_DISCONNECT_TIMEOUT_NS = 1000000000; + // Default number of messages to store in eviction log - static const size_t DEFAULT_EVICTION_LOG_LENGTH = 50; + static const size_t DEFAULT_EVENT_LOG_LENGTH = 100; enum { // Default last user id @@ -492,6 +495,7 @@ private: // Circular buffer for storing event logging for dumps RingBuffer<String8> mEventLog; + Mutex mLogLock; // UID of last user. int mLastUserId; @@ -546,14 +550,45 @@ private: void doUserSwitch(int newUserId); /** - * Add a event log message that a client has been disconnected. + * Add an event log message. + */ + void logEvent(const char* event); + + /** + * Add an event log message that a client has been disconnected. + */ + void logDisconnected(const char* cameraId, int clientPid, const char* clientPackage); + + /** + * Add an event log message that a client has been connected. + */ + void logConnected(const char* cameraId, int clientPid, const char* clientPackage); + + /** + * Add an event log message that a client's connect attempt has been rejected. */ - void logDisconnected(const String8& cameraId, int clientPid, const String8& clientPackage); + void logRejected(const char* cameraId, int clientPid, const char* clientPackage, + const char* reason); /** - * Add a event log message that a client has been connected. + * Add an event log message that the current device user has been switched. */ - void logConnected(const String8& cameraId, int clientPid, const String8& clientPackage); + void logUserSwitch(int oldUserId, int newUserId); + + /** + * Add an event log message that a device has been removed by the HAL + */ + void logDeviceRemoved(const char* cameraId, const char* reason); + + /** + * Add an event log message that a device has been added by the HAL + */ + void logDeviceAdded(const char* cameraId, const char* reason); + + /** + * Add an event log message that a client has unexpectedly died. + */ + void logClientDied(int clientPid, const char* reason); int mNumberOfCameras; @@ -714,9 +749,10 @@ status_t CameraService::connectHelper(const sp<CALLBACK>& cameraCb, const String String8 clientName8(clientPackageName); int clientPid = getCallingPid(); - ALOGI("CameraService::connect call E (PID %d \"%s\", camera ID %s) for HAL version %d and " + ALOGI("CameraService::connect call (PID %d \"%s\", camera ID %s) for HAL version %s and " "Camera API version %d", clientPid, clientName8.string(), cameraId.string(), - halVersion, static_cast<int>(effectiveApiLevel)); + (halVersion == -1) ? "default" : std::to_string(halVersion).c_str(), + static_cast<int>(effectiveApiLevel)); sp<CLIENT> client = nullptr; { @@ -734,7 +770,15 @@ status_t CameraService::connectHelper(const sp<CALLBACK>& cameraCb, const String if((ret = validateConnectLocked(cameraId, /*inout*/clientUid)) != NO_ERROR) { return ret; } - mLastUserId = multiuser_get_user_id(clientUid); + int userId = multiuser_get_user_id(clientUid); + + if (userId != mLastUserId && clientPid != getpid() ) { + // If no previous user ID had been set, set to the user of the caller. + logUserSwitch(mLastUserId, userId); + LOG_ALWAYS_FATAL_IF(mLastUserId != DEFAULT_LAST_USER_ID, + "Invalid state: Should never update user ID here unless was default"); + mLastUserId = userId; + } // Check the shim parameters after acquiring lock, if they have already been updated and // we were doing a shim update, return immediately diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp index 6f44aee..f53f425 100644 --- a/services/camera/libcameraservice/api1/Camera2Client.cpp +++ b/services/camera/libcameraservice/api1/Camera2Client.cpp @@ -121,7 +121,8 @@ status_t Camera2Client::initialize(CameraModule *module) } case CAMERA_DEVICE_API_VERSION_3_0: case CAMERA_DEVICE_API_VERSION_3_1: - case CAMERA_DEVICE_API_VERSION_3_2: { + case CAMERA_DEVICE_API_VERSION_3_2: + case CAMERA_DEVICE_API_VERSION_3_3: { sp<ZslProcessor3> zslProc = new ZslProcessor3(this, mCaptureSequencer); mZslProcessor = zslProc; diff --git a/services/camera/libcameraservice/utils/ClientManager.h b/services/camera/libcameraservice/utils/ClientManager.h index ad5486d..aa40a2d 100644 --- a/services/camera/libcameraservice/utils/ClientManager.h +++ b/services/camera/libcameraservice/utils/ClientManager.h @@ -17,7 +17,9 @@ #ifndef ANDROID_SERVICE_UTILS_EVICTION_POLICY_MANAGER_H #define ANDROID_SERVICE_UTILS_EVICTION_POLICY_MANAGER_H +#include <utils/Condition.h> #include <utils/Mutex.h> +#include <utils/Timers.h> #include <algorithm> #include <utility> @@ -263,6 +265,16 @@ public: */ std::shared_ptr<ClientDescriptor<KEY, VALUE>> get(const KEY& key) const; + /** + * Block until the given client is no longer in the active clients list, or the timeout + * occurred. + * + * Returns NO_ERROR if this succeeded, -ETIMEDOUT on a timeout, or a negative error code on + * failure. + */ + status_t waitUntilRemoved(const std::shared_ptr<ClientDescriptor<KEY, VALUE>> client, + nsecs_t timeout) const; + protected: ~ClientManager(); @@ -284,6 +296,7 @@ private: int64_t getCurrentCostLocked() const; mutable Mutex mLock; + mutable Condition mRemovedCondition; int32_t mMaxCost; // LRU ordered, most recent at end std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> mClients; @@ -430,6 +443,7 @@ std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> ClientManager<KEY, VA }), mClients.end()); mClients.push_back(client); + mRemovedCondition.broadcast(); return evicted; } @@ -487,6 +501,7 @@ template<class KEY, class VALUE> void ClientManager<KEY, VALUE>::removeAll() { Mutex::Autolock lock(mLock); mClients.clear(); + mRemovedCondition.broadcast(); } template<class KEY, class VALUE> @@ -505,6 +520,39 @@ std::shared_ptr<ClientDescriptor<KEY, VALUE>> ClientManager<KEY, VALUE>::remove( return false; }), mClients.end()); + mRemovedCondition.broadcast(); + return ret; +} + +template<class KEY, class VALUE> +status_t ClientManager<KEY, VALUE>::waitUntilRemoved( + const std::shared_ptr<ClientDescriptor<KEY, VALUE>> client, + nsecs_t timeout) const { + status_t ret = NO_ERROR; + Mutex::Autolock lock(mLock); + + bool isRemoved = false; + + // Figure out what time in the future we should hit the timeout + nsecs_t failTime = systemTime(SYSTEM_TIME_MONOTONIC) + timeout; + + while (!isRemoved) { + isRemoved = true; + for (const auto& i : mClients) { + if (i == client) { + isRemoved = false; + } + } + + if (!isRemoved) { + ret = mRemovedCondition.waitRelative(mLock, timeout); + if (ret != NO_ERROR) { + break; + } + timeout = failTime - systemTime(SYSTEM_TIME_MONOTONIC); + } + } + return ret; } @@ -520,6 +568,7 @@ void ClientManager<KEY, VALUE>::remove( } return false; }), mClients.end()); + mRemovedCondition.broadcast(); } template<class KEY, class VALUE> |