diff options
Diffstat (limited to 'media')
78 files changed, 4041 insertions, 476 deletions
diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c index 4ee05f2..a39d837 100644 --- a/media/libeffects/downmix/EffectDownmix.c +++ b/media/libeffects/downmix/EffectDownmix.c @@ -629,7 +629,9 @@ int Downmix_Configure(downmix_module_t *pDwmModule, effect_config_t *pConfig, bo return -EINVAL; } - memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t)); + if (&pDwmModule->config != pConfig) { + memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t)); + } if (init) { pDownmixer->type = DOWNMIX_TYPE_FOLD; @@ -697,7 +699,7 @@ int Downmix_Reset(downmix_object_t *pDownmixer, bool init) { * *---------------------------------------------------------------------------- */ -int Downmix_setParameter(downmix_object_t *pDownmixer, int32_t param, size_t size, void *pValue) { +int Downmix_setParameter(downmix_object_t *pDownmixer, int32_t param, uint32_t size, void *pValue) { int16_t value16; ALOGV("Downmix_setParameter, context %p, param %d, value16 %d, value32 %d", @@ -707,7 +709,7 @@ int Downmix_setParameter(downmix_object_t *pDownmixer, int32_t param, size_t siz case DOWNMIX_PARAM_TYPE: if (size != sizeof(downmix_type_t)) { - ALOGE("Downmix_setParameter(DOWNMIX_PARAM_TYPE) invalid size %zu, should be %zu", + ALOGE("Downmix_setParameter(DOWNMIX_PARAM_TYPE) invalid size %u, should be %zu", size, sizeof(downmix_type_t)); return -EINVAL; } @@ -753,7 +755,7 @@ int Downmix_setParameter(downmix_object_t *pDownmixer, int32_t param, size_t siz * *---------------------------------------------------------------------------- */ -int Downmix_getParameter(downmix_object_t *pDownmixer, int32_t param, size_t *pSize, void *pValue) { +int Downmix_getParameter(downmix_object_t *pDownmixer, int32_t param, uint32_t *pSize, void *pValue) { int16_t *pValue16; switch (param) { diff --git a/media/libeffects/downmix/EffectDownmix.h b/media/libeffects/downmix/EffectDownmix.h index cb6b957..fcb3c9e 100644 --- a/media/libeffects/downmix/EffectDownmix.h +++ b/media/libeffects/downmix/EffectDownmix.h @@ -93,8 +93,8 @@ static int Downmix_GetDescriptor(effect_handle_t self, int Downmix_Init(downmix_module_t *pDwmModule); int Downmix_Configure(downmix_module_t *pDwmModule, effect_config_t *pConfig, bool init); int Downmix_Reset(downmix_object_t *pDownmixer, bool init); -int Downmix_setParameter(downmix_object_t *pDownmixer, int32_t param, size_t size, void *pValue); -int Downmix_getParameter(downmix_object_t *pDownmixer, int32_t param, size_t *pSize, void *pValue); +int Downmix_setParameter(downmix_object_t *pDownmixer, int32_t param, uint32_t size, void *pValue); +int Downmix_getParameter(downmix_object_t *pDownmixer, int32_t param, uint32_t *pSize, void *pValue); void Downmix_foldFromQuad(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate); void Downmix_foldFromSurround(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate); diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.c b/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.c index 32c4ce0..35e5bc8 100644 --- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.c +++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.c @@ -178,7 +178,7 @@ LVDBE_ReturnStatus_en LVDBE_Init(LVDBE_Handle_t *phInstance, { return(LVDBE_NULLADDRESS); } - if (((LVM_UINT32)pMemoryTable->Region[i].pBaseAddress % pMemoryTable->Region[i].Alignment)!=0){ + if (((uintptr_t)pMemoryTable->Region[i].pBaseAddress % pMemoryTable->Region[i].Alignment)!=0){ return(LVDBE_ALIGNMENTERROR); } } diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Process.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Process.c index 794271b..f5a01f3 100644 --- a/media/libeffects/lvm/lib/Bundle/src/LVM_Process.c +++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Process.c @@ -99,7 +99,7 @@ LVM_ReturnStatus_en LVM_Process(LVM_Handle_t hInstance, /* * Check the buffer alignment */ - if((((LVM_UINT32)pInData % 4) != 0) || (((LVM_UINT32)pOutData % 4) != 0)) + if((((uintptr_t)pInData % 4) != 0) || (((uintptr_t)pOutData % 4) != 0)) { return(LVM_ALIGNMENTERROR); } diff --git a/media/libeffects/lvm/lib/Common/lib/InstAlloc.h b/media/libeffects/lvm/lib/Common/lib/InstAlloc.h index c6954f2..7f725f4 100644 --- a/media/libeffects/lvm/lib/Common/lib/InstAlloc.h +++ b/media/libeffects/lvm/lib/Common/lib/InstAlloc.h @@ -29,7 +29,7 @@ extern "C" { typedef struct { LVM_UINT32 TotalSize; /* Accumulative total memory size */ - LVM_UINT32 pNextMember; /* Pointer to the next instance member to be allocated */ + uintptr_t pNextMember; /* Pointer to the next instance member to be allocated */ } INST_ALLOC; diff --git a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h index 81655dd..0c6fb25 100644 --- a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h +++ b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h @@ -29,6 +29,7 @@ extern "C" { #endif /* __cplusplus */ +#include <stdint.h> /****************************************************************************************/ /* */ @@ -85,14 +86,14 @@ extern "C" { typedef char LVM_CHAR; /* ASCII character */ -typedef char LVM_INT8; /* Signed 8-bit word */ -typedef unsigned char LVM_UINT8; /* Unsigned 8-bit word */ +typedef int8_t LVM_INT8; /* Signed 8-bit word */ +typedef uint8_t LVM_UINT8; /* Unsigned 8-bit word */ -typedef short LVM_INT16; /* Signed 16-bit word */ -typedef unsigned short LVM_UINT16; /* Unsigned 16-bit word */ +typedef int16_t LVM_INT16; /* Signed 16-bit word */ +typedef uint16_t LVM_UINT16; /* Unsigned 16-bit word */ -typedef long LVM_INT32; /* Signed 32-bit word */ -typedef unsigned long LVM_UINT32; /* Unsigned 32-bit word */ +typedef int32_t LVM_INT32; /* Signed 32-bit word */ +typedef uint32_t LVM_UINT32; /* Unsigned 32-bit word */ /****************************************************************************************/ diff --git a/media/libeffects/lvm/lib/Common/src/InstAlloc.c b/media/libeffects/lvm/lib/Common/src/InstAlloc.c index 481df84..a89a5c3 100644 --- a/media/libeffects/lvm/lib/Common/src/InstAlloc.c +++ b/media/libeffects/lvm/lib/Common/src/InstAlloc.c @@ -30,7 +30,7 @@ void InstAlloc_Init( INST_ALLOC *pms, void *StartAddr ) { pms->TotalSize = 3; - pms->pNextMember = (LVM_UINT32)(((LVM_UINT32)StartAddr + 3) & 0xFFFFFFFC);/* This code will fail if the platform address space is more than 32-bits*/ + pms->pNextMember = (((uintptr_t)StartAddr + 3) & (uintptr_t)~3); } @@ -51,7 +51,7 @@ void* InstAlloc_AddMember( INST_ALLOC *pms, void *NewMemberAddress; /* Variable to temporarily store the return value */ NewMemberAddress = (void*)pms->pNextMember; - Size = ((Size + 3) & 0xFFFFFFFC); /* Ceil the size to a multiple of four */ + Size = ((Size + 3) & (LVM_UINT32)~3); /* Ceil the size to a multiple of four */ pms->TotalSize += Size; pms->pNextMember += Size; @@ -84,30 +84,30 @@ LVM_UINT32 InstAlloc_GetTotal( INST_ALLOC *pms) void InstAlloc_InitAll( INST_ALLOC *pms, LVM_MemoryTable_st *pMemoryTable) { - LVM_UINT32 StartAddr; + uintptr_t StartAddr; - StartAddr = (LVM_UINT32)pMemoryTable->Region[LVM_PERSISTENT_SLOW_DATA].pBaseAddress; + StartAddr = (uintptr_t)pMemoryTable->Region[LVM_PERSISTENT_SLOW_DATA].pBaseAddress; pms[0].TotalSize = 3; - pms[0].pNextMember = (LVM_UINT32)(((LVM_UINT32)StartAddr + 3) & 0xFFFFFFFC); + pms[0].pNextMember = ((StartAddr + 3) & (uintptr_t)~3); - StartAddr = (LVM_UINT32)pMemoryTable->Region[LVM_PERSISTENT_FAST_DATA].pBaseAddress; + StartAddr = (uintptr_t)pMemoryTable->Region[LVM_PERSISTENT_FAST_DATA].pBaseAddress; pms[1].TotalSize = 3; - pms[1].pNextMember = (LVM_UINT32)(((LVM_UINT32)StartAddr + 3) & 0xFFFFFFFC); + pms[1].pNextMember = ((StartAddr + 3) & (uintptr_t)~3); - StartAddr = (LVM_UINT32)pMemoryTable->Region[LVM_PERSISTENT_FAST_COEF].pBaseAddress; + StartAddr = (uintptr_t)pMemoryTable->Region[LVM_PERSISTENT_FAST_COEF].pBaseAddress; pms[2].TotalSize = 3; - pms[2].pNextMember = (LVM_UINT32)(((LVM_UINT32)StartAddr + 3) & 0xFFFFFFFC); + pms[2].pNextMember = ((StartAddr + 3) & (uintptr_t)~3); - StartAddr = (LVM_UINT32)pMemoryTable->Region[LVM_TEMPORARY_FAST].pBaseAddress; + StartAddr = (uintptr_t)pMemoryTable->Region[LVM_TEMPORARY_FAST].pBaseAddress; pms[3].TotalSize = 3; - pms[3].pNextMember = (LVM_UINT32)(((LVM_UINT32)StartAddr + 3) & 0xFFFFFFFC); + pms[3].pNextMember = ((StartAddr + 3) & (uintptr_t)~3); } diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.c index ac3c740..58f58ed 100644 --- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.c +++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.c @@ -77,7 +77,7 @@ LVEQNB_ReturnStatus_en LVEQNB_Process(LVEQNB_Handle_t hInstance, } /* Check if the input and output data buffers are 32-bit aligned */ - if ((((LVM_INT32)pInData % 4) != 0) || (((LVM_INT32)pOutData % 4) != 0)) + if ((((uintptr_t)pInData % 4) != 0) || (((uintptr_t)pOutData % 4) != 0)) { return LVEQNB_ALIGNMENTERROR; } diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp index 58d7767..db5c78f 100644 --- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp +++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp @@ -2813,9 +2813,9 @@ int Effect_command(effect_handle_t self, if(pContext->EffectType == LVM_BASS_BOOST){ if (pCmdData == NULL || - cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || + cmdSize < (sizeof(effect_param_t) + sizeof(int32_t)) || pReplyData == NULL || - *replySize < (int) (sizeof(effect_param_t) + sizeof(int32_t))){ + *replySize < (sizeof(effect_param_t) + sizeof(int32_t))){ ALOGV("\tLVM_ERROR : BassBoost_command cmdCode Case: " "EFFECT_CMD_GET_PARAM: ERROR"); return -EINVAL; @@ -2844,9 +2844,9 @@ int Effect_command(effect_handle_t self, if(pContext->EffectType == LVM_VIRTUALIZER){ if (pCmdData == NULL || - cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || + cmdSize < (sizeof(effect_param_t) + sizeof(int32_t)) || pReplyData == NULL || - *replySize < (int) (sizeof(effect_param_t) + sizeof(int32_t))){ + *replySize < (sizeof(effect_param_t) + sizeof(int32_t))){ ALOGV("\tLVM_ERROR : Virtualizer_command cmdCode Case: " "EFFECT_CMD_GET_PARAM: ERROR"); return -EINVAL; @@ -2876,7 +2876,7 @@ int Effect_command(effect_handle_t self, //ALOGV("\tEqualizer_command cmdCode Case: " // "EFFECT_CMD_GET_PARAM start"); if (pCmdData == NULL || - cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || + cmdSize < (sizeof(effect_param_t) + sizeof(int32_t)) || pReplyData == NULL || *replySize < (int) (sizeof(effect_param_t) + sizeof(int32_t))) { ALOGV("\tLVM_ERROR : Equalizer_command cmdCode Case: " @@ -2908,7 +2908,7 @@ int Effect_command(effect_handle_t self, if(pContext->EffectType == LVM_VOLUME){ //ALOGV("\tVolume_command cmdCode Case: EFFECT_CMD_GET_PARAM start"); if (pCmdData == NULL || - cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || + cmdSize < (sizeof(effect_param_t) + sizeof(int32_t)) || pReplyData == NULL || *replySize < (int) (sizeof(effect_param_t) + sizeof(int32_t))){ ALOGV("\tLVM_ERROR : Volume_command cmdCode Case: " @@ -2947,7 +2947,7 @@ int Effect_command(effect_handle_t self, // *(int16_t *)((char *)pCmdData + sizeof(effect_param_t) + sizeof(int32_t))); if (pCmdData == NULL|| - cmdSize != (int)(sizeof(effect_param_t) + sizeof(int32_t) +sizeof(int16_t))|| + cmdSize != (sizeof(effect_param_t) + sizeof(int32_t) +sizeof(int16_t))|| pReplyData == NULL|| *replySize != sizeof(int32_t)){ ALOGV("\tLVM_ERROR : BassBoost_command cmdCode Case: " @@ -2980,7 +2980,7 @@ int Effect_command(effect_handle_t self, // *(int16_t *)((char *)pCmdData + sizeof(effect_param_t) + sizeof(int32_t))); if (pCmdData == NULL|| - cmdSize != (int)(sizeof(effect_param_t) + sizeof(int32_t) +sizeof(int16_t))|| + cmdSize != (sizeof(effect_param_t) + sizeof(int32_t) +sizeof(int16_t))|| pReplyData == NULL|| *replySize != sizeof(int32_t)){ ALOGV("\tLVM_ERROR : Virtualizer_command cmdCode Case: " @@ -3014,7 +3014,7 @@ int Effect_command(effect_handle_t self, // *replySize, // *(int16_t *)((char *)pCmdData + sizeof(effect_param_t) + sizeof(int32_t))); - if (pCmdData == NULL || cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || + if (pCmdData == NULL || cmdSize < (sizeof(effect_param_t) + sizeof(int32_t)) || pReplyData == NULL || *replySize != sizeof(int32_t)) { ALOGV("\tLVM_ERROR : Equalizer_command cmdCode Case: " "EFFECT_CMD_SET_PARAM: ERROR"); @@ -3034,7 +3034,7 @@ int Effect_command(effect_handle_t self, // *(int16_t *)((char *)pCmdData + sizeof(effect_param_t) +sizeof(int32_t))); if ( pCmdData == NULL|| - cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t))|| + cmdSize < (sizeof(effect_param_t) + sizeof(int32_t))|| pReplyData == NULL|| *replySize != sizeof(int32_t)){ ALOGV("\tLVM_ERROR : Volume_command cmdCode Case: " diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp index 0367302..c6d3759 100644 --- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp +++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp @@ -181,7 +181,7 @@ void Reverb_getConfig (ReverbContext *pContext, effect_config_t *pConfig); int Reverb_setParameter (ReverbContext *pContext, void *pParam, void *pValue); int Reverb_getParameter (ReverbContext *pContext, void *pParam, - size_t *pValueSize, + uint32_t *pValueSize, void *pValue); int Reverb_LoadPreset (ReverbContext *pContext); @@ -1534,7 +1534,7 @@ int Reverb_LoadPreset(ReverbContext *pContext) int Reverb_getParameter(ReverbContext *pContext, void *pParam, - size_t *pValueSize, + uint32_t *pValueSize, void *pValue){ int status = 0; int32_t *pParamTemp = (int32_t *)pParam; @@ -1956,9 +1956,9 @@ int Reverb_command(effect_handle_t self, //ALOGV("\tReverb_command cmdCode Case: " // "EFFECT_CMD_GET_PARAM start"); if (pCmdData == NULL || - cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || + cmdSize < (sizeof(effect_param_t) + sizeof(int32_t)) || pReplyData == NULL || - *replySize < (int) (sizeof(effect_param_t) + sizeof(int32_t))){ + *replySize < (sizeof(effect_param_t) + sizeof(int32_t))){ ALOGV("\tLVM_ERROR : Reverb_command cmdCode Case: " "EFFECT_CMD_GET_PARAM: ERROR"); return -EINVAL; @@ -1973,7 +1973,7 @@ int Reverb_command(effect_handle_t self, p->status = android::Reverb_getParameter(pContext, (void *)p->data, - (size_t *)&p->vsize, + &p->vsize, p->data + voffset); *replySize = sizeof(effect_param_t) + voffset + p->vsize; @@ -1994,8 +1994,8 @@ int Reverb_command(effect_handle_t self, // *replySize, // *(int16_t *)((char *)pCmdData + sizeof(effect_param_t) + sizeof(int32_t))); - if (pCmdData == NULL || (cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t))) - || pReplyData == NULL || *replySize != (int)sizeof(int32_t)) { + if (pCmdData == NULL || (cmdSize < (sizeof(effect_param_t) + sizeof(int32_t))) + || pReplyData == NULL || *replySize != sizeof(int32_t)) { ALOGV("\tLVM_ERROR : Reverb_command cmdCode Case: " "EFFECT_CMD_SET_PARAM: ERROR"); return -EINVAL; diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp index c56ff72..a96a703 100644 --- a/media/libeffects/preprocessing/PreProcessing.cpp +++ b/media/libeffects/preprocessing/PreProcessing.cpp @@ -77,7 +77,7 @@ struct preproc_ops_s { void (* enable)(preproc_effect_t *fx); void (* disable)(preproc_effect_t *fx); int (* set_parameter)(preproc_effect_t *fx, void *param, void *value); - int (* get_parameter)(preproc_effect_t *fx, void *param, size_t *size, void *value); + int (* get_parameter)(preproc_effect_t *fx, void *param, uint32_t *size, void *value); int (* set_device)(preproc_effect_t *fx, uint32_t device); }; @@ -291,7 +291,7 @@ int AgcCreate(preproc_effect_t *effect) int AgcGetParameter(preproc_effect_t *effect, void *pParam, - size_t *pValueSize, + uint32_t *pValueSize, void *pValue) { int status = 0; @@ -452,9 +452,9 @@ int AecCreate(preproc_effect_t *effect) return 0; } -int AecGetParameter(preproc_effect_t *effect, +int AecGetParameter(preproc_effect_t *effect, void *pParam, - size_t *pValueSize, + uint32_t *pValueSize, void *pValue) { int status = 0; @@ -575,9 +575,9 @@ int NsCreate(preproc_effect_t *effect) return 0; } -int NsGetParameter(preproc_effect_t *effect, +int NsGetParameter(preproc_effect_t *effect, void *pParam, - size_t *pValueSize, + uint32_t *pValueSize, void *pValue) { int status = 0; @@ -1453,7 +1453,7 @@ int PreProcessingFx_Command(effect_handle_t self, if (effect->ops->get_parameter) { p->status = effect->ops->get_parameter(effect, p->data, - (size_t *)&p->vsize, + &p->vsize, p->data + voffset); *replySize = sizeof(effect_param_t) + voffset + p->vsize; } diff --git a/media/libeffects/testlibs/EffectEqualizer.cpp b/media/libeffects/testlibs/EffectEqualizer.cpp index 8d00206..3cb13f2 100644 --- a/media/libeffects/testlibs/EffectEqualizer.cpp +++ b/media/libeffects/testlibs/EffectEqualizer.cpp @@ -115,7 +115,7 @@ struct EqualizerContext { int Equalizer_init(EqualizerContext *pContext); int Equalizer_setConfig(EqualizerContext *pContext, effect_config_t *pConfig); -int Equalizer_getParameter(AudioEqualizer * pEqualizer, int32_t *pParam, size_t *pValueSize, void *pValue); +int Equalizer_getParameter(AudioEqualizer * pEqualizer, int32_t *pParam, uint32_t *pValueSize, void *pValue); int Equalizer_setParameter(AudioEqualizer * pEqualizer, int32_t *pParam, void *pValue); @@ -360,7 +360,7 @@ int Equalizer_init(EqualizerContext *pContext) // //---------------------------------------------------------------------------- -int Equalizer_getParameter(AudioEqualizer * pEqualizer, int32_t *pParam, size_t *pValueSize, void *pValue) +int Equalizer_getParameter(AudioEqualizer * pEqualizer, int32_t *pParam, uint32_t *pValueSize, void *pValue) { int status = 0; int32_t param = *pParam++; @@ -662,8 +662,8 @@ extern "C" int Equalizer_command(effect_handle_t self, uint32_t cmdCode, uint32_ Equalizer_setConfig(pContext, &pContext->config); break; case EFFECT_CMD_GET_PARAM: { - if (pCmdData == NULL || cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || - pReplyData == NULL || *replySize < (int) (sizeof(effect_param_t) + sizeof(int32_t))) { + if (pCmdData == NULL || cmdSize < (sizeof(effect_param_t) + sizeof(int32_t)) || + pReplyData == NULL || *replySize < (sizeof(effect_param_t) + sizeof(int32_t))) { return -EINVAL; } effect_param_t *p = (effect_param_t *)pCmdData; @@ -682,7 +682,7 @@ extern "C" int Equalizer_command(effect_handle_t self, uint32_t cmdCode, uint32_ case EFFECT_CMD_SET_PARAM: { ALOGV("Equalizer_command EFFECT_CMD_SET_PARAM cmdSize %d pCmdData %p, *replySize %d, pReplyData %p", cmdSize, pCmdData, *replySize, pReplyData); - if (pCmdData == NULL || cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || + if (pCmdData == NULL || cmdSize < (sizeof(effect_param_t) + sizeof(int32_t)) || pReplyData == NULL || *replySize != sizeof(int32_t)) { return -EINVAL; } diff --git a/media/libeffects/testlibs/EffectReverb.c b/media/libeffects/testlibs/EffectReverb.c index c37f392..f056d19 100644 --- a/media/libeffects/testlibs/EffectReverb.c +++ b/media/libeffects/testlibs/EffectReverb.c @@ -750,7 +750,7 @@ void Reverb_Reset(reverb_object_t *pReverb, bool init) { * *---------------------------------------------------------------------------- */ -int Reverb_getParameter(reverb_object_t *pReverb, int32_t param, size_t *pSize, +int Reverb_getParameter(reverb_object_t *pReverb, int32_t param, uint32_t *pSize, void *pValue) { int32_t *pValue32; int16_t *pValue16; @@ -758,7 +758,7 @@ int Reverb_getParameter(reverb_object_t *pReverb, int32_t param, size_t *pSize, int32_t i; int32_t temp; int32_t temp2; - size_t size; + uint32_t size; if (pReverb->m_Preset) { if (param != REVERB_PARAM_PRESET || *pSize < sizeof(int16_t)) { @@ -1033,7 +1033,7 @@ int Reverb_getParameter(reverb_object_t *pReverb, int32_t param, size_t *pSize, * *---------------------------------------------------------------------------- */ -int Reverb_setParameter(reverb_object_t *pReverb, int32_t param, size_t size, +int Reverb_setParameter(reverb_object_t *pReverb, int32_t param, uint32_t size, void *pValue) { int32_t value32; int16_t value16; @@ -1044,7 +1044,7 @@ int Reverb_setParameter(reverb_object_t *pReverb, int32_t param, size_t size, reverb_preset_t *pPreset; int maxSamples; int32_t averageDelay; - size_t paramSize; + uint32_t paramSize; ALOGV("Reverb_setParameter, context %p, param %d, value16 %d, value32 %d", pReverb, param, *(int16_t *)pValue, *(int32_t *)pValue); diff --git a/media/libeffects/testlibs/EffectReverb.h b/media/libeffects/testlibs/EffectReverb.h index e5248fe..756c5ea 100644 --- a/media/libeffects/testlibs/EffectReverb.h +++ b/media/libeffects/testlibs/EffectReverb.h @@ -330,8 +330,8 @@ int Reverb_setConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig, bool void Reverb_getConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig); void Reverb_Reset(reverb_object_t *pReverb, bool init); -int Reverb_setParameter (reverb_object_t *pReverb, int32_t param, size_t size, void *pValue); -int Reverb_getParameter(reverb_object_t *pReverb, int32_t param, size_t *pSize, void *pValue); +int Reverb_setParameter (reverb_object_t *pReverb, int32_t param, uint32_t size, void *pValue); +int Reverb_getParameter(reverb_object_t *pReverb, int32_t param, uint32_t *pSize, void *pValue); /*---------------------------------------------------------------------------- * ReverbUpdateXfade diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk index dd2d306..c92c543 100644 --- a/media/libeffects/visualizer/Android.mk +++ b/media/libeffects/visualizer/Android.mk @@ -17,7 +17,6 @@ LOCAL_MODULE_RELATIVE_PATH := soundfx LOCAL_MODULE:= libvisualizer LOCAL_C_INCLUDES := \ - $(call include-path-for, graphics corecg) \ $(call include-path-for, audio-effects) diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp index 2d66eef..5bdaa03 100644 --- a/media/libeffects/visualizer/EffectVisualizer.cpp +++ b/media/libeffects/visualizer/EffectVisualizer.cpp @@ -544,56 +544,57 @@ int Visualizer_command(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize, break; - case VISUALIZER_CMD_CAPTURE: - if (pReplyData == NULL || *replySize != pContext->mCaptureSize) { - ALOGV("VISUALIZER_CMD_CAPTURE() error *replySize %d pContext->mCaptureSize %d", - *replySize, pContext->mCaptureSize); + case VISUALIZER_CMD_CAPTURE: { + int32_t captureSize = pContext->mCaptureSize; + if (pReplyData == NULL || *replySize != captureSize) { + ALOGV("VISUALIZER_CMD_CAPTURE() error *replySize %d captureSize %d", + *replySize, captureSize); return -EINVAL; } if (pContext->mState == VISUALIZER_STATE_ACTIVE) { - int32_t latencyMs = pContext->mLatency; const uint32_t deltaMs = Visualizer_getDeltaTimeMsFromUpdatedTime(pContext); - latencyMs -= deltaMs; - if (latencyMs < 0) { - latencyMs = 0; - } - const uint32_t deltaSmpl = pContext->mConfig.inputCfg.samplingRate * latencyMs / 1000; - - int32_t capturePoint = pContext->mCaptureIdx - pContext->mCaptureSize - deltaSmpl; - int32_t captureSize = pContext->mCaptureSize; - if (capturePoint < 0) { - int32_t size = -capturePoint; - if (size > captureSize) { - size = captureSize; - } - memcpy(pReplyData, - pContext->mCaptureBuf + CAPTURE_BUF_SIZE + capturePoint, - size); - pReplyData = (char *)pReplyData + size; - captureSize -= size; - capturePoint = 0; - } - memcpy(pReplyData, - pContext->mCaptureBuf + capturePoint, - captureSize); - // if audio framework has stopped playing audio although the effect is still // active we must clear the capture buffer to return silence if ((pContext->mLastCaptureIdx == pContext->mCaptureIdx) && - (pContext->mBufferUpdateTime.tv_sec != 0)) { - if (deltaMs > MAX_STALL_TIME_MS) { + (pContext->mBufferUpdateTime.tv_sec != 0) && + (deltaMs > MAX_STALL_TIME_MS)) { ALOGV("capture going to idle"); pContext->mBufferUpdateTime.tv_sec = 0; - memset(pReplyData, 0x80, pContext->mCaptureSize); + memset(pReplyData, 0x80, captureSize); + } else { + int32_t latencyMs = pContext->mLatency; + latencyMs -= deltaMs; + if (latencyMs < 0) { + latencyMs = 0; } + const uint32_t deltaSmpl = + pContext->mConfig.inputCfg.samplingRate * latencyMs / 1000; + int32_t capturePoint = pContext->mCaptureIdx - captureSize - deltaSmpl; + + if (capturePoint < 0) { + int32_t size = -capturePoint; + if (size > captureSize) { + size = captureSize; + } + memcpy(pReplyData, + pContext->mCaptureBuf + CAPTURE_BUF_SIZE + capturePoint, + size); + pReplyData = (char *)pReplyData + size; + captureSize -= size; + capturePoint = 0; + } + memcpy(pReplyData, + pContext->mCaptureBuf + capturePoint, + captureSize); } + pContext->mLastCaptureIdx = pContext->mCaptureIdx; } else { - memset(pReplyData, 0x80, pContext->mCaptureSize); + memset(pReplyData, 0x80, captureSize); } - break; + } break; case VISUALIZER_CMD_MEASURE: { uint16_t peakU16 = 0; diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk index e0acae6..f3770e4 100644 --- a/media/libmedia/Android.mk +++ b/media/libmedia/Android.mk @@ -72,7 +72,6 @@ LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper LOCAL_MODULE:= libmedia LOCAL_C_INCLUDES := \ - $(call include-path-for, graphics corecg) \ $(TOP)/frameworks/native/include/media/openmax \ external/icu4c/common \ external/icu4c/i18n \ diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 700718d..961b0a2 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -41,30 +41,22 @@ status_t AudioRecord::getMinFrameCount( return BAD_VALUE; } - // default to 0 in case of error - *frameCount = 0; - - size_t size = 0; + size_t size; status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size); if (status != NO_ERROR) { - ALOGE("AudioSystem could not query the input buffer size; status %d", status); - return NO_INIT; + ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, " + "channelMask %#x; status %d", sampleRate, format, channelMask, status); + return status; } - if (size == 0) { + // We double the size of input buffer for ping pong use of record buffer. + // Assumes audio_is_linear_pcm(format) + if ((*frameCount = (size * 2) / (popcount(channelMask) * audio_bytes_per_sample(format))) == 0) { ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x", sampleRate, format, channelMask); return BAD_VALUE; } - // We double the size of input buffer for ping pong use of record buffer. - size <<= 1; - - // Assumes audio_is_linear_pcm(format) - uint32_t channelCount = popcount(channelMask); - size /= channelCount * audio_bytes_per_sample(format); - - *frameCount = size; return NO_ERROR; } @@ -81,10 +73,10 @@ AudioRecord::AudioRecord( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, callback_t cbf, void* user, - int notificationFrames, + uint32_t notificationFrames, int sessionId, transfer_type transferType, audio_input_flags_t flags __unused) @@ -110,10 +102,8 @@ AudioRecord::~AudioRecord() mAudioRecordThread->requestExitAndWait(); mAudioRecordThread.clear(); } - if (mAudioRecord != 0) { - mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); - mAudioRecord.clear(); - } + mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); + mAudioRecord.clear(); IPCThreadState::self()->flushCommands(); AudioSystem::releaseAudioSessionId(mSessionId, -1); } @@ -124,15 +114,20 @@ status_t AudioRecord::set( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCountInt, + size_t frameCount, callback_t cbf, void* user, - int notificationFrames, + uint32_t notificationFrames, bool threadCanCallJava, int sessionId, transfer_type transferType, audio_input_flags_t flags) { + ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " + "notificationFrames %u, sessionId %d, transferType %d, flags %#x", + inputSource, sampleRate, format, channelMask, frameCount, notificationFrames, + sessionId, transferType, flags); + switch (transferType) { case TRANSFER_DEFAULT: if (cbf == NULL || threadCanCallJava) { @@ -156,23 +151,15 @@ status_t AudioRecord::set( } mTransfer = transferType; - // FIXME "int" here is legacy and will be replaced by size_t later - if (frameCountInt < 0) { - ALOGE("Invalid frame count %d", frameCountInt); - return BAD_VALUE; - } - size_t frameCount = frameCountInt; - - ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask, - frameCount); - AutoMutex lock(mLock); + // invariant that mAudioRecord != 0 is true only after set() returns successfully if (mAudioRecord != 0) { ALOGE("Track already in use"); return INVALID_OPERATION; } + // handle default values first. if (inputSource == AUDIO_SOURCE_DEFAULT) { inputSource = AUDIO_SOURCE_MIC; } @@ -209,15 +196,19 @@ status_t AudioRecord::set( uint32_t channelCount = popcount(channelMask); mChannelCount = channelCount; - // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t) - mFrameSize = channelCount * audio_bytes_per_sample(format); + if (audio_is_linear_pcm(format)) { + mFrameSize = channelCount * audio_bytes_per_sample(format); + } else { + mFrameSize = sizeof(uint8_t); + } // validate framecount - size_t minFrameCount = 0; + size_t minFrameCount; status_t status = AudioRecord::getMinFrameCount(&minFrameCount, sampleRate, format, channelMask); if (status != NO_ERROR) { - ALOGE("getMinFrameCount() failed; status %d", status); + ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d", + sampleRate, format, channelMask, status); return status; } ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount); @@ -242,23 +233,27 @@ status_t AudioRecord::set( ALOGV("set(): mSessionId %d", mSessionId); mFlags = flags; + mCbf = cbf; + + if (cbf != NULL) { + mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava); + mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO); + } // create the IAudioRecord status = openRecord_l(0 /*epoch*/); + if (status != NO_ERROR) { + if (mAudioRecordThread != 0) { + mAudioRecordThread->requestExit(); // see comment in AudioRecord.h + mAudioRecordThread->requestExitAndWait(); + mAudioRecordThread.clear(); + } return status; } - if (cbf != NULL) { - mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava); - mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO); - } - mStatus = NO_ERROR; - mActive = false; - mCbf = cbf; - mRefreshRemaining = true; mUserData = user; // TODO: add audio hardware input latency here mLatency = (1000*mFrameCount) / sampleRate; @@ -433,22 +428,37 @@ status_t AudioRecord::openRecord_l(size_t epoch) return NO_INIT; } - IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; - pid_t tid = -1; + // Fast tracks must be at the primary _output_ [sic] sampling rate, + // because there is currently no concept of a primary input sampling rate + uint32_t afSampleRate = AudioSystem::getPrimaryOutputSamplingRate(); + if (afSampleRate == 0) { + ALOGW("getPrimaryOutputSamplingRate failed"); + } // Client can only express a preference for FAST. Server will perform additional tests. - // The only supported use case for FAST is callback transfer mode. + if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !( + // use case: callback transfer mode + (mTransfer == TRANSFER_CALLBACK) && + // matching sample rate + (mSampleRate == afSampleRate))) { + ALOGW("AUDIO_INPUT_FLAG_FAST denied by client"); + // once denied, do not request again if IAudioRecord is re-created + mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); + } + + IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; + + pid_t tid = -1; if (mFlags & AUDIO_INPUT_FLAG_FAST) { - if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) { - ALOGW("AUDIO_INPUT_FLAG_FAST denied by client"); - // once denied, do not request again if IAudioRecord is re-created - mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); - } else { - trackFlags |= IAudioFlinger::TRACK_FAST; + trackFlags |= IAudioFlinger::TRACK_FAST; + if (mAudioRecordThread != 0) { tid = mAudioRecordThread->getTid(); } } + // FIXME Assume double buffering, because we don't know the true HAL sample rate + const uint32_t nBuffering = 2; + mNotificationFramesAct = mNotificationFramesReq; size_t frameCount = mReqFrameCount; @@ -485,10 +495,12 @@ status_t AudioRecord::openRecord_l(size_t epoch) ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, "session ID changed from %d to %d", originalSessionId, mSessionId); - if (record == 0 || status != NO_ERROR) { + if (status != NO_ERROR) { ALOGE("AudioFlinger could not create record track, status: %d", status); goto release; } + ALOG_ASSERT(record != 0); + // AudioFlinger now owns the reference to the I/O handle, // so we are no longer responsible for releasing it. @@ -502,52 +514,55 @@ status_t AudioRecord::openRecord_l(size_t epoch) ALOGE("Could not get control block pointer"); return NO_INIT; } + // invariant that mAudioRecord != 0 is true only after set() returns successfully if (mAudioRecord != 0) { mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); mDeathNotifier.clear(); } - - // We retain a copy of the I/O handle, but don't own the reference - mInput = input; mAudioRecord = record; + mCblkMemory = iMem; audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); mCblk = cblk; - // note that temp is the (possibly revised) value of mFrameCount + // note that temp is the (possibly revised) value of frameCount if (temp < frameCount || (frameCount == 0 && temp == 0)) { ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); } frameCount = temp; - // If IAudioRecord is re-created, don't let the requested frameCount - // decrease. This can confuse clients that cache frameCount(). - if (frameCount > mReqFrameCount) { - mReqFrameCount = frameCount; - } - // FIXME missing fast track frameCount logic mAwaitBoost = false; if (mFlags & AUDIO_INPUT_FLAG_FAST) { if (trackFlags & IAudioFlinger::TRACK_FAST) { - ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", mFrameCount); + ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", frameCount); mAwaitBoost = true; - // double-buffering is not required for fast tracks, due to tighter scheduling - if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount) { - mNotificationFramesAct = mFrameCount; - } } else { - ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", mFrameCount); + ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", frameCount); // once denied, do not request again if IAudioRecord is re-created mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); - if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) { - mNotificationFramesAct = mFrameCount/2; - } + } + // Theoretically double-buffering is not required for fast tracks, + // due to tighter scheduling. But in practice, to accomodate kernels with + // scheduling jitter, and apps with computation jitter, we use double-buffering. + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { + mNotificationFramesAct = frameCount/nBuffering; } } - // starting address of buffers in shared memory + // We retain a copy of the I/O handle, but don't own the reference + mInput = input; + mRefreshRemaining = true; + + // Starting address of buffers in shared memory, immediately after the control block. This + // address is for the mapping within client address space. AudioFlinger::TrackBase::mBuffer + // is for the server address space. void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); mFrameCount = frameCount; + // If IAudioRecord is re-created, don't let the requested frameCount + // decrease. This can confuse clients that cache frameCount(). + if (frameCount > mReqFrameCount) { + mReqFrameCount = frameCount; + } // update proxy mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize); @@ -799,7 +814,7 @@ nsecs_t AudioRecord::processAudioBuffer() } // Cache other fields that will be needed soon - size_t notificationFrames = mNotificationFramesAct; + uint32_t notificationFrames = mNotificationFramesAct; if (mRefreshRemaining) { mRefreshRemaining = false; mRemainingFrames = notificationFrames; diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 5c62260..ae47201 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -99,7 +99,8 @@ AudioTrack::AudioTrack() : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT) + mPreviousSchedulingGroup(SP_DEFAULT), + mPausedPosition(0) { } @@ -108,11 +109,11 @@ AudioTrack::AudioTrack( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, audio_output_flags_t flags, callback_t cbf, void* user, - int notificationFrames, + uint32_t notificationFrames, int sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, @@ -121,7 +122,8 @@ AudioTrack::AudioTrack( : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT) + mPreviousSchedulingGroup(SP_DEFAULT), + mPausedPosition(0) { mStatus = set(streamType, sampleRate, format, channelMask, frameCount, flags, cbf, user, notificationFrames, @@ -138,7 +140,7 @@ AudioTrack::AudioTrack( audio_output_flags_t flags, callback_t cbf, void* user, - int notificationFrames, + uint32_t notificationFrames, int sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, @@ -147,7 +149,8 @@ AudioTrack::AudioTrack( : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT) + mPreviousSchedulingGroup(SP_DEFAULT), + mPausedPosition(0) { mStatus = set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags, cbf, user, notificationFrames, @@ -182,11 +185,11 @@ status_t AudioTrack::set( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCountInt, + size_t frameCount, audio_output_flags_t flags, callback_t cbf, void* user, - int notificationFrames, + uint32_t notificationFrames, const sp<IMemory>& sharedBuffer, bool threadCanCallJava, int sessionId, @@ -195,6 +198,11 @@ status_t AudioTrack::set( int uid, pid_t pid) { + ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " + "flags #%x, notificationFrames %u, sessionId %d, transferType %d", + streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, + sessionId, transferType); + switch (transferType) { case TRANSFER_DEFAULT: if (sharedBuffer != 0) { @@ -231,13 +239,6 @@ status_t AudioTrack::set( mSharedBuffer = sharedBuffer; mTransfer = transferType; - // FIXME "int" here is legacy and will be replaced by size_t later - if (frameCountInt < 0) { - ALOGE("Invalid frame count %d", frameCountInt); - return BAD_VALUE; - } - size_t frameCount = frameCountInt; - ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); @@ -288,6 +289,9 @@ status_t AudioTrack::set( ALOGE("Invalid channel mask %#x", channelMask); return BAD_VALUE; } + mChannelMask = channelMask; + uint32_t channelCount = popcount(channelMask); + mChannelCount = channelCount; // AudioFlinger does not currently support 8-bit data in shared memory if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { @@ -311,10 +315,6 @@ status_t AudioTrack::set( flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); } - mChannelMask = channelMask; - uint32_t channelCount = popcount(channelMask); - mChannelCount = channelCount; - if (audio_is_linear_pcm(format)) { mFrameSize = channelCount * audio_bytes_per_sample(format); mFrameSizeAF = channelCount * sizeof(int16_t); @@ -554,6 +554,16 @@ void AudioTrack::pause() } mProxy->interrupt(); mAudioTrack->pause(); + + if (isOffloaded_l()) { + if (mOutput != 0) { + uint32_t halFrames; + // OffloadThread sends HAL pause in its threadLoop.. time saved + // here can be slightly off + AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); + ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); + } + } } status_t AudioTrack::setVolume(float left, float right) @@ -773,6 +783,12 @@ status_t AudioTrack::getPosition(uint32_t *position) const if (isOffloaded_l()) { uint32_t dspFrames = 0; + if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) { + ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); + *position = mPausedPosition; + return NO_ERROR; + } + if (mOutput != 0) { uint32_t halFrames; AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); @@ -888,8 +904,8 @@ status_t AudioTrack::createTrack_l(size_t epoch) // either of these use cases: // use case 1: shared buffer (mSharedBuffer != 0) || - // use case 2: callback handler - (mCbf != NULL)) && + // use case 2: callback transfer mode + (mTransfer == TRANSFER_CALLBACK)) && // matching sample rate (mSampleRate == afSampleRate))) { ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); @@ -1012,10 +1028,12 @@ status_t AudioTrack::createTrack_l(size_t epoch) mClientUid, &status); - if (track == 0) { + if (status != NO_ERROR) { ALOGE("AudioFlinger could not create track, status: %d", status); goto release; } + ALOG_ASSERT(track != 0); + // AudioFlinger now owns the reference to the I/O handle, // so we are no longer responsible for releasing it. @@ -1035,6 +1053,7 @@ status_t AudioTrack::createTrack_l(size_t epoch) mDeathNotifier.clear(); } mAudioTrack = track; + mCblkMemory = iMem; audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); mCblk = cblk; @@ -1046,6 +1065,7 @@ status_t AudioTrack::createTrack_l(size_t epoch) ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); } frameCount = temp; + mAwaitBoost = false; if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { if (trackFlags & IAudioFlinger::TRACK_FAST) { @@ -1099,6 +1119,7 @@ status_t AudioTrack::createTrack_l(size_t epoch) mAudioTrack->attachAuxEffect(mAuxEffectId); // FIXME don't believe this lie mLatency = afLatency + (1000*frameCount) / mSampleRate; + mFrameCount = frameCount; // If IAudioTrack is re-created, don't let the requested frameCount // decrease. This can confuse clients that cache frameCount(). @@ -1478,7 +1499,7 @@ nsecs_t AudioTrack::processAudioBuffer() // Cache other fields that will be needed soon uint32_t loopPeriod = mLoopPeriod; uint32_t sampleRate = mSampleRate; - size_t notificationFrames = mNotificationFramesAct; + uint32_t notificationFrames = mNotificationFramesAct; if (mRefreshRemaining) { mRefreshRemaining = false; mRemainingFrames = notificationFrames; @@ -1486,6 +1507,7 @@ nsecs_t AudioTrack::processAudioBuffer() } size_t misalignment = mProxy->getMisalignment(); uint32_t sequence = mSequence; + sp<AudioTrackClientProxy> proxy = mProxy; // These fields don't need to be cached, because they are assigned only by set(): // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags @@ -1494,35 +1516,32 @@ nsecs_t AudioTrack::processAudioBuffer() mLock.unlock(); if (waitStreamEnd) { - AutoMutex lock(mLock); - - sp<AudioTrackClientProxy> proxy = mProxy; - sp<IMemory> iMem = mCblkMemory; - struct timespec timeout; timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; timeout.tv_nsec = 0; - mLock.unlock(); - status_t status = mProxy->waitStreamEndDone(&timeout); - mLock.lock(); + status_t status = proxy->waitStreamEndDone(&timeout); switch (status) { case NO_ERROR: case DEAD_OBJECT: case TIMED_OUT: - mLock.unlock(); mCbf(EVENT_STREAM_END, mUserData, NULL); - mLock.lock(); - if (mState == STATE_STOPPING) { - mState = STATE_STOPPED; - if (status != DEAD_OBJECT) { - return NS_INACTIVE; + { + AutoMutex lock(mLock); + // The previously assigned value of waitStreamEnd is no longer valid, + // since the mutex has been unlocked and either the callback handler + // or another thread could have re-started the AudioTrack during that time. + waitStreamEnd = mState == STATE_STOPPING; + if (waitStreamEnd) { + mState = STATE_STOPPED; } } - return 0; - default: - return 0; + if (waitStreamEnd && status != DEAD_OBJECT) { + return NS_INACTIVE; + } + break; } + return 0; } // perform callbacks while unlocked diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index e696323..a9a9f1a 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -58,7 +58,7 @@ enum { RESTORE_OUTPUT, OPEN_INPUT, CLOSE_INPUT, - SET_STREAM_OUTPUT, + INVALIDATE_STREAM, SET_VOICE_VOLUME, GET_RENDER_POSITION, GET_INPUT_FRAMES_LOST, @@ -545,13 +545,12 @@ public: return reply.readInt32(); } - virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) + virtual status_t invalidateStream(audio_stream_type_t stream) { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); data.writeInt32((int32_t) stream); - data.writeInt32((int32_t) output); - remote()->transact(SET_STREAM_OUTPUT, data, &reply); + remote()->transact(INVALIDATE_STREAM, data, &reply); return reply.readInt32(); } @@ -1044,11 +1043,10 @@ status_t BnAudioFlinger::onTransact( reply->writeInt32(closeInput((audio_io_handle_t) data.readInt32())); return NO_ERROR; } break; - case SET_STREAM_OUTPUT: { + case INVALIDATE_STREAM: { CHECK_INTERFACE(IAudioFlinger, data, reply); - uint32_t stream = data.readInt32(); - audio_io_handle_t output = (audio_io_handle_t) data.readInt32(); - reply->writeInt32(setStreamOutput((audio_stream_type_t) stream, output)); + audio_stream_type_t stream = (audio_stream_type_t) data.readInt32(); + reply->writeInt32(invalidateStream(stream)); return NO_ERROR; } break; case SET_VOICE_VOLUME: { diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp index 4be3c09..1a027a6 100644 --- a/media/libmedia/IAudioPolicyService.cpp +++ b/media/libmedia/IAudioPolicyService.cpp @@ -476,10 +476,11 @@ status_t BnAudioPolicyService::onTransact( case START_OUTPUT: { CHECK_INTERFACE(IAudioPolicyService, data, reply); audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32()); - uint32_t stream = data.readInt32(); + audio_stream_type_t stream = + static_cast <audio_stream_type_t>(data.readInt32()); int session = data.readInt32(); reply->writeInt32(static_cast <uint32_t>(startOutput(output, - (audio_stream_type_t)stream, + stream, session))); return NO_ERROR; } break; @@ -487,10 +488,11 @@ status_t BnAudioPolicyService::onTransact( case STOP_OUTPUT: { CHECK_INTERFACE(IAudioPolicyService, data, reply); audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32()); - uint32_t stream = data.readInt32(); + audio_stream_type_t stream = + static_cast <audio_stream_type_t>(data.readInt32()); int session = data.readInt32(); reply->writeInt32(static_cast <uint32_t>(stopOutput(output, - (audio_stream_type_t)stream, + stream, session))); return NO_ERROR; } break; @@ -633,7 +635,7 @@ status_t BnAudioPolicyService::onTransact( CHECK_INTERFACE(IAudioPolicyService, data, reply); audio_stream_type_t stream = (audio_stream_type_t) data.readInt32(); uint32_t inPastMs = (uint32_t)data.readInt32(); - reply->writeInt32( isStreamActive((audio_stream_type_t) stream, inPastMs) ); + reply->writeInt32( isStreamActive(stream, inPastMs) ); return NO_ERROR; } break; @@ -641,7 +643,7 @@ status_t BnAudioPolicyService::onTransact( CHECK_INTERFACE(IAudioPolicyService, data, reply); audio_stream_type_t stream = (audio_stream_type_t) data.readInt32(); uint32_t inPastMs = (uint32_t)data.readInt32(); - reply->writeInt32( isStreamActiveRemotely((audio_stream_type_t) stream, inPastMs) ); + reply->writeInt32( isStreamActiveRemotely(stream, inPastMs) ); return NO_ERROR; } break; diff --git a/media/libmedia/IMediaHTTPConnection.cpp b/media/libmedia/IMediaHTTPConnection.cpp index 22c470a..7e26ee6 100644 --- a/media/libmedia/IMediaHTTPConnection.cpp +++ b/media/libmedia/IMediaHTTPConnection.cpp @@ -95,7 +95,10 @@ struct BpMediaHTTPConnection : public BpInterface<IMediaHTTPConnection> { data.writeInt32(size); status_t err = remote()->transact(READ_AT, data, &reply); - CHECK_EQ(err, (status_t)OK); + if (err != OK) { + ALOGE("remote readAt failed"); + return UNKNOWN_ERROR; + } int32_t exceptionCode = reply.readExceptionCode(); diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp index e914b34..f0f1832 100644 --- a/media/libmedia/JetPlayer.cpp +++ b/media/libmedia/JetPlayer.cpp @@ -90,7 +90,7 @@ int JetPlayer::init() pLibConfig->sampleRate, AUDIO_FORMAT_PCM_16_BIT, audio_channel_out_mask_from_count(pLibConfig->numChannels), - mTrackBufferSize, + (size_t) mTrackBufferSize, AUDIO_OUTPUT_FLAG_NONE); // create render and playback thread diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp index 4885b4f..a55e09c 100644 --- a/media/libmedia/SoundPool.cpp +++ b/media/libmedia/SoundPool.cpp @@ -587,7 +587,7 @@ void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftV uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5); uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate; uint32_t bufferFrames = (totalFrames + (kDefaultBufferCount - 1)) / kDefaultBufferCount; - uint32_t frameCount = 0; + size_t frameCount = 0; if (loop) { frameCount = sample->size()/numChannels/ diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk index 8f21632..4189a5e 100644 --- a/media/libmediaplayerservice/Android.mk +++ b/media/libmediaplayerservice/Android.mk @@ -45,7 +45,6 @@ LOCAL_STATIC_LIBRARIES := \ libstagefright_rtsp \ LOCAL_C_INCLUDES := \ - $(call include-path-for, graphics corecg) \ $(TOP)/frameworks/av/media/libstagefright/include \ $(TOP)/frameworks/av/media/libstagefright/rtsp \ $(TOP)/frameworks/av/media/libstagefright/wifi-display \ diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp index 142788d..200c561 100644 --- a/media/libmediaplayerservice/MediaPlayerService.cpp +++ b/media/libmediaplayerservice/MediaPlayerService.cpp @@ -1455,7 +1455,7 @@ status_t MediaPlayerService::AudioOutput::open( format, bufferCount, mSessionId, flags); uint32_t afSampleRate; size_t afFrameCount; - uint32_t frameCount; + size_t frameCount; // offloading is only supported in callback mode for now. // offloadInfo must be present if offload flag is set diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp index 845a589..5b7a236 100644 --- a/media/libmediaplayerservice/StagefrightRecorder.cpp +++ b/media/libmediaplayerservice/StagefrightRecorder.cpp @@ -748,7 +748,7 @@ status_t StagefrightRecorder::setClientName(const String16& clientName) { return OK; } -status_t StagefrightRecorder::prepare() { +status_t StagefrightRecorder::prepareInternal() { ALOGV("prepare"); if (mOutputFd < 0) { ALOGE("Output file descriptor is invalid"); @@ -794,6 +794,13 @@ status_t StagefrightRecorder::prepare() { return status; } +status_t StagefrightRecorder::prepare() { + if (mVideoSource == VIDEO_SOURCE_SURFACE) { + return prepareInternal(); + } + return OK; +} + status_t StagefrightRecorder::start() { ALOGV("start"); if (mOutputFd < 0) { @@ -801,15 +808,20 @@ status_t StagefrightRecorder::start() { return INVALID_OPERATION; } - // Get UID here for permission checking - mClientUid = IPCThreadState::self()->getCallingUid(); + status_t status = OK; + + if (mVideoSource != VIDEO_SOURCE_SURFACE) { + status = prepareInternal(); + if (status != OK) { + return status; + } + } + if (mWriter == NULL) { ALOGE("File writer is not avaialble"); return UNKNOWN_ERROR; } - status_t status = OK; - switch (mOutputFormat) { case OUTPUT_FORMAT_DEFAULT: case OUTPUT_FORMAT_THREE_GPP: diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h index 7d6abd3..377d168 100644 --- a/media/libmediaplayerservice/StagefrightRecorder.h +++ b/media/libmediaplayerservice/StagefrightRecorder.h @@ -127,6 +127,7 @@ private: sp<IGraphicBufferProducer> mGraphicBufferProducer; sp<ALooper> mLooper; + status_t prepareInternal(); status_t setupMPEG4Recording(); void setupMPEG4MetaData(sp<MetaData> *meta); status_t setupAMRRecording(); diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp index d47ac98..a750ad0 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp @@ -1006,7 +1006,14 @@ status_t NuPlayer::feedDecoderInputData(bool audio, const sp<AMessage> &msg) { &NuPlayer::performScanSources)); } - flushDecoder(audio, formatChange); + sp<AMessage> newFormat = mSource->getFormat(audio); + sp<Decoder> &decoder = audio ? mAudioDecoder : mVideoDecoder; + if (formatChange && !decoder->supportsSeamlessFormatChange(newFormat)) { + flushDecoder(audio, /* needShutdown = */ true); + } else { + flushDecoder(audio, /* needShutdown = */ false); + err = OK; + } } else { // This stream is unaffected by the discontinuity diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp index 22f699e..2423fd5 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp @@ -67,6 +67,7 @@ void NuPlayer::Decoder::configure(const sp<AMessage> &format) { // queue. bool needDedicatedLooper = !strncasecmp(mime.c_str(), "video/", 6); + mFormat = format; mCodec = new ACodec; if (needDedicatedLooper && mCodecLooper == NULL) { @@ -147,5 +148,65 @@ void NuPlayer::Decoder::initiateShutdown() { } } +bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const { + if (targetFormat == NULL) { + return true; + } + + AString mime; + if (!targetFormat->findString("mime", &mime)) { + return false; + } + + if (!strcasecmp(mime.c_str(), MEDIA_MIMETYPE_AUDIO_AAC)) { + // field-by-field comparison + const char * keys[] = { "channel-count", "sample-rate", "is-adts" }; + for (unsigned int i = 0; i < sizeof(keys) / sizeof(keys[0]); i++) { + int32_t oldVal, newVal; + if (!mFormat->findInt32(keys[i], &oldVal) || !targetFormat->findInt32(keys[i], &newVal) + || oldVal != newVal) { + return false; + } + } + + sp<ABuffer> oldBuf, newBuf; + if (mFormat->findBuffer("csd-0", &oldBuf) && targetFormat->findBuffer("csd-0", &newBuf)) { + if (oldBuf->size() != newBuf->size()) { + return false; + } + return !memcmp(oldBuf->data(), newBuf->data(), oldBuf->size()); + } + } + return false; +} + +bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetFormat) const { + if (mFormat == NULL) { + return false; + } + + if (targetFormat == NULL) { + return true; + } + + AString oldMime, newMime; + if (!mFormat->findString("mime", &oldMime) + || !targetFormat->findString("mime", &newMime) + || !(oldMime == newMime)) { + return false; + } + + bool audio = !strncasecmp(oldMime.c_str(), "audio/", strlen("audio/")); + bool seamless; + if (audio) { + seamless = supportsSeamlessAudioFormatChange(targetFormat); + } else { + seamless = mCodec != NULL && mCodec->isConfiguredForAdaptivePlayback(); + } + + ALOGV("%s seamless support for %s", seamless ? "yes" : "no", oldMime.c_str()); + return seamless; +} + } // namespace android diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h index a876148..78ea74a 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h @@ -36,6 +36,8 @@ struct NuPlayer::Decoder : public AHandler { void signalResume(); void initiateShutdown(); + bool supportsSeamlessFormatChange(const sp<AMessage> &to) const; + protected: virtual ~Decoder(); @@ -49,6 +51,7 @@ private: sp<AMessage> mNotify; sp<NativeWindowWrapper> mNativeWindow; + sp<AMessage> mFormat; sp<ACodec> mCodec; sp<ALooper> mCodecLooper; @@ -59,6 +62,8 @@ private: void onFillThisBuffer(const sp<AMessage> &msg); + bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const; + DISALLOW_EVIL_CONSTRUCTORS(Decoder); }; diff --git a/media/libnbaio/AudioBufferProviderSource.cpp b/media/libnbaio/AudioBufferProviderSource.cpp index 4a69104..551f516 100644 --- a/media/libnbaio/AudioBufferProviderSource.cpp +++ b/media/libnbaio/AudioBufferProviderSource.cpp @@ -68,7 +68,7 @@ ssize_t AudioBufferProviderSource::read(void *buffer, } // count could be zero, either because count was zero on entry or // available is zero, but both are unlikely so don't check for that - memcpy(buffer, (char *) mBuffer.raw + (mConsumed << mBitShift), count << mBitShift); + memcpy(buffer, (char *) mBuffer.raw + (mConsumed * mFrameSize), count * mFrameSize); if (CC_UNLIKELY((mConsumed += count) >= mBuffer.frameCount)) { mProvider->releaseBuffer(&mBuffer); mBuffer.raw = NULL; @@ -120,7 +120,7 @@ ssize_t AudioBufferProviderSource::readVia(readVia_t via, size_t total, void *us count = available; } if (CC_LIKELY(count > 0)) { - char* readTgt = (char *) mBuffer.raw + (mConsumed << mBitShift); + char* readTgt = (char *) mBuffer.raw + (mConsumed * mFrameSize); ssize_t ret = via(user, readTgt, count, readPTS); if (CC_UNLIKELY(ret <= 0)) { if (CC_LIKELY(accumulator > 0)) { diff --git a/media/libnbaio/AudioStreamInSource.cpp b/media/libnbaio/AudioStreamInSource.cpp index ae8fac8..80bf61a 100644 --- a/media/libnbaio/AudioStreamInSource.cpp +++ b/media/libnbaio/AudioStreamInSource.cpp @@ -43,13 +43,11 @@ ssize_t AudioStreamInSource::negotiate(const NBAIO_Format offers[], size_t numOf if (!Format_isValid(mFormat)) { mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common); audio_format_t streamFormat = mStream->common.get_format(&mStream->common); - if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) { - uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); - audio_channel_mask_t channelMask = - (audio_channel_mask_t) mStream->common.get_channels(&mStream->common); - mFormat = Format_from_SR_C(sampleRate, popcount(channelMask)); - mBitShift = Format_frameBitShift(mFormat); - } + uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); + audio_channel_mask_t channelMask = + (audio_channel_mask_t) mStream->common.get_channels(&mStream->common); + mFormat = Format_from_SR_C(sampleRate, popcount(channelMask), streamFormat); + mFrameSize = Format_frameSize(mFormat); } return NBAIO_Source::negotiate(offers, numOffers, counterOffers, numCounterOffers); } @@ -70,9 +68,9 @@ ssize_t AudioStreamInSource::read(void *buffer, size_t count) if (CC_UNLIKELY(!Format_isValid(mFormat))) { return NEGOTIATE; } - ssize_t bytesRead = mStream->read(mStream, buffer, count << mBitShift); + ssize_t bytesRead = mStream->read(mStream, buffer, count * mFrameSize); if (bytesRead > 0) { - size_t framesRead = bytesRead >> mBitShift; + size_t framesRead = bytesRead / mFrameSize; mFramesRead += framesRead; return framesRead; } else { diff --git a/media/libnbaio/AudioStreamOutSink.cpp b/media/libnbaio/AudioStreamOutSink.cpp index aa9810e..c28d34d 100644 --- a/media/libnbaio/AudioStreamOutSink.cpp +++ b/media/libnbaio/AudioStreamOutSink.cpp @@ -40,13 +40,11 @@ ssize_t AudioStreamOutSink::negotiate(const NBAIO_Format offers[], size_t numOff if (!Format_isValid(mFormat)) { mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common); audio_format_t streamFormat = mStream->common.get_format(&mStream->common); - if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) { - uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); - audio_channel_mask_t channelMask = - (audio_channel_mask_t) mStream->common.get_channels(&mStream->common); - mFormat = Format_from_SR_C(sampleRate, popcount(channelMask)); - mBitShift = Format_frameBitShift(mFormat); - } + uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); + audio_channel_mask_t channelMask = + (audio_channel_mask_t) mStream->common.get_channels(&mStream->common); + mFormat = Format_from_SR_C(sampleRate, popcount(channelMask), streamFormat); + mFrameSize = Format_frameSize(mFormat); } return NBAIO_Sink::negotiate(offers, numOffers, counterOffers, numCounterOffers); } @@ -57,9 +55,9 @@ ssize_t AudioStreamOutSink::write(const void *buffer, size_t count) return NEGOTIATE; } ALOG_ASSERT(Format_isValid(mFormat)); - ssize_t ret = mStream->write(mStream, buffer, count << mBitShift); + ssize_t ret = mStream->write(mStream, buffer, count * mFrameSize); if (ret > 0) { - ret >>= mBitShift; + ret /= mFrameSize; mFramesWritten += ret; } else { // FIXME verify HAL implementations are returning the correct error codes e.g. WOULD_BLOCK diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp index b23967b..9c8461c 100644 --- a/media/libnbaio/MonoPipe.cpp +++ b/media/libnbaio/MonoPipe.cpp @@ -115,11 +115,11 @@ ssize_t MonoPipe::write(const void *buffer, size_t count) part1 = written; } if (CC_LIKELY(part1 > 0)) { - memcpy((char *) mBuffer + (rear << mBitShift), buffer, part1 << mBitShift); + memcpy((char *) mBuffer + (rear * mFrameSize), buffer, part1 * mFrameSize); if (CC_UNLIKELY(rear + part1 == mMaxFrames)) { size_t part2 = written - part1; if (CC_LIKELY(part2 > 0)) { - memcpy(mBuffer, (char *) buffer + (part1 << mBitShift), part2 << mBitShift); + memcpy(mBuffer, (char *) buffer + (part1 * mFrameSize), part2 * mFrameSize); } } android_atomic_release_store(written + mRear, &mRear); @@ -129,7 +129,7 @@ ssize_t MonoPipe::write(const void *buffer, size_t count) break; } count -= written; - buffer = (char *) buffer + (written << mBitShift); + buffer = (char *) buffer + (written * mFrameSize); // Simulate blocking I/O by sleeping at different rates, depending on a throttle. // The throttle tries to keep the mean pipe depth near the setpoint, with a slight jitter. uint32_t ns; diff --git a/media/libnbaio/MonoPipeReader.cpp b/media/libnbaio/MonoPipeReader.cpp index 851341a..de82229 100644 --- a/media/libnbaio/MonoPipeReader.cpp +++ b/media/libnbaio/MonoPipeReader.cpp @@ -73,11 +73,11 @@ ssize_t MonoPipeReader::read(void *buffer, size_t count, int64_t readPTS) part1 = red; } if (CC_LIKELY(part1 > 0)) { - memcpy(buffer, (char *) mPipe->mBuffer + (front << mBitShift), part1 << mBitShift); + memcpy(buffer, (char *) mPipe->mBuffer + (front * mFrameSize), part1 * mFrameSize); if (CC_UNLIKELY(front + part1 == mPipe->mMaxFrames)) { size_t part2 = red - part1; if (CC_LIKELY(part2 > 0)) { - memcpy((char *) buffer + (part1 << mBitShift), mPipe->mBuffer, part2 << mBitShift); + memcpy((char *) buffer + (part1 * mFrameSize), mPipe->mBuffer, part2 * mFrameSize); } } mPipe->updateFrontAndNRPTS(red + mPipe->mFront, nextReadPTS); diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp index 51514de..ff3284c 100644 --- a/media/libnbaio/NBAIO.cpp +++ b/media/libnbaio/NBAIO.cpp @@ -24,63 +24,17 @@ namespace android { size_t Format_frameSize(const NBAIO_Format& format) { - // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT - return Format_channelCount(format) * sizeof(short); + return format.mFrameSize; } -int Format_frameBitShift(const NBAIO_Format& format) -{ - // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT - // sizeof(short) == 2, so frame size == 1 << channels - return Format_channelCount(format); - // FIXME must return -1 for non-power of 2 -} - -const NBAIO_Format Format_Invalid = { 0 }; - -enum { - Format_SR_8000, - Format_SR_11025, - Format_SR_16000, - Format_SR_22050, - Format_SR_24000, - Format_SR_32000, - Format_SR_44100, - Format_SR_48000, - Format_SR_Mask = 7 -}; - -enum { - Format_C_1 = 0x08, - Format_C_2 = 0x10, - Format_C_Mask = 0x18 -}; +const NBAIO_Format Format_Invalid = { 0, 0, AUDIO_FORMAT_INVALID, 0 }; unsigned Format_sampleRate(const NBAIO_Format& format) { if (!Format_isValid(format)) { return 0; } - switch (format.mPacked & Format_SR_Mask) { - case Format_SR_8000: - return 8000; - case Format_SR_11025: - return 11025; - case Format_SR_16000: - return 16000; - case Format_SR_22050: - return 22050; - case Format_SR_24000: - return 24000; - case Format_SR_32000: - return 32000; - case Format_SR_44100: - return 44100; - case Format_SR_48000: - return 48000; - default: - return 0; - } + return format.mSampleRate; } unsigned Format_channelCount(const NBAIO_Format& format) @@ -88,59 +42,21 @@ unsigned Format_channelCount(const NBAIO_Format& format) if (!Format_isValid(format)) { return 0; } - switch (format.mPacked & Format_C_Mask) { - case Format_C_1: - return 1; - case Format_C_2: - return 2; - default: - return 0; - } + return format.mChannelCount; } -NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount) +NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount, + audio_format_t format) { - unsigned format; - switch (sampleRate) { - case 8000: - format = Format_SR_8000; - break; - case 11025: - format = Format_SR_11025; - break; - case 16000: - format = Format_SR_16000; - break; - case 22050: - format = Format_SR_22050; - break; - case 24000: - format = Format_SR_24000; - break; - case 32000: - format = Format_SR_32000; - break; - case 44100: - format = Format_SR_44100; - break; - case 48000: - format = Format_SR_48000; - break; - default: - return Format_Invalid; - } - switch (channelCount) { - case 1: - format |= Format_C_1; - break; - case 2: - format |= Format_C_2; - break; - default: + if (sampleRate == 0 || channelCount == 0 || !audio_is_valid_format(format)) { return Format_Invalid; } NBAIO_Format ret; - ret.mPacked = format; + ret.mSampleRate = sampleRate; + ret.mChannelCount = channelCount; + ret.mFormat = format; + ret.mFrameSize = audio_is_linear_pcm(format) ? + channelCount * audio_bytes_per_sample(format) : sizeof(uint8_t); return ret; } @@ -242,12 +158,15 @@ ssize_t NBAIO_Port::negotiate(const NBAIO_Format offers[], size_t numOffers, bool Format_isValid(const NBAIO_Format& format) { - return format.mPacked != Format_Invalid.mPacked; + return format.mSampleRate != 0 && format.mChannelCount != 0 && + format.mFormat != AUDIO_FORMAT_INVALID && format.mFrameSize != 0; } bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2) { - return format1.mPacked == format2.mPacked; + return format1.mSampleRate == format2.mSampleRate && + format1.mChannelCount == format2.mChannelCount && format1.mFormat == format2.mFormat && + format1.mFrameSize == format2.mFrameSize; } } // namespace android diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp index 115f311..28a034c 100644 --- a/media/libnbaio/Pipe.cpp +++ b/media/libnbaio/Pipe.cpp @@ -52,13 +52,13 @@ ssize_t Pipe::write(const void *buffer, size_t count) if (CC_LIKELY(written > count)) { written = count; } - memcpy((char *) mBuffer + (rear << mBitShift), buffer, written << mBitShift); + memcpy((char *) mBuffer + (rear * mFrameSize), buffer, written * mFrameSize); if (CC_UNLIKELY(rear + written == mMaxFrames)) { if (CC_UNLIKELY((count -= written) > rear)) { count = rear; } if (CC_LIKELY(count > 0)) { - memcpy(mBuffer, (char *) buffer + (written << mBitShift), count << mBitShift); + memcpy(mBuffer, (char *) buffer + (written * mFrameSize), count * mFrameSize); written += count; } } diff --git a/media/libnbaio/PipeReader.cpp b/media/libnbaio/PipeReader.cpp index 24da1bd..c8e4953 100644 --- a/media/libnbaio/PipeReader.cpp +++ b/media/libnbaio/PipeReader.cpp @@ -76,14 +76,14 @@ ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS __unused) red = count; } // In particular, an overrun during the memcpy will result in reading corrupt data - memcpy(buffer, (char *) mPipe.mBuffer + (front << mBitShift), red << mBitShift); + memcpy(buffer, (char *) mPipe.mBuffer + (front * mFrameSize), red * mFrameSize); // We could re-read the rear pointer here to detect the corruption, but why bother? if (CC_UNLIKELY(front + red == mPipe.mMaxFrames)) { if (CC_UNLIKELY((count -= red) > front)) { count = front; } if (CC_LIKELY(count > 0)) { - memcpy((char *) buffer + (red << mBitShift), mPipe.mBuffer, count << mBitShift); + memcpy((char *) buffer + (red * mFrameSize), mPipe.mBuffer, count * mFrameSize); red += count; } } diff --git a/media/libnbaio/SourceAudioBufferProvider.cpp b/media/libnbaio/SourceAudioBufferProvider.cpp index 062fa0f..e21ef48 100644 --- a/media/libnbaio/SourceAudioBufferProvider.cpp +++ b/media/libnbaio/SourceAudioBufferProvider.cpp @@ -24,7 +24,7 @@ namespace android { SourceAudioBufferProvider::SourceAudioBufferProvider(const sp<NBAIO_Source>& source) : mSource(source), - // mFrameBitShiftFormat below + // mFrameSize below mAllocated(NULL), mSize(0), mOffset(0), mRemaining(0), mGetCount(0), mFramesReleased(0) { ALOG_ASSERT(source != 0); @@ -37,7 +37,7 @@ SourceAudioBufferProvider::SourceAudioBufferProvider(const sp<NBAIO_Source>& sou numCounterOffers = 0; index = source->negotiate(counterOffers, 1, NULL, numCounterOffers); ALOG_ASSERT(index == 0); - mFrameBitShift = Format_frameBitShift(source->format()); + mFrameSize = Format_frameSize(source->format()); } SourceAudioBufferProvider::~SourceAudioBufferProvider() @@ -54,14 +54,14 @@ status_t SourceAudioBufferProvider::getNextBuffer(Buffer *buffer, int64_t pts) if (mRemaining < buffer->frameCount) { buffer->frameCount = mRemaining; } - buffer->raw = (char *) mAllocated + (mOffset << mFrameBitShift); + buffer->raw = (char *) mAllocated + (mOffset * mFrameSize); mGetCount = buffer->frameCount; return OK; } // do we need to reallocate? if (buffer->frameCount > mSize) { free(mAllocated); - mAllocated = malloc(buffer->frameCount << mFrameBitShift); + mAllocated = malloc(buffer->frameCount * mFrameSize); mSize = buffer->frameCount; } // read from source @@ -84,7 +84,7 @@ status_t SourceAudioBufferProvider::getNextBuffer(Buffer *buffer, int64_t pts) void SourceAudioBufferProvider::releaseBuffer(Buffer *buffer) { ALOG_ASSERT((buffer != NULL) && - (buffer->raw == (char *) mAllocated + (mOffset << mFrameBitShift)) && + (buffer->raw == (char *) mAllocated + (mOffset * mFrameSize)) && (buffer->frameCount <= mGetCount) && (mGetCount <= mRemaining) && (mOffset + mRemaining <= mSize)); diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index 4450d62..9c48587 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -964,6 +964,8 @@ status_t ACodec::setComponentRole( "audio_decoder.aac", "audio_encoder.aac" }, { MEDIA_MIMETYPE_AUDIO_VORBIS, "audio_decoder.vorbis", "audio_encoder.vorbis" }, + { MEDIA_MIMETYPE_AUDIO_OPUS, + "audio_decoder.opus", "audio_encoder.opus" }, { MEDIA_MIMETYPE_AUDIO_G711_MLAW, "audio_decoder.g711mlaw", "audio_encoder.g711mlaw" }, { MEDIA_MIMETYPE_AUDIO_G711_ALAW, diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk index 0636dcc..714b5e0 100644 --- a/media/libstagefright/Android.mk +++ b/media/libstagefright/Android.mk @@ -81,6 +81,7 @@ LOCAL_SHARED_LIBRARIES := \ libicuuc \ liblog \ libmedia \ + libopus \ libsonivox \ libssl \ libstagefright_omx \ @@ -96,6 +97,7 @@ LOCAL_STATIC_LIBRARIES := \ libstagefright_color_conversion \ libstagefright_aacenc \ libstagefright_matroska \ + libstagefright_webm \ libstagefright_timedtext \ libvpx \ libwebm \ diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp index df7da0a..d0e0e8e 100644 --- a/media/libstagefright/AudioSource.cpp +++ b/media/libstagefright/AudioSource.cpp @@ -65,7 +65,7 @@ AudioSource::AudioSource( if (status == OK) { // make sure that the AudioRecord callback never returns more than the maximum // buffer size - int frameCount = kMaxBufferSize / sizeof(int16_t) / channelCount; + uint32_t frameCount = kMaxBufferSize / sizeof(int16_t) / channelCount; // make sure that the AudioRecord total buffer size is large enough size_t bufCount = 2; @@ -76,10 +76,10 @@ AudioSource::AudioSource( mRecord = new AudioRecord( inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT, audio_channel_in_mask_from_count(channelCount), - bufCount * frameCount, + (size_t) (bufCount * frameCount), AudioRecordCallbackFunction, this, - frameCount); + frameCount /*notificationFrames*/); mInitCheck = mRecord->initCheck(); } else { mInitCheck = status; diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp index e83ec62..4bad14b 100644 --- a/media/libstagefright/AwesomePlayer.cpp +++ b/media/libstagefright/AwesomePlayer.cpp @@ -2217,6 +2217,10 @@ status_t AwesomePlayer::finishSetDataSource_l() { mLock.unlock(); status_t err = mConnectingDataSource->connect(mUri, &mUriHeaders); + // force connection at this point, to avoid a race condition between getMIMEType and the + // caching datasource constructed below, which could result in multiple requests to the + // server, and/or failed connections. + String8 contentType = mConnectingDataSource->getMIMEType(); mLock.lock(); if (err != OK) { @@ -2247,8 +2251,6 @@ status_t AwesomePlayer::finishSetDataSource_l() { mConnectingDataSource.clear(); - String8 contentType = dataSource->getMIMEType(); - if (strncasecmp(contentType.string(), "audio/", 6)) { // We're not doing this for streams that appear to be audio-only // streams to ensure that even low bandwidth streams start diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp index f80772a..2a3fa04 100644 --- a/media/libstagefright/MPEG4Extractor.cpp +++ b/media/libstagefright/MPEG4Extractor.cpp @@ -913,6 +913,8 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('e', 'l', 's', 't'): { + *offset += chunk_size; + // See 14496-12 8.6.6 uint8_t version; if (mDataSource->readAt(data_offset, &version, 1) < 1) { @@ -975,12 +977,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->meta->setInt32(kKeyEncoderPadding, paddingsamples); } } - *offset += chunk_size; break; } case FOURCC('f', 'r', 'm', 'a'): { + *offset += chunk_size; + uint32_t original_fourcc; if (mDataSource->readAt(data_offset, &original_fourcc, 4) < 4) { return ERROR_IO; @@ -994,12 +997,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->meta->setInt32(kKeyChannelCount, num_channels); mLastTrack->meta->setInt32(kKeySampleRate, sample_rate); } - *offset += chunk_size; break; } case FOURCC('t', 'e', 'n', 'c'): { + *offset += chunk_size; + if (chunk_size < 32) { return ERROR_MALFORMED; } @@ -1044,23 +1048,25 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->meta->setInt32(kKeyCryptoMode, defaultAlgorithmId); mLastTrack->meta->setInt32(kKeyCryptoDefaultIVSize, defaultIVSize); mLastTrack->meta->setData(kKeyCryptoKey, 'tenc', defaultKeyId, 16); - *offset += chunk_size; break; } case FOURCC('t', 'k', 'h', 'd'): { + *offset += chunk_size; + status_t err; if ((err = parseTrackHeader(data_offset, chunk_data_size)) != OK) { return err; } - *offset += chunk_size; break; } case FOURCC('p', 's', 's', 'h'): { + *offset += chunk_size; + PsshInfo pssh; if (mDataSource->readAt(data_offset + 4, &pssh.uuid, 16) < 16) { @@ -1086,12 +1092,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { } mPssh.push_back(pssh); - *offset += chunk_size; break; } case FOURCC('m', 'd', 'h', 'd'): { + *offset += chunk_size; + if (chunk_data_size < 4) { return ERROR_MALFORMED; } @@ -1172,7 +1179,6 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->meta->setCString( kKeyMediaLanguage, lang_code); - *offset += chunk_size; break; } @@ -1339,11 +1345,12 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->sampleTable->setChunkOffsetParams( chunk_type, data_offset, chunk_data_size); + *offset += chunk_size; + if (err != OK) { return err; } - *offset += chunk_size; break; } @@ -1353,11 +1360,12 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->sampleTable->setSampleToChunkParams( data_offset, chunk_data_size); + *offset += chunk_size; + if (err != OK) { return err; } - *offset += chunk_size; break; } @@ -1368,6 +1376,8 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->sampleTable->setSampleSizeParams( chunk_type, data_offset, chunk_data_size); + *offset += chunk_size; + if (err != OK) { return err; } @@ -1408,7 +1418,6 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { } mLastTrack->meta->setInt32(kKeyMaxInputSize, max_size); } - *offset += chunk_size; // NOTE: setting another piece of metadata invalidates any pointers (such as the // mimetype) previously obtained, so don't cache them. @@ -1432,6 +1441,8 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('s', 't', 't', 's'): { + *offset += chunk_size; + status_t err = mLastTrack->sampleTable->setTimeToSampleParams( data_offset, chunk_data_size); @@ -1440,12 +1451,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return err; } - *offset += chunk_size; break; } case FOURCC('c', 't', 't', 's'): { + *offset += chunk_size; + status_t err = mLastTrack->sampleTable->setCompositionTimeToSampleParams( data_offset, chunk_data_size); @@ -1454,12 +1466,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return err; } - *offset += chunk_size; break; } case FOURCC('s', 't', 's', 's'): { + *offset += chunk_size; + status_t err = mLastTrack->sampleTable->setSyncSampleParams( data_offset, chunk_data_size); @@ -1468,13 +1481,14 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { return err; } - *offset += chunk_size; break; } // @xyz case FOURCC('\xA9', 'x', 'y', 'z'): { + *offset += chunk_size; + // Best case the total data length inside "@xyz" box // would be 8, for instance "@xyz" + "\x00\x04\x15\xc7" + "0+0/", // where "\x00\x04" is the text string length with value = 4, @@ -1503,12 +1517,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { buffer[location_length] = '\0'; mFileMetaData->setCString(kKeyLocation, buffer); - *offset += chunk_size; break; } case FOURCC('e', 's', 'd', 's'): { + *offset += chunk_size; + if (chunk_data_size < 4) { return ERROR_MALFORMED; } @@ -1546,12 +1561,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { } } - *offset += chunk_size; break; } case FOURCC('a', 'v', 'c', 'C'): { + *offset += chunk_size; + sp<ABuffer> buffer = new ABuffer(chunk_data_size); if (mDataSource->readAt( @@ -1562,12 +1578,12 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->meta->setData( kKeyAVCC, kTypeAVCC, buffer->data(), chunk_data_size); - *offset += chunk_size; break; } case FOURCC('d', '2', '6', '3'): { + *offset += chunk_size; /* * d263 contains a fixed 7 bytes part: * vendor - 4 bytes @@ -1593,7 +1609,6 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->meta->setData(kKeyD263, kTypeD263, buffer, chunk_data_size); - *offset += chunk_size; break; } @@ -1601,11 +1616,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { { uint8_t buffer[4]; if (chunk_data_size < (off64_t)sizeof(buffer)) { + *offset += chunk_size; return ERROR_MALFORMED; } if (mDataSource->readAt( data_offset, buffer, 4) < 4) { + *offset += chunk_size; return ERROR_IO; } @@ -1639,6 +1656,8 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('n', 'a', 'm', 'e'): case FOURCC('d', 'a', 't', 'a'): { + *offset += chunk_size; + if (mPath.size() == 6 && underMetaDataPath(mPath)) { status_t err = parseITunesMetaData(data_offset, chunk_data_size); @@ -1647,12 +1666,13 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { } } - *offset += chunk_size; break; } case FOURCC('m', 'v', 'h', 'd'): { + *offset += chunk_size; + if (chunk_data_size < 24) { return ERROR_MALFORMED; } @@ -1680,7 +1700,6 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mFileMetaData->setCString(kKeyDate, s.string()); - *offset += chunk_size; break; } @@ -1701,6 +1720,8 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('h', 'd', 'l', 'r'): { + *offset += chunk_size; + uint32_t buffer; if (mDataSource->readAt( data_offset + 8, &buffer, 4) < 4) { @@ -1715,7 +1736,6 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { mLastTrack->meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_TEXT_3GPP); } - *offset += chunk_size; break; } @@ -1740,6 +1760,8 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { delete[] buffer; buffer = NULL; + // advance read pointer so we don't end up reading this again + *offset += chunk_size; return ERROR_IO; } @@ -1754,6 +1776,8 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('c', 'o', 'v', 'r'): { + *offset += chunk_size; + if (mFileMetaData != NULL) { ALOGV("chunk_data_size = %lld and data_offset = %lld", chunk_data_size, data_offset); @@ -1768,7 +1792,6 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { buffer->data() + kSkipBytesOfDataBox, chunk_data_size - kSkipBytesOfDataBox); } - *offset += chunk_size; break; } @@ -1779,25 +1802,27 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('a', 'l', 'b', 'm'): case FOURCC('y', 'r', 'r', 'c'): { + *offset += chunk_size; + status_t err = parse3GPPMetaData(data_offset, chunk_data_size, depth); if (err != OK) { return err; } - *offset += chunk_size; break; } case FOURCC('I', 'D', '3', '2'): { + *offset += chunk_size; + if (chunk_data_size < 6) { return ERROR_MALFORMED; } parseID3v2MetaData(data_offset + 6); - *offset += chunk_size; break; } diff --git a/media/libstagefright/MediaDefs.cpp b/media/libstagefright/MediaDefs.cpp index 340cba7..c670bb4 100644 --- a/media/libstagefright/MediaDefs.cpp +++ b/media/libstagefright/MediaDefs.cpp @@ -36,6 +36,7 @@ const char *MEDIA_MIMETYPE_AUDIO_MPEG_LAYER_II = "audio/mpeg-L2"; const char *MEDIA_MIMETYPE_AUDIO_AAC = "audio/mp4a-latm"; const char *MEDIA_MIMETYPE_AUDIO_QCELP = "audio/qcelp"; const char *MEDIA_MIMETYPE_AUDIO_VORBIS = "audio/vorbis"; +const char *MEDIA_MIMETYPE_AUDIO_OPUS = "audio/opus"; const char *MEDIA_MIMETYPE_AUDIO_G711_ALAW = "audio/g711-alaw"; const char *MEDIA_MIMETYPE_AUDIO_G711_MLAW = "audio/g711-mlaw"; const char *MEDIA_MIMETYPE_AUDIO_RAW = "audio/raw"; diff --git a/media/libstagefright/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp index d87e910..90335ee 100644 --- a/media/libstagefright/MediaMuxer.cpp +++ b/media/libstagefright/MediaMuxer.cpp @@ -16,6 +16,9 @@ //#define LOG_NDEBUG 0 #define LOG_TAG "MediaMuxer" + +#include "webm/WebmWriter.h" + #include <utils/Log.h> #include <media/stagefright/MediaMuxer.h> @@ -36,19 +39,30 @@ namespace android { MediaMuxer::MediaMuxer(const char *path, OutputFormat format) - : mState(UNINITIALIZED) { + : mFormat(format), + mState(UNINITIALIZED) { if (format == OUTPUT_FORMAT_MPEG_4) { mWriter = new MPEG4Writer(path); + } else if (format == OUTPUT_FORMAT_WEBM) { + mWriter = new WebmWriter(path); + } + + if (mWriter != NULL) { mFileMeta = new MetaData; mState = INITIALIZED; } - } MediaMuxer::MediaMuxer(int fd, OutputFormat format) - : mState(UNINITIALIZED) { + : mFormat(format), + mState(UNINITIALIZED) { if (format == OUTPUT_FORMAT_MPEG_4) { mWriter = new MPEG4Writer(fd); + } else if (format == OUTPUT_FORMAT_WEBM) { + mWriter = new WebmWriter(fd); + } + + if (mWriter != NULL) { mFileMeta = new MetaData; mState = INITIALIZED; } @@ -109,8 +123,13 @@ status_t MediaMuxer::setLocation(int latitude, int longitude) { ALOGE("setLocation() must be called before start()."); return INVALID_OPERATION; } + if (mFormat != OUTPUT_FORMAT_MPEG_4) { + ALOGE("setLocation() is only supported for .mp4 output."); + return INVALID_OPERATION; + } + ALOGV("Setting location: latitude = %d, longitude = %d", latitude, longitude); - return mWriter->setGeoData(latitude, longitude); + return static_cast<MPEG4Writer*>(mWriter.get())->setGeoData(latitude, longitude); } status_t MediaMuxer::start() { diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp index 625922f..4d3b5bd 100644 --- a/media/libstagefright/OMXCodec.cpp +++ b/media/libstagefright/OMXCodec.cpp @@ -489,6 +489,13 @@ status_t OMXCodec::configureCodec(const sp<MetaData> &meta) { CHECK(meta->findData(kKeyVorbisBooks, &type, &data, &size)); addCodecSpecificData(data, size); + } else if (meta->findData(kKeyOpusHeader, &type, &data, &size)) { + addCodecSpecificData(data, size); + + CHECK(meta->findData(kKeyOpusCodecDelay, &type, &data, &size)); + addCodecSpecificData(data, size); + CHECK(meta->findData(kKeyOpusSeekPreRoll, &type, &data, &size)); + addCodecSpecificData(data, size); } } @@ -1387,6 +1394,8 @@ void OMXCodec::setComponentRole( "audio_decoder.aac", "audio_encoder.aac" }, { MEDIA_MIMETYPE_AUDIO_VORBIS, "audio_decoder.vorbis", "audio_encoder.vorbis" }, + { MEDIA_MIMETYPE_AUDIO_OPUS, + "audio_decoder.opus", "audio_encoder.opus" }, { MEDIA_MIMETYPE_AUDIO_G711_MLAW, "audio_decoder.g711mlaw", "audio_encoder.g711mlaw" }, { MEDIA_MIMETYPE_AUDIO_G711_ALAW, @@ -4125,6 +4134,7 @@ static const char *audioCodingTypeString(OMX_AUDIO_CODINGTYPE type) { "OMX_AUDIO_CodingMP3", "OMX_AUDIO_CodingSBC", "OMX_AUDIO_CodingVORBIS", + "OMX_AUDIO_CodingOPUS", "OMX_AUDIO_CodingWMA", "OMX_AUDIO_CodingRA", "OMX_AUDIO_CodingMIDI", diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp index 451e907..4ff805f 100644 --- a/media/libstagefright/Utils.cpp +++ b/media/libstagefright/Utils.cpp @@ -251,6 +251,13 @@ status_t convertMetaDataToMessage( buffer->meta()->setInt32("csd", true); buffer->meta()->setInt64("timeUs", 0); msg->setBuffer("csd-1", buffer); + } else if (meta->findData(kKeyOpusHeader, &type, &data, &size)) { + sp<ABuffer> buffer = new ABuffer(size); + memcpy(buffer->data(), data, size); + + buffer->meta()->setInt32("csd", true); + buffer->meta()->setInt64("timeUs", 0); + msg->setBuffer("csd-0", buffer); } *format = msg; @@ -528,6 +535,7 @@ static const struct mime_conv_t mimeLookup[] = { { MEDIA_MIMETYPE_AUDIO_AMR_WB, AUDIO_FORMAT_AMR_WB }, { MEDIA_MIMETYPE_AUDIO_AAC, AUDIO_FORMAT_AAC }, { MEDIA_MIMETYPE_AUDIO_VORBIS, AUDIO_FORMAT_VORBIS }, + { MEDIA_MIMETYPE_AUDIO_OPUS, AUDIO_FORMAT_OPUS}, { 0, AUDIO_FORMAT_INVALID } }; diff --git a/media/libstagefright/codecs/common/Config.mk b/media/libstagefright/codecs/common/Config.mk index a6d4286..a843cef 100644 --- a/media/libstagefright/codecs/common/Config.mk +++ b/media/libstagefright/codecs/common/Config.mk @@ -14,8 +14,10 @@ VOTT := pc endif # Do we also need to check on ARCH_ARM_HAVE_ARMV7A? - probably not -ifeq ($(ARCH_ARM_HAVE_NEON),true) -VOTT := v7 +ifeq ($(TARGET_ARCH),arm) + ifeq ($(ARCH_ARM_HAVE_NEON),true) + VOTT := v7 + endif endif VOTEST := 0 diff --git a/media/libstagefright/codecs/on2/h264dec/Android.mk b/media/libstagefright/codecs/on2/h264dec/Android.mk index 655b2ab..bf03ad9 100644 --- a/media/libstagefright/codecs/on2/h264dec/Android.mk +++ b/media/libstagefright/codecs/on2/h264dec/Android.mk @@ -84,8 +84,8 @@ MY_OMXDL_ASM_SRC := \ ./omxdl/arm_neon/vc/m4p10/src_gcc/omxVCM4P10_DequantTransformResidualFromPairAndAdd_s.S \ ./omxdl/arm_neon/vc/m4p10/src_gcc/omxVCM4P10_TransformDequantChromaDCFromPair_s.S \ - -ifeq ($(ARCH_ARM_HAVE_NEON),true) +ifeq ($(TARGET_ARCH),arm) + ifeq ($(ARCH_ARM_HAVE_NEON),true) LOCAL_ARM_NEON := true # LOCAL_CFLAGS := -std=c99 -D._NEON -D._OMXDL LOCAL_CFLAGS := -DH264DEC_NEON -DH264DEC_OMXDL @@ -94,6 +94,7 @@ ifeq ($(ARCH_ARM_HAVE_NEON),true) LOCAL_C_INCLUDES += $(LOCAL_PATH)/./omxdl/arm_neon/api \ $(LOCAL_PATH)/./omxdl/arm_neon/vc/api \ $(LOCAL_PATH)/./omxdl/arm_neon/vc/m4p10/api + endif endif LOCAL_SHARED_LIBRARIES := \ diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c index 15eabfb..52c85e5 100755 --- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c +++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c @@ -1110,7 +1110,7 @@ void Intra16x16PlanePrediction(u8 *data, u8 *above, u8 *left) /* Variables */ - u32 i, j; + i32 i, j; i32 a, b, c; i32 tmp; @@ -1123,20 +1123,20 @@ void Intra16x16PlanePrediction(u8 *data, u8 *above, u8 *left) a = 16 * (above[15] + left[15]); for (i = 0, b = 0; i < 8; i++) - b += ((i32)i + 1) * (above[8+i] - above[6-i]); + b += (i + 1) * (above[8+i] - above[6-i]); b = (5 * b + 32) >> 6; for (i = 0, c = 0; i < 7; i++) - c += ((i32)i + 1) * (left[8+i] - left[6-i]); + c += (i + 1) * (left[8+i] - left[6-i]); /* p[-1,-1] has to be accessed through above pointer */ - c += ((i32)i + 1) * (left[8+i] - above[-1]); + c += (i + 1) * (left[8+i] - above[-1]); c = (5 * c + 32) >> 6; for (i = 0; i < 16; i++) { for (j = 0; j < 16; j++) { - tmp = (a + b * ((i32)j - 7) + c * ((i32)i - 7) + 16) >> 5; + tmp = (a + b * (j - 7) + c * (i - 7) + 16) >> 5; data[i*16+j] = (u8)CLIP1(tmp); } } diff --git a/media/libstagefright/codecs/opus/Android.mk b/media/libstagefright/codecs/opus/Android.mk new file mode 100644 index 0000000..365b179 --- /dev/null +++ b/media/libstagefright/codecs/opus/Android.mk @@ -0,0 +1,4 @@ +LOCAL_PATH:= $(call my-dir) +include $(CLEAR_VARS) + +include $(call all-makefiles-under,$(LOCAL_PATH))
\ No newline at end of file diff --git a/media/libstagefright/codecs/opus/dec/Android.mk b/media/libstagefright/codecs/opus/dec/Android.mk new file mode 100644 index 0000000..2379c5f --- /dev/null +++ b/media/libstagefright/codecs/opus/dec/Android.mk @@ -0,0 +1,19 @@ +LOCAL_PATH:= $(call my-dir) +include $(CLEAR_VARS) + +LOCAL_SRC_FILES := \ + SoftOpus.cpp + +LOCAL_C_INCLUDES := \ + external/libopus/include \ + frameworks/av/media/libstagefright/include \ + frameworks/native/include/media/openmax \ + +LOCAL_SHARED_LIBRARIES := \ + libopus libstagefright libstagefright_omx \ + libstagefright_foundation libutils liblog + +LOCAL_MODULE := libstagefright_soft_opusdec +LOCAL_MODULE_TAGS := optional + +include $(BUILD_SHARED_LIBRARY)
\ No newline at end of file diff --git a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp new file mode 100644 index 0000000..b8084ae --- /dev/null +++ b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp @@ -0,0 +1,540 @@ +/* + * Copyright (C) 2014 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 "SoftOpus" +#include <utils/Log.h> + +#include "SoftOpus.h" +#include <OMX_AudioExt.h> +#include <OMX_IndexExt.h> + +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/MediaDefs.h> + +extern "C" { + #include <opus.h> + #include <opus_multistream.h> +} + +namespace android { + +static const int kRate = 48000; + +template<class T> +static void InitOMXParams(T *params) { + params->nSize = sizeof(T); + params->nVersion.s.nVersionMajor = 1; + params->nVersion.s.nVersionMinor = 0; + params->nVersion.s.nRevision = 0; + params->nVersion.s.nStep = 0; +} + +SoftOpus::SoftOpus( + const char *name, + const OMX_CALLBACKTYPE *callbacks, + OMX_PTR appData, + OMX_COMPONENTTYPE **component) + : SimpleSoftOMXComponent(name, callbacks, appData, component), + mInputBufferCount(0), + mDecoder(NULL), + mHeader(NULL), + mCodecDelay(0), + mSeekPreRoll(0), + mAnchorTimeUs(0), + mNumFramesOutput(0), + mOutputPortSettingsChange(NONE) { + initPorts(); + CHECK_EQ(initDecoder(), (status_t)OK); +} + +SoftOpus::~SoftOpus() { + if (mDecoder != NULL) { + opus_multistream_decoder_destroy(mDecoder); + mDecoder = NULL; + } + if (mHeader != NULL) { + delete mHeader; + mHeader = NULL; + } +} + +void SoftOpus::initPorts() { + OMX_PARAM_PORTDEFINITIONTYPE def; + InitOMXParams(&def); + + def.nPortIndex = 0; + def.eDir = OMX_DirInput; + def.nBufferCountMin = kNumBuffers; + def.nBufferCountActual = def.nBufferCountMin; + def.nBufferSize = 960 * 6; + def.bEnabled = OMX_TRUE; + def.bPopulated = OMX_FALSE; + def.eDomain = OMX_PortDomainAudio; + def.bBuffersContiguous = OMX_FALSE; + def.nBufferAlignment = 1; + + def.format.audio.cMIMEType = + const_cast<char *>(MEDIA_MIMETYPE_AUDIO_OPUS); + + def.format.audio.pNativeRender = NULL; + def.format.audio.bFlagErrorConcealment = OMX_FALSE; + def.format.audio.eEncoding = + (OMX_AUDIO_CODINGTYPE)OMX_AUDIO_CodingAndroidOPUS; + + addPort(def); + + def.nPortIndex = 1; + def.eDir = OMX_DirOutput; + def.nBufferCountMin = kNumBuffers; + def.nBufferCountActual = def.nBufferCountMin; + def.nBufferSize = kMaxNumSamplesPerBuffer * sizeof(int16_t); + def.bEnabled = OMX_TRUE; + def.bPopulated = OMX_FALSE; + def.eDomain = OMX_PortDomainAudio; + def.bBuffersContiguous = OMX_FALSE; + def.nBufferAlignment = 2; + + def.format.audio.cMIMEType = const_cast<char *>("audio/raw"); + def.format.audio.pNativeRender = NULL; + def.format.audio.bFlagErrorConcealment = OMX_FALSE; + def.format.audio.eEncoding = OMX_AUDIO_CodingPCM; + + addPort(def); +} + +status_t SoftOpus::initDecoder() { + return OK; +} + +OMX_ERRORTYPE SoftOpus::internalGetParameter( + OMX_INDEXTYPE index, OMX_PTR params) { + switch ((int)index) { + case OMX_IndexParamAudioAndroidOpus: + { + OMX_AUDIO_PARAM_ANDROID_OPUSTYPE *opusParams = + (OMX_AUDIO_PARAM_ANDROID_OPUSTYPE *)params; + + if (opusParams->nPortIndex != 0) { + return OMX_ErrorUndefined; + } + + opusParams->nAudioBandWidth = 0; + opusParams->nSampleRate = kRate; + opusParams->nBitRate = 0; + + if (!isConfigured()) { + opusParams->nChannels = 1; + } else { + opusParams->nChannels = mHeader->channels; + } + + return OMX_ErrorNone; + } + + case OMX_IndexParamAudioPcm: + { + OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams = + (OMX_AUDIO_PARAM_PCMMODETYPE *)params; + + if (pcmParams->nPortIndex != 1) { + return OMX_ErrorUndefined; + } + + pcmParams->eNumData = OMX_NumericalDataSigned; + pcmParams->eEndian = OMX_EndianBig; + pcmParams->bInterleaved = OMX_TRUE; + pcmParams->nBitPerSample = 16; + pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear; + pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF; + pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF; + pcmParams->nSamplingRate = kRate; + + if (!isConfigured()) { + pcmParams->nChannels = 1; + } else { + pcmParams->nChannels = mHeader->channels; + } + + return OMX_ErrorNone; + } + + default: + return SimpleSoftOMXComponent::internalGetParameter(index, params); + } +} + +OMX_ERRORTYPE SoftOpus::internalSetParameter( + OMX_INDEXTYPE index, const OMX_PTR params) { + switch ((int)index) { + case OMX_IndexParamStandardComponentRole: + { + const OMX_PARAM_COMPONENTROLETYPE *roleParams = + (const OMX_PARAM_COMPONENTROLETYPE *)params; + + if (strncmp((const char *)roleParams->cRole, + "audio_decoder.opus", + OMX_MAX_STRINGNAME_SIZE - 1)) { + return OMX_ErrorUndefined; + } + + return OMX_ErrorNone; + } + + case OMX_IndexParamAudioAndroidOpus: + { + const OMX_AUDIO_PARAM_ANDROID_OPUSTYPE *opusParams = + (const OMX_AUDIO_PARAM_ANDROID_OPUSTYPE *)params; + + if (opusParams->nPortIndex != 0) { + return OMX_ErrorUndefined; + } + + return OMX_ErrorNone; + } + + default: + return SimpleSoftOMXComponent::internalSetParameter(index, params); + } +} + +bool SoftOpus::isConfigured() const { + return mInputBufferCount >= 1; +} + +static uint16_t ReadLE16(const uint8_t *data, size_t data_size, + uint32_t read_offset) { + if (read_offset + 1 > data_size) + return 0; + uint16_t val; + val = data[read_offset]; + val |= data[read_offset + 1] << 8; + return val; +} + +// Opus uses Vorbis channel mapping, and Vorbis channel mapping specifies +// mappings for up to 8 channels. This information is part of the Vorbis I +// Specification: +// http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html +static const int kMaxChannels = 8; + +// Maximum packet size used in Xiph's opusdec. +static const int kMaxOpusOutputPacketSizeSamples = 960 * 6; + +// Default audio output channel layout. Used to initialize |stream_map| in +// OpusHeader, and passed to opus_multistream_decoder_create() when the header +// does not contain mapping information. The values are valid only for mono and +// stereo output: Opus streams with more than 2 channels require a stream map. +static const int kMaxChannelsWithDefaultLayout = 2; +static const uint8_t kDefaultOpusChannelLayout[kMaxChannelsWithDefaultLayout] = { 0, 1 }; + +// Parses Opus Header. Header spec: http://wiki.xiph.org/OggOpus#ID_Header +static bool ParseOpusHeader(const uint8_t *data, size_t data_size, + OpusHeader* header) { + // Size of the Opus header excluding optional mapping information. + const size_t kOpusHeaderSize = 19; + + // Offset to the channel count byte in the Opus header. + const size_t kOpusHeaderChannelsOffset = 9; + + // Offset to the pre-skip value in the Opus header. + const size_t kOpusHeaderSkipSamplesOffset = 10; + + // Offset to the gain value in the Opus header. + const size_t kOpusHeaderGainOffset = 16; + + // Offset to the channel mapping byte in the Opus header. + const size_t kOpusHeaderChannelMappingOffset = 18; + + // Opus Header contains a stream map. The mapping values are in the header + // beyond the always present |kOpusHeaderSize| bytes of data. The mapping + // data contains stream count, coupling information, and per channel mapping + // values: + // - Byte 0: Number of streams. + // - Byte 1: Number coupled. + // - Byte 2: Starting at byte 2 are |header->channels| uint8 mapping + // values. + const size_t kOpusHeaderNumStreamsOffset = kOpusHeaderSize; + const size_t kOpusHeaderNumCoupledOffset = kOpusHeaderNumStreamsOffset + 1; + const size_t kOpusHeaderStreamMapOffset = kOpusHeaderNumStreamsOffset + 2; + + if (data_size < kOpusHeaderSize) { + ALOGV("Header size is too small."); + return false; + } + header->channels = *(data + kOpusHeaderChannelsOffset); + + if (header->channels <= 0 || header->channels > kMaxChannels) { + ALOGV("Invalid Header, wrong channel count: %d", header->channels); + return false; + } + header->skip_samples = ReadLE16(data, data_size, + kOpusHeaderSkipSamplesOffset); + header->gain_db = static_cast<int16_t>( + ReadLE16(data, data_size, + kOpusHeaderGainOffset)); + header->channel_mapping = *(data + kOpusHeaderChannelMappingOffset); + if (!header->channel_mapping) { + if (header->channels > kMaxChannelsWithDefaultLayout) { + ALOGV("Invalid Header, missing stream map."); + return false; + } + header->num_streams = 1; + header->num_coupled = header->channels > 1; + header->stream_map[0] = 0; + header->stream_map[1] = 1; + return true; + } + if (data_size < kOpusHeaderStreamMapOffset + header->channels) { + ALOGV("Invalid stream map; insufficient data for current channel " + "count: %d", header->channels); + return false; + } + header->num_streams = *(data + kOpusHeaderNumStreamsOffset); + header->num_coupled = *(data + kOpusHeaderNumCoupledOffset); + if (header->num_streams + header->num_coupled != header->channels) { + ALOGV("Inconsistent channel mapping."); + return false; + } + for (int i = 0; i < header->channels; ++i) + header->stream_map[i] = *(data + kOpusHeaderStreamMapOffset + i); + return true; +} + +// Convert nanoseconds to number of samples. +static uint64_t ns_to_samples(uint64_t ns, int kRate) { + return static_cast<double>(ns) * kRate / 1000000000; +} + +void SoftOpus::onQueueFilled(OMX_U32 portIndex) { + List<BufferInfo *> &inQueue = getPortQueue(0); + List<BufferInfo *> &outQueue = getPortQueue(1); + + if (mOutputPortSettingsChange != NONE) { + return; + } + + if (portIndex == 0 && mInputBufferCount < 3) { + BufferInfo *info = *inQueue.begin(); + OMX_BUFFERHEADERTYPE *header = info->mHeader; + + const uint8_t *data = header->pBuffer + header->nOffset; + size_t size = header->nFilledLen; + + if (mInputBufferCount == 0) { + CHECK(mHeader == NULL); + mHeader = new OpusHeader(); + memset(mHeader, 0, sizeof(*mHeader)); + if (!ParseOpusHeader(data, size, mHeader)) { + ALOGV("Parsing Opus Header failed."); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + return; + } + + uint8_t channel_mapping[kMaxChannels] = {0}; + memcpy(&channel_mapping, + kDefaultOpusChannelLayout, + kMaxChannelsWithDefaultLayout); + + int status = OPUS_INVALID_STATE; + mDecoder = opus_multistream_decoder_create(kRate, + mHeader->channels, + mHeader->num_streams, + mHeader->num_coupled, + channel_mapping, + &status); + if (!mDecoder || status != OPUS_OK) { + ALOGV("opus_multistream_decoder_create failed status=%s", + opus_strerror(status)); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + return; + } + status = + opus_multistream_decoder_ctl(mDecoder, + OPUS_SET_GAIN(mHeader->gain_db)); + if (status != OPUS_OK) { + ALOGV("Failed to set OPUS header gain; status=%s", + opus_strerror(status)); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + return; + } + } else if (mInputBufferCount == 1) { + mCodecDelay = ns_to_samples( + *(reinterpret_cast<int64_t*>(header->pBuffer + + header->nOffset)), + kRate); + mSamplesToDiscard = mCodecDelay; + } else { + mSeekPreRoll = ns_to_samples( + *(reinterpret_cast<int64_t*>(header->pBuffer + + header->nOffset)), + kRate); + notify(OMX_EventPortSettingsChanged, 1, 0, NULL); + mOutputPortSettingsChange = AWAITING_DISABLED; + } + + inQueue.erase(inQueue.begin()); + info->mOwnedByUs = false; + notifyEmptyBufferDone(header); + ++mInputBufferCount; + return; + } + + while (!inQueue.empty() && !outQueue.empty()) { + BufferInfo *inInfo = *inQueue.begin(); + OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + + BufferInfo *outInfo = *outQueue.begin(); + OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; + + if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + inQueue.erase(inQueue.begin()); + inInfo->mOwnedByUs = false; + notifyEmptyBufferDone(inHeader); + + outHeader->nFilledLen = 0; + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + + outQueue.erase(outQueue.begin()); + outInfo->mOwnedByUs = false; + notifyFillBufferDone(outHeader); + return; + } + + if (inHeader->nOffset == 0) { + mAnchorTimeUs = inHeader->nTimeStamp; + mNumFramesOutput = 0; + } + + // When seeking to zero, |mCodecDelay| samples has to be discarded + // instead of |mSeekPreRoll| samples (as we would when seeking to any + // other timestamp). + if (inHeader->nTimeStamp == 0) { + mSamplesToDiscard = mCodecDelay; + } + + const uint8_t *data = inHeader->pBuffer + inHeader->nOffset; + const uint32_t size = inHeader->nFilledLen; + + int numFrames = opus_multistream_decode(mDecoder, + data, + size, + (int16_t *)outHeader->pBuffer, + kMaxOpusOutputPacketSizeSamples, + 0); + if (numFrames < 0) { + ALOGE("opus_multistream_decode returned %d", numFrames); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + return; + } + + outHeader->nOffset = 0; + if (mSamplesToDiscard > 0) { + if (mSamplesToDiscard > numFrames) { + mSamplesToDiscard -= numFrames; + numFrames = 0; + } else { + numFrames -= mSamplesToDiscard; + outHeader->nOffset = mSamplesToDiscard * sizeof(int16_t) * + mHeader->channels; + mSamplesToDiscard = 0; + } + } + + outHeader->nFilledLen = numFrames * sizeof(int16_t) * mHeader->channels; + outHeader->nFlags = 0; + + outHeader->nTimeStamp = mAnchorTimeUs + + (mNumFramesOutput * 1000000ll) / + kRate; + + mNumFramesOutput += numFrames; + + inInfo->mOwnedByUs = false; + inQueue.erase(inQueue.begin()); + inInfo = NULL; + notifyEmptyBufferDone(inHeader); + inHeader = NULL; + + outInfo->mOwnedByUs = false; + outQueue.erase(outQueue.begin()); + outInfo = NULL; + notifyFillBufferDone(outHeader); + outHeader = NULL; + + ++mInputBufferCount; + } +} + +void SoftOpus::onPortFlushCompleted(OMX_U32 portIndex) { + if (portIndex == 0 && mDecoder != NULL) { + // Make sure that the next buffer output does not still + // depend on fragments from the last one decoded. + mNumFramesOutput = 0; + opus_multistream_decoder_ctl(mDecoder, OPUS_RESET_STATE); + mAnchorTimeUs = 0; + mSamplesToDiscard = mSeekPreRoll; + } +} + +void SoftOpus::onReset() { + mInputBufferCount = 0; + mNumFramesOutput = 0; + if (mDecoder != NULL) { + opus_multistream_decoder_destroy(mDecoder); + mDecoder = NULL; + } + if (mHeader != NULL) { + delete mHeader; + mHeader = NULL; + } + + mOutputPortSettingsChange = NONE; +} + +void SoftOpus::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { + if (portIndex != 1) { + return; + } + + switch (mOutputPortSettingsChange) { + case NONE: + break; + + case AWAITING_DISABLED: + { + CHECK(!enabled); + mOutputPortSettingsChange = AWAITING_ENABLED; + break; + } + + default: + { + CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED); + CHECK(enabled); + mOutputPortSettingsChange = NONE; + break; + } + } +} + +} // namespace android + +android::SoftOMXComponent *createSoftOMXComponent( + const char *name, const OMX_CALLBACKTYPE *callbacks, + OMX_PTR appData, OMX_COMPONENTTYPE **component) { + return new android::SoftOpus(name, callbacks, appData, component); +} diff --git a/media/libstagefright/codecs/opus/dec/SoftOpus.h b/media/libstagefright/codecs/opus/dec/SoftOpus.h new file mode 100644 index 0000000..97f6561 --- /dev/null +++ b/media/libstagefright/codecs/opus/dec/SoftOpus.h @@ -0,0 +1,94 @@ +/* + * Copyright (C) 2014 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. + */ + +/* + * The Opus specification is part of IETF RFC 6716: + * http://tools.ietf.org/html/rfc6716 + */ + +#ifndef SOFT_OPUS_H_ + +#define SOFT_OPUS_H_ + +#include "SimpleSoftOMXComponent.h" + +struct OpusMSDecoder; + +namespace android { + +struct OpusHeader { + int channels; + int skip_samples; + int channel_mapping; + int num_streams; + int num_coupled; + int16_t gain_db; + uint8_t stream_map[8]; +}; + +struct SoftOpus : public SimpleSoftOMXComponent { + SoftOpus(const char *name, + const OMX_CALLBACKTYPE *callbacks, + OMX_PTR appData, + OMX_COMPONENTTYPE **component); + +protected: + virtual ~SoftOpus(); + + virtual OMX_ERRORTYPE internalGetParameter( + OMX_INDEXTYPE index, OMX_PTR params); + + virtual OMX_ERRORTYPE internalSetParameter( + OMX_INDEXTYPE index, const OMX_PTR params); + + virtual void onQueueFilled(OMX_U32 portIndex); + virtual void onPortFlushCompleted(OMX_U32 portIndex); + virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled); + virtual void onReset(); + +private: + enum { + kNumBuffers = 4, + kMaxNumSamplesPerBuffer = 960 * 6 + }; + + size_t mInputBufferCount; + + OpusMSDecoder *mDecoder; + OpusHeader *mHeader; + + int64_t mCodecDelay; + int64_t mSeekPreRoll; + int64_t mSamplesToDiscard; + int64_t mAnchorTimeUs; + int64_t mNumFramesOutput; + + enum { + NONE, + AWAITING_DISABLED, + AWAITING_ENABLED + } mOutputPortSettingsChange; + + void initPorts(); + status_t initDecoder(); + bool isConfigured() const; + + DISALLOW_EVIL_CONSTRUCTORS(SoftOpus); +}; + +} // namespace android + +#endif // SOFT_OPUS_H_ diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index bc26de1..ceb3c8f 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -40,6 +40,8 @@ #include <media/stagefright/MetaData.h> #include <media/stagefright/Utils.h> +#include <utils/Mutex.h> + #include <ctype.h> #include <openssl/aes.h> #include <openssl/md5.h> @@ -56,11 +58,16 @@ LiveSession::LiveSession( mHTTPDataSource(new MediaHTTP(mHTTPService->makeHTTPConnection())), mPrevBandwidthIndex(-1), mStreamMask(0), + mNewStreamMask(0), + mSwapMask(0), mCheckBandwidthGeneration(0), + mSwitchGeneration(0), mLastDequeuedTimeUs(0ll), mRealTimeBaseUs(0ll), mReconfigurationInProgress(false), - mDisconnectReplyID(0) { + mSwitchInProgress(false), + mDisconnectReplyID(0), + mSeekReplyID(0) { mStreams[kAudioIndex] = StreamItem("audio"); mStreams[kVideoIndex] = StreamItem("video"); @@ -68,16 +75,37 @@ LiveSession::LiveSession( for (size_t i = 0; i < kMaxStreams; ++i) { mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */)); + mPacketSources2.add(indexToType(i), new AnotherPacketSource(NULL /* meta */)); } } LiveSession::~LiveSession() { } +sp<ABuffer> LiveSession::createFormatChangeBuffer(bool swap) { + ABuffer *discontinuity = new ABuffer(0); + discontinuity->meta()->setInt32("discontinuity", ATSParser::DISCONTINUITY_FORMATCHANGE); + discontinuity->meta()->setInt32("swapPacketSource", swap); + discontinuity->meta()->setInt32("switchGeneration", mSwitchGeneration); + discontinuity->meta()->setInt64("timeUs", -1); + return discontinuity; +} + +void LiveSession::swapPacketSource(StreamType stream) { + sp<AnotherPacketSource> &aps = mPacketSources.editValueFor(stream); + sp<AnotherPacketSource> &aps2 = mPacketSources2.editValueFor(stream); + sp<AnotherPacketSource> tmp = aps; + aps = aps2; + aps2 = tmp; + aps2->clear(); +} + status_t LiveSession::dequeueAccessUnit( StreamType stream, sp<ABuffer> *accessUnit) { if (!(mStreamMask & stream)) { - return UNKNOWN_ERROR; + // return -EWOULDBLOCK to avoid halting the decoder + // when switching between audio/video and audio only. + return -EWOULDBLOCK; } sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream); @@ -117,6 +145,25 @@ status_t LiveSession::dequeueAccessUnit( streamStr, type, extra == NULL ? "NULL" : extra->debugString().c_str()); + + int32_t swap; + if (type == ATSParser::DISCONTINUITY_FORMATCHANGE + && (*accessUnit)->meta()->findInt32("swapPacketSource", &swap) + && swap) { + + int32_t switchGeneration; + CHECK((*accessUnit)->meta()->findInt32("switchGeneration", &switchGeneration)); + { + Mutex::Autolock lock(mSwapMutex); + if (switchGeneration == mSwitchGeneration) { + swapPacketSource(stream); + sp<AMessage> msg = new AMessage(kWhatSwapped, id()); + msg->setInt32("stream", stream); + msg->setInt32("switchGeneration", switchGeneration); + msg->post(); + } + } + } } else if (err == OK) { if (stream == STREAMTYPE_AUDIO || stream == STREAMTYPE_VIDEO) { int64_t timeUs; @@ -138,6 +185,7 @@ status_t LiveSession::dequeueAccessUnit( } status_t LiveSession::getStreamFormat(StreamType stream, sp<AMessage> *format) { + // No swapPacketSource race condition; called from the same thread as dequeueAccessUnit. if (!(mStreamMask & stream)) { return UNKNOWN_ERROR; } @@ -183,6 +231,10 @@ status_t LiveSession::seekTo(int64_t timeUs) { sp<AMessage> response; status_t err = msg->postAndAwaitResponse(&response); + uint32_t replyID; + CHECK(response == mSeekReply && 0 != mSeekReplyID); + mSeekReply.clear(); + mSeekReplyID = 0; return err; } @@ -208,15 +260,12 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { case kWhatSeek: { - uint32_t replyID; - CHECK(msg->senderAwaitsResponse(&replyID)); + CHECK(msg->senderAwaitsResponse(&mSeekReplyID)); status_t err = onSeek(msg); - sp<AMessage> response = new AMessage; - response->setInt32("err", err); - - response->postReply(replyID); + mSeekReply = new AMessage; + mSeekReply->setInt32("err", err); break; } @@ -234,13 +283,23 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { if (what == PlaylistFetcher::kWhatStopped) { AString uri; CHECK(msg->findString("uri", &uri)); - mFetcherInfos.removeItem(uri); + if (mFetcherInfos.removeItem(uri) < 0) { + // ignore duplicated kWhatStopped messages. + break; + } + + tryToFinishBandwidthSwitch(); } if (mContinuation != NULL) { CHECK_GT(mContinuationCounter, 0); if (--mContinuationCounter == 0) { mContinuation->post(); + + if (mSeekReplyID != 0) { + CHECK(mSeekReply != NULL); + mSeekReply->postReply(mSeekReplyID); + } } } break; @@ -270,6 +329,8 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { postPrepared(err); } + cancelBandwidthSwitch(); + mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(err); mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(err); @@ -308,6 +369,27 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { break; } + case PlaylistFetcher::kWhatStartedAt: + { + int32_t switchGeneration; + CHECK(msg->findInt32("switchGeneration", &switchGeneration)); + + if (switchGeneration != mSwitchGeneration) { + break; + } + + // 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. + for (size_t i = 0; i < mFetcherInfos.size(); i++) { + const FetcherInfo info = mFetcherInfos.valueAt(i); + if (info.mToBeRemoved) { + info.mFetcher->resumeUntilAsync(msg); + } + } + break; + } + default: TRESPASS(); } @@ -352,6 +434,11 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { break; } + case kWhatSwapped: + { + onSwapped(msg); + break; + } default: TRESPASS(); break; @@ -462,6 +549,10 @@ void LiveSession::finishDisconnect() { // during disconnection either. cancelCheckBandwidthEvent(); + // Protect mPacketSources from a swapPacketSource race condition through disconnect. + // (finishDisconnect, onFinishDisconnect2) + cancelBandwidthSwitch(); + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { mFetcherInfos.valueAt(i).mFetcher->stopAsync(); } @@ -501,11 +592,13 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) { sp<AMessage> notify = new AMessage(kWhatFetcherNotify, id()); notify->setString("uri", uri); + notify->setInt32("switchGeneration", mSwitchGeneration); FetcherInfo info; info.mFetcher = new PlaylistFetcher(notify, this, uri); info.mDurationUs = -1ll; info.mIsPrepared = false; + info.mToBeRemoved = false; looper()->registerHandler(info.mFetcher); mFetcherInfos.add(uri, info); @@ -845,8 +938,25 @@ status_t LiveSession::selectTrack(size_t index, bool select) { return err; } +bool LiveSession::canSwitchUp() { + // Allow upwards bandwidth switch when a stream has buffered at least 10 seconds. + status_t err = OK; + for (size_t i = 0; i < mPacketSources.size(); ++i) { + sp<AnotherPacketSource> source = mPacketSources.valueAt(i); + int64_t dur = source->getBufferedDurationUs(&err); + if (err == OK && dur > 10000000) { + return true; + } + } + return false; +} + void LiveSession::changeConfiguration( int64_t timeUs, size_t bandwidthIndex, bool pickTrack) { + // Protect mPacketSources from a swapPacketSource race condition through reconfiguration. + // (changeConfiguration, onChangeConfiguration2, onChangeConfiguration3). + cancelBandwidthSwitch(); + CHECK(!mReconfigurationInProgress); mReconfigurationInProgress = true; @@ -862,7 +972,8 @@ void LiveSession::changeConfiguration( CHECK_LT(bandwidthIndex, mBandwidthItems.size()); const BandwidthItem &item = mBandwidthItems.itemAt(bandwidthIndex); - uint32_t streamMask = 0; + uint32_t streamMask = 0; // streams that should be fetched by the new fetcher + uint32_t resumeMask = 0; // streams that should be fetched by the original fetcher AString URIs[kMaxStreams]; for (size_t i = 0; i < kMaxStreams; ++i) { @@ -880,9 +991,14 @@ void LiveSession::changeConfiguration( // If we're seeking all current fetchers are discarded. if (timeUs < 0ll) { + // delay fetcher removal + discardFetcher = false; + for (size_t j = 0; j < kMaxStreams; ++j) { - if ((streamMask & indexToType(j)) && uri == URIs[j]) { - discardFetcher = false; + StreamType type = indexToType(j); + if ((streamMask & type) && uri == URIs[j]) { + resumeMask |= type; + streamMask &= ~type; } } } @@ -894,8 +1010,15 @@ void LiveSession::changeConfiguration( } } - sp<AMessage> msg = new AMessage(kWhatChangeConfiguration2, id()); + sp<AMessage> msg; + if (timeUs < 0ll) { + // skip onChangeConfiguration2 (decoder destruction) if switching. + msg = new AMessage(kWhatChangeConfiguration3, id()); + } else { + msg = new AMessage(kWhatChangeConfiguration2, id()); + } msg->setInt32("streamMask", streamMask); + msg->setInt32("resumeMask", resumeMask); msg->setInt64("timeUs", timeUs); for (size_t i = 0; i < kMaxStreams; ++i) { if (streamMask & indexToType(i)) { @@ -912,6 +1035,11 @@ void LiveSession::changeConfiguration( if (mContinuationCounter == 0) { msg->post(); + + if (mSeekReplyID != 0) { + CHECK(mSeekReply != NULL); + mSeekReply->postReply(mSeekReplyID); + } } } @@ -978,11 +1106,13 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) { } void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { + mContinuation.clear(); // All remaining fetchers are still suspended, the player has shutdown // any decoders that needed it. - uint32_t streamMask; + uint32_t streamMask, resumeMask; CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask)); + CHECK(msg->findInt32("resumeMask", (int32_t *)&resumeMask)); for (size_t i = 0; i < kMaxStreams; ++i) { if (streamMask & indexToType(i)) { @@ -991,38 +1121,39 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { } int64_t timeUs; + bool switching = false; CHECK(msg->findInt64("timeUs", &timeUs)); if (timeUs < 0ll) { timeUs = mLastDequeuedTimeUs; + switching = true; } mRealTimeBaseUs = ALooper::GetNowUs() - timeUs; - mStreamMask = streamMask; + mNewStreamMask = streamMask; - // Resume all existing fetchers and assign them packet sources. + // Of all existing fetchers: + // * Resume fetchers that are still needed and assign them original packet sources. + // * Mark otherwise unneeded fetchers for removal. + ALOGV("resuming fetchers for mask 0x%08x", resumeMask); for (size_t i = 0; i < mFetcherInfos.size(); ++i) { const AString &uri = mFetcherInfos.keyAt(i); - uint32_t resumeMask = 0; - sp<AnotherPacketSource> sources[kMaxStreams]; - // TRICKY: looping from i as earlier streams are already removed from streamMask - for (size_t j = i; j < kMaxStreams; ++j) { - if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) { + for (size_t j = 0; j < kMaxStreams; ++j) { + if ((resumeMask & indexToType(j)) && uri == mStreams[j].mUri) { sources[j] = mPacketSources.valueFor(indexToType(j)); - resumeMask |= indexToType(j); } } - CHECK_NE(resumeMask, 0u); - - ALOGV("resuming fetchers for mask 0x%08x", resumeMask); - - streamMask &= ~resumeMask; - - mFetcherInfos.valueAt(i).mFetcher->startAsync( - sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]); + FetcherInfo &info = mFetcherInfos.editValueAt(i); + if (sources[kAudioIndex] != NULL || sources[kVideoIndex] != NULL + || sources[kSubtitleIndex] != NULL) { + info.mFetcher->startAsync( + sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]); + } else { + info.mToBeRemoved = true; + } } // streamMask now only contains the types that need a new fetcher created. @@ -1031,6 +1162,8 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { ALOGV("creating new fetchers for mask 0x%08x", streamMask); } + // Find out when the original fetchers have buffered up to and start the new fetchers + // at a later timestamp. for (size_t i = 0; i < kMaxStreams; i++) { if (!(indexToType(i) & streamMask)) { continue; @@ -1042,12 +1175,40 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { sp<PlaylistFetcher> fetcher = addFetcher(uri.c_str()); CHECK(fetcher != NULL); + int32_t latestSeq = -1; + int64_t latestTimeUs = 0ll; sp<AnotherPacketSource> sources[kMaxStreams]; + // TRICKY: looping from i as earlier streams are already removed from streamMask for (size_t j = i; j < kMaxStreams; ++j) { if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) { sources[j] = mPacketSources.valueFor(indexToType(j)); - sources[j]->clear(); + + if (!switching) { + sources[j]->clear(); + } else { + int32_t type, seq; + int64_t srcTimeUs; + sp<AMessage> meta = sources[j]->getLatestMeta(); + + if (meta != NULL && !meta->findInt32("discontinuity", &type)) { + CHECK(meta->findInt32("seq", &seq)); + if (seq > latestSeq) { + latestSeq = seq; + } + CHECK(meta->findInt64("timeUs", &srcTimeUs)); + if (srcTimeUs > latestTimeUs) { + latestTimeUs = srcTimeUs; + } + } + + sources[j] = mPacketSources2.valueFor(indexToType(j)); + sources[j]->clear(); + uint32_t extraStreams = mNewStreamMask & (~mStreamMask); + if (extraStreams & indexToType(j)) { + sources[j]->queueAccessUnit(createFormatChangeBuffer(/* swap = */ false)); + } + } streamMask &= ~indexToType(j); } @@ -1057,7 +1218,9 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex], - timeUs); + timeUs, + latestTimeUs /* min start time(us) */, + latestSeq >= 0 ? latestSeq + 1 : -1 /* starting sequence number hint */ ); } // All fetchers have now been started, the configuration change @@ -1066,14 +1229,61 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { scheduleCheckBandwidthEvent(); ALOGV("XXX configuration change completed."); - mReconfigurationInProgress = false; + if (switching) { + mSwitchInProgress = true; + } else { + mStreamMask = mNewStreamMask; + } if (mDisconnectReplyID != 0) { finishDisconnect(); } } +void LiveSession::onSwapped(const sp<AMessage> &msg) { + int32_t switchGeneration; + CHECK(msg->findInt32("switchGeneration", &switchGeneration)); + if (switchGeneration != mSwitchGeneration) { + return; + } + + int32_t stream; + CHECK(msg->findInt32("stream", &stream)); + mSwapMask |= stream; + if (mSwapMask != mStreamMask) { + return; + } + + // Check if new variant contains extra streams. + uint32_t extraStreams = mNewStreamMask & (~mStreamMask); + while (extraStreams) { + StreamType extraStream = (StreamType) (extraStreams & ~(extraStreams - 1)); + swapPacketSource(extraStream); + extraStreams &= ~extraStream; + } + + tryToFinishBandwidthSwitch(); +} + +// Mark switch done when: +// 1. all old buffers are swapped out, AND +// 2. all old fetchers are removed. +void LiveSession::tryToFinishBandwidthSwitch() { + bool needToRemoveFetchers = false; + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { + if (mFetcherInfos.valueAt(i).mToBeRemoved) { + needToRemoveFetchers = true; + break; + } + } + if (!needToRemoveFetchers && mSwapMask == mStreamMask) { + mStreamMask = mNewStreamMask; + mSwitchInProgress = false; + mSwapMask = 0; + } +} + void LiveSession::scheduleCheckBandwidthEvent() { sp<AMessage> msg = new AMessage(kWhatCheckBandwidth, id()); msg->setInt32("generation", mCheckBandwidthGeneration); @@ -1084,16 +1294,37 @@ void LiveSession::cancelCheckBandwidthEvent() { ++mCheckBandwidthGeneration; } -void LiveSession::onCheckBandwidth() { - if (mReconfigurationInProgress) { - scheduleCheckBandwidthEvent(); - return; +void LiveSession::cancelBandwidthSwitch() { + Mutex::Autolock lock(mSwapMutex); + mSwitchGeneration++; + mSwitchInProgress = false; + mSwapMask = 0; +} + +bool LiveSession::canSwitchBandwidthTo(size_t bandwidthIndex) { + if (mReconfigurationInProgress || mSwitchInProgress) { + return false; + } + + if (mPrevBandwidthIndex < 0) { + return true; } + if (bandwidthIndex == (size_t)mPrevBandwidthIndex) { + return false; + } else if (bandwidthIndex > (size_t)mPrevBandwidthIndex) { + return canSwitchUp(); + } else { + return true; + } +} + +void LiveSession::onCheckBandwidth() { size_t bandwidthIndex = getBandwidthIndex(); - if (mPrevBandwidthIndex < 0 - || bandwidthIndex != (size_t)mPrevBandwidthIndex) { + if (canSwitchBandwidthTo(bandwidthIndex)) { changeConfiguration(-1ll /* timeUs */, bandwidthIndex); + } else { + scheduleCheckBandwidthEvent(); } // Handling the kWhatCheckBandwidth even here does _not_ automatically diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index c4d125c..f489ec4 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -83,6 +83,11 @@ struct LiveSession : public AHandler { kWhatPreparationFailed, }; + // create a format-change discontinuity + // + // swap: + // whether is format-change discontinuity should trigger a buffer swap + sp<ABuffer> createFormatChangeBuffer(bool swap = true); protected: virtual ~LiveSession(); @@ -101,6 +106,7 @@ private: kWhatChangeConfiguration2 = 'chC2', kWhatChangeConfiguration3 = 'chC3', kWhatFinishDisconnect2 = 'fin2', + kWhatSwapped = 'swap', }; struct BandwidthItem { @@ -112,6 +118,7 @@ private: sp<PlaylistFetcher> mFetcher; int64_t mDurationUs; bool mIsPrepared; + bool mToBeRemoved; }; struct StreamItem { @@ -146,18 +153,38 @@ private: KeyedVector<AString, FetcherInfo> mFetcherInfos; uint32_t mStreamMask; + // Masks used during reconfiguration: + // mNewStreamMask: streams in the variant playlist we're switching to; + // we don't want to immediately overwrite the original value. + uint32_t mNewStreamMask; + + // mSwapMask: streams that have started to playback content in the new variant playlist; + // we use this to track reconfiguration progress. + uint32_t mSwapMask; + KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources; + // A second set of packet sources that buffer content for the variant we're switching to. + KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources2; + + // A mutex used to serialize two sets of events: + // * the swapping of packet sources in dequeueAccessUnit on the player thread, AND + // * a forced bandwidth switch termination in cancelSwitch on the live looper. + Mutex mSwapMutex; int32_t mCheckBandwidthGeneration; + int32_t mSwitchGeneration; size_t mContinuationCounter; sp<AMessage> mContinuation; + sp<AMessage> mSeekReply; int64_t mLastDequeuedTimeUs; int64_t mRealTimeBaseUs; bool mReconfigurationInProgress; + bool mSwitchInProgress; uint32_t mDisconnectReplyID; + uint32_t mSeekReplyID; sp<PlaylistFetcher> addFetcher(const char *uri); @@ -199,16 +226,27 @@ private: void onChangeConfiguration(const sp<AMessage> &msg); void onChangeConfiguration2(const sp<AMessage> &msg); void onChangeConfiguration3(const sp<AMessage> &msg); + void onSwapped(const sp<AMessage> &msg); + void tryToFinishBandwidthSwitch(); void scheduleCheckBandwidthEvent(); void cancelCheckBandwidthEvent(); + // cancelBandwidthSwitch is atomic wrt swapPacketSource; call it to prevent packet sources + // from being swapped out on stale discontinuities while manipulating + // mPacketSources/mPacketSources2. + void cancelBandwidthSwitch(); + + bool canSwitchBandwidthTo(size_t bandwidthIndex); void onCheckBandwidth(); void finishDisconnect(); void postPrepared(status_t err); + void swapPacketSource(StreamType stream); + bool canSwitchUp(); + DISALLOW_EVIL_CONSTRUCTORS(LiveSession); }; diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 030cbde..9d7cb99 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -48,16 +48,20 @@ namespace android { // static const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll; const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll; +const int32_t PlaylistFetcher::kNumSkipFrames = 10; PlaylistFetcher::PlaylistFetcher( const sp<AMessage> ¬ify, const sp<LiveSession> &session, const char *uri) : mNotify(notify), + mStartTimeUsNotify(notify->dup()), mSession(session), mURI(uri), mStreamTypeMask(0), mStartTimeUs(-1ll), + mMinStartTimeUs(0ll), + mStopParams(NULL), mLastPlaylistFetchTimeUs(-1ll), mSeqNumber(-1), mNumRetries(0), @@ -69,6 +73,8 @@ PlaylistFetcher::PlaylistFetcher( mFirstPTSValid(false), mAbsoluteTimeAnchorUs(0ll) { memset(mPlaylistHash, 0, sizeof(mPlaylistHash)); + mStartTimeUsNotify->setInt32("what", kWhatStartedAt); + mStartTimeUsNotify->setInt32("streamMask", 0); } PlaylistFetcher::~PlaylistFetcher() { @@ -325,7 +331,9 @@ void PlaylistFetcher::startAsync( const sp<AnotherPacketSource> &audioSource, const sp<AnotherPacketSource> &videoSource, const sp<AnotherPacketSource> &subtitleSource, - int64_t startTimeUs) { + int64_t startTimeUs, + int64_t minStartTimeUs, + int32_t startSeqNumberHint) { sp<AMessage> msg = new AMessage(kWhatStart, id()); uint32_t streamTypeMask = 0ul; @@ -347,6 +355,8 @@ void PlaylistFetcher::startAsync( msg->setInt32("streamTypeMask", streamTypeMask); msg->setInt64("startTimeUs", startTimeUs); + msg->setInt64("minStartTimeUs", minStartTimeUs); + msg->setInt32("startSeqNumberHint", startSeqNumberHint); msg->post(); } @@ -354,8 +364,16 @@ void PlaylistFetcher::pauseAsync() { (new AMessage(kWhatPause, id()))->post(); } -void PlaylistFetcher::stopAsync() { - (new AMessage(kWhatStop, id()))->post(); +void PlaylistFetcher::stopAsync(bool selfTriggered) { + sp<AMessage> msg = new AMessage(kWhatStop, id()); + msg->setInt32("selfTriggered", selfTriggered); + msg->post(); +} + +void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> ¶ms) { + AMessage* msg = new AMessage(kWhatResumeUntil, id()); + msg->setMessage("params", params); + msg->post(); } void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { @@ -383,7 +401,7 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { case kWhatStop: { - onStop(); + onStop(msg); sp<AMessage> notify = mNotify->dup(); notify->setInt32("what", kWhatStopped); @@ -392,6 +410,7 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { } case kWhatMonitorQueue: + case kWhatDownloadNext: { int32_t generation; CHECK(msg->findInt32("generation", &generation)); @@ -401,7 +420,17 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { break; } - onMonitorQueue(); + if (msg->what() == kWhatMonitorQueue) { + onMonitorQueue(); + } else { + onDownloadNext(); + } + break; + } + + case kWhatResumeUntil: + { + onResumeUntil(msg); break; } @@ -417,7 +446,10 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask)); int64_t startTimeUs; + int32_t startSeqNumberHint; CHECK(msg->findInt64("startTimeUs", &startTimeUs)); + CHECK(msg->findInt64("minStartTimeUs", (int64_t *) &mMinStartTimeUs)); + CHECK(msg->findInt32("startSeqNumberHint", &startSeqNumberHint)); if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) { void *ptr; @@ -455,6 +487,10 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { mPrepared = false; } + if (startSeqNumberHint >= 0) { + mSeqNumber = startSeqNumberHint; + } + postMonitorQueue(); return OK; @@ -462,22 +498,83 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { void PlaylistFetcher::onPause() { cancelMonitorQueue(); - - mPacketSources.clear(); - mStreamTypeMask = 0; } -void PlaylistFetcher::onStop() { +void PlaylistFetcher::onStop(const sp<AMessage> &msg) { cancelMonitorQueue(); - for (size_t i = 0; i < mPacketSources.size(); ++i) { - mPacketSources.valueAt(i)->clear(); + int32_t selfTriggered; + CHECK(msg->findInt32("selfTriggered", &selfTriggered)); + if (!selfTriggered) { + // Self triggered stops only happen during switching, in which case we do not want + // to clear the discontinuities queued at the end of packet sources. + for (size_t i = 0; i < mPacketSources.size(); i++) { + sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + packetSource->clear(); + } } mPacketSources.clear(); mStreamTypeMask = 0; } +// Resume until we have reached the boundary timestamps listed in `msg`; when +// the remaining time is too short (within a resume threshold) stop immediately +// instead. +status_t PlaylistFetcher::onResumeUntil(const sp<AMessage> &msg) { + sp<AMessage> params; + CHECK(msg->findMessage("params", ¶ms)); + + bool stop = false; + for (size_t i = 0; i < mPacketSources.size(); i++) { + sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + + const char *stopKey; + int streamType = mPacketSources.keyAt(i); + switch (streamType) { + case LiveSession::STREAMTYPE_VIDEO: + stopKey = "timeUsVideo"; + break; + + case LiveSession::STREAMTYPE_AUDIO: + stopKey = "timeUsAudio"; + break; + + case LiveSession::STREAMTYPE_SUBTITLES: + stopKey = "timeUsSubtitle"; + break; + + default: + TRESPASS(); + } + + // Don't resume if we would stop within a resume threshold. + int64_t latestTimeUs = 0, stopTimeUs = 0; + sp<AMessage> latestMeta = packetSource->getLatestMeta(); + if (latestMeta != NULL + && (latestMeta->findInt64("timeUs", &latestTimeUs) + && params->findInt64(stopKey, &stopTimeUs))) { + int64_t diffUs = stopTimeUs - latestTimeUs; + if (diffUs < resumeThreshold(latestMeta)) { + stop = true; + } + } + } + + if (stop) { + for (size_t i = 0; i < mPacketSources.size(); i++) { + mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer()); + } + stopAsync(/* selfTriggered = */ true); + return OK; + } + + mStopParams = params; + postMonitorQueue(); + + return OK; +} + void PlaylistFetcher::notifyError(status_t err) { sp<AMessage> notify = mNotify->dup(); notify->setInt32("what", kWhatError); @@ -519,8 +616,9 @@ void PlaylistFetcher::onMonitorQueue() { packetSource->getBufferedDurationUs(&finalResult); finalResult = OK; } else { - bool first = true; - + // Use max stream duration to prevent us from waiting on a non-existent stream; + // when we cannot make out from the manifest what streams are included in a playlist + // we might assume extra streams. for (size_t i = 0; i < mPacketSources.size(); ++i) { if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) { continue; @@ -528,9 +626,10 @@ void PlaylistFetcher::onMonitorQueue() { int64_t bufferedStreamDurationUs = mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); - if (first || bufferedStreamDurationUs < bufferedDurationUs) { + ALOGV("buffered %lld for stream %d", + bufferedStreamDurationUs, mPacketSources.keyAt(i)); + if (bufferedStreamDurationUs > bufferedDurationUs) { bufferedDurationUs = bufferedStreamDurationUs; - first = false; } } } @@ -550,7 +649,12 @@ void PlaylistFetcher::onMonitorQueue() { if (finalResult == OK && downloadMore) { ALOGV("monitoring, buffered=%lld < %lld", bufferedDurationUs, durationToBufferUs); - onDownloadNext(); + // delay the next download slightly; hopefully this gives other concurrent fetchers + // a better chance to run. + // onDownloadNext(); + sp<AMessage> msg = new AMessage(kWhatDownloadNext, id()); + msg->setInt32("generation", mMonitorQueueGeneration); + msg->post(1000l); } else { // Nothing to do yet, try again in a second. @@ -617,6 +721,12 @@ void PlaylistFetcher::onDownloadNext() { const int32_t lastSeqNumberInPlaylist = firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; + if (mStartup && mSeqNumber >= 0 + && (mSeqNumber < firstSeqNumberInPlaylist || mSeqNumber > lastSeqNumberInPlaylist)) { + // in case we guessed wrong during reconfiguration, try fetching the latest content. + mSeqNumber = lastSeqNumberInPlaylist; + } + if (mSeqNumber < 0) { CHECK_GE(mStartTimeUs, 0ll); @@ -762,6 +872,18 @@ void PlaylistFetcher::onDownloadNext() { err = extractAndQueueAccessUnits(buffer, itemMeta); + if (err == -EAGAIN) { + // bad starting sequence number hint + postMonitorQueue(); + return; + } + + if (err == ERROR_OUT_OF_RANGE) { + // reached stopping point + stopAsync(/* selfTriggered = */ true); + return; + } + if (err != OK) { notifyError(err); return; @@ -818,12 +940,15 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( } if (mTSParser == NULL) { - mTSParser = new ATSParser; + // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers. + mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE); } if (mNextPTSTimeUs >= 0ll) { sp<AMessage> extra = new AMessage; - extra->setInt64(IStreamListener::kKeyMediaTimeUs, mNextPTSTimeUs); + // Since we are using absolute timestamps, signal an offset of 0 to prevent + // ATSParser from skewing the timestamps of access units. + extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0); mTSParser->signalDiscontinuity( ATSParser::DISCONTINUITY_SEEK, extra); @@ -842,17 +967,23 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( offset += 188; } + status_t err = OK; for (size_t i = mPacketSources.size(); i-- > 0;) { sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + const char *key; ATSParser::SourceType type; - switch (mPacketSources.keyAt(i)) { + const LiveSession::StreamType stream = mPacketSources.keyAt(i); + switch (stream) { + case LiveSession::STREAMTYPE_VIDEO: type = ATSParser::VIDEO; + key = "timeUsVideo"; break; case LiveSession::STREAMTYPE_AUDIO: type = ATSParser::AUDIO; + key = "timeUsAudio"; break; case LiveSession::STREAMTYPE_SUBTITLES: @@ -879,19 +1010,87 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( continue; } + int64_t timeUs; sp<ABuffer> accessUnit; status_t finalResult; while (source->hasBufferAvailable(&finalResult) && source->dequeueAccessUnit(&accessUnit) == OK) { - // Note that we do NOT dequeue any discontinuities. + + CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); + if (mMinStartTimeUs > 0) { + if (timeUs < mMinStartTimeUs) { + // TODO untested path + // try a later ts + int32_t targetDuration; + mPlaylist->meta()->findInt32("target-duration", &targetDuration); + int32_t incr = (mMinStartTimeUs - timeUs) / 1000000 / targetDuration; + if (incr == 0) { + // increment mSeqNumber by at least one + incr = 1; + } + mSeqNumber += incr; + err = -EAGAIN; + break; + } else { + int64_t startTimeUs; + if (mStartTimeUsNotify != NULL + && !mStartTimeUsNotify->findInt64(key, &startTimeUs)) { + mStartTimeUsNotify->setInt64(key, timeUs); + + uint32_t streamMask = 0; + mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask); + streamMask |= mPacketSources.keyAt(i); + mStartTimeUsNotify->setInt32("streamMask", streamMask); + + if (streamMask == mStreamTypeMask) { + mStartTimeUsNotify->post(); + mStartTimeUsNotify.clear(); + } + } + } + } + + if (mStopParams != NULL) { + // Queue discontinuity in original stream. + int64_t stopTimeUs; + if (!mStopParams->findInt64(key, &stopTimeUs) || timeUs >= stopTimeUs) { + packetSource->queueAccessUnit(mSession->createFormatChangeBuffer()); + mStreamTypeMask &= ~stream; + mPacketSources.removeItemsAt(i); + break; + } + } + + // Note that we do NOT dequeue any discontinuities except for format change. // for simplicity, store a reference to the format in each unit sp<MetaData> format = source->getFormat(); if (format != NULL) { accessUnit->meta()->setObject("format", format); } + + // Stash the sequence number so we can hint future fetchers where to start at. + accessUnit->meta()->setInt32("seq", mSeqNumber); packetSource->queueAccessUnit(accessUnit); } + + if (err != OK) { + break; + } + } + + if (err != OK) { + for (size_t i = mPacketSources.size(); i-- > 0;) { + sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + packetSource->clear(); + } + return err; + } + + if (!mStreamTypeMask) { + // Signal gap is filled between original and new stream. + ALOGV("ERROR OUT OF RANGE"); + return ERROR_OUT_OF_RANGE; } return OK; @@ -908,6 +1107,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( CHECK(itemMeta->findInt64("durationUs", &durationUs)); buffer->meta()->setInt64("timeUs", getSegmentStartTimeUs(mSeqNumber)); buffer->meta()->setInt64("durationUs", durationUs); + buffer->meta()->setInt32("seq", mSeqNumber); packetSource->queueAccessUnit(buffer); return OK; @@ -1033,6 +1233,18 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( | (adtsHeader[4] << 3) | (adtsHeader[5] >> 5); + if (aac_frame_length == 0) { + const uint8_t *id3Header = adtsHeader; + if (!memcmp(id3Header, "ID3", 3)) { + ID3 id3(id3Header, buffer->size() - offset, true); + if (id3.isValid()) { + offset += id3.rawSize(); + continue; + }; + } + return ERROR_MALFORMED; + } + CHECK_LE(offset + aac_frame_length, buffer->size()); sp<ABuffer> unit = new ABuffer(aac_frame_length); @@ -1044,6 +1256,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( // Each AAC frame encodes 1024 samples. numSamples += 1024; + unit->meta()->setInt32("seq", mSeqNumber); packetSource->queueAccessUnit(unit); offset += aac_frame_length; @@ -1071,4 +1284,33 @@ void PlaylistFetcher::updateDuration() { msg->post(); } +int64_t PlaylistFetcher::resumeThreshold(const sp<AMessage> &msg) { + int64_t durationUs, threshold; + if (msg->findInt64("durationUs", &durationUs)) { + return kNumSkipFrames * durationUs; + } + + sp<RefBase> obj; + msg->findObject("format", &obj); + MetaData *format = static_cast<MetaData *>(obj.get()); + + const char *mime; + CHECK(format->findCString(kKeyMIMEType, &mime)); + bool audio = !strncasecmp(mime, "audio/", 6); + if (audio) { + // Assumes 1000 samples per frame. + int32_t sampleRate; + CHECK(format->findInt32(kKeySampleRate, &sampleRate)); + return kNumSkipFrames /* frames */ * 1000 /* samples */ + * (1000000 / sampleRate) /* sample duration (us) */; + } else { + int32_t frameRate; + if (format->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) { + return kNumSkipFrames * (1000000 / frameRate); + } + } + + return 500000ll; +} + } // namespace android diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index ac04a77..8404b8d 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -43,6 +43,7 @@ struct PlaylistFetcher : public AHandler { kWhatTemporarilyDoneFetching, kWhatPrepared, kWhatPreparationFailed, + kWhatStartedAt, }; PlaylistFetcher( @@ -56,11 +57,15 @@ struct PlaylistFetcher : public AHandler { const sp<AnotherPacketSource> &audioSource, const sp<AnotherPacketSource> &videoSource, const sp<AnotherPacketSource> &subtitleSource, - int64_t startTimeUs = -1ll); + int64_t startTimeUs = -1ll, + int64_t minStartTimeUs = 0ll /* start after this timestamp */, + int32_t startSeqNumberHint = -1 /* try starting at this sequence number */); void pauseAsync(); - void stopAsync(); + void stopAsync(bool selfTriggered = false); + + void resumeUntilAsync(const sp<AMessage> ¶ms); protected: virtual ~PlaylistFetcher(); @@ -76,17 +81,25 @@ private: kWhatPause = 'paus', kWhatStop = 'stop', kWhatMonitorQueue = 'moni', + kWhatResumeUntil = 'rsme', + kWhatDownloadNext = 'dlnx', }; static const int64_t kMinBufferedDurationUs; static const int64_t kMaxMonitorDelayUs; + static const int32_t kNumSkipFrames; + // notifications to mSession sp<AMessage> mNotify; + sp<AMessage> mStartTimeUsNotify; + sp<LiveSession> mSession; AString mURI; uint32_t mStreamTypeMask; int64_t mStartTimeUs; + int64_t mMinStartTimeUs; // start fetching no earlier than this value + sp<AMessage> mStopParams; // message containing the latest timestamps we should fetch. KeyedVector<LiveSession::StreamType, sp<AnotherPacketSource> > mPacketSources; @@ -149,10 +162,13 @@ private: status_t onStart(const sp<AMessage> &msg); void onPause(); - void onStop(); + void onStop(const sp<AMessage> &msg); void onMonitorQueue(); void onDownloadNext(); + // Resume a fetcher to continue until the stopping point stored in msg. + status_t onResumeUntil(const sp<AMessage> &msg); + status_t extractAndQueueAccessUnits( const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta); @@ -165,6 +181,10 @@ private: void updateDuration(); + // Before resuming a fetcher in onResume, check the remaining duration is longer than that + // returned by resumeThreshold. + int64_t resumeThreshold(const sp<AMessage> &msg); + DISALLOW_EVIL_CONSTRUCTORS(PlaylistFetcher); }; diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp index 6f69d0b..6ec9263 100644 --- a/media/libstagefright/matroska/MatroskaExtractor.cpp +++ b/media/libstagefright/matroska/MatroskaExtractor.cpp @@ -313,7 +313,7 @@ void BlockIterator::seek( *actualFrameTimeUs = -1ll; - const int64_t seekTimeNs = seekTimeUs * 1000ll; + const int64_t seekTimeNs = seekTimeUs * 1000ll - mExtractor->mSeekPreRollNs; mkvparser::Segment* const pSegment = mExtractor->mSegment; @@ -628,7 +628,8 @@ MatroskaExtractor::MatroskaExtractor(const sp<DataSource> &source) mReader(new DataSourceReader(mDataSource)), mSegment(NULL), mExtractedThumbnails(false), - mIsWebm(false) { + mIsWebm(false), + mSeekPreRollNs(0) { off64_t size; mIsLiveStreaming = (mDataSource->flags() @@ -919,6 +920,12 @@ void MatroskaExtractor::addTracks() { err = addVorbisCodecInfo( meta, codecPrivate, codecPrivateSize); + } else if (!strcmp("A_OPUS", codecID)) { + meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_OPUS); + meta->setData(kKeyOpusHeader, 0, codecPrivate, codecPrivateSize); + meta->setInt64(kKeyOpusCodecDelay, track->GetCodecDelay()); + meta->setInt64(kKeyOpusSeekPreRoll, track->GetSeekPreRoll()); + mSeekPreRollNs = track->GetSeekPreRoll(); } else if (!strcmp("A_MPEG/L3", codecID)) { meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG); } else { diff --git a/media/libstagefright/matroska/MatroskaExtractor.h b/media/libstagefright/matroska/MatroskaExtractor.h index 1294b4f..cf200f3 100644 --- a/media/libstagefright/matroska/MatroskaExtractor.h +++ b/media/libstagefright/matroska/MatroskaExtractor.h @@ -69,6 +69,7 @@ private: bool mExtractedThumbnails; bool mIsLiveStreaming; bool mIsWebm; + int64_t mSeekPreRollNs; void addTracks(); void findThumbnails(); diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp index d49e50b..65f5404 100644 --- a/media/libstagefright/omx/SoftOMXPlugin.cpp +++ b/media/libstagefright/omx/SoftOMXPlugin.cpp @@ -50,6 +50,7 @@ static const struct { { "OMX.google.mpeg4.encoder", "mpeg4enc", "video_encoder.mpeg4" }, { "OMX.google.mp3.decoder", "mp3dec", "audio_decoder.mp3" }, { "OMX.google.vorbis.decoder", "vorbisdec", "audio_decoder.vorbis" }, + { "OMX.google.opus.decoder", "opusdec", "audio_decoder.opus" }, { "OMX.google.vp8.decoder", "vpxdec", "video_decoder.vp8" }, { "OMX.google.vp9.decoder", "vpxdec", "video_decoder.vp9" }, { "OMX.google.vp8.encoder", "vpxenc", "video_encoder.vp8" }, diff --git a/media/libstagefright/omx/tests/OMXHarness.cpp b/media/libstagefright/omx/tests/OMXHarness.cpp index 03725df..f4dfd6b 100644 --- a/media/libstagefright/omx/tests/OMXHarness.cpp +++ b/media/libstagefright/omx/tests/OMXHarness.cpp @@ -463,6 +463,7 @@ static const char *GetMimeFromComponentRole(const char *componentRole) { { "audio_decoder.aac", "audio/mp4a-latm" }, { "audio_decoder.mp3", "audio/mpeg" }, { "audio_decoder.vorbis", "audio/vorbis" }, + { "audio_decoder.opus", "audio/opus" }, { "audio_decoder.g711alaw", MEDIA_MIMETYPE_AUDIO_G711_ALAW }, { "audio_decoder.g711mlaw", MEDIA_MIMETYPE_AUDIO_G711_MLAW }, }; @@ -495,6 +496,7 @@ static const char *GetURLForMime(const char *mime) { { "audio/mpeg", "file:///sdcard/media_api/music/MP3_48KHz_128kbps_s_1_17_CBR.mp3" }, { "audio/vorbis", NULL }, + { "audio/opus", NULL }, { "video/x-vnd.on2.vp8", "file:///sdcard/media_api/video/big-buck-bunny_trailer.webm" }, { MEDIA_MIMETYPE_AUDIO_G711_ALAW, "file:///sdcard/M1F1-Alaw-AFsp.wav" }, diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp index aeecdbc..a3093d0 100644 --- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp +++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp @@ -35,7 +35,6 @@ #include <gui/SurfaceComposerClient.h> #include <binder/ProcessState.h> -#include <ui/FramebufferNativeWindow.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/MediaBufferGroup.h> diff --git a/media/libstagefright/webm/Android.mk b/media/libstagefright/webm/Android.mk new file mode 100644 index 0000000..7081463 --- /dev/null +++ b/media/libstagefright/webm/Android.mk @@ -0,0 +1,23 @@ +LOCAL_PATH:= $(call my-dir) +include $(CLEAR_VARS) + +LOCAL_CPPFLAGS += -D__STDINT_LIMITS \ + -Werror + +LOCAL_SRC_FILES:= EbmlUtil.cpp \ + WebmElement.cpp \ + WebmFrame.cpp \ + WebmFrameThread.cpp \ + WebmWriter.cpp + + +LOCAL_C_INCLUDES += $(TOP)/frameworks/av/include + +LOCAL_SHARED_LIBRARIES += libstagefright_foundation \ + libstagefright \ + libutils \ + liblog + +LOCAL_MODULE:= libstagefright_webm + +include $(BUILD_STATIC_LIBRARY) diff --git a/media/libstagefright/webm/EbmlUtil.cpp b/media/libstagefright/webm/EbmlUtil.cpp new file mode 100644 index 0000000..449fec6 --- /dev/null +++ b/media/libstagefright/webm/EbmlUtil.cpp @@ -0,0 +1,108 @@ +/* + * Copyright (C) 2014 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. + */ + +#include <stdint.h> + +namespace { + +// Table for Seal's algorithm for Number of Trailing Zeros. Hacker's Delight +// online, Figure 5-18 (http://www.hackersdelight.org/revisions.pdf) +// The entries whose value is -1 are never referenced. +int NTZ_TABLE[] = { + 32, 0, 1, 12, 2, 6, -1, 13, 3, -1, 7, -1, -1, -1, -1, 14, + 10, 4, -1, -1, 8, -1, -1, 25, -1, -1, -1, -1, -1, 21, 27, 15, + 31, 11, 5, -1, -1, -1, -1, -1, 9, -1, -1, 24, -1, -1, 20, 26, + 30, -1, -1, -1, -1, 23, -1, 19, 29, -1, 22, 18, 28, 17, 16, -1 +}; + +int numberOfTrailingZeros32(int32_t i) { + uint32_t u = (i & -i) * 0x0450FBAF; + return NTZ_TABLE[(u) >> 26]; +} + +uint64_t highestOneBit(uint64_t n) { + n |= (n >> 1); + n |= (n >> 2); + n |= (n >> 4); + n |= (n >> 8); + n |= (n >> 16); + n |= (n >> 32); + return n - (n >> 1); +} + +uint64_t _powerOf2(uint64_t u) { + uint64_t powerOf2 = highestOneBit(u); + return powerOf2 ? powerOf2 : 1; +} + +// Based on Long.numberOfTrailingZeros in Long.java +int numberOfTrailingZeros(uint64_t u) { + int32_t low = u; + return low !=0 ? numberOfTrailingZeros32(low) + : 32 + numberOfTrailingZeros32((int32_t) (u >> 32)); +} +} + +namespace webm { + +// Encode the id and/or size of an EBML element bytes by setting a leading length descriptor bit: +// +// 1xxxxxxx - 1-byte values +// 01xxxxxx xxxxxxxx - +// 001xxxxx xxxxxxxx xxxxxxxx - +// 0001xxxx xxxxxxxx xxxxxxxx xxxxxxxx - ... +// 00001xxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx - +// 000001xx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx - +// 0000001x xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx - +// 00000001 xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx - 8-byte values +// +// This function uses the least the number of bytes possible. +uint64_t encodeUnsigned(uint64_t u) { + uint64_t powerOf2 = _powerOf2(u); + if (u + 1 == powerOf2 << 1) + powerOf2 <<= 1; + int shiftWidth = (7 + numberOfTrailingZeros(powerOf2)) / 7 * 7; + long lengthDescriptor = 1 << shiftWidth; + return lengthDescriptor | u; +} + +// Like above but pads the input value with leading zeros up to the specified width. The length +// descriptor is calculated based on width. +uint64_t encodeUnsigned(uint64_t u, int width) { + int shiftWidth = 7 * width; + uint64_t lengthDescriptor = 1; + lengthDescriptor <<= shiftWidth; + return lengthDescriptor | u; +} + +// Calculate the length of an EBML coded id or size from its length descriptor. +int sizeOf(uint64_t u) { + uint64_t powerOf2 = _powerOf2(u); + int unsignedLength = numberOfTrailingZeros(powerOf2) / 8 + 1; + return unsignedLength; +} + +// Serialize an EBML coded id or size in big-endian order. +int serializeCodedUnsigned(uint64_t u, uint8_t* bary) { + int unsignedLength = sizeOf(u); + for (int i = unsignedLength - 1; i >= 0; i--) { + bary[i] = u & 0xff; + u >>= 8; + } + return unsignedLength; +} + +} diff --git a/media/libstagefright/webm/EbmlUtil.h b/media/libstagefright/webm/EbmlUtil.h new file mode 100644 index 0000000..eb9c37c --- /dev/null +++ b/media/libstagefright/webm/EbmlUtil.h @@ -0,0 +1,50 @@ +/* + * Copyright (C) 2014 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 EBMLUTIL_H_ +#define EBMLUTIL_H_ + +#include <stdint.h> + +namespace webm { + +// Encode the id and/or size of an EBML element bytes by setting a leading length descriptor bit: +// +// 1xxxxxxx - 1-byte values +// 01xxxxxx xxxxxxxx - +// 001xxxxx xxxxxxxx xxxxxxxx - +// 0001xxxx xxxxxxxx xxxxxxxx xxxxxxxx - ... +// 00001xxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx - +// 000001xx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx - +// 0000001x xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx - +// 00000001 xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx - 8-byte values +// +// This function uses the least the number of bytes possible. +uint64_t encodeUnsigned(uint64_t u); + +// Like above but pads the input value with leading zeros up to the specified width. The length +// descriptor is calculated based on width. +uint64_t encodeUnsigned(uint64_t u, int width); + +// Serialize an EBML coded id or size in big-endian order. +int serializeCodedUnsigned(uint64_t u, uint8_t* bary); + +// Calculate the length of an EBML coded id or size from its length descriptor. +int sizeOf(uint64_t u); + +} + +#endif /* EBMLUTIL_H_ */ diff --git a/media/libstagefright/webm/LinkedBlockingQueue.h b/media/libstagefright/webm/LinkedBlockingQueue.h new file mode 100644 index 0000000..0b6a9a1 --- /dev/null +++ b/media/libstagefright/webm/LinkedBlockingQueue.h @@ -0,0 +1,79 @@ +/* + * Copyright (C) 2014 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 LINKEDBLOCKINGQUEUE_H_ +#define LINKEDBLOCKINGQUEUE_H_ + +#include <utils/List.h> +#include <utils/Mutex.h> +#include <utils/Condition.h> + +namespace android { + +template<typename T> +class LinkedBlockingQueue { + List<T> mList; + Mutex mLock; + Condition mContentAvailableCondition; + + T front(bool remove) { + Mutex::Autolock autolock(mLock); + while (mList.empty()) { + mContentAvailableCondition.wait(mLock); + } + T e = *(mList.begin()); + if (remove) { + mList.erase(mList.begin()); + } + return e; + } + + DISALLOW_EVIL_CONSTRUCTORS(LinkedBlockingQueue); + +public: + LinkedBlockingQueue() { + } + + ~LinkedBlockingQueue() { + } + + bool empty() { + Mutex::Autolock autolock(mLock); + return mList.empty(); + } + + void clear() { + Mutex::Autolock autolock(mLock); + mList.clear(); + } + + T peek() { + return front(false); + } + + T take() { + return front(true); + } + + void push(T e) { + Mutex::Autolock autolock(mLock); + mList.push_back(e); + mContentAvailableCondition.signal(); + } +}; + +} /* namespace android */ +#endif /* LINKEDBLOCKINGQUEUE_H_ */ diff --git a/media/libstagefright/webm/WebmConstants.h b/media/libstagefright/webm/WebmConstants.h new file mode 100644 index 0000000..c53f458 --- /dev/null +++ b/media/libstagefright/webm/WebmConstants.h @@ -0,0 +1,133 @@ +/* + * Copyright (C) 2014 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 WEBMCONSTANTS_H_ +#define WEBMCONSTANTS_H_ + +#include <stdint.h> + +namespace webm { + +const int kMinEbmlVoidSize = 2; +const int64_t kMaxMetaSeekSize = 64; +const int64_t kMkvUnknownLength = 0x01ffffffffffffffl; + +// EBML element id's from http://matroska.org/technical/specs/index.html +enum Mkv { + kMkvEbml = 0x1A45DFA3, + kMkvEbmlVersion = 0x4286, + kMkvEbmlReadVersion = 0x42F7, + kMkvEbmlMaxIdlength = 0x42F2, + kMkvEbmlMaxSizeLength = 0x42F3, + kMkvDocType = 0x4282, + kMkvDocTypeVersion = 0x4287, + kMkvDocTypeReadVersion = 0x4285, + kMkvVoid = 0xEC, + kMkvSignatureSlot = 0x1B538667, + kMkvSignatureAlgo = 0x7E8A, + kMkvSignatureHash = 0x7E9A, + kMkvSignaturePublicKey = 0x7EA5, + kMkvSignature = 0x7EB5, + kMkvSignatureElements = 0x7E5B, + kMkvSignatureElementList = 0x7E7B, + kMkvSignedElement = 0x6532, + kMkvSegment = 0x18538067, + kMkvSeekHead = 0x114D9B74, + kMkvSeek = 0x4DBB, + kMkvSeekId = 0x53AB, + kMkvSeekPosition = 0x53AC, + kMkvInfo = 0x1549A966, + kMkvTimecodeScale = 0x2AD7B1, + kMkvSegmentDuration = 0x4489, + kMkvDateUtc = 0x4461, + kMkvMuxingApp = 0x4D80, + kMkvWritingApp = 0x5741, + kMkvCluster = 0x1F43B675, + kMkvTimecode = 0xE7, + kMkvPrevSize = 0xAB, + kMkvBlockGroup = 0xA0, + kMkvBlock = 0xA1, + kMkvBlockAdditions = 0x75A1, + kMkvBlockMore = 0xA6, + kMkvBlockAddId = 0xEE, + kMkvBlockAdditional = 0xA5, + kMkvBlockDuration = 0x9B, + kMkvReferenceBlock = 0xFB, + kMkvLaceNumber = 0xCC, + kMkvSimpleBlock = 0xA3, + kMkvTracks = 0x1654AE6B, + kMkvTrackEntry = 0xAE, + kMkvTrackNumber = 0xD7, + kMkvTrackUid = 0x73C5, + kMkvTrackType = 0x83, + kMkvFlagEnabled = 0xB9, + kMkvFlagDefault = 0x88, + kMkvFlagForced = 0x55AA, + kMkvFlagLacing = 0x9C, + kMkvDefaultDuration = 0x23E383, + kMkvMaxBlockAdditionId = 0x55EE, + kMkvName = 0x536E, + kMkvLanguage = 0x22B59C, + kMkvCodecId = 0x86, + kMkvCodecPrivate = 0x63A2, + kMkvCodecName = 0x258688, + kMkvVideo = 0xE0, + kMkvFlagInterlaced = 0x9A, + kMkvStereoMode = 0x53B8, + kMkvAlphaMode = 0x53C0, + kMkvPixelWidth = 0xB0, + kMkvPixelHeight = 0xBA, + kMkvPixelCropBottom = 0x54AA, + kMkvPixelCropTop = 0x54BB, + kMkvPixelCropLeft = 0x54CC, + kMkvPixelCropRight = 0x54DD, + kMkvDisplayWidth = 0x54B0, + kMkvDisplayHeight = 0x54BA, + kMkvDisplayUnit = 0x54B2, + kMkvAspectRatioType = 0x54B3, + kMkvFrameRate = 0x2383E3, + kMkvAudio = 0xE1, + kMkvSamplingFrequency = 0xB5, + kMkvOutputSamplingFrequency = 0x78B5, + kMkvChannels = 0x9F, + kMkvBitDepth = 0x6264, + kMkvCues = 0x1C53BB6B, + kMkvCuePoint = 0xBB, + kMkvCueTime = 0xB3, + kMkvCueTrackPositions = 0xB7, + kMkvCueTrack = 0xF7, + kMkvCueClusterPosition = 0xF1, + kMkvCueBlockNumber = 0x5378 +}; + +enum TrackTypes { + kInvalidType = -1, + kVideoType = 0x1, + kAudioType = 0x2, + kComplexType = 0x3, + kLogoType = 0x10, + kSubtitleType = 0x11, + kButtonsType = 0x12, + kControlType = 0x20 +}; + +enum TrackNum { + kVideoTrackNum = 0x1, + kAudioTrackNum = 0x2 +}; +} + +#endif /* WEBMCONSTANTS_H_ */ diff --git a/media/libstagefright/webm/WebmElement.cpp b/media/libstagefright/webm/WebmElement.cpp new file mode 100644 index 0000000..c978966 --- /dev/null +++ b/media/libstagefright/webm/WebmElement.cpp @@ -0,0 +1,367 @@ +/* + * Copyright (C) 2014 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 "WebmElement" + +#include "EbmlUtil.h" +#include "WebmElement.h" +#include "WebmConstants.h" + +#include <media/stagefright/foundation/ADebug.h> +#include <utils/Log.h> + +#include <string.h> +#include <unistd.h> +#include <errno.h> +#include <fcntl.h> +#include <sys/mman.h> + +using namespace android; +using namespace webm; + +namespace { + +int64_t voidSize(int64_t totalSize) { + if (totalSize < 2) { + return -1; + } + if (totalSize < 9) { + return totalSize - 2; + } + return totalSize - 9; +} + +uint64_t childrenSum(const List<sp<WebmElement> >& children) { + uint64_t total = 0; + for (List<sp<WebmElement> >::const_iterator it = children.begin(); + it != children.end(); ++it) { + total += (*it)->totalSize(); + } + return total; +} + +void populateCommonTrackEntries( + int num, + uint64_t uid, + bool lacing, + const char *lang, + const char *codec, + TrackTypes type, + List<sp<WebmElement> > &ls) { + ls.push_back(new WebmUnsigned(kMkvTrackNumber, num)); + ls.push_back(new WebmUnsigned(kMkvTrackUid, uid)); + ls.push_back(new WebmUnsigned(kMkvFlagLacing, lacing)); + ls.push_back(new WebmString(kMkvLanguage, lang)); + ls.push_back(new WebmString(kMkvCodecId, codec)); + ls.push_back(new WebmUnsigned(kMkvTrackType, type)); +} +} + +namespace android { + +WebmElement::WebmElement(uint64_t id, uint64_t size) + : mId(id), mSize(size) { +} + +WebmElement::~WebmElement() { +} + +int WebmElement::serializePayloadSize(uint8_t *buf) { + return serializeCodedUnsigned(encodeUnsigned(mSize), buf); +} + +uint64_t WebmElement::serializeInto(uint8_t *buf) { + uint8_t *cur = buf; + int head = serializeCodedUnsigned(mId, cur); + cur += head; + int neck = serializePayloadSize(cur); + cur += neck; + serializePayload(cur); + cur += mSize; + return cur - buf; +} + +uint64_t WebmElement::totalSize() { + uint8_t buf[8]; + //............... + sizeOf(encodeUnsigned(size)) + return sizeOf(mId) + serializePayloadSize(buf) + mSize; +} + +uint8_t *WebmElement::serialize(uint64_t& size) { + size = totalSize(); + uint8_t *buf = new uint8_t[size]; + serializeInto(buf); + return buf; +} + +int WebmElement::write(int fd, uint64_t& size) { + uint8_t buf[8]; + size = totalSize(); + off64_t off = ::lseek64(fd, (size - 1), SEEK_CUR) - (size - 1); + ::write(fd, buf, 1); // extend file + + off64_t curOff = off + size; + off64_t alignedOff = off & ~(::sysconf(_SC_PAGE_SIZE) - 1); + off64_t mapSize = curOff - alignedOff; + off64_t pageOff = off - alignedOff; + void *dst = ::mmap64(NULL, mapSize, PROT_WRITE, MAP_SHARED, fd, alignedOff); + if ((int) dst == -1) { + ALOGE("mmap64 failed; errno = %d", errno); + ALOGE("fd %d; flags: %o", fd, ::fcntl(fd, F_GETFL, 0)); + return errno; + } else { + serializeInto((uint8_t*) dst + pageOff); + ::msync(dst, mapSize, MS_SYNC); + return ::munmap(dst, mapSize); + } +} + +//================================================================================================= + +WebmUnsigned::WebmUnsigned(uint64_t id, uint64_t value) + : WebmElement(id, sizeOf(value)), mValue(value) { +} + +void WebmUnsigned::serializePayload(uint8_t *buf) { + serializeCodedUnsigned(mValue, buf); +} + +//================================================================================================= + +WebmFloat::WebmFloat(uint64_t id, double value) + : WebmElement(id, sizeof(double)), mValue(value) { +} + +WebmFloat::WebmFloat(uint64_t id, float value) + : WebmElement(id, sizeof(float)), mValue(value) { +} + +void WebmFloat::serializePayload(uint8_t *buf) { + uint64_t data; + if (mSize == sizeof(float)) { + float f = mValue; + data = *reinterpret_cast<const uint32_t*>(&f); + } else { + data = *reinterpret_cast<const uint64_t*>(&mValue); + } + for (int i = mSize - 1; i >= 0; --i) { + buf[i] = data & 0xff; + data >>= 8; + } +} + +//================================================================================================= + +WebmBinary::WebmBinary(uint64_t id, const sp<ABuffer> &ref) + : WebmElement(id, ref->size()), mRef(ref) { +} + +void WebmBinary::serializePayload(uint8_t *buf) { + memcpy(buf, mRef->data(), mRef->size()); +} + +//================================================================================================= + +WebmString::WebmString(uint64_t id, const char *str) + : WebmElement(id, strlen(str)), mStr(str) { +} + +void WebmString::serializePayload(uint8_t *buf) { + memcpy(buf, mStr, strlen(mStr)); +} + +//================================================================================================= + +WebmSimpleBlock::WebmSimpleBlock( + int trackNum, + int16_t relTimecode, + bool key, + const sp<ABuffer>& orig) + // ............................ trackNum*1 + timecode*2 + flags*1 + // ^^^ + // Only the least significant byte of trackNum is encoded + : WebmElement(kMkvSimpleBlock, orig->size() + 4), + mTrackNum(trackNum), + mRelTimecode(relTimecode), + mKey(key), + mRef(orig) { +} + +void WebmSimpleBlock::serializePayload(uint8_t *buf) { + serializeCodedUnsigned(encodeUnsigned(mTrackNum), buf); + buf[1] = (mRelTimecode & 0xff00) >> 8; + buf[2] = mRelTimecode & 0xff; + buf[3] = mKey ? 0x80 : 0; + memcpy(buf + 4, mRef->data(), mSize - 4); +} + +//================================================================================================= + +EbmlVoid::EbmlVoid(uint64_t totalSize) + : WebmElement(kMkvVoid, voidSize(totalSize)), + mSizeWidth(totalSize - sizeOf(kMkvVoid) - voidSize(totalSize)) { + CHECK_GE(voidSize(totalSize), 0); +} + +int EbmlVoid::serializePayloadSize(uint8_t *buf) { + return serializeCodedUnsigned(encodeUnsigned(mSize, mSizeWidth), buf); +} + +void EbmlVoid::serializePayload(uint8_t *buf) { + ::memset(buf, 0, mSize); + return; +} + +//================================================================================================= + +WebmMaster::WebmMaster(uint64_t id, const List<sp<WebmElement> >& children) + : WebmElement(id, childrenSum(children)), mChildren(children) { +} + +WebmMaster::WebmMaster(uint64_t id) + : WebmElement(id, 0) { +} + +int WebmMaster::serializePayloadSize(uint8_t *buf) { + if (mSize == 0){ + return serializeCodedUnsigned(kMkvUnknownLength, buf); + } + return WebmElement::serializePayloadSize(buf); +} + +void WebmMaster::serializePayload(uint8_t *buf) { + uint64_t off = 0; + for (List<sp<WebmElement> >::const_iterator it = mChildren.begin(); it != mChildren.end(); + ++it) { + sp<WebmElement> child = (*it); + child->serializeInto(buf + off); + off += child->totalSize(); + } +} + +//================================================================================================= + +sp<WebmElement> WebmElement::CuePointEntry(uint64_t time, int track, uint64_t off) { + List<sp<WebmElement> > cuePointEntryFields; + cuePointEntryFields.push_back(new WebmUnsigned(kMkvCueTrack, track)); + cuePointEntryFields.push_back(new WebmUnsigned(kMkvCueClusterPosition, off)); + WebmElement *cueTrackPositions = new WebmMaster(kMkvCueTrackPositions, cuePointEntryFields); + + cuePointEntryFields.clear(); + cuePointEntryFields.push_back(new WebmUnsigned(kMkvCueTime, time)); + cuePointEntryFields.push_back(cueTrackPositions); + return new WebmMaster(kMkvCuePoint, cuePointEntryFields); +} + +sp<WebmElement> WebmElement::SeekEntry(uint64_t id, uint64_t off) { + List<sp<WebmElement> > seekEntryFields; + seekEntryFields.push_back(new WebmUnsigned(kMkvSeekId, id)); + seekEntryFields.push_back(new WebmUnsigned(kMkvSeekPosition, off)); + return new WebmMaster(kMkvSeek, seekEntryFields); +} + +sp<WebmElement> WebmElement::EbmlHeader( + int ver, + int readVer, + int maxIdLen, + int maxSizeLen, + int docVer, + int docReadVer) { + List<sp<WebmElement> > headerFields; + headerFields.push_back(new WebmUnsigned(kMkvEbmlVersion, ver)); + headerFields.push_back(new WebmUnsigned(kMkvEbmlReadVersion, readVer)); + headerFields.push_back(new WebmUnsigned(kMkvEbmlMaxIdlength, maxIdLen)); + headerFields.push_back(new WebmUnsigned(kMkvEbmlMaxSizeLength, maxSizeLen)); + headerFields.push_back(new WebmString(kMkvDocType, "webm")); + headerFields.push_back(new WebmUnsigned(kMkvDocTypeVersion, docVer)); + headerFields.push_back(new WebmUnsigned(kMkvDocTypeReadVersion, docReadVer)); + return new WebmMaster(kMkvEbml, headerFields); +} + +sp<WebmElement> WebmElement::SegmentInfo(uint64_t scale, double dur) { + List<sp<WebmElement> > segmentInfo; + // place duration first; easier to patch + segmentInfo.push_back(new WebmFloat(kMkvSegmentDuration, dur)); + segmentInfo.push_back(new WebmUnsigned(kMkvTimecodeScale, scale)); + segmentInfo.push_back(new WebmString(kMkvMuxingApp, "android")); + segmentInfo.push_back(new WebmString(kMkvWritingApp, "android")); + return new WebmMaster(kMkvInfo, segmentInfo); +} + +sp<WebmElement> WebmElement::AudioTrackEntry( + int chans, + double rate, + const sp<ABuffer> &buf, + int bps, + uint64_t uid, + bool lacing, + const char *lang) { + if (uid == 0) { + uid = kAudioTrackNum; + } + + List<sp<WebmElement> > trackEntryFields; + populateCommonTrackEntries( + kAudioTrackNum, + uid, + lacing, + lang, + "A_VORBIS", + kAudioType, + trackEntryFields); + + List<sp<WebmElement> > audioInfo; + audioInfo.push_back(new WebmUnsigned(kMkvChannels, chans)); + audioInfo.push_back(new WebmFloat(kMkvSamplingFrequency, rate)); + if (bps) { + WebmElement *bitDepth = new WebmUnsigned(kMkvBitDepth, bps); + audioInfo.push_back(bitDepth); + } + + trackEntryFields.push_back(new WebmMaster(kMkvAudio, audioInfo)); + trackEntryFields.push_back(new WebmBinary(kMkvCodecPrivate, buf)); + return new WebmMaster(kMkvTrackEntry, trackEntryFields); +} + +sp<WebmElement> WebmElement::VideoTrackEntry( + uint64_t width, + uint64_t height, + uint64_t uid, + bool lacing, + const char *lang) { + if (uid == 0) { + uid = kVideoTrackNum; + } + + List<sp<WebmElement> > trackEntryFields; + populateCommonTrackEntries( + kVideoTrackNum, + uid, + lacing, + lang, + "V_VP8", + kVideoType, + trackEntryFields); + + List<sp<WebmElement> > videoInfo; + videoInfo.push_back(new WebmUnsigned(kMkvPixelWidth, width)); + videoInfo.push_back(new WebmUnsigned(kMkvPixelHeight, height)); + + trackEntryFields.push_back(new WebmMaster(kMkvVideo, videoInfo)); + return new WebmMaster(kMkvTrackEntry, trackEntryFields); +} +} /* namespace android */ diff --git a/media/libstagefright/webm/WebmElement.h b/media/libstagefright/webm/WebmElement.h new file mode 100644 index 0000000..f19933e --- /dev/null +++ b/media/libstagefright/webm/WebmElement.h @@ -0,0 +1,127 @@ +/* + * Copyright (C) 2014 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 WEBMELEMENT_H_ +#define WEBMELEMENT_H_ + +#include <media/stagefright/MediaBuffer.h> +#include <media/stagefright/foundation/ABase.h> +#include <media/stagefright/foundation/ABuffer.h> +#include <utils/List.h> + +namespace android { + +struct WebmElement : public LightRefBase<WebmElement> { + const uint64_t mId, mSize; + + WebmElement(uint64_t id, uint64_t size); + virtual ~WebmElement(); + + virtual int serializePayloadSize(uint8_t *buf); + virtual void serializePayload(uint8_t *buf)=0; + uint64_t totalSize(); + uint64_t serializeInto(uint8_t *buf); + uint8_t *serialize(uint64_t& size); + int write(int fd, uint64_t& size); + + static sp<WebmElement> EbmlHeader( + int ver = 1, + int readVer = 1, + int maxIdLen = 4, + int maxSizeLen = 8, + int docVer = 2, + int docReadVer = 2); + + static sp<WebmElement> SegmentInfo(uint64_t scale = 1000000, double dur = 0); + + static sp<WebmElement> AudioTrackEntry( + int chans, + double rate, + const sp<ABuffer> &buf, + int bps = 0, + uint64_t uid = 0, + bool lacing = false, + const char *lang = "und"); + + static sp<WebmElement> VideoTrackEntry( + uint64_t width, + uint64_t height, + uint64_t uid = 0, + bool lacing = false, + const char *lang = "und"); + + static sp<WebmElement> SeekEntry(uint64_t id, uint64_t off); + static sp<WebmElement> CuePointEntry(uint64_t time, int track, uint64_t off); + static sp<WebmElement> SimpleBlock( + int trackNum, + int16_t timecode, + bool key, + const uint8_t *data, + uint64_t dataSize); +}; + +struct WebmUnsigned : public WebmElement { + WebmUnsigned(uint64_t id, uint64_t value); + const uint64_t mValue; + void serializePayload(uint8_t *buf); +}; + +struct WebmFloat : public WebmElement { + const double mValue; + WebmFloat(uint64_t id, float value); + WebmFloat(uint64_t id, double value); + void serializePayload(uint8_t *buf); +}; + +struct WebmBinary : public WebmElement { + const sp<ABuffer> mRef; + WebmBinary(uint64_t id, const sp<ABuffer> &ref); + void serializePayload(uint8_t *buf); +}; + +struct WebmString : public WebmElement { + const char *const mStr; + WebmString(uint64_t id, const char *str); + void serializePayload(uint8_t *buf); +}; + +struct WebmSimpleBlock : public WebmElement { + const int mTrackNum; + const int16_t mRelTimecode; + const bool mKey; + const sp<ABuffer> mRef; + + WebmSimpleBlock(int trackNum, int16_t timecode, bool key, const sp<ABuffer>& orig); + void serializePayload(uint8_t *buf); +}; + +struct EbmlVoid : public WebmElement { + const uint64_t mSizeWidth; + EbmlVoid(uint64_t totalSize); + int serializePayloadSize(uint8_t *buf); + void serializePayload(uint8_t *buf); +}; + +struct WebmMaster : public WebmElement { + const List<sp<WebmElement> > mChildren; + WebmMaster(uint64_t id); + WebmMaster(uint64_t id, const List<sp<WebmElement> > &children); + int serializePayloadSize(uint8_t *buf); + void serializePayload(uint8_t *buf); +}; + +} /* namespace android */ +#endif /* WEBMELEMENT_H_ */ diff --git a/media/libstagefright/webm/WebmFrame.cpp b/media/libstagefright/webm/WebmFrame.cpp new file mode 100644 index 0000000..e5134ed --- /dev/null +++ b/media/libstagefright/webm/WebmFrame.cpp @@ -0,0 +1,83 @@ +/* + * Copyright (C) 2014 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 "WebmFrame" + +#include "WebmFrame.h" +#include "WebmConstants.h" + +#include <media/stagefright/foundation/ADebug.h> +#include <unistd.h> + +using namespace android; +using namespace webm; + +namespace { +sp<ABuffer> toABuffer(MediaBuffer *mbuf) { + sp<ABuffer> abuf = new ABuffer(mbuf->range_length()); + memcpy(abuf->data(), (uint8_t*) mbuf->data() + mbuf->range_offset(), mbuf->range_length()); + return abuf; +} +} + +namespace android { + +const sp<WebmFrame> WebmFrame::EOS = new WebmFrame(); + +WebmFrame::WebmFrame() + : mType(kInvalidType), + mKey(false), + mAbsTimecode(UINT64_MAX), + mData(new ABuffer(0)), + mEos(true) { +} + +WebmFrame::WebmFrame(int type, bool key, uint64_t absTimecode, MediaBuffer *mbuf) + : mType(type), + mKey(key), + mAbsTimecode(absTimecode), + mData(toABuffer(mbuf)), + mEos(false) { +} + +sp<WebmElement> WebmFrame::SimpleBlock(uint64_t baseTimecode) const { + return new WebmSimpleBlock( + mType == kVideoType ? kVideoTrackNum : kAudioTrackNum, + mAbsTimecode - baseTimecode, + mKey, + mData); +} + +bool WebmFrame::operator<(const WebmFrame &other) const { + if (this->mEos) { + return false; + } + if (other.mEos) { + return true; + } + if (this->mAbsTimecode == other.mAbsTimecode) { + if (this->mType == kAudioType && other.mType == kVideoType) { + return true; + } + if (this->mType == kVideoType && other.mType == kAudioType) { + return false; + } + return false; + } + return this->mAbsTimecode < other.mAbsTimecode; +} +} /* namespace android */ diff --git a/media/libstagefright/webm/WebmFrame.h b/media/libstagefright/webm/WebmFrame.h new file mode 100644 index 0000000..4f0b055 --- /dev/null +++ b/media/libstagefright/webm/WebmFrame.h @@ -0,0 +1,46 @@ +/* + * Copyright (C) 2014 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 WEBMFRAME_H_ +#define WEBMFRAME_H_ + +#include "WebmElement.h" + +namespace android { + +struct WebmFrame : LightRefBase<WebmFrame> { +public: + const int mType; + const bool mKey; + const uint64_t mAbsTimecode; + const sp<ABuffer> mData; + const bool mEos; + + WebmFrame(); + WebmFrame(int type, bool key, uint64_t absTimecode, MediaBuffer *buf); + ~WebmFrame() {} + + sp<WebmElement> SimpleBlock(uint64_t baseTimecode) const; + + bool operator<(const WebmFrame &other) const; + + static const sp<WebmFrame> EOS; +private: + DISALLOW_EVIL_CONSTRUCTORS(WebmFrame); +}; + +} /* namespace android */ +#endif /* WEBMFRAME_H_ */ diff --git a/media/libstagefright/webm/WebmFrameThread.cpp b/media/libstagefright/webm/WebmFrameThread.cpp new file mode 100644 index 0000000..5addd3c --- /dev/null +++ b/media/libstagefright/webm/WebmFrameThread.cpp @@ -0,0 +1,399 @@ +/* + * Copyright (C) 2014 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 "WebmFrameThread" + +#include "WebmConstants.h" +#include "WebmFrameThread.h" + +#include <media/stagefright/MetaData.h> +#include <media/stagefright/foundation/ADebug.h> + +#include <utils/Log.h> +#include <inttypes.h> + +using namespace webm; + +namespace android { + +void *WebmFrameThread::wrap(void *arg) { + WebmFrameThread *worker = reinterpret_cast<WebmFrameThread*>(arg); + worker->run(); + return NULL; +} + +status_t WebmFrameThread::start() { + pthread_attr_t attr; + pthread_attr_init(&attr); + pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE); + pthread_create(&mThread, &attr, WebmFrameThread::wrap, this); + pthread_attr_destroy(&attr); + return OK; +} + +status_t WebmFrameThread::stop() { + void *status; + pthread_join(mThread, &status); + return (status_t) status; +} + +//================================================================================================= + +WebmFrameSourceThread::WebmFrameSourceThread( + int type, + LinkedBlockingQueue<const sp<WebmFrame> >& sink) + : mType(type), mSink(sink) { +} + +//================================================================================================= + +WebmFrameSinkThread::WebmFrameSinkThread( + const int& fd, + const uint64_t& off, + sp<WebmFrameSourceThread> videoThread, + sp<WebmFrameSourceThread> audioThread, + List<sp<WebmElement> >& cues) + : mFd(fd), + mSegmentDataStart(off), + mVideoFrames(videoThread->mSink), + mAudioFrames(audioThread->mSink), + mCues(cues), + mDone(true) { +} + +WebmFrameSinkThread::WebmFrameSinkThread( + const int& fd, + const uint64_t& off, + LinkedBlockingQueue<const sp<WebmFrame> >& videoSource, + LinkedBlockingQueue<const sp<WebmFrame> >& audioSource, + List<sp<WebmElement> >& cues) + : mFd(fd), + mSegmentDataStart(off), + mVideoFrames(videoSource), + mAudioFrames(audioSource), + mCues(cues), + mDone(true) { +} + +// Initializes a webm cluster with its starting timecode. +// +// frames: +// sequence of input audio/video frames received from the source. +// +// clusterTimecodeL: +// the starting timecode of the cluster; this is the timecode of the first +// frame since frames are ordered by timestamp. +// +// children: +// list to hold child elements in a webm cluster (start timecode and +// simple blocks). +// +// static +void WebmFrameSinkThread::initCluster( + List<const sp<WebmFrame> >& frames, + uint64_t& clusterTimecodeL, + List<sp<WebmElement> >& children) { + CHECK(!frames.empty() && children.empty()); + + const sp<WebmFrame> f = *(frames.begin()); + clusterTimecodeL = f->mAbsTimecode; + WebmUnsigned *clusterTimecode = new WebmUnsigned(kMkvTimecode, clusterTimecodeL); + children.clear(); + children.push_back(clusterTimecode); +} + +void WebmFrameSinkThread::writeCluster(List<sp<WebmElement> >& children) { + // children must contain at least one simpleblock and its timecode + CHECK_GE(children.size(), 2); + + uint64_t size; + sp<WebmElement> cluster = new WebmMaster(kMkvCluster, children); + cluster->write(mFd, size); + children.clear(); +} + +// Write out (possibly multiple) webm cluster(s) from frames split on video key frames. +// +// last: +// current flush is triggered by EOS instead of a second outstanding video key frame. +void WebmFrameSinkThread::flushFrames(List<const sp<WebmFrame> >& frames, bool last) { + if (frames.empty()) { + return; + } + + uint64_t clusterTimecodeL; + List<sp<WebmElement> > children; + initCluster(frames, clusterTimecodeL, children); + + uint64_t cueTime = clusterTimecodeL; + off_t fpos = ::lseek(mFd, 0, SEEK_CUR); + size_t n = frames.size(); + if (!last) { + // If we are not flushing the last sequence of outstanding frames, flushFrames + // must have been called right after we have pushed a second outstanding video key + // frame (the last frame), which belongs to the next cluster; also hold back on + // flushing the second to last frame before we check its type. A audio frame + // should precede the aforementioned video key frame in the next sequence, a video + // frame should be the last frame in the current (to-be-flushed) sequence. + CHECK_GE(n, 2); + n -= 2; + } + + for (size_t i = 0; i < n; i++) { + const sp<WebmFrame> f = *(frames.begin()); + if (f->mType == kVideoType && f->mKey) { + cueTime = f->mAbsTimecode; + } + + if (f->mAbsTimecode - clusterTimecodeL > INT16_MAX) { + writeCluster(children); + initCluster(frames, clusterTimecodeL, children); + } + + frames.erase(frames.begin()); + children.push_back(f->SimpleBlock(clusterTimecodeL)); + } + + // equivalent to last==false + if (!frames.empty()) { + // decide whether to write out the second to last frame. + const sp<WebmFrame> secondLastFrame = *(frames.begin()); + if (secondLastFrame->mType == kVideoType) { + frames.erase(frames.begin()); + children.push_back(secondLastFrame->SimpleBlock(clusterTimecodeL)); + } + } + + writeCluster(children); + sp<WebmElement> cuePoint = WebmElement::CuePointEntry(cueTime, 1, fpos - mSegmentDataStart); + mCues.push_back(cuePoint); +} + +status_t WebmFrameSinkThread::start() { + mDone = false; + return WebmFrameThread::start(); +} + +status_t WebmFrameSinkThread::stop() { + mDone = true; + mVideoFrames.push(WebmFrame::EOS); + mAudioFrames.push(WebmFrame::EOS); + return WebmFrameThread::stop(); +} + +void WebmFrameSinkThread::run() { + int numVideoKeyFrames = 0; + List<const sp<WebmFrame> > outstandingFrames; + while (!mDone) { + ALOGV("wait v frame"); + const sp<WebmFrame> videoFrame = mVideoFrames.peek(); + ALOGV("v frame: %p", videoFrame.get()); + + ALOGV("wait a frame"); + const sp<WebmFrame> audioFrame = mAudioFrames.peek(); + ALOGV("a frame: %p", audioFrame.get()); + + if (videoFrame->mEos && audioFrame->mEos) { + break; + } + + if (*audioFrame < *videoFrame) { + ALOGV("take a frame"); + mAudioFrames.take(); + outstandingFrames.push_back(audioFrame); + } else { + ALOGV("take v frame"); + mVideoFrames.take(); + outstandingFrames.push_back(videoFrame); + if (videoFrame->mKey) + numVideoKeyFrames++; + } + + if (numVideoKeyFrames == 2) { + flushFrames(outstandingFrames, /* last = */ false); + numVideoKeyFrames--; + } + } + ALOGV("flushing last cluster (size %zu)", outstandingFrames.size()); + flushFrames(outstandingFrames, /* last = */ true); + mDone = true; +} + +//================================================================================================= + +static const int64_t kInitialDelayTimeUs = 700000LL; + +void WebmFrameMediaSourceThread::clearFlags() { + mDone = false; + mPaused = false; + mResumed = false; + mStarted = false; + mReachedEOS = false; +} + +WebmFrameMediaSourceThread::WebmFrameMediaSourceThread( + const sp<MediaSource>& source, + int type, + LinkedBlockingQueue<const sp<WebmFrame> >& sink, + uint64_t timeCodeScale, + int64_t startTimeRealUs, + int32_t startTimeOffsetMs, + int numTracks, + bool realTimeRecording) + : WebmFrameSourceThread(type, sink), + mSource(source), + mTimeCodeScale(timeCodeScale), + mTrackDurationUs(0) { + clearFlags(); + mStartTimeUs = startTimeRealUs; + if (realTimeRecording && numTracks > 1) { + /* + * Copied from MPEG4Writer + * + * This extra delay of accepting incoming audio/video signals + * helps to align a/v start time at the beginning of a recording + * session, and it also helps eliminate the "recording" sound for + * camcorder applications. + * + * If client does not set the start time offset, we fall back to + * use the default initial delay value. + */ + int64_t startTimeOffsetUs = startTimeOffsetMs * 1000LL; + if (startTimeOffsetUs < 0) { // Start time offset was not set + startTimeOffsetUs = kInitialDelayTimeUs; + } + mStartTimeUs += startTimeOffsetUs; + ALOGI("Start time offset: %" PRId64 " us", startTimeOffsetUs); + } +} + +status_t WebmFrameMediaSourceThread::start() { + sp<MetaData> meta = new MetaData; + meta->setInt64(kKeyTime, mStartTimeUs); + status_t err = mSource->start(meta.get()); + if (err != OK) { + mDone = true; + mReachedEOS = true; + return err; + } else { + mStarted = true; + return WebmFrameThread::start(); + } +} + +status_t WebmFrameMediaSourceThread::resume() { + if (!mDone && mPaused) { + mPaused = false; + mResumed = true; + } + return OK; +} + +status_t WebmFrameMediaSourceThread::pause() { + if (mStarted) { + mPaused = true; + } + return OK; +} + +status_t WebmFrameMediaSourceThread::stop() { + if (mStarted) { + mStarted = false; + mDone = true; + mSource->stop(); + return WebmFrameThread::stop(); + } + return OK; +} + +void WebmFrameMediaSourceThread::run() { + int32_t count = 0; + int64_t timestampUs = 0xdeadbeef; + int64_t lastTimestampUs = 0; // Previous sample time stamp + int64_t lastDurationUs = 0; // Previous sample duration + int64_t previousPausedDurationUs = 0; + + const uint64_t kUninitialized = 0xffffffffffffffffL; + mStartTimeUs = kUninitialized; + + status_t err = OK; + MediaBuffer *buffer; + while (!mDone && (err = mSource->read(&buffer, NULL)) == OK) { + if (buffer->range_length() == 0) { + buffer->release(); + buffer = NULL; + continue; + } + + sp<MetaData> md = buffer->meta_data(); + CHECK(md->findInt64(kKeyTime, ×tampUs)); + if (mStartTimeUs == kUninitialized) { + mStartTimeUs = timestampUs; + } + timestampUs -= mStartTimeUs; + + if (mPaused && !mResumed) { + lastDurationUs = timestampUs - lastTimestampUs; + lastTimestampUs = timestampUs; + buffer->release(); + buffer = NULL; + continue; + } + ++count; + + // adjust time-stamps after pause/resume + if (mResumed) { + int64_t durExcludingEarlierPausesUs = timestampUs - previousPausedDurationUs; + CHECK_GE(durExcludingEarlierPausesUs, 0ll); + int64_t pausedDurationUs = durExcludingEarlierPausesUs - mTrackDurationUs; + CHECK_GE(pausedDurationUs, lastDurationUs); + previousPausedDurationUs += pausedDurationUs - lastDurationUs; + mResumed = false; + } + timestampUs -= previousPausedDurationUs; + CHECK_GE(timestampUs, 0ll); + + int32_t isSync = false; + md->findInt32(kKeyIsSyncFrame, &isSync); + const sp<WebmFrame> f = new WebmFrame( + mType, + isSync, + timestampUs * 1000 / mTimeCodeScale, + buffer); + mSink.push(f); + + ALOGV( + "%s %s frame at %" PRId64 " size %zu\n", + mType == kVideoType ? "video" : "audio", + isSync ? "I" : "P", + timestampUs * 1000 / mTimeCodeScale, + buffer->range_length()); + + buffer->release(); + buffer = NULL; + + if (timestampUs > mTrackDurationUs) { + mTrackDurationUs = timestampUs; + } + lastDurationUs = timestampUs - lastTimestampUs; + lastTimestampUs = timestampUs; + } + + mTrackDurationUs += lastDurationUs; + mSink.push(WebmFrame::EOS); +} +} diff --git a/media/libstagefright/webm/WebmFrameThread.h b/media/libstagefright/webm/WebmFrameThread.h new file mode 100644 index 0000000..d65d9b7 --- /dev/null +++ b/media/libstagefright/webm/WebmFrameThread.h @@ -0,0 +1,160 @@ +/* + * Copyright (C) 2014 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 WEBMFRAMETHREAD_H_ +#define WEBMFRAMETHREAD_H_ + +#include "WebmFrame.h" +#include "LinkedBlockingQueue.h" + +#include <media/stagefright/FileSource.h> +#include <media/stagefright/MediaSource.h> + +#include <utils/List.h> +#include <utils/Errors.h> + +#include <pthread.h> + +namespace android { + +class WebmFrameThread : public LightRefBase<WebmFrameThread> { +public: + virtual void run() = 0; + virtual bool running() { return false; } + virtual status_t start(); + virtual status_t pause() { return OK; } + virtual status_t resume() { return OK; } + virtual status_t stop(); + virtual ~WebmFrameThread() { stop(); } + static void *wrap(void *arg); + +protected: + WebmFrameThread() + : mThread(0) { + } + +private: + pthread_t mThread; + DISALLOW_EVIL_CONSTRUCTORS(WebmFrameThread); +}; + +//================================================================================================= + +class WebmFrameSourceThread; +class WebmFrameSinkThread : public WebmFrameThread { +public: + WebmFrameSinkThread( + const int& fd, + const uint64_t& off, + sp<WebmFrameSourceThread> videoThread, + sp<WebmFrameSourceThread> audioThread, + List<sp<WebmElement> >& cues); + + WebmFrameSinkThread( + const int& fd, + const uint64_t& off, + LinkedBlockingQueue<const sp<WebmFrame> >& videoSource, + LinkedBlockingQueue<const sp<WebmFrame> >& audioSource, + List<sp<WebmElement> >& cues); + + void run(); + bool running() { + return !mDone; + } + status_t start(); + status_t stop(); + +private: + const int& mFd; + const uint64_t& mSegmentDataStart; + LinkedBlockingQueue<const sp<WebmFrame> >& mVideoFrames; + LinkedBlockingQueue<const sp<WebmFrame> >& mAudioFrames; + List<sp<WebmElement> >& mCues; + + volatile bool mDone; + + static void initCluster( + List<const sp<WebmFrame> >& frames, + uint64_t& clusterTimecodeL, + List<sp<WebmElement> >& children); + void writeCluster(List<sp<WebmElement> >& children); + void flushFrames(List<const sp<WebmFrame> >& frames, bool last); +}; + +//================================================================================================= + +class WebmFrameSourceThread : public WebmFrameThread { +public: + WebmFrameSourceThread(int type, LinkedBlockingQueue<const sp<WebmFrame> >& sink); + virtual int64_t getDurationUs() = 0; +protected: + const int mType; + LinkedBlockingQueue<const sp<WebmFrame> >& mSink; + + friend class WebmFrameSinkThread; +}; + +//================================================================================================= + +class WebmFrameEmptySourceThread : public WebmFrameSourceThread { +public: + WebmFrameEmptySourceThread(int type, LinkedBlockingQueue<const sp<WebmFrame> >& sink) + : WebmFrameSourceThread(type, sink) { + } + void run() { mSink.push(WebmFrame::EOS); } + int64_t getDurationUs() { return 0; } +}; + +//================================================================================================= + +class WebmFrameMediaSourceThread: public WebmFrameSourceThread { +public: + WebmFrameMediaSourceThread( + const sp<MediaSource>& source, + int type, + LinkedBlockingQueue<const sp<WebmFrame> >& sink, + uint64_t timeCodeScale, + int64_t startTimeRealUs, + int32_t startTimeOffsetMs, + int numPeers, + bool realTimeRecording); + + void run(); + status_t start(); + status_t resume(); + status_t pause(); + status_t stop(); + int64_t getDurationUs() { + return mTrackDurationUs; + } + +private: + const sp<MediaSource> mSource; + const uint64_t mTimeCodeScale; + uint64_t mStartTimeUs; + + volatile bool mDone; + volatile bool mPaused; + volatile bool mResumed; + volatile bool mStarted; + volatile bool mReachedEOS; + int64_t mTrackDurationUs; + + void clearFlags(); +}; +} /* namespace android */ + +#endif /* WEBMFRAMETHREAD_H_ */ diff --git a/media/libstagefright/webm/WebmWriter.cpp b/media/libstagefright/webm/WebmWriter.cpp new file mode 100644 index 0000000..03cf92a --- /dev/null +++ b/media/libstagefright/webm/WebmWriter.cpp @@ -0,0 +1,551 @@ +/* + * Copyright (C) 2014 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 "WebmWriter" + +#include "EbmlUtil.h" +#include "WebmWriter.h" + +#include <media/stagefright/MetaData.h> +#include <media/stagefright/MediaDefs.h> +#include <media/stagefright/foundation/ADebug.h> + +#include <utils/Errors.h> + +#include <unistd.h> +#include <fcntl.h> +#include <sys/stat.h> +#include <inttypes.h> + +using namespace webm; + +namespace { +size_t XiphLaceCodeLen(size_t size) { + return size / 0xff + 1; +} + +size_t XiphLaceEnc(uint8_t *buf, size_t size) { + size_t i; + for (i = 0; size >= 0xff; ++i, size -= 0xff) { + buf[i] = 0xff; + } + buf[i++] = size; + return i; +} +} + +namespace android { + +static const int64_t kMinStreamableFileSizeInBytes = 5 * 1024 * 1024; + +WebmWriter::WebmWriter(int fd) + : mFd(dup(fd)), + mInitCheck(mFd < 0 ? NO_INIT : OK), + mTimeCodeScale(1000000), + mStartTimestampUs(0), + mStartTimeOffsetMs(0), + mSegmentOffset(0), + mSegmentDataStart(0), + mInfoOffset(0), + mInfoSize(0), + mTracksOffset(0), + mCuesOffset(0), + mPaused(false), + mStarted(false), + mIsFileSizeLimitExplicitlyRequested(false), + mIsRealTimeRecording(false), + mStreamableFile(true), + mEstimatedCuesSize(0) { + mStreams[kAudioIndex] = WebmStream(kAudioType, "Audio", &WebmWriter::audioTrack); + mStreams[kVideoIndex] = WebmStream(kVideoType, "Video", &WebmWriter::videoTrack); + mSinkThread = new WebmFrameSinkThread( + mFd, + mSegmentDataStart, + mStreams[kVideoIndex].mSink, + mStreams[kAudioIndex].mSink, + mCuePoints); +} + +WebmWriter::WebmWriter(const char *filename) + : mInitCheck(NO_INIT), + mTimeCodeScale(1000000), + mStartTimestampUs(0), + mStartTimeOffsetMs(0), + mSegmentOffset(0), + mSegmentDataStart(0), + mInfoOffset(0), + mInfoSize(0), + mTracksOffset(0), + mCuesOffset(0), + mPaused(false), + mStarted(false), + mIsFileSizeLimitExplicitlyRequested(false), + mIsRealTimeRecording(false), + mStreamableFile(true), + mEstimatedCuesSize(0) { + mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); + if (mFd >= 0) { + ALOGV("fd %d; flags: %o", mFd, fcntl(mFd, F_GETFL, 0)); + mInitCheck = OK; + } + mStreams[kAudioIndex] = WebmStream(kAudioType, "Audio", &WebmWriter::audioTrack); + mStreams[kVideoIndex] = WebmStream(kVideoType, "Video", &WebmWriter::videoTrack); + mSinkThread = new WebmFrameSinkThread( + mFd, + mSegmentDataStart, + mStreams[kVideoIndex].mSink, + mStreams[kAudioIndex].mSink, + mCuePoints); +} + +// static +sp<WebmElement> WebmWriter::videoTrack(const sp<MetaData>& md) { + int32_t width, height; + CHECK(md->findInt32(kKeyWidth, &width)); + CHECK(md->findInt32(kKeyHeight, &height)); + return WebmElement::VideoTrackEntry(width, height); +} + +// static +sp<WebmElement> WebmWriter::audioTrack(const sp<MetaData>& md) { + int32_t nChannels, samplerate; + uint32_t type; + const void *headerData1; + const char headerData2[] = { 3, 'v', 'o', 'r', 'b', 'i', 's', 7, 0, 0, 0, + 'a', 'n', 'd', 'r', 'o', 'i', 'd', 0, 0, 0, 0, 1 }; + const void *headerData3; + size_t headerSize1, headerSize2 = sizeof(headerData2), headerSize3; + + CHECK(md->findInt32(kKeyChannelCount, &nChannels)); + CHECK(md->findInt32(kKeySampleRate, &samplerate)); + CHECK(md->findData(kKeyVorbisInfo, &type, &headerData1, &headerSize1)); + CHECK(md->findData(kKeyVorbisBooks, &type, &headerData3, &headerSize3)); + + size_t codecPrivateSize = 1; + codecPrivateSize += XiphLaceCodeLen(headerSize1); + codecPrivateSize += XiphLaceCodeLen(headerSize2); + codecPrivateSize += headerSize1 + headerSize2 + headerSize3; + + off_t off = 0; + sp<ABuffer> codecPrivateBuf = new ABuffer(codecPrivateSize); + uint8_t *codecPrivateData = codecPrivateBuf->data(); + codecPrivateData[off++] = 2; + + off += XiphLaceEnc(codecPrivateData + off, headerSize1); + off += XiphLaceEnc(codecPrivateData + off, headerSize2); + + memcpy(codecPrivateData + off, headerData1, headerSize1); + off += headerSize1; + memcpy(codecPrivateData + off, headerData2, headerSize2); + off += headerSize2; + memcpy(codecPrivateData + off, headerData3, headerSize3); + + sp<WebmElement> entry = WebmElement::AudioTrackEntry( + nChannels, + samplerate, + codecPrivateBuf); + return entry; +} + +size_t WebmWriter::numTracks() { + Mutex::Autolock autolock(mLock); + + size_t numTracks = 0; + for (size_t i = 0; i < kMaxStreams; ++i) { + if (mStreams[i].mTrackEntry != NULL) { + numTracks++; + } + } + + return numTracks; +} + +uint64_t WebmWriter::estimateCuesSize(int32_t bitRate) { + // This implementation is based on estimateMoovBoxSize in MPEG4Writer. + // + // Statistical analysis shows that metadata usually accounts + // for a small portion of the total file size, usually < 0.6%. + + // The default MIN_MOOV_BOX_SIZE is set to 0.6% x 1MB / 2, + // where 1MB is the common file size limit for MMS application. + // The default MAX _MOOV_BOX_SIZE value is based on about 3 + // minute video recording with a bit rate about 3 Mbps, because + // statistics also show that most of the video captured are going + // to be less than 3 minutes. + + // If the estimation is wrong, we will pay the price of wasting + // some reserved space. This should not happen so often statistically. + static const int32_t factor = 2; + static const int64_t MIN_CUES_SIZE = 3 * 1024; // 3 KB + static const int64_t MAX_CUES_SIZE = (180 * 3000000 * 6LL / 8000); + int64_t size = MIN_CUES_SIZE; + + // Max file size limit is set + if (mMaxFileSizeLimitBytes != 0 && mIsFileSizeLimitExplicitlyRequested) { + size = mMaxFileSizeLimitBytes * 6 / 1000; + } + + // Max file duration limit is set + if (mMaxFileDurationLimitUs != 0) { + if (bitRate > 0) { + int64_t size2 = ((mMaxFileDurationLimitUs * bitRate * 6) / 1000 / 8000000); + if (mMaxFileSizeLimitBytes != 0 && mIsFileSizeLimitExplicitlyRequested) { + // When both file size and duration limits are set, + // we use the smaller limit of the two. + if (size > size2) { + size = size2; + } + } else { + // Only max file duration limit is set + size = size2; + } + } + } + + if (size < MIN_CUES_SIZE) { + size = MIN_CUES_SIZE; + } + + // Any long duration recording will be probably end up with + // non-streamable webm file. + if (size > MAX_CUES_SIZE) { + size = MAX_CUES_SIZE; + } + + ALOGV("limits: %" PRId64 "/%" PRId64 " bytes/us," + " bit rate: %d bps and the estimated cues size %" PRId64 " bytes", + mMaxFileSizeLimitBytes, mMaxFileDurationLimitUs, bitRate, size); + return factor * size; +} + +void WebmWriter::initStream(size_t idx) { + if (mStreams[idx].mThread != NULL) { + return; + } + if (mStreams[idx].mSource == NULL) { + ALOGV("adding dummy source ... "); + mStreams[idx].mThread = new WebmFrameEmptySourceThread( + mStreams[idx].mType, mStreams[idx].mSink); + } else { + ALOGV("adding source %p", mStreams[idx].mSource.get()); + mStreams[idx].mThread = new WebmFrameMediaSourceThread( + mStreams[idx].mSource, + mStreams[idx].mType, + mStreams[idx].mSink, + mTimeCodeScale, + mStartTimestampUs, + mStartTimeOffsetMs, + numTracks(), + mIsRealTimeRecording); + } +} + +void WebmWriter::release() { + close(mFd); + mFd = -1; + mInitCheck = NO_INIT; + mStarted = false; +} + +status_t WebmWriter::reset() { + if (mInitCheck != OK) { + return OK; + } else { + if (!mStarted) { + release(); + return OK; + } + } + + status_t err = OK; + int64_t maxDurationUs = 0; + int64_t minDurationUs = 0x7fffffffffffffffLL; + for (int i = 0; i < kMaxStreams; ++i) { + if (mStreams[i].mThread == NULL) { + continue; + } + + status_t status = mStreams[i].mThread->stop(); + if (err == OK && status != OK) { + err = status; + } + + int64_t durationUs = mStreams[i].mThread->getDurationUs(); + if (durationUs > maxDurationUs) { + maxDurationUs = durationUs; + } + if (durationUs < minDurationUs) { + minDurationUs = durationUs; + } + } + + if (numTracks() > 1) { + ALOGD("Duration from tracks range is [%" PRId64 ", %" PRId64 "] us", minDurationUs, maxDurationUs); + } + + mSinkThread->stop(); + + // Do not write out movie header on error. + if (err != OK) { + release(); + return err; + } + + sp<WebmElement> cues = new WebmMaster(kMkvCues, mCuePoints); + uint64_t cuesSize = cues->totalSize(); + // TRICKY Even when the cues do fit in the space we reserved, if they do not fit + // perfectly, we still need to check if there is enough "extra space" to write an + // EBML void element. + if (cuesSize != mEstimatedCuesSize && cuesSize > mEstimatedCuesSize - kMinEbmlVoidSize) { + mCuesOffset = ::lseek(mFd, 0, SEEK_CUR); + cues->write(mFd, cuesSize); + } else { + uint64_t spaceSize; + ::lseek(mFd, mCuesOffset, SEEK_SET); + cues->write(mFd, cuesSize); + sp<WebmElement> space = new EbmlVoid(mEstimatedCuesSize - cuesSize); + space->write(mFd, spaceSize); + } + + mCuePoints.clear(); + mStreams[kVideoIndex].mSink.clear(); + mStreams[kAudioIndex].mSink.clear(); + + uint8_t bary[sizeof(uint64_t)]; + uint64_t totalSize = ::lseek(mFd, 0, SEEK_END); + uint64_t segmentSize = totalSize - mSegmentDataStart; + ::lseek(mFd, mSegmentOffset + sizeOf(kMkvSegment), SEEK_SET); + uint64_t segmentSizeCoded = encodeUnsigned(segmentSize, sizeOf(kMkvUnknownLength)); + serializeCodedUnsigned(segmentSizeCoded, bary); + ::write(mFd, bary, sizeOf(kMkvUnknownLength)); + + uint64_t size; + uint64_t durationOffset = mInfoOffset + sizeOf(kMkvInfo) + sizeOf(mInfoSize) + + sizeOf(kMkvSegmentDuration) + sizeOf(sizeof(double)); + sp<WebmElement> duration = new WebmFloat( + kMkvSegmentDuration, + (double) (maxDurationUs * 1000 / mTimeCodeScale)); + duration->serializePayload(bary); + ::lseek(mFd, durationOffset, SEEK_SET); + ::write(mFd, bary, sizeof(double)); + + List<sp<WebmElement> > seekEntries; + seekEntries.push_back(WebmElement::SeekEntry(kMkvInfo, mInfoOffset - mSegmentDataStart)); + seekEntries.push_back(WebmElement::SeekEntry(kMkvTracks, mTracksOffset - mSegmentDataStart)); + seekEntries.push_back(WebmElement::SeekEntry(kMkvCues, mCuesOffset - mSegmentDataStart)); + sp<WebmElement> seekHead = new WebmMaster(kMkvSeekHead, seekEntries); + + uint64_t metaSeekSize; + ::lseek(mFd, mSegmentDataStart, SEEK_SET); + seekHead->write(mFd, metaSeekSize); + + uint64_t spaceSize; + sp<WebmElement> space = new EbmlVoid(kMaxMetaSeekSize - metaSeekSize); + space->write(mFd, spaceSize); + + release(); + return err; +} + +status_t WebmWriter::addSource(const sp<MediaSource> &source) { + Mutex::Autolock l(mLock); + if (mStarted) { + ALOGE("Attempt to add source AFTER recording is started"); + return UNKNOWN_ERROR; + } + + // At most 2 tracks can be supported. + if (mStreams[kVideoIndex].mTrackEntry != NULL + && mStreams[kAudioIndex].mTrackEntry != NULL) { + ALOGE("Too many tracks (2) to add"); + return ERROR_UNSUPPORTED; + } + + CHECK(source != NULL); + + // A track of type other than video or audio is not supported. + const char *mime; + source->getFormat()->findCString(kKeyMIMEType, &mime); + const char *vp8 = MEDIA_MIMETYPE_VIDEO_VP8; + const char *vorbis = MEDIA_MIMETYPE_AUDIO_VORBIS; + + size_t streamIndex; + if (!strncasecmp(mime, vp8, strlen(vp8))) { + streamIndex = kVideoIndex; + } else if (!strncasecmp(mime, vorbis, strlen(vorbis))) { + streamIndex = kAudioIndex; + } else { + ALOGE("Track (%s) other than %s or %s is not supported", mime, vp8, vorbis); + return ERROR_UNSUPPORTED; + } + + // No more than one video or one audio track is supported. + if (mStreams[streamIndex].mTrackEntry != NULL) { + ALOGE("%s track already exists", mStreams[streamIndex].mName); + return ERROR_UNSUPPORTED; + } + + // This is the first track of either audio or video. + // Go ahead to add the track. + mStreams[streamIndex].mSource = source; + mStreams[streamIndex].mTrackEntry = mStreams[streamIndex].mMakeTrack(source->getFormat()); + + return OK; +} + +status_t WebmWriter::start(MetaData *params) { + if (mInitCheck != OK) { + return UNKNOWN_ERROR; + } + + if (mStreams[kVideoIndex].mTrackEntry == NULL + && mStreams[kAudioIndex].mTrackEntry == NULL) { + ALOGE("No source added"); + return INVALID_OPERATION; + } + + if (mMaxFileSizeLimitBytes != 0) { + mIsFileSizeLimitExplicitlyRequested = true; + } + + if (params) { + int32_t isRealTimeRecording; + params->findInt32(kKeyRealTimeRecording, &isRealTimeRecording); + mIsRealTimeRecording = isRealTimeRecording; + } + + if (mStarted) { + if (mPaused) { + mPaused = false; + mStreams[kAudioIndex].mThread->resume(); + mStreams[kVideoIndex].mThread->resume(); + } + return OK; + } + + if (params) { + int32_t tcsl; + if (params->findInt32(kKeyTimeScale, &tcsl)) { + mTimeCodeScale = tcsl; + } + } + CHECK_GT(mTimeCodeScale, 0); + ALOGV("movie time scale: %" PRIu64, mTimeCodeScale); + + /* + * When the requested file size limit is small, the priority + * is to meet the file size limit requirement, rather than + * to make the file streamable. mStreamableFile does not tell + * whether the actual recorded file is streamable or not. + */ + mStreamableFile = (!mMaxFileSizeLimitBytes) + || (mMaxFileSizeLimitBytes >= kMinStreamableFileSizeInBytes); + + /* + * Write various metadata. + */ + sp<WebmElement> ebml, segment, info, seekHead, tracks, cues; + ebml = WebmElement::EbmlHeader(); + segment = new WebmMaster(kMkvSegment); + seekHead = new EbmlVoid(kMaxMetaSeekSize); + info = WebmElement::SegmentInfo(mTimeCodeScale, 0); + + List<sp<WebmElement> > children; + for (size_t i = 0; i < kMaxStreams; ++i) { + if (mStreams[i].mTrackEntry != NULL) { + children.push_back(mStreams[i].mTrackEntry); + } + } + tracks = new WebmMaster(kMkvTracks, children); + + if (!mStreamableFile) { + cues = NULL; + } else { + int32_t bitRate = -1; + if (params) { + params->findInt32(kKeyBitRate, &bitRate); + } + mEstimatedCuesSize = estimateCuesSize(bitRate); + CHECK_GE(mEstimatedCuesSize, 8); + cues = new EbmlVoid(mEstimatedCuesSize); + } + + sp<WebmElement> elems[] = { ebml, segment, seekHead, info, tracks, cues }; + size_t nElems = sizeof(elems) / sizeof(elems[0]); + uint64_t offsets[nElems]; + uint64_t sizes[nElems]; + for (uint32_t i = 0; i < nElems; i++) { + WebmElement *e = elems[i].get(); + if (!e) { + continue; + } + + uint64_t size; + offsets[i] = ::lseek(mFd, 0, SEEK_CUR); + sizes[i] = e->mSize; + e->write(mFd, size); + } + + mSegmentOffset = offsets[1]; + mSegmentDataStart = offsets[2]; + mInfoOffset = offsets[3]; + mInfoSize = sizes[3]; + mTracksOffset = offsets[4]; + mCuesOffset = offsets[5]; + + // start threads + if (params) { + params->findInt64(kKeyTime, &mStartTimestampUs); + } + + initStream(kAudioIndex); + initStream(kVideoIndex); + + mStreams[kAudioIndex].mThread->start(); + mStreams[kVideoIndex].mThread->start(); + mSinkThread->start(); + + mStarted = true; + return OK; +} + +status_t WebmWriter::pause() { + if (mInitCheck != OK) { + return OK; + } + mPaused = true; + status_t err = OK; + for (int i = 0; i < kMaxStreams; ++i) { + if (mStreams[i].mThread == NULL) { + continue; + } + status_t status = mStreams[i].mThread->pause(); + if (status != OK) { + err = status; + } + } + return err; +} + +status_t WebmWriter::stop() { + return reset(); +} + +bool WebmWriter::reachedEOS() { + return !mSinkThread->running(); +} +} /* namespace android */ diff --git a/media/libstagefright/webm/WebmWriter.h b/media/libstagefright/webm/WebmWriter.h new file mode 100644 index 0000000..529dec8 --- /dev/null +++ b/media/libstagefright/webm/WebmWriter.h @@ -0,0 +1,130 @@ +/* + * Copyright (C) 2014 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 WEBMWRITER_H_ +#define WEBMWRITER_H_ + +#include "WebmConstants.h" +#include "WebmFrameThread.h" +#include "LinkedBlockingQueue.h" + +#include <media/stagefright/MediaSource.h> +#include <media/stagefright/MediaWriter.h> + +#include <utils/Errors.h> +#include <utils/Mutex.h> +#include <utils/StrongPointer.h> + +#include <stdint.h> + +using namespace webm; + +namespace android { + +class WebmWriter : public MediaWriter { +public: + WebmWriter(int fd); + WebmWriter(const char *filename); + ~WebmWriter() { reset(); } + + + status_t addSource(const sp<MediaSource> &source); + status_t start(MetaData *param = NULL); + status_t stop(); + status_t pause(); + bool reachedEOS(); + + void setStartTimeOffsetMs(int ms) { mStartTimeOffsetMs = ms; } + int32_t getStartTimeOffsetMs() const { return mStartTimeOffsetMs; } + +private: + int mFd; + status_t mInitCheck; + + uint64_t mTimeCodeScale; + int64_t mStartTimestampUs; + int32_t mStartTimeOffsetMs; + + uint64_t mSegmentOffset; + uint64_t mSegmentDataStart; + uint64_t mInfoOffset; + uint64_t mInfoSize; + uint64_t mTracksOffset; + uint64_t mCuesOffset; + + bool mPaused; + bool mStarted; + bool mIsFileSizeLimitExplicitlyRequested; + bool mIsRealTimeRecording; + bool mStreamableFile; + uint64_t mEstimatedCuesSize; + + Mutex mLock; + List<sp<WebmElement> > mCuePoints; + + enum { + kAudioIndex = 0, + kVideoIndex = 1, + kMaxStreams = 2, + }; + + struct WebmStream { + int mType; + const char *mName; + sp<WebmElement> (*mMakeTrack)(const sp<MetaData>&); + + sp<MediaSource> mSource; + sp<WebmElement> mTrackEntry; + sp<WebmFrameSourceThread> mThread; + LinkedBlockingQueue<const sp<WebmFrame> > mSink; + + WebmStream() + : mType(kInvalidType), + mName("Invalid"), + mMakeTrack(NULL) { + } + + WebmStream(int type, const char *name, sp<WebmElement> (*makeTrack)(const sp<MetaData>&)) + : mType(type), + mName(name), + mMakeTrack(makeTrack) { + } + + WebmStream &operator=(const WebmStream &other) { + mType = other.mType; + mName = other.mName; + mMakeTrack = other.mMakeTrack; + return *this; + } + }; + WebmStream mStreams[kMaxStreams]; + + sp<WebmFrameSinkThread> mSinkThread; + + size_t numTracks(); + uint64_t estimateCuesSize(int32_t bitRate); + void initStream(size_t idx); + void release(); + status_t reset(); + + static sp<WebmElement> videoTrack(const sp<MetaData>& md); + static sp<WebmElement> audioTrack(const sp<MetaData>& md); + + DISALLOW_EVIL_CONSTRUCTORS(WebmWriter); +}; + +} /* namespace android */ +#endif /* WEBMWRITER_H_ */ |