diff --git a/miniaudio.h b/miniaudio.h index 008e26c7..54555416 100644 --- a/miniaudio.h +++ b/miniaudio.h @@ -1,6 +1,6 @@ /* Audio playback and capture library. Choice of public domain or MIT-0. See license statements at the end of this file. -miniaudio - v0.10.42 - 2021-08-22 +miniaudio - v0.10.42 - 2021-08-22 David Reid - mackron@gmail.com @@ -3286,7 +3286,7 @@ extern "C" { #define MA_VERSION_MAJOR 0 #define MA_VERSION_MINOR 10 -#define MA_VERSION_REVISION 42 +#define MA_VERSION_REVISION 42 #define MA_VERSION_STRING MA_XSTRINGIFY(MA_VERSION_MAJOR) "." MA_XSTRINGIFY(MA_VERSION_MINOR) "." MA_XSTRINGIFY(MA_VERSION_REVISION) #if defined(_MSC_VER) && !defined(__clang__) @@ -3302,6 +3302,23 @@ extern "C" { #endif #endif +#if defined (__STDC_VERSION__) && (__STDC_VERSION__ >= 201112L) + #include + #define MA_ALIGN_TYPE(n) alignas(n) + #define MA_ALIGN_MEMBER(align, type) MA_ALIGN_TYPE(align) type +#else + #if defined(__GNUC__) + #define MA_ALIGN_TYPE(n) __attribute__((aligned(n))) + #define MA_ALIGN_MEMBER(align, type) type MA_ALIGN_TYPE(align) + #elif defined(_MSC_VER) + #define MA_ALIGN_TYPE(n) __declspec(align(n)) + #define MA_ALIGN_MEMBER(align, type) MA_ALIGN_TYPE(align) type + #else + #define MA_ALIGN_TYPE(n) /* disabled */ + #define MA_ALIGN_MEMBER(align, type) /* disabled */ + #endif +#endif + /* Platform/backend detection. */ #ifdef _WIN32 #define MA_WIN32 @@ -8898,7 +8915,7 @@ typedef struct } breakup; ma_uint64 allocation; } toc; /* 8 bytes. We encode the job code into the slot allocation data to save space. */ - ma_uint64 next; /* refcount + slot for the next item. Does not include the job code. */ + MA_ATOMIC MA_ALIGN_MEMBER(8, ma_uint64) next; /* refcount + slot for the next item. Does not include the job code. */ ma_uint32 order; /* Execution order. Used to create a data dependency and ensure a job is executed in order. Usage is contextual depending on the job type. */ union @@ -9003,7 +9020,7 @@ typedef struct { ma_uint32 flags; /* Flags passed in at initialization time. */ ma_uint32 capacity; /* The maximum number of jobs that can fit in the queue at a time. Set by the config. */ - MA_ATOMIC ma_uint64 head; /* The first item in the list. Required for removing from the top of the list. */ + MA_ATOMIC MA_ALIGN_MEMBER(8, ma_uint64) head; /* The first item in the list. Required for removing from the top of the list. */ MA_ATOMIC ma_uint64 tail; /* The last item in the list. Required for appending to the end of the list. */ #ifndef MA_NO_THREADING ma_semaphore sem; /* Only used when MA_RESOURCE_MANAGER_JOB_QUEUE_FLAG_NON_BLOCKING is unset. */ @@ -9133,7 +9150,7 @@ struct ma_resource_manager_data_stream ma_bool32 isDecoderInitialized; /* Required for determining whether or not the decoder should be uninitialized in MA_RESOURCE_MANAGER_JOB_FREE_DATA_STREAM. */ ma_uint64 totalLengthInPCMFrames; /* This is calculated when first loaded by the MA_RESOURCE_MANAGER_JOB_LOAD_DATA_STREAM. */ ma_uint32 relativeCursor; /* The playback cursor, relative to the current page. Only ever accessed by the public API. Never accessed by the job thread. */ - ma_uint64 absoluteCursor; /* The playback cursor, in absolute position starting from the start of the file. */ + MA_ATOMIC MA_ALIGN_MEMBER(8, ma_uint64) absoluteCursor; /* The playback cursor, in absolute position starting from the start of the file. */ ma_uint32 currentPageIndex; /* Toggles between 0 and 1. Index 0 is the first half of pPageData. Index 1 is the second half. Only ever accessed by the public API. Never accessed by the job thread. */ MA_ATOMIC ma_uint32 executionCounter; /* For allocating execution orders for jobs. */ MA_ATOMIC ma_uint32 executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */ @@ -9423,8 +9440,8 @@ struct ma_node_base /* These variables are read and written between different threads. */ MA_ATOMIC ma_node_state state; /* When set to stopped, nothing will be read, regardless of the times in stateTimes. */ - MA_ATOMIC ma_uint64 stateTimes[2]; /* Indexed by ma_node_state. Specifies the time based on the global clock that a node should be considered to be in the relevant state. */ - MA_ATOMIC ma_uint64 localTime; /* The node's local clock. This is just a running sum of the number of output frames that have been processed. Can be modified by any thread with `ma_node_set_time()`. */ + MA_ATOMIC MA_ALIGN_MEMBER(8, ma_uint64) stateTimes[2]; /* Indexed by ma_node_state. Specifies the time based on the global clock that a node should be considered to be in the relevant state. */ + MA_ATOMIC MA_ALIGN_MEMBER(8, ma_uint64) localTime; /* The node's local clock. This is just a running sum of the number of output frames that have been processed. Can be modified by any thread with `ma_node_set_time()`. */ ma_uint32 inputBusCount; ma_uint32 outputBusCount; ma_node_input_bus* pInputBuses; @@ -16524,11 +16541,11 @@ static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice) } } - /* Make sure we don't get stuck in the inner loop. */ - if (capturedDeviceFramesProcessed == 0) { - break; - } - + /* Make sure we don't get stuck in the inner loop. */ + if (capturedDeviceFramesProcessed == 0) { + break; + } + totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; } } break; @@ -16552,11 +16569,11 @@ static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice) break; } - /* Make sure we don't get stuck in the inner loop. */ - if (framesProcessed == 0) { - break; - } - + /* Make sure we don't get stuck in the inner loop. */ + if (framesProcessed == 0) { + break; + } + ma_device__send_frames_to_client(pDevice, framesProcessed, capturedDeviceData); framesReadThisPeriod += framesProcessed; @@ -16584,11 +16601,11 @@ static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice) break; } - /* Make sure we don't get stuck in the inner loop. */ - if (framesProcessed == 0) { - break; - } - + /* Make sure we don't get stuck in the inner loop. */ + if (framesProcessed == 0) { + break; + } + framesWrittenThisPeriod += framesProcessed; } } break; @@ -28589,6 +28606,13 @@ References #if defined(TARGET_OS_WATCH) && TARGET_OS_WATCH == 1 #define MA_APPLE_WATCH #endif + #if __has_feature(objc_arc) + #define MA_BRIDGE_TRANSFER __bridge_transfer + #define MA_BRIDGE_RETAINED __bridge_retained + #else + #define MA_BRIDGE_TRANSFER + #define MA_BRIDGE_RETAINED + #endif #else #define MA_APPLE_DESKTOP #endif @@ -30164,7 +30188,7 @@ static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFl MA_ASSERT(pDevice != NULL); - ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Output Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pBufferList->mNumberBuffers); + ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Output Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", (int)busNumber, (int)frameCount, (int)pBufferList->mNumberBuffers); /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */ layout = ma_stream_layout_interleaved; @@ -30182,7 +30206,7 @@ static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFl ma_device_handle_backend_data_callback(pDevice, pBufferList->mBuffers[iBuffer].mData, NULL, frameCountForThisBuffer); } - ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pBufferList->mBuffers[iBuffer].mNumberChannels, pBufferList->mBuffers[iBuffer].mDataByteSize); + ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pBufferList->mBuffers[iBuffer].mNumberChannels, (int)pBufferList->mBuffers[iBuffer].mDataByteSize); } else { /* This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's @@ -30190,7 +30214,7 @@ static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFl output silence here. */ MA_ZERO_MEMORY(pBufferList->mBuffers[iBuffer].mData, pBufferList->mBuffers[iBuffer].mDataByteSize); - ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pBufferList->mBuffers[iBuffer].mNumberChannels, pBufferList->mBuffers[iBuffer].mDataByteSize); + ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pBufferList->mBuffers[iBuffer].mNumberChannels, (int)pBufferList->mBuffers[iBuffer].mDataByteSize); } } } else { @@ -30259,7 +30283,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla layout = ma_stream_layout_deinterleaved; } - ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pRenderedBufferList->mNumberBuffers); + ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", (int)busNumber, (int)frameCount, (int)pRenderedBufferList->mNumberBuffers); /* There has been a situation reported where frame count passed into this function is greater than the capacity of @@ -30287,7 +30311,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList); if (status != noErr) { - ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " ERROR: AudioUnitRender() failed with %d\n", status); + ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " ERROR: AudioUnitRender() failed with %d\n", (int)status); return status; } @@ -30295,7 +30319,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) { if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) { ma_device_handle_backend_data_callback(pDevice, NULL, pRenderedBufferList->mBuffers[iBuffer].mData, frameCount); - ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " mDataByteSize=%d\n", pRenderedBufferList->mBuffers[iBuffer].mDataByteSize); + ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " mDataByteSize=%d\n", (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize); } else { /* This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's @@ -30318,7 +30342,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla framesRemaining -= framesToSend; } - ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pRenderedBufferList->mBuffers[iBuffer].mNumberChannels, pRenderedBufferList->mBuffers[iBuffer].mDataByteSize); + ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pRenderedBufferList->mBuffers[iBuffer].mNumberChannels, (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize); } } } else { @@ -30672,6 +30696,7 @@ static ma_result ma_device__untrack__coreaudio(ma_device* pDevice) -(void)dealloc { [self remove_handler]; + [super dealloc]; } -(void)remove_handler @@ -30766,7 +30791,7 @@ static ma_result ma_device_uninit__coreaudio(ma_device* pDevice) #endif #if defined(MA_APPLE_MOBILE) if (pDevice->coreaudio.pRouteChangeHandler != NULL) { - ma_router_change_handler* pRouteChangeHandler = (__bridge_transfer ma_router_change_handler*)pDevice->coreaudio.pRouteChangeHandler; + ma_router_change_handler* pRouteChangeHandler = (MA_BRIDGE_TRANSFER ma_router_change_handler*)pDevice->coreaudio.pRouteChangeHandler; [pRouteChangeHandler remove_handler]; } #endif @@ -31087,7 +31112,7 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev } #else /* TODO: Figure out how to get the channel map using AVAudioSession. */ - ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMap, ma_countof(pData->channelMap), pData->channelsOut); + ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut); #endif @@ -31440,7 +31465,7 @@ static ma_result ma_device_init__coreaudio(ma_device* pDevice, const ma_device_c differently on non-Desktop Apple platforms. */ #if defined(MA_APPLE_MOBILE) - pDevice->coreaudio.pRouteChangeHandler = (__bridge_retained void*)[[ma_router_change_handler alloc] init:pDevice]; + pDevice->coreaudio.pRouteChangeHandler = (MA_BRIDGE_RETAINED void*)[[ma_router_change_handler alloc] init:pDevice]; #endif return MA_SUCCESS; @@ -39057,7 +39082,7 @@ MA_API ma_result ma_slot_allocator_free(ma_slot_allocator* pAllocator, ma_uint64 MA_ASSERT(iBit < 32); /* This must be true due to the logic we used to actually calculate it. */ - while (c89atomic_load_32(&pAllocator->count) > 0) { + while (c89atomic_load_32(&pAllocator->count) > 0) { /* CAS */ ma_uint32 oldBitfield; ma_uint32 newBitfield; @@ -55246,7 +55271,7 @@ extern "C" { #define DRFLAC_XSTRINGIFY(x) DRFLAC_STRINGIFY(x) #define DRFLAC_VERSION_MAJOR 0 #define DRFLAC_VERSION_MINOR 12 -#define DRFLAC_VERSION_REVISION 31 +#define DRFLAC_VERSION_REVISION 31 #define DRFLAC_VERSION_STRING DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MAJOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MINOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_REVISION) #include typedef signed char drflac_int8; @@ -55607,7 +55632,7 @@ extern "C" { #define DRMP3_XSTRINGIFY(x) DRMP3_STRINGIFY(x) #define DRMP3_VERSION_MAJOR 0 #define DRMP3_VERSION_MINOR 6 -#define DRMP3_VERSION_REVISION 31 +#define DRMP3_VERSION_REVISION 31 #define DRMP3_VERSION_STRING DRMP3_XSTRINGIFY(DRMP3_VERSION_MAJOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_MINOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_REVISION) #include typedef signed char drmp3_int8; @@ -58534,7 +58559,7 @@ MA_API ma_result ma_stbvorbis_read_pcm_frames(ma_stbvorbis* pVorbis, void* pFram framesRead = stb_vorbis_get_samples_float_interleaved(pVorbis->stb, channels, (float*)ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, format, channels), (int)framesRemaining * channels); /* Safe cast. */ totalFramesRead += framesRead; - if (framesRead < framesRemaining) { + if (framesRead < (int)framesRemaining) { break; /* Nothing left to read. Get out. */ } } @@ -65813,8 +65838,8 @@ MA_API void ma_debug_fill_pcm_frames_with_sine_wave(float* pFramesOut, ma_uint32 (void)sampleRate; #if defined(MA_DEBUG_OUTPUT) { - #if _MSC_VER - #pragma message ("ma_debug_fill_pcm_frames_with_sine_wave() will do nothing because MA_NO_GENERATION is enabled.") + #if _MSC_VER + #pragma message ("ma_debug_fill_pcm_frames_with_sine_wave() will do nothing because MA_NO_GENERATION is enabled.") #endif } #endif @@ -84040,7 +84065,7 @@ static type* drflac__full_read_and_close_ ## extension (drflac* pFlac, unsigned DRFLAC_ZERO_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), (size_t)(sampleDataBufferSize - totalPCMFrameCount*pFlac->channels*sizeof(type))); \ } else { \ drflac_uint64 dataSize = totalPCMFrameCount*pFlac->channels*sizeof(type); \ - if (dataSize > (drflac_uint64)DRFLAC_SIZE_MAX) { \ + if (dataSize > (drflac_uint64)DRFLAC_SIZE_MAX) { \ goto on_error; \ } \ \ @@ -84483,29 +84508,29 @@ static __inline__ __attribute__((always_inline)) drmp3_int32 drmp3_clip_int16_ar #else #define DRMP3_HAVE_ARMV6 0 #endif -#ifndef DRMP3_ASSERT -#include -#define DRMP3_ASSERT(expression) assert(expression) -#endif -#ifndef DRMP3_COPY_MEMORY -#define DRMP3_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) -#endif -#ifndef DRMP3_MOVE_MEMORY -#define DRMP3_MOVE_MEMORY(dst, src, sz) memmove((dst), (src), (sz)) -#endif -#ifndef DRMP3_ZERO_MEMORY -#define DRMP3_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) -#endif -#define DRMP3_ZERO_OBJECT(p) DRMP3_ZERO_MEMORY((p), sizeof(*(p))) -#ifndef DRMP3_MALLOC -#define DRMP3_MALLOC(sz) malloc((sz)) -#endif -#ifndef DRMP3_REALLOC -#define DRMP3_REALLOC(p, sz) realloc((p), (sz)) -#endif -#ifndef DRMP3_FREE -#define DRMP3_FREE(p) free((p)) -#endif +#ifndef DRMP3_ASSERT +#include +#define DRMP3_ASSERT(expression) assert(expression) +#endif +#ifndef DRMP3_COPY_MEMORY +#define DRMP3_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRMP3_MOVE_MEMORY +#define DRMP3_MOVE_MEMORY(dst, src, sz) memmove((dst), (src), (sz)) +#endif +#ifndef DRMP3_ZERO_MEMORY +#define DRMP3_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#define DRMP3_ZERO_OBJECT(p) DRMP3_ZERO_MEMORY((p), sizeof(*(p))) +#ifndef DRMP3_MALLOC +#define DRMP3_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRMP3_REALLOC +#define DRMP3_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRMP3_FREE +#define DRMP3_FREE(p) free((p)) +#endif typedef struct { const drmp3_uint8 *buf; @@ -84745,7 +84770,7 @@ static int drmp3_L12_dequantize_granule(float *grbuf, drmp3_bs *bs, drmp3_L12_sc static void drmp3_L12_apply_scf_384(drmp3_L12_scale_info *sci, const float *scf, float *dst) { int i, k; - DRMP3_COPY_MEMORY(dst + 576 + sci->stereo_bands*18, dst + sci->stereo_bands*18, (sci->total_bands - sci->stereo_bands)*18*sizeof(float)); + DRMP3_COPY_MEMORY(dst + 576 + sci->stereo_bands*18, dst + sci->stereo_bands*18, (sci->total_bands - sci->stereo_bands)*18*sizeof(float)); for (i = 0; i < sci->total_bands; i++, dst += 18, scf += 6) { for (k = 0; k < 12; k++) @@ -84883,14 +84908,14 @@ static void drmp3_L3_read_scalefactors(drmp3_uint8 *scf, drmp3_uint8 *ist_pos, c int cnt = scf_count[i]; if (scfsi & 8) { - DRMP3_COPY_MEMORY(scf, ist_pos, cnt); + DRMP3_COPY_MEMORY(scf, ist_pos, cnt); } else { int bits = scf_size[i]; if (!bits) { - DRMP3_ZERO_MEMORY(scf, cnt); - DRMP3_ZERO_MEMORY(ist_pos, cnt); + DRMP3_ZERO_MEMORY(scf, cnt); + DRMP3_ZERO_MEMORY(ist_pos, cnt); } else { int max_scf = (scfsi < 0) ? (1 << bits) - 1 : -1; @@ -85243,7 +85268,7 @@ static void drmp3_L3_reorder(float *grbuf, float *scratch, const drmp3_uint8 *sf *dst++ = src[2*len]; } } - DRMP3_COPY_MEMORY(grbuf, scratch, (dst - scratch)*sizeof(float)); + DRMP3_COPY_MEMORY(grbuf, scratch, (dst - scratch)*sizeof(float)); } static void drmp3_L3_antialias(float *grbuf, int nbands) { @@ -85393,8 +85418,8 @@ static void drmp3_L3_imdct_short(float *grbuf, float *overlap, int nbands) for (;nbands > 0; nbands--, overlap += 9, grbuf += 18) { float tmp[18]; - DRMP3_COPY_MEMORY(tmp, grbuf, sizeof(tmp)); - DRMP3_COPY_MEMORY(grbuf, overlap, 6*sizeof(float)); + DRMP3_COPY_MEMORY(tmp, grbuf, sizeof(tmp)); + DRMP3_COPY_MEMORY(grbuf, overlap, 6*sizeof(float)); drmp3_L3_imdct12(tmp, grbuf + 6, overlap + 6); drmp3_L3_imdct12(tmp + 1, grbuf + 12, overlap + 6); drmp3_L3_imdct12(tmp + 2, overlap, overlap + 6); @@ -85435,7 +85460,7 @@ static void drmp3_L3_save_reservoir(drmp3dec *h, drmp3dec_scratch *s) } if (remains > 0) { - DRMP3_MOVE_MEMORY(h->reserv_buf, s->maindata + pos, remains); + DRMP3_MOVE_MEMORY(h->reserv_buf, s->maindata + pos, remains); } h->reserv = remains; } @@ -85443,8 +85468,8 @@ static int drmp3_L3_restore_reservoir(drmp3dec *h, drmp3_bs *bs, drmp3dec_scratc { int frame_bytes = (bs->limit - bs->pos)/8; int bytes_have = DRMP3_MIN(h->reserv, main_data_begin); - DRMP3_COPY_MEMORY(s->maindata, h->reserv_buf + DRMP3_MAX(0, h->reserv - main_data_begin), DRMP3_MIN(h->reserv, main_data_begin)); - DRMP3_COPY_MEMORY(s->maindata + bytes_have, bs->buf + bs->pos/8, frame_bytes); + DRMP3_COPY_MEMORY(s->maindata, h->reserv_buf + DRMP3_MAX(0, h->reserv - main_data_begin), DRMP3_MIN(h->reserv, main_data_begin)); + DRMP3_COPY_MEMORY(s->maindata + bytes_have, bs->buf + bs->pos/8, frame_bytes); drmp3_bs_init(&s->bs, s->maindata, bytes_have + frame_bytes); return h->reserv >= main_data_begin; } @@ -85822,7 +85847,7 @@ static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int { drmp3d_DCT_II(grbuf + 576*i, nbands); } - DRMP3_COPY_MEMORY(lins, qmf_state, sizeof(float)*15*64); + DRMP3_COPY_MEMORY(lins, qmf_state, sizeof(float)*15*64); for (i = 0; i < nbands; i += 2) { drmp3d_synth(grbuf + i, pcm + 32*nch*i, nch, lins + i*64); @@ -85837,7 +85862,7 @@ static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int } else #endif { - DRMP3_COPY_MEMORY(qmf_state, lins + nbands*64, sizeof(float)*15*64); + DRMP3_COPY_MEMORY(qmf_state, lins + nbands*64, sizeof(float)*15*64); } } static int drmp3d_match_frame(const drmp3_uint8 *hdr, int mp3_bytes, int frame_bytes) @@ -85908,7 +85933,7 @@ DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int m } if (!frame_size) { - DRMP3_ZERO_MEMORY(dec, sizeof(drmp3dec)); + DRMP3_ZERO_MEMORY(dec, sizeof(drmp3dec)); i = drmp3d_find_frame(mp3, mp3_bytes, &dec->free_format_bytes, &frame_size); if (!frame_size || i + frame_size > mp3_bytes) { @@ -85917,7 +85942,7 @@ DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int m } } hdr = mp3 + i; - DRMP3_COPY_MEMORY(dec->header, hdr, DRMP3_HDR_SIZE); + DRMP3_COPY_MEMORY(dec->header, hdr, DRMP3_HDR_SIZE); info->frame_bytes = i + frame_size; info->channels = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; info->hz = drmp3_hdr_sample_rate_hz(hdr); @@ -85941,7 +85966,7 @@ DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int m { for (igr = 0; igr < (DRMP3_HDR_TEST_MPEG1(hdr) ? 2 : 1); igr++, pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*576*info->channels)) { - DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); + DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); drmp3_L3_decode(dec, &scratch, scratch.gr_info + igr*info->channels, info->channels); drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 18, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]); } @@ -85957,7 +85982,7 @@ DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int m return drmp3_hdr_frame_samples(hdr); } drmp3_L12_read_scale_info(hdr, bs_frame, sci); - DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); + DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); for (i = 0, igr = 0; igr < 3; igr++) { if (12 == (i += drmp3_L12_dequantize_granule(scratch.grbuf[0] + i, bs_frame, sci, info->layer | 1))) @@ -85965,7 +85990,7 @@ DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int m i = 0; drmp3_L12_apply_scf_384(sci, sci->scf + igr, scratch.grbuf[0]); drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 12, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]); - DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); + DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*384*info->channels); } if (bs_frame->pos > bs_frame->limit) @@ -86219,7 +86244,7 @@ static drmp3_uint32 drmp3_decode_next_frame_ex__callbacks(drmp3* pMP3, drmp3d_sa if (pMP3->dataSize < DRMP3_MIN_DATA_CHUNK_SIZE) { size_t bytesRead; if (pMP3->pData != NULL) { - DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize); + DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize); } pMP3->dataConsumed = 0; if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) { @@ -86262,7 +86287,7 @@ static drmp3_uint32 drmp3_decode_next_frame_ex__callbacks(drmp3* pMP3, drmp3d_sa break; } else if (info.frame_bytes == 0) { size_t bytesRead; - DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize); + DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize); pMP3->dataConsumed = 0; if (pMP3->dataCapacity == pMP3->dataSize) { drmp3_uint8* pNewData; @@ -86294,20 +86319,20 @@ static drmp3_uint32 drmp3_decode_next_frame_ex__memory(drmp3* pMP3, drmp3d_sampl if (pMP3->atEnd) { return 0; } - for (;;) { - pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->memory.pData + pMP3->memory.currentReadPos, (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos), pPCMFrames, &info); - if (pcmFramesRead > 0) { - pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header); - pMP3->pcmFramesConsumedInMP3Frame = 0; - pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; - pMP3->mp3FrameChannels = info.channels; - pMP3->mp3FrameSampleRate = info.hz; - break; - } else if (info.frame_bytes > 0) { - pMP3->memory.currentReadPos += (size_t)info.frame_bytes; - } else { - break; - } + for (;;) { + pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->memory.pData + pMP3->memory.currentReadPos, (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos), pPCMFrames, &info); + if (pcmFramesRead > 0) { + pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header); + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; + pMP3->mp3FrameChannels = info.channels; + pMP3->mp3FrameSampleRate = info.hz; + break; + } else if (info.frame_bytes > 0) { + pMP3->memory.currentReadPos += (size_t)info.frame_bytes; + } else { + break; + } } pMP3->memory.currentReadPos += (size_t)info.frame_bytes; return pcmFramesRead; @@ -86352,7 +86377,7 @@ static drmp3_bool32 drmp3_init_internal(drmp3* pMP3, drmp3_read_proc onRead, drm if (pMP3->allocationCallbacks.onFree == NULL || (pMP3->allocationCallbacks.onMalloc == NULL && pMP3->allocationCallbacks.onRealloc == NULL)) { return DRMP3_FALSE; } - if (drmp3_decode_next_frame(pMP3) == 0) { + if (drmp3_decode_next_frame(pMP3) == 0) { drmp3__free_from_callbacks(pMP3->pData, &pMP3->allocationCallbacks); return DRMP3_FALSE; } @@ -87434,7 +87459,7 @@ static float* drmp3__full_read_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, } oldFramesBufferSize = framesCapacity * pMP3->channels * sizeof(float); newFramesBufferSize = newFramesCap * pMP3->channels * sizeof(float); - if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) { + if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) { break; } pNewFrames = (float*)drmp3__realloc_from_callbacks(pFrames, (size_t)newFramesBufferSize, (size_t)oldFramesBufferSize, &pMP3->allocationCallbacks); @@ -87485,7 +87510,7 @@ static drmp3_int16* drmp3__full_read_and_close_s16(drmp3* pMP3, drmp3_config* pC } oldFramesBufferSize = framesCapacity * pMP3->channels * sizeof(drmp3_int16); newFramesBufferSize = newFramesCap * pMP3->channels * sizeof(drmp3_int16); - if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) { + if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) { break; } pNewFrames = (drmp3_int16*)drmp3__realloc_from_callbacks(pFrames, (size_t)newFramesBufferSize, (size_t)oldFramesBufferSize, &pMP3->allocationCallbacks); @@ -87909,9 +87934,9 @@ issues with certain devices and configurations. These can be individually enable /* REVISION HISTORY ================ -v0.10.42 - 2021-08-22 - - Fix a possible deadlock when stopping devices. - +v0.10.42 - 2021-08-22 + - Fix a possible deadlock when stopping devices. + v0.10.41 - 2021-08-15 - Core Audio: Fix some deadlock errors.