mirror of
https://github.com/mackron/miniaudio.git
synced 2026-04-21 15:56:58 +02:00
Merge pull request #372 from orx/dev-0.11
Fixed compilation warnings & errors on iOS
This commit is contained in:
David Reid
GitHub
+126
-101
@@ -1,6 +1,6 @@
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/*
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Audio playback and capture library. Choice of public domain or MIT-0. See license statements at the end of this file.
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miniaudio - v0.10.42 - 2021-08-22
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miniaudio - v0.10.42 - 2021-08-22
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David Reid - mackron@gmail.com
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@@ -3286,7 +3286,7 @@ extern "C" {
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#define MA_VERSION_MAJOR 0
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#define MA_VERSION_MINOR 10
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#define MA_VERSION_REVISION 42
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#define MA_VERSION_REVISION 42
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#define MA_VERSION_STRING MA_XSTRINGIFY(MA_VERSION_MAJOR) "." MA_XSTRINGIFY(MA_VERSION_MINOR) "." MA_XSTRINGIFY(MA_VERSION_REVISION)
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#if defined(_MSC_VER) && !defined(__clang__)
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@@ -3302,6 +3302,23 @@ extern "C" {
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#endif
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#endif
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#if defined (__STDC_VERSION__) && (__STDC_VERSION__ >= 201112L)
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#include <stdalign.h>
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#define MA_ALIGN_TYPE(n) alignas(n)
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#define MA_ALIGN_MEMBER(align, type) MA_ALIGN_TYPE(align) type
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#else
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#if defined(__GNUC__)
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#define MA_ALIGN_TYPE(n) __attribute__((aligned(n)))
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#define MA_ALIGN_MEMBER(align, type) type MA_ALIGN_TYPE(align)
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#elif defined(_MSC_VER)
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#define MA_ALIGN_TYPE(n) __declspec(align(n))
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#define MA_ALIGN_MEMBER(align, type) MA_ALIGN_TYPE(align) type
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#else
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#define MA_ALIGN_TYPE(n) /* disabled */
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#define MA_ALIGN_MEMBER(align, type) /* disabled */
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#endif
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#endif
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/* Platform/backend detection. */
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#ifdef _WIN32
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#define MA_WIN32
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@@ -8898,7 +8915,7 @@ typedef struct
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} breakup;
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ma_uint64 allocation;
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} toc; /* 8 bytes. We encode the job code into the slot allocation data to save space. */
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ma_uint64 next; /* refcount + slot for the next item. Does not include the job code. */
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MA_ATOMIC MA_ALIGN_MEMBER(8, ma_uint64) next; /* refcount + slot for the next item. Does not include the job code. */
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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. */
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union
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@@ -9003,7 +9020,7 @@ typedef struct
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{
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ma_uint32 flags; /* Flags passed in at initialization time. */
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ma_uint32 capacity; /* The maximum number of jobs that can fit in the queue at a time. Set by the config. */
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MA_ATOMIC ma_uint64 head; /* The first item in the list. Required for removing from the top of the list. */
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MA_ATOMIC MA_ALIGN_MEMBER(8, ma_uint64) head; /* The first item in the list. Required for removing from the top of the list. */
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MA_ATOMIC ma_uint64 tail; /* The last item in the list. Required for appending to the end of the list. */
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#ifndef MA_NO_THREADING
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ma_semaphore sem; /* Only used when MA_RESOURCE_MANAGER_JOB_QUEUE_FLAG_NON_BLOCKING is unset. */
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@@ -9133,7 +9150,7 @@ struct ma_resource_manager_data_stream
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ma_bool32 isDecoderInitialized; /* Required for determining whether or not the decoder should be uninitialized in MA_RESOURCE_MANAGER_JOB_FREE_DATA_STREAM. */
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ma_uint64 totalLengthInPCMFrames; /* This is calculated when first loaded by the MA_RESOURCE_MANAGER_JOB_LOAD_DATA_STREAM. */
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ma_uint32 relativeCursor; /* The playback cursor, relative to the current page. Only ever accessed by the public API. Never accessed by the job thread. */
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ma_uint64 absoluteCursor; /* The playback cursor, in absolute position starting from the start of the file. */
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MA_ATOMIC MA_ALIGN_MEMBER(8, ma_uint64) absoluteCursor; /* The playback cursor, in absolute position starting from the start of the file. */
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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. */
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MA_ATOMIC ma_uint32 executionCounter; /* For allocating execution orders for jobs. */
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MA_ATOMIC ma_uint32 executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
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@@ -9423,8 +9440,8 @@ struct ma_node_base
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/* These variables are read and written between different threads. */
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MA_ATOMIC ma_node_state state; /* When set to stopped, nothing will be read, regardless of the times in stateTimes. */
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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. */
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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()`. */
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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. */
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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()`. */
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ma_uint32 inputBusCount;
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ma_uint32 outputBusCount;
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ma_node_input_bus* pInputBuses;
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@@ -16524,11 +16541,11 @@ static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice)
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}
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}
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/* Make sure we don't get stuck in the inner loop. */
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if (capturedDeviceFramesProcessed == 0) {
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break;
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}
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/* Make sure we don't get stuck in the inner loop. */
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if (capturedDeviceFramesProcessed == 0) {
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break;
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}
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totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed;
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}
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} break;
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@@ -16552,11 +16569,11 @@ static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice)
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break;
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}
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/* Make sure we don't get stuck in the inner loop. */
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if (framesProcessed == 0) {
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break;
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}
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/* Make sure we don't get stuck in the inner loop. */
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if (framesProcessed == 0) {
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break;
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}
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ma_device__send_frames_to_client(pDevice, framesProcessed, capturedDeviceData);
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framesReadThisPeriod += framesProcessed;
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@@ -16584,11 +16601,11 @@ static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice)
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break;
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}
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/* Make sure we don't get stuck in the inner loop. */
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if (framesProcessed == 0) {
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break;
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}
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/* Make sure we don't get stuck in the inner loop. */
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if (framesProcessed == 0) {
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break;
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}
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framesWrittenThisPeriod += framesProcessed;
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}
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} break;
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@@ -28589,6 +28606,13 @@ References
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#if defined(TARGET_OS_WATCH) && TARGET_OS_WATCH == 1
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#define MA_APPLE_WATCH
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#endif
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#if __has_feature(objc_arc)
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#define MA_BRIDGE_TRANSFER __bridge_transfer
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#define MA_BRIDGE_RETAINED __bridge_retained
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#else
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#define MA_BRIDGE_TRANSFER
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#define MA_BRIDGE_RETAINED
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#endif
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#else
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#define MA_APPLE_DESKTOP
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#endif
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@@ -30164,7 +30188,7 @@ static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFl
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MA_ASSERT(pDevice != NULL);
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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);
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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);
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/* 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. */
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layout = ma_stream_layout_interleaved;
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@@ -30182,7 +30206,7 @@ static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFl
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ma_device_handle_backend_data_callback(pDevice, pBufferList->mBuffers[iBuffer].mData, NULL, frameCountForThisBuffer);
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}
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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);
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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);
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} else {
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/*
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This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
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@@ -30190,7 +30214,7 @@ static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFl
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output silence here.
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*/
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MA_ZERO_MEMORY(pBufferList->mBuffers[iBuffer].mData, pBufferList->mBuffers[iBuffer].mDataByteSize);
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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);
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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);
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}
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}
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} else {
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@@ -30259,7 +30283,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
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layout = ma_stream_layout_deinterleaved;
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}
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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);
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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);
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/*
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There has been a situation reported where frame count passed into this function is greater than the capacity of
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@@ -30287,7 +30311,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
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status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList);
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if (status != noErr) {
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ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " ERROR: AudioUnitRender() failed with %d\n", status);
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ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " ERROR: AudioUnitRender() failed with %d\n", (int)status);
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return status;
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}
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@@ -30295,7 +30319,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
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for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
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if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) {
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ma_device_handle_backend_data_callback(pDevice, NULL, pRenderedBufferList->mBuffers[iBuffer].mData, frameCount);
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ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " mDataByteSize=%d\n", pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);
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ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " mDataByteSize=%d\n", (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);
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} else {
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/*
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This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
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@@ -30318,7 +30342,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
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framesRemaining -= framesToSend;
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}
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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);
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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);
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}
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}
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} else {
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@@ -30672,6 +30696,7 @@ static ma_result ma_device__untrack__coreaudio(ma_device* pDevice)
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-(void)dealloc
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{
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[self remove_handler];
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[super dealloc];
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}
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|
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-(void)remove_handler
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@@ -30766,7 +30791,7 @@ static ma_result ma_device_uninit__coreaudio(ma_device* pDevice)
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#endif
|
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#if defined(MA_APPLE_MOBILE)
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if (pDevice->coreaudio.pRouteChangeHandler != NULL) {
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ma_router_change_handler* pRouteChangeHandler = (__bridge_transfer ma_router_change_handler*)pDevice->coreaudio.pRouteChangeHandler;
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ma_router_change_handler* pRouteChangeHandler = (MA_BRIDGE_TRANSFER ma_router_change_handler*)pDevice->coreaudio.pRouteChangeHandler;
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[pRouteChangeHandler remove_handler];
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}
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#endif
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@@ -31087,7 +31112,7 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev
|
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}
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#else
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/* TODO: Figure out how to get the channel map using AVAudioSession. */
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ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMap, ma_countof(pData->channelMap), pData->channelsOut);
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ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut);
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#endif
|
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|
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|
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@@ -31440,7 +31465,7 @@ static ma_result ma_device_init__coreaudio(ma_device* pDevice, const ma_device_c
|
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differently on non-Desktop Apple platforms.
|
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*/
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#if defined(MA_APPLE_MOBILE)
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pDevice->coreaudio.pRouteChangeHandler = (__bridge_retained void*)[[ma_router_change_handler alloc] init:pDevice];
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pDevice->coreaudio.pRouteChangeHandler = (MA_BRIDGE_RETAINED void*)[[ma_router_change_handler alloc] init:pDevice];
|
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#endif
|
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|
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return MA_SUCCESS;
|
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@@ -39057,7 +39082,7 @@ MA_API ma_result ma_slot_allocator_free(ma_slot_allocator* pAllocator, ma_uint64
|
||||
|
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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) {
|
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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 <stddef.h>
|
||||
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 <stddef.h>
|
||||
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 <assert.h>
|
||||
#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 <assert.h>
|
||||
#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.
|
||||
|
||||
|
||||
Reference in New Issue
Block a user