mirrors/SDL
1
0
Fork 0
mirror of https://github.com/libsdl-org/SDL.git synced 2025-10-05 01:16:26 +00:00
Code Issues Packages Projects Releases Wiki Activity

Use C++ style comments consistently in SDL source code

Browse Source 
Implemented using this script:

find . -type f -exec sed -i'' -e 's,/\* *\([^*]*\)\*/ *$,// \1,' -e 's, \+$,,' {} \;
git checkout \
    core/linux/SDL_evdev_kbd_default_keymap.h \
    events/imKStoUCS.* \
    hidapi \
    joystick/controller_type.c \
    joystick/controller_type.h \
    joystick/hidapi/steam/controller_constants.h \
    joystick/hidapi/steam/controller_structs.h \
    joystick/SDL_gamepad_db.h \
    libm \
    render/*/*Shader*.h \
    render/vitagxm/SDL_render_vita_gxm_shaders.h \
    render/metal/SDL_shaders_metal_*.h \
    stdlib/SDL_malloc.c \
    stdlib/SDL_qsort.c \
    stdlib/SDL_strtokr.c \
    test/ \
    video/directx/SDL_d3d12_xbox_cmacros.h \
    video/directx/d3d12.h \
    video/directx/d3d12sdklayers.h \
    video/khronos \
    video/x11/edid-parse.c \
    video/x11/xsettings-client.* \
    video/yuv2rgb
sed -i'' -e 's,/\* *\([^*]*\)\*/ *$,// \1,' -e 's, \+$,,' hidapi/SDL_hidapi.c
This commit is contained in:
Sam Lantinga
2024-08-22 10:30:45 -07:00
parent 658fc3db0f
commit 6501e90018
743 changed files with 11882 additions and 11882 deletions
Download Patch File Download Diff File Expand all files Collapse all files

246
src/audio/SDL_wave.c
View File

@@ -24,14 +24,14 @@
#include <limits.h>
#endif
#ifndef INT_MAX
/* Make a lucky guess. */
// Make a lucky guess.
#define INT_MAX SDL_MAX_SINT32
#endif
#ifndef SIZE_MAX
#define SIZE_MAX ((size_t)-1)
#endif
/* Microsoft WAVE file loading routines */
// Microsoft WAVE file loading routines
#include "SDL_wave.h"
#include "SDL_sysaudio.h"
@@ -52,17 +52,17 @@ static int SafeMult(size_t *f1, size_t f2)
typedef struct ADPCM_DecoderState
{
Uint32 channels; /* Number of channels. */
size_t blocksize; /* Size of an ADPCM block in bytes. */
size_t blockheadersize; /* Size of an ADPCM block header in bytes. */
size_t samplesperblock; /* Number of samples per channel in an ADPCM block. */
size_t framesize; /* Size of a sample frame (16-bit PCM) in bytes. */
Sint64 framestotal; /* Total number of sample frames. */
Sint64 framesleft; /* Number of sample frames still to be decoded. */
void *ddata; /* Decoder data from initialization. */
void *cstate; /* Decoding state for each channel. */
Uint32 channels; // Number of channels.
size_t blocksize; // Size of an ADPCM block in bytes.
size_t blockheadersize; // Size of an ADPCM block header in bytes.
size_t samplesperblock; // Number of samples per channel in an ADPCM block.
size_t framesize; // Size of a sample frame (16-bit PCM) in bytes.
Sint64 framestotal; // Total number of sample frames.
Sint64 framesleft; // Number of sample frames still to be decoded.
void *ddata; // Decoder data from initialization.
void *cstate; // Decoding state for each channel.
/* ADPCM data. */
// ADPCM data.
struct
{
Uint8 *data;
@@ -70,7 +70,7 @@ typedef struct ADPCM_DecoderState
size_t pos;
} input;
/* Current ADPCM block in the ADPCM data above. */
// Current ADPCM block in the ADPCM data above.
struct
{
Uint8 *data;
@@ -78,7 +78,7 @@ typedef struct ADPCM_DecoderState
size_t pos;
} block;
/* Decoded 16-bit PCM data. */
// Decoded 16-bit PCM data.
struct
{
Sint16 *data;
@@ -91,7 +91,7 @@ typedef struct MS_ADPCM_CoeffData
{
Uint16 coeffcount;
Sint16 *coeff;
Sint16 aligndummy; /* Has to be last member. */
Sint16 aligndummy; // Has to be last member.
} MS_ADPCM_CoeffData;
typedef struct MS_ADPCM_ChannelState
@@ -342,18 +342,18 @@ static int MS_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
const size_t trailingdata = datalength % file->format.blockalign;
if (file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict) {
/* The size of the data chunk must be a multiple of the block size. */
// The size of the data chunk must be a multiple of the block size.
if (datalength < blockheadersize || trailingdata > 0) {
return SDL_SetError("Truncated MS ADPCM block");
}
}
/* Calculate number of sample frames that will be decoded. */
// Calculate number of sample frames that will be decoded.
file->sampleframes = (Sint64)availableblocks * format->samplesperblock;
if (trailingdata > 0) {
/* The last block is truncated. Check if we can get any samples out of it. */
// The last block is truncated. Check if we can get any samples out of it.
if (file->trunchint == TruncDropFrame) {
/* Drop incomplete sample frame. */
// Drop incomplete sample frame.
if (trailingdata >= blockheadersize) {
size_t trailingsamples = 2 + (trailingdata - blockheadersize) * 8 / blockframebitsize;
if (trailingsamples > format->samplesperblock) {
@@ -384,7 +384,7 @@ static int MS_ADPCM_Init(WaveFile *file, size_t datalength)
size_t i, coeffcount;
MS_ADPCM_CoeffData *coeffdata;
/* Sanity checks. */
// Sanity checks.
/* While it's clear how IMA ADPCM handles more than two channels, the nibble
* order of MS ADPCM makes it awkward. The Standards Update does not talk
@@ -398,7 +398,7 @@ static int MS_ADPCM_Init(WaveFile *file, size_t datalength)
return SDL_SetError("Invalid MS ADPCM bits per sample of %u", (unsigned int)format->bitspersample);
}
/* The block size must be big enough to contain the block header. */
// The block size must be big enough to contain the block header.
if (format->blockalign < blockheadersize) {
return SDL_SetError("Invalid MS ADPCM block size (nBlockAlign)");
}
@@ -419,7 +419,7 @@ static int MS_ADPCM_Init(WaveFile *file, size_t datalength)
}
format->samplesperblock = chunk->data[18] | ((Uint16)chunk->data[19] << 8);
/* Number of coefficient pairs. A pair has two 16-bit integers. */
// Number of coefficient pairs. A pair has two 16-bit integers.
coeffcount = chunk->data[20] | ((size_t)chunk->data[21] << 8);
/* bPredictor, the integer offset into the coefficients array, is only
* 8 bits. It can only address the first 256 coefficients. Let's limit
@@ -438,14 +438,14 @@ static int MS_ADPCM_Init(WaveFile *file, size_t datalength)
}
coeffdata = (MS_ADPCM_CoeffData *)SDL_malloc(sizeof(MS_ADPCM_CoeffData) + coeffcount * 4);
file->decoderdata = coeffdata; /* Freed in cleanup. */
file->decoderdata = coeffdata; // Freed in cleanup.
if (!coeffdata) {
return -1;
}
coeffdata->coeff = &coeffdata->aligndummy;
coeffdata->coeffcount = (Uint16)coeffcount;
/* Copy the 16-bit pairs. */
// Copy the 16-bit pairs.
for (i = 0; i < coeffcount * 2; i++) {
Sint32 c = chunk->data[22 + i * 2] | ((Sint32)chunk->data[23 + i * 2] << 8);
if (c >= 0x8000) {
@@ -504,7 +504,7 @@ static Sint16 MS_ADPCM_ProcessNibble(MS_ADPCM_ChannelState *cstate, Sint32 sampl
Uint32 delta = cstate->delta;
new_sample = (sample1 * cstate->coeff1 + sample2 * cstate->coeff2) / 256;
/* The nibble is a signed 4-bit error delta. */
// The nibble is a signed 4-bit error delta.
errordelta = (Sint32)nybble - (nybble >= 0x08 ? 0x10 : 0);
new_sample += (Sint32)delta * errordelta;
if (new_sample < min_audioval) {
@@ -538,7 +538,7 @@ static int MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
for (c = 0; c < channels; c++) {
size_t o = c;
/* Load the coefficient pair into the channel state. */
// Load the coefficient pair into the channel state.
coeffindex = state->block.data[o];
if (coeffindex > ddata->coeffcount) {
return SDL_SetError("Invalid MS ADPCM coefficient index in block header");
@@ -546,7 +546,7 @@ static int MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
cstate[c].coeff1 = ddata->coeff[coeffindex * 2];
cstate[c].coeff2 = ddata->coeff[coeffindex * 2 + 1];
/* Initial delta value. */
// Initial delta value.
o = (size_t)channels + c * 2;
cstate[c].delta = state->block.data[o] | ((Uint16)state->block.data[o + 1] << 8);
@@ -572,10 +572,10 @@ static int MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
state->block.pos += state->blockheadersize;
/* Skip second sample frame that came from the header. */
// Skip second sample frame that came from the header.
state->output.pos += state->channels;
/* Header provided two sample frames. */
// Header provided two sample frames.
state->framesleft -= 2;
return 0;
@@ -611,12 +611,12 @@ static int MS_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
} else if (blockpos < blocksize) {
nybble = state->block.data[blockpos++] | 0x4000;
} else {
/* Out of input data. Drop the incomplete frame and return. */
// Out of input data. Drop the incomplete frame and return.
state->output.pos = outpos - c;
return -1;
}
/* Load previous samples which may come from the block header. */
// Load previous samples which may come from the block header.
sample1 = state->output.data[outpos - channels];
sample2 = state->output.data[outpos - channels * 2];
@@ -645,13 +645,13 @@ static int MS_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
SDL_zeroa(cstate);
if (chunk->size != chunk->length) {
/* Could not read everything. Recalculate number of sample frames. */
// Could not read everything. Recalculate number of sample frames.
if (MS_ADPCM_CalculateSampleFrames(file, chunk->size) < 0) {
return -1;
}
}
/* Nothing to decode, nothing to return. */
// Nothing to decode, nothing to return.
if (file->sampleframes == 0) {
*audio_buf = NULL;
*audio_len = 0;
@@ -671,7 +671,7 @@ static int MS_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
state.input.size = chunk->size;
state.input.pos = 0;
/* The output size in bytes. May get modified if data is truncated. */
// The output size in bytes. May get modified if data is truncated.
outputsize = (size_t)state.framestotal;
if (SafeMult(&outputsize, state.framesize)) {
return SDL_SetError("WAVE file too big");
@@ -688,7 +688,7 @@ static int MS_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
state.cstate = cstate;
/* Decode block by block. A truncated block will stop the decoding. */
// Decode block by block. A truncated block will stop the decoding.
bytesleft = state.input.size - state.input.pos;
while (state.framesleft > 0 && bytesleft >= state.blockheadersize) {
state.block.data = state.input.data + state.input.pos;
@@ -696,29 +696,29 @@ static int MS_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
state.block.pos = 0;
if (state.output.size - state.output.pos < (Uint64)state.framesleft * state.channels) {
/* Somehow didn't allocate enough space for the output. */
// Somehow didn't allocate enough space for the output.
SDL_free(state.output.data);
return SDL_SetError("Unexpected overflow in MS ADPCM decoder");
}
/* Initialize decoder with the values from the block header. */
// Initialize decoder with the values from the block header.
result = MS_ADPCM_DecodeBlockHeader(&state);
if (result < 0) {
SDL_free(state.output.data);
return -1;
}
/* Decode the block data. It stores the samples directly in the output. */
// Decode the block data. It stores the samples directly in the output.
result = MS_ADPCM_DecodeBlockData(&state);
if (result < 0) {
/* Unexpected end. Stop decoding and return partial data if necessary. */
// Unexpected end. Stop decoding and return partial data if necessary.
if (file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict) {
SDL_free(state.output.data);
return SDL_SetError("Truncated data chunk");
} else if (file->trunchint != TruncDropFrame) {
state.output.pos -= state.output.pos % (state.samplesperblock * state.channels);
}
outputsize = state.output.pos * sizeof(Sint16); /* Can't overflow, is always smaller. */
outputsize = state.output.pos * sizeof(Sint16); // Can't overflow, is always smaller.
break;
}
@@ -741,16 +741,16 @@ static int IMA_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
const size_t trailingdata = datalength % format->blockalign;
if (file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict) {
/* The size of the data chunk must be a multiple of the block size. */
// The size of the data chunk must be a multiple of the block size.
if (datalength < blockheadersize || trailingdata > 0) {
return SDL_SetError("Truncated IMA ADPCM block");
}
}
/* Calculate number of sample frames that will be decoded. */
// Calculate number of sample frames that will be decoded.
file->sampleframes = (Uint64)availableblocks * format->samplesperblock;
if (trailingdata > 0) {
/* The last block is truncated. Check if we can get any samples out of it. */
// The last block is truncated. Check if we can get any samples out of it.
if (file->trunchint == TruncDropFrame && trailingdata > blockheadersize - 2) {
/* The sample frame in the header of the truncated block is present.
* Drop incomplete sample frames.
@@ -758,7 +758,7 @@ static int IMA_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
size_t trailingsamples = 1;
if (trailingdata > blockheadersize) {
/* More data following after the header. */
// More data following after the header.
const size_t trailingblockdata = trailingdata - blockheadersize;
const size_t trailingsubblockdata = trailingblockdata % subblockframesize;
trailingsamples += (trailingblockdata / subblockframesize) * 8;
@@ -794,9 +794,9 @@ static int IMA_ADPCM_Init(WaveFile *file, size_t datalength)
const size_t blockframebitsize = (size_t)format->bitspersample * format->channels;
const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize;
/* Sanity checks. */
// Sanity checks.
/* IMA ADPCM can also have 3-bit samples, but it's not supported by SDL at this time. */
// IMA ADPCM can also have 3-bit samples, but it's not supported by SDL at this time.
if (format->bitspersample == 3) {
return SDL_SetError("3-bit IMA ADPCM currently not supported");
} else if (format->bitspersample != 4) {
@@ -815,7 +815,7 @@ static int IMA_ADPCM_Init(WaveFile *file, size_t datalength)
* format because the extensible header has wSampePerBlocks too.
*/
} else {
/* The Standards Update says there 'should' be 2 bytes for wSamplesPerBlock. */
// The Standards Update says there 'should' be 2 bytes for wSamplesPerBlock.
if (chunk->size >= 20 && format->extsize >= 2) {
format->samplesperblock = chunk->data[18] | ((Uint16)chunk->data[19] << 8);
}
@@ -874,17 +874,17 @@ static Sint16 IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 ny
Sint32 sample, delta;
Sint8 index = *cindex;
/* Clamp index into valid range. */
// Clamp index into valid range.
if (index > 88) {
index = 88;
} else if (index < 0) {
index = 0;
}
/* explicit cast to avoid gcc warning about using 'char' as array index */
// explicit cast to avoid gcc warning about using 'char' as array index
step = step_table[(size_t)index];
/* Update index value */
// Update index value
*cindex = index + index_table_4b[nybble];
/* This calculation uses shifts and additions because multiplications were
@@ -909,7 +909,7 @@ static Sint16 IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 ny
sample = lastsample + delta;
/* Clamp output sample */
// Clamp output sample
if (sample > max_audioval) {
sample = max_audioval;
} else if (sample < min_audioval) {
@@ -928,18 +928,18 @@ static int IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
for (c = 0; c < state->channels; c++) {
size_t o = state->block.pos + c * 4;
/* Extract the sample from the header. */
// Extract the sample from the header.
Sint32 sample = state->block.data[o] | ((Sint32)state->block.data[o + 1] << 8);
if (sample >= 0x8000) {
sample -= 0x10000;
}
state->output.data[state->output.pos++] = (Sint16)sample;
/* Channel step index. */
// Channel step index.
step = (Sint16)state->block.data[o + 2];
cstate[c] = (Sint8)(step > 0x80 ? step - 0x100 : step);
/* Reserved byte in block header, should be 0. */
// Reserved byte in block header, should be 0.
if (state->block.data[o + 3] != 0) {
/* Uh oh, corrupt data? Buggy code? */;
}
@@ -947,7 +947,7 @@ static int IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
state->block.pos += state->blockheadersize;
/* Header provided one sample frame. */
// Header provided one sample frame.
state->framesleft--;
return 0;
@@ -980,14 +980,14 @@ static int IMA_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
bytesrequired = (blockframesleft + 7) / 8 * subblockframesize;
if (blockleft < bytesrequired) {
/* Data truncated. Calculate how many samples we can get out if it. */
// Data truncated. Calculate how many samples we can get out if it.
const size_t guaranteedframes = blockleft / subblockframesize;
const size_t remainingbytes = blockleft % subblockframesize;
blockframesleft = guaranteedframes;
if (remainingbytes > subblockframesize - 4) {
blockframesleft += (Sint64)(remainingbytes % 4) * 2;
}
/* Signal the truncation. */
// Signal the truncation.
retval = -1;
}
@@ -1001,7 +1001,7 @@ static int IMA_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
for (c = 0; c < channels; c++) {
Uint8 nybble = 0;
/* Load previous sample which may come from the block header. */
// Load previous sample which may come from the block header.
Sint16 sample = state->output.data[outpos + c - channels];
for (i = 0; i < subblocksamples; i++) {
@@ -1036,13 +1036,13 @@ static int IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len
Sint8 *cstate;
if (chunk->size != chunk->length) {
/* Could not read everything. Recalculate number of sample frames. */
// Could not read everything. Recalculate number of sample frames.
if (IMA_ADPCM_CalculateSampleFrames(file, chunk->size) < 0) {
return -1;
}
}
/* Nothing to decode, nothing to return. */
// Nothing to decode, nothing to return.
if (file->sampleframes == 0) {
*audio_buf = NULL;
*audio_len = 0;
@@ -1062,7 +1062,7 @@ static int IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len
state.input.size = chunk->size;
state.input.pos = 0;
/* The output size in bytes. May get modified if data is truncated. */
// The output size in bytes. May get modified if data is truncated.
outputsize = (size_t)state.framestotal;
if (SafeMult(&outputsize, state.framesize)) {
return SDL_SetError("WAVE file too big");
@@ -1084,7 +1084,7 @@ static int IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len
}
state.cstate = cstate;
/* Decode block by block. A truncated block will stop the decoding. */
// Decode block by block. A truncated block will stop the decoding.
bytesleft = state.input.size - state.input.pos;
while (state.framesleft > 0 && bytesleft >= state.blockheadersize) {
state.block.data = state.input.data + state.input.pos;
@@ -1092,21 +1092,21 @@ static int IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len
state.block.pos = 0;
if (state.output.size - state.output.pos < (Uint64)state.framesleft * state.channels) {
/* Somehow didn't allocate enough space for the output. */
// Somehow didn't allocate enough space for the output.
SDL_free(state.output.data);
SDL_free(cstate);
return SDL_SetError("Unexpected overflow in IMA ADPCM decoder");
}
/* Initialize decoder with the values from the block header. */
// Initialize decoder with the values from the block header.
result = IMA_ADPCM_DecodeBlockHeader(&state);
if (result == 0) {
/* Decode the block data. It stores the samples directly in the output. */
// Decode the block data. It stores the samples directly in the output.
result = IMA_ADPCM_DecodeBlockData(&state);
}
if (result < 0) {
/* Unexpected end. Stop decoding and return partial data if necessary. */
// Unexpected end. Stop decoding and return partial data if necessary.
if (file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict) {
SDL_free(state.output.data);
SDL_free(cstate);
@@ -1114,7 +1114,7 @@ static int IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len
} else if (file->trunchint != TruncDropFrame) {
state.output.pos -= state.output.pos % (state.samplesperblock * state.channels);
}
outputsize = state.output.pos * sizeof(Sint16); /* Can't overflow, is always smaller. */
outputsize = state.output.pos * sizeof(Sint16); // Can't overflow, is always smaller.
break;
}
@@ -1134,12 +1134,12 @@ static int LAW_Init(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
/* Standards Update requires this to be 8. */
// Standards Update requires this to be 8.
if (format->bitspersample != 8) {
return SDL_SetError("Invalid companded bits per sample of %u", (unsigned int)format->bitspersample);
}
/* Not going to bother with weird padding. */
// Not going to bother with weird padding.
if (format->blockalign != format->channels) {
return SDL_SetError("Unsupported block alignment");
}
@@ -1212,7 +1212,7 @@ static int LAW_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
}
}
/* Nothing to decode, nothing to return. */
// Nothing to decode, nothing to return.
if (file->sampleframes == 0) {
*audio_buf = NULL;
*audio_len = 0;
@@ -1231,7 +1231,7 @@ static int LAW_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
return SDL_SetError("WAVE file too big");
}
/* 1 to avoid allocating zero bytes, to keep static analysis happy. */
// 1 to avoid allocating zero bytes, to keep static analysis happy.
src = (Uint8 *)SDL_realloc(chunk->data, expanded_len ? expanded_len : 1);
if (!src) {
return -1;
@@ -1310,7 +1310,7 @@ static int PCM_Init(WaveFile *file, size_t datalength)
case 16:
case 24:
case 32:
/* These are supported. */
// These are supported.
break;
default:
return SDL_SetError("%u-bit PCM format not supported", (unsigned int)format->bitspersample);
@@ -1324,7 +1324,7 @@ static int PCM_Init(WaveFile *file, size_t datalength)
/* It wouldn't be that hard to support more exotic block sizes, but
* the most common formats should do for now.
*/
/* Make sure we're a multiple of the blockalign, at least. */
// Make sure we're a multiple of the blockalign, at least.
if ((format->channels * format->bitspersample) % (format->blockalign * 8)) {
return SDL_SetError("Unsupported block alignment");
}
@@ -1362,20 +1362,20 @@ static int PCM_ConvertSint24ToSint32(WaveFile *file, Uint8 **audio_buf, Uint32 *
return SDL_SetError("WAVE file too big");
}
/* 1 to avoid allocating zero bytes, to keep static analysis happy. */
// 1 to avoid allocating zero bytes, to keep static analysis happy.
ptr = (Uint8 *)SDL_realloc(chunk->data, expanded_len ? expanded_len : 1);
if (!ptr) {
return -1;
}
/* This pointer is now invalid. */
// This pointer is now invalid.
chunk->data = NULL;
chunk->size = 0;
*audio_buf = ptr;
*audio_len = (Uint32)expanded_len;
/* work from end to start, since we're expanding in-place. */
// work from end to start, since we're expanding in-place.
for (i = sample_count; i > 0; i--) {
const size_t o = i - 1;
uint8_t b[4];
@@ -1407,14 +1407,14 @@ static int PCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
}
}
/* Nothing to decode, nothing to return. */
// Nothing to decode, nothing to return.
if (file->sampleframes == 0) {
*audio_buf = NULL;
*audio_len = 0;
return 0;
}
/* 24-bit samples get shifted to 32 bits. */
// 24-bit samples get shifted to 32 bits.
if (format->encoding == PCM_CODE && format->bitspersample == 24) {
return PCM_ConvertSint24ToSint32(file, audio_buf, audio_len);
}
@@ -1429,7 +1429,7 @@ static int PCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
*audio_buf = chunk->data;
*audio_len = (Uint32)outputsize;
/* This pointer is going to be returned to the caller. Prevent free in cleanup. */
// This pointer is going to be returned to the caller. Prevent free in cleanup.
chunk->data = NULL;
chunk->size = 0;
@@ -1507,21 +1507,21 @@ static int WaveNextChunk(SDL_IOStream *src, WaveChunk *chunk)
Uint32 chunkheader[2];
Sint64 nextposition = chunk->position + chunk->length;
/* Data is no longer valid after this function returns. */
// Data is no longer valid after this function returns.
WaveFreeChunkData(chunk);
/* Error on overflows. */
// Error on overflows.
if (SDL_MAX_SINT64 - chunk->length < chunk->position || SDL_MAX_SINT64 - 8 < nextposition) {
return -1;
}
/* RIFF chunks have a 2-byte alignment. Skip padding byte. */
// RIFF chunks have a 2-byte alignment. Skip padding byte.
if (chunk->length & 1) {
nextposition++;
}
if (SDL_SeekIO(src, nextposition, SDL_IO_SEEK_SET) != nextposition) {
/* Not sure how we ended up here. Just abort. */
// Not sure how we ended up here. Just abort.
return -2;
} else if (SDL_ReadIO(src, chunkheader, sizeof(Uint32) * 2) != (sizeof(Uint32) * 2)) {
return -1;
@@ -1549,13 +1549,13 @@ static int WaveReadPartialChunkData(SDL_IOStream *src, WaveChunk *chunk, size_t
}
if (SDL_SeekIO(src, chunk->position, SDL_IO_SEEK_SET) != chunk->position) {
/* Not sure how we ended up here. Just abort. */
// Not sure how we ended up here. Just abort.
return -2;
}
chunk->size = SDL_ReadIO(src, chunk->data, length);
if (chunk->size != length) {
/* Expected to be handled by the caller. */
// Expected to be handled by the caller.
}
}
@@ -1573,7 +1573,7 @@ typedef struct WaveExtensibleGUID
Uint8 guid[16];
} WaveExtensibleGUID;
/* Some of the GUIDs that are used by WAVEFORMATEXTENSIBLE. */
// Some of the GUIDs that are used by WAVEFORMATEXTENSIBLE.
#define WAVE_FORMATTAG_GUID(tag) \
{ \
(tag) & 0xff, (tag) >> 8, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113 \
@@ -1606,7 +1606,7 @@ static int WaveReadFormat(WaveFile *file)
size_t fmtlen = chunk->size;
if (fmtlen > SDL_MAX_SINT32) {
/* Limit given by SDL_IOFromConstMem. */
// Limit given by SDL_IOFromConstMem.
return SDL_SetError("Data of WAVE fmt chunk too big");
}
fmtsrc = SDL_IOFromConstMem(chunk->data, (int)chunk->size);
@@ -1623,7 +1623,7 @@ static int WaveReadFormat(WaveFile *file)
}
format->encoding = format->formattag;
/* This is PCM specific in the first version of the specification. */
// This is PCM specific in the first version of the specification.
if (fmtlen >= 16) {
if (!SDL_ReadU16LE(fmtsrc, &format->bitspersample)) {
return -1;
@@ -1633,7 +1633,7 @@ static int WaveReadFormat(WaveFile *file)
return SDL_SetError("Missing wBitsPerSample field in WAVE fmt chunk");
}
/* The earlier versions also don't have this field. */
// The earlier versions also don't have this field.
if (fmtlen >= 18) {
if (!SDL_ReadU16LE(fmtsrc, &format->extsize)) {
return -1;
@@ -1647,7 +1647,7 @@ static int WaveReadFormat(WaveFile *file)
* to be useful working when they use this format flag.
*/
/* Extensible header must be at least 22 bytes. */
// Extensible header must be at least 22 bytes.
if (fmtlen < 40 || format->extsize < 22) {
SDL_CloseIO(fmtsrc);
return SDL_SetError("Extensible WAVE header too small");
@@ -1670,7 +1670,7 @@ static int WaveCheckFormat(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
/* Check for some obvious issues. */
// Check for some obvious issues.
if (format->channels == 0) {
return SDL_SetError("Invalid number of channels");
@@ -1682,7 +1682,7 @@ static int WaveCheckFormat(WaveFile *file, size_t datalength)
return SDL_SetError("Sample rate exceeds limit of %d", INT_MAX);
}
/* Reject invalid fact chunks in strict mode. */
// Reject invalid fact chunks in strict mode.
if (file->facthint == FactStrict && file->fact.status == -1) {
return SDL_SetError("Invalid fact chunk in WAVE file");
}
@@ -1697,22 +1697,22 @@ static int WaveCheckFormat(WaveFile *file, size_t datalength)
case MULAW_CODE:
case MS_ADPCM_CODE:
case IMA_ADPCM_CODE:
/* These formats require a fact chunk. */
// These formats require a fact chunk.
if (file->facthint == FactStrict && file->fact.status <= 0) {
return SDL_SetError("Missing fact chunk in WAVE file");
}
SDL_FALLTHROUGH;
case PCM_CODE:
/* All supported formats require a non-zero bit depth. */
// All supported formats require a non-zero bit depth.
if (file->chunk.size < 16) {
return SDL_SetError("Missing wBitsPerSample field in WAVE fmt chunk");
} else if (format->bitspersample == 0) {
return SDL_SetError("Invalid bits per sample");
}
/* All supported formats must have a proper block size. */
// All supported formats must have a proper block size.
if (format->blockalign == 0) {
format->blockalign = 1; /* force it to 1 if it was unset. */
format->blockalign = 1; // force it to 1 if it was unset.
}
/* If the fact chunk is valid and the appropriate hint is set, the
@@ -1727,7 +1727,7 @@ static int WaveCheckFormat(WaveFile *file, size_t datalength)
}
}
/* Check the format for encoding specific issues and initialize decoders. */
// Check the format for encoding specific issues and initialize decoders.
switch (format->encoding) {
case PCM_CODE:
case IEEE_FLOAT_CODE:
@@ -1805,24 +1805,24 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
return SDL_SetError("Could not read RIFF header");
}
/* Check main WAVE file identifiers. */
// Check main WAVE file identifiers.
if (RIFFchunk.fourcc == RIFF) {
Uint32 formtype;
/* Read the form type. "WAVE" expected. */
// Read the form type. "WAVE" expected.
if (!SDL_ReadU32LE(src, &formtype)) {
return SDL_SetError("Could not read RIFF form type");
} else if (formtype != WAVE) {
return SDL_SetError("RIFF form type is not WAVE (not a Waveform file)");
}
} else if (RIFFchunk.fourcc == WAVE) {
/* RIFF chunk missing or skipped. Length unknown. */
// RIFF chunk missing or skipped. Length unknown.
RIFFchunk.position = 0;
RIFFchunk.length = 0;
} else {
return SDL_SetError("Could not find RIFF or WAVE identifiers (not a Waveform file)");
}
/* The 4-byte form type is immediately followed by the first chunk.*/
// The 4-byte form type is immediately followed by the first chunk.
chunk->position = RIFFchunk.position + 4;
/* Use the RIFF chunk size to limit the search for the chunks. This is not
@@ -1854,18 +1854,18 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
* currently also ignores cue, list, and slnt chunks.
*/
while ((Uint64)RIFFend > (Uint64)chunk->position + chunk->length + (chunk->length & 1)) {
/* Abort after too many chunks or else corrupt files may waste time. */
// Abort after too many chunks or else corrupt files may waste time.
if (chunkcount++ >= chunkcountlimit) {
return SDL_SetError("Chunk count in WAVE file exceeds limit of %" SDL_PRIu32, chunkcountlimit);
}
result = WaveNextChunk(src, chunk);
if (result < 0) {
/* Unexpected EOF. Corrupt file or I/O issues. */
// Unexpected EOF. Corrupt file or I/O issues.
if (file->trunchint == TruncVeryStrict) {
return SDL_SetError("Unexpected end of WAVE file");
}
/* Let the checks after this loop sort this issue out. */
// Let the checks after this loop sort this issue out.
break;
} else if (result == -2) {
return SDL_SetError("Could not seek to WAVE chunk header");
@@ -1873,9 +1873,9 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
if (chunk->fourcc == FMT) {
if (fmtchunk.fourcc == FMT) {
/* Multiple fmt chunks. Ignore or error? */
// Multiple fmt chunks. Ignore or error?
} else {
/* The fmt chunk must occur before the data chunk. */
// The fmt chunk must occur before the data chunk.
if (datachunk.fourcc == DATA) {
return SDL_SetError("fmt chunk after data chunk in WAVE file");
}
@@ -1896,7 +1896,7 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
if (chunk->length < 4) {
file->fact.status = -1;
} else {
/* Let's use src directly, it's just too convenient. */
// Let's use src directly, it's just too convenient.
Sint64 position = SDL_SeekIO(src, chunk->position, SDL_IO_SEEK_SET);
if (position == chunk->position && SDL_ReadU32LE(src, &file->fact.samplelength)) {
file->fact.status = 1;
@@ -1926,17 +1926,17 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
*/
lastchunkpos = chunk->position + chunk->length;
/* The fmt chunk is mandatory. */
// The fmt chunk is mandatory.
if (fmtchunk.fourcc != FMT) {
return SDL_SetError("Missing fmt chunk in WAVE file");
}
/* A data chunk must be present. */
// A data chunk must be present.
if (datachunk.fourcc != DATA) {
return SDL_SetError("Missing data chunk in WAVE file");
}
/* Check if the last chunk has all of its data in verystrict mode. */
// Check if the last chunk has all of its data in verystrict mode.
if (file->trunchint == TruncVeryStrict) {
/* data chunk is handled later. */
// data chunk is handled later.
if (chunk->fourcc != DATA && chunk->length > 0) {
Uint8 tmp;
Uint64 position = (Uint64)chunk->position + chunk->length - 1;
@@ -1948,7 +1948,7 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
}
}
/* Process fmt chunk. */
// Process fmt chunk.
*chunk = fmtchunk;
/* No need to read more than 1046 bytes of the fmt chunk data with the
@@ -1980,7 +1980,7 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
WaveFreeChunkData(chunk);
/* Process data chunk. */
// Process data chunk.
*chunk = datachunk;
if (chunk->length > 0) {
@@ -1993,14 +1993,14 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
}
if (chunk->length != chunk->size) {
/* I/O issues or corrupt file. */
// I/O issues or corrupt file.
if (file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict) {
return SDL_SetError("Could not read data of WAVE data chunk");
}
/* The decoders handle this truncation. */
// The decoders handle this truncation.
}
/* Decode or convert the data if necessary. */
// Decode or convert the data if necessary.
switch (format->encoding) {
case PCM_CODE:
case IEEE_FLOAT_CODE:
@@ -2038,7 +2038,7 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
case IMA_ADPCM_CODE:
case ALAW_CODE:
case MULAW_CODE:
/* These can be easily stored in the byte order of the system. */
// These can be easily stored in the byte order of the system.
spec->format = SDL_AUDIO_S16;
break;
case IEEE_FLOAT_CODE:
@@ -2052,12 +2052,12 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
case 16:
spec->format = SDL_AUDIO_S16LE;
break;
case 24: /* Has been shifted to 32 bits. */
case 24: // Has been shifted to 32 bits.
case 32:
spec->format = SDL_AUDIO_S32LE;
break;
default:
/* Just in case something unexpected happened in the checks. */
// Just in case something unexpected happened in the checks.
return SDL_SetError("Unexpected %u-bit PCM data format", (unsigned int)format->bitspersample);
}
break;
@@ -2065,7 +2065,7 @@ static int WaveLoad(SDL_IOStream *src, WaveFile *file, SDL_AudioSpec *spec, Uint
return SDL_SetError("Unexpected data format");
}
/* Report the end position back to the cleanup code. */
// Report the end position back to the cleanup code.
if (RIFFlengthknown) {
chunk->position = RIFFend;
} else {
@@ -2090,9 +2090,9 @@ int SDL_LoadWAV_IO(SDL_IOStream *src, SDL_bool closeio, SDL_AudioSpec *spec, Uin
*audio_len = 0;
}
/* Make sure we are passed a valid data source */
// Make sure we are passed a valid data source
if (!src) {
goto done; /* Error may come from SDL_IOStream. */
goto done; // Error may come from SDL_IOStream.
} else if (!spec) {
SDL_InvalidParamError("spec");
goto done;
@@ -2116,7 +2116,7 @@ int SDL_LoadWAV_IO(SDL_IOStream *src, SDL_bool closeio, SDL_AudioSpec *spec, Uin
audio_len = 0;
}
/* Cleanup */
// Cleanup
if (!closeio) {
SDL_SeekIO(src, file.chunk.position, SDL_IO_SEEK_SET);
}