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-rw-r--r--examples/alffplay.cpp86
-rw-r--r--examples/alhrtf.c13
-rw-r--r--examples/allatency.c4
-rw-r--r--examples/alloopback.c2
-rw-r--r--examples/almultireverb.c18
-rw-r--r--examples/alplay.c4
-rw-r--r--examples/alrecord.c24
-rw-r--r--examples/alreverb.c8
-rw-r--r--examples/alstream.c17
-rw-r--r--examples/altonegen.c8
-rw-r--r--examples/common/alhelpers.c4
-rw-r--r--utils/makemhr/loadsofa.cpp2
-rw-r--r--utils/openal-info.c2
13 files changed, 99 insertions, 93 deletions
diff --git a/examples/alffplay.cpp b/examples/alffplay.cpp
index cdb228e1..655ffc96 100644
--- a/examples/alffplay.cpp
+++ b/examples/alffplay.cpp
@@ -213,7 +213,7 @@ class PacketQueue {
void pop()
{
AVPacket *pkt = &mPackets.front();
- mTotalSize -= pkt->size;
+ mTotalSize -= static_cast<unsigned int>(pkt->size);
av_packet_unref(pkt);
mPackets.pop_front();
}
@@ -267,7 +267,7 @@ public:
return true;
}
- mTotalSize += mPackets.back().size;
+ mTotalSize += static_cast<unsigned int>(mPackets.back().size);
}
mCondVar.notify_one();
return true;
@@ -299,7 +299,7 @@ struct AudioState {
SwrContextPtr mSwresCtx;
/* Conversion format, for what gets fed to OpenAL */
- int mDstChanLayout{0};
+ uint64_t mDstChanLayout{0};
AVSampleFormat mDstSampleFmt{AV_SAMPLE_FMT_NONE};
/* Storage of converted samples */
@@ -310,14 +310,14 @@ struct AudioState {
/* OpenAL format */
ALenum mFormat{AL_NONE};
- ALsizei mFrameSize{0};
+ ALuint mFrameSize{0};
std::mutex mSrcMutex;
std::condition_variable mSrcCond;
std::atomic_flag mConnected;
ALuint mSource{0};
std::vector<ALuint> mBuffers;
- ALsizei mBufferIdx{0};
+ ALuint mBufferIdx{0};
AudioState(MovieState &movie) : mMovie(movie)
{ mConnected.test_and_set(std::memory_order_relaxed); }
@@ -326,7 +326,7 @@ struct AudioState {
if(mSource)
alDeleteSources(1, &mSource);
if(!mBuffers.empty())
- alDeleteBuffers(mBuffers.size(), mBuffers.data());
+ alDeleteBuffers(static_cast<ALsizei>(mBuffers.size()), mBuffers.data());
av_freep(&mSamples);
}
@@ -348,7 +348,7 @@ struct AudioState {
int getSync();
int decodeFrame();
- bool readAudio(uint8_t *samples, int length);
+ bool readAudio(uint8_t *samples, unsigned int length);
int handler();
};
@@ -441,7 +441,7 @@ struct MovieState {
nanoseconds getDuration();
- int streamComponentOpen(int stream_index);
+ int streamComponentOpen(unsigned int stream_index);
int parse_handler();
};
@@ -618,17 +618,17 @@ int AudioState::decodeFrame()
* multiple of the template type size.
*/
template<typename T>
-static void sample_dup(uint8_t *out, const uint8_t *in, int count, int frame_size)
+static void sample_dup(uint8_t *out, const uint8_t *in, unsigned int count, size_t frame_size)
{
- const T *sample = reinterpret_cast<const T*>(in);
- T *dst = reinterpret_cast<T*>(out);
+ auto *sample = reinterpret_cast<const T*>(in);
+ auto *dst = reinterpret_cast<T*>(out);
if(frame_size == sizeof(T))
std::fill_n(dst, count, *sample);
else
{
/* NOTE: frame_size is a multiple of sizeof(T). */
- int type_mult = frame_size / sizeof(T);
- int i = 0;
+ size_t type_mult{frame_size / sizeof(T)};
+ size_t i{0};
std::generate_n(dst, count*type_mult,
[sample,type_mult,&i]() -> T
{
@@ -641,10 +641,10 @@ static void sample_dup(uint8_t *out, const uint8_t *in, int count, int frame_siz
}
-bool AudioState::readAudio(uint8_t *samples, int length)
+bool AudioState::readAudio(uint8_t *samples, unsigned int length)
{
- int sample_skip = getSync();
- int audio_size = 0;
+ int sample_skip{getSync()};
+ unsigned int audio_size{0};
/* Read the next chunk of data, refill the buffer, and queue it
* on the source */
@@ -669,16 +669,17 @@ bool AudioState::readAudio(uint8_t *samples, int length)
continue;
}
- int rem = length - audio_size;
+ unsigned int rem{length - audio_size};
if(mSamplesPos >= 0)
{
- int len = mSamplesLen - mSamplesPos;
+ const auto len = static_cast<unsigned int>(mSamplesLen - mSamplesPos);
if(rem > len) rem = len;
- memcpy(samples, mSamples + mSamplesPos*mFrameSize, rem*mFrameSize);
+ std::copy_n(mSamples + static_cast<unsigned int>(mSamplesPos)*mFrameSize,
+ rem*mFrameSize, samples);
}
else
{
- rem = std::min(rem, -mSamplesPos);
+ rem = std::min(rem, static_cast<unsigned int>(-mSamplesPos));
/* Add samples by copying the first sample */
if((mFrameSize&7) == 0)
@@ -692,7 +693,7 @@ bool AudioState::readAudio(uint8_t *samples, int length)
}
mSamplesPos += rem;
- mCurrentPts += nanoseconds(seconds(rem)) / mCodecCtx->sample_rate;
+ mCurrentPts += nanoseconds{seconds{rem}} / mCodecCtx->sample_rate;
samples += rem*mFrameSize;
audio_size += rem;
}
@@ -701,10 +702,10 @@ bool AudioState::readAudio(uint8_t *samples, int length)
if(audio_size < length)
{
- int rem = length - audio_size;
+ const unsigned int rem{length - audio_size};
std::fill_n(samples, rem*mFrameSize,
(mDstSampleFmt == AV_SAMPLE_FMT_U8) ? 0x80 : 0x00);
- mCurrentPts += nanoseconds(seconds(rem)) / mCodecCtx->sample_rate;
+ mCurrentPts += nanoseconds{seconds{rem}} / mCodecCtx->sample_rate;
audio_size += rem;
}
return true;
@@ -928,8 +929,8 @@ int AudioState::handler()
}
}
void *samples{nullptr};
- ALsizei buffer_len = std::chrono::duration_cast<std::chrono::duration<int>>(
- mCodecCtx->sample_rate * AudioBufferTime).count() * mFrameSize;
+ ALsizei buffer_len = static_cast<int>(std::chrono::duration_cast<seconds>(
+ mCodecCtx->sample_rate * AudioBufferTime).count() * mFrameSize);
mSamples = nullptr;
mSamplesMax = 0;
@@ -968,9 +969,9 @@ int AudioState::handler()
}
else
mSwresCtx.reset(swr_alloc_set_opts(nullptr,
- mDstChanLayout, mDstSampleFmt, mCodecCtx->sample_rate,
- mCodecCtx->channel_layout ? mCodecCtx->channel_layout :
- static_cast<uint64_t>(av_get_default_channel_layout(mCodecCtx->channels)),
+ static_cast<int64_t>(mDstChanLayout), mDstSampleFmt, mCodecCtx->sample_rate,
+ mCodecCtx->channel_layout ? static_cast<int64_t>(mCodecCtx->channel_layout) :
+ av_get_default_channel_layout(mCodecCtx->channels),
mCodecCtx->sample_fmt, mCodecCtx->sample_rate,
0, nullptr));
if(!mSwresCtx || swr_init(mSwresCtx.get()) != 0)
@@ -980,7 +981,7 @@ int AudioState::handler()
}
mBuffers.assign(AudioBufferTotalTime / AudioBufferTime, 0);
- alGenBuffers(mBuffers.size(), mBuffers.data());
+ alGenBuffers(static_cast<ALsizei>(mBuffers.size()), mBuffers.data());
alGenSources(1, &mSource);
if(EnableDirectOut)
@@ -1003,12 +1004,12 @@ int AudioState::handler()
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Failed to use mapped buffers\n");
- samples = av_malloc(buffer_len);
+ samples = av_malloc(static_cast<ALuint>(buffer_len));
}
}
else
#endif
- samples = av_malloc(buffer_len);
+ samples = av_malloc(static_cast<ALuint>(buffer_len));
/* Prefill the codec buffer. */
do {
@@ -1053,14 +1054,15 @@ int AudioState::handler()
{
auto ptr = static_cast<uint8_t*>(alMapBufferSOFT(bufid, 0, buffer_len,
AL_MAP_WRITE_BIT_SOFT));
- bool got_audio{readAudio(ptr, buffer_len)};
+ bool got_audio{readAudio(ptr, static_cast<unsigned int>(buffer_len))};
alUnmapBufferSOFT(bufid);
if(!got_audio) break;
}
else
#endif
{
- if(!readAudio(static_cast<uint8_t*>(samples), buffer_len))
+ auto ptr = static_cast<uint8_t*>(samples);
+ if(!readAudio(ptr, static_cast<unsigned int>(buffer_len)))
break;
alBufferData(bufid, mFormat, samples, buffer_len, mCodecCtx->sample_rate);
}
@@ -1138,27 +1140,27 @@ void VideoState::display(SDL_Window *screen, SDL_Renderer *renderer)
if(!mImage)
return;
- float aspect_ratio;
+ double aspect_ratio;
int win_w, win_h;
int w, h, x, y;
if(mCodecCtx->sample_aspect_ratio.num == 0)
- aspect_ratio = 0.0f;
+ aspect_ratio = 0.0;
else
{
aspect_ratio = av_q2d(mCodecCtx->sample_aspect_ratio) * mCodecCtx->width /
mCodecCtx->height;
}
- if(aspect_ratio <= 0.0f)
- aspect_ratio = static_cast<float>(mCodecCtx->width) / static_cast<float>(mCodecCtx->height);
+ if(aspect_ratio <= 0.0)
+ aspect_ratio = static_cast<double>(mCodecCtx->width) / mCodecCtx->height;
SDL_GetWindowSize(screen, &win_w, &win_h);
h = win_h;
- w = (static_cast<int>(rint(h * aspect_ratio)) + 3) & ~3;
+ w = (static_cast<int>(std::rint(h * aspect_ratio)) + 3) & ~3;
if(w > win_w)
{
w = win_w;
- h = (static_cast<int>(rint(w / aspect_ratio)) + 3) & ~3;
+ h = (static_cast<int>(std::rint(w / aspect_ratio)) + 3) & ~3;
}
x = (win_w - w) / 2;
y = (win_h - h) / 2;
@@ -1460,9 +1462,9 @@ nanoseconds MovieState::getMasterClock()
nanoseconds MovieState::getDuration()
{ return std::chrono::duration<int64_t,std::ratio<1,AV_TIME_BASE>>(mFormatCtx->duration); }
-int MovieState::streamComponentOpen(int stream_index)
+int MovieState::streamComponentOpen(unsigned int stream_index)
{
- if(stream_index < 0 || static_cast<unsigned int>(stream_index) >= mFormatCtx->nb_streams)
+ if(stream_index >= mFormatCtx->nb_streams)
return -1;
/* Get a pointer to the codec context for the stream, and open the
@@ -1499,7 +1501,7 @@ int MovieState::streamComponentOpen(int stream_index)
return -1;
}
- return stream_index;
+ return static_cast<int>(stream_index);
}
int MovieState::parse_handler()
diff --git a/examples/alhrtf.c b/examples/alhrtf.c
index 96cf0255..f09f3e99 100644
--- a/examples/alhrtf.c
+++ b/examples/alhrtf.c
@@ -112,7 +112,7 @@ static ALuint LoadSound(const char *filename)
* close the file. */
buffer = 0;
alGenBuffers(1, &buffer);
- alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate);
+ alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
Sound_FreeSample(sample);
/* Check if an error occured, and clean up if so. */
@@ -132,6 +132,7 @@ static ALuint LoadSound(const char *filename)
int main(int argc, char **argv)
{
ALCdevice *device;
+ ALCcontext *context;
ALboolean has_angle_ext;
ALuint source, buffer;
const char *soundname;
@@ -153,7 +154,8 @@ int main(int argc, char **argv)
if(InitAL(&argv, &argc) != 0)
return 1;
- device = alcGetContextsDevice(alcGetCurrentContext());
+ context = alcGetCurrentContext();
+ device = alcGetContextsDevice(context);
if(!alcIsExtensionPresent(device, "ALC_SOFT_HRTF"))
{
fprintf(stderr, "Error: ALC_SOFT_HRTF not supported\n");
@@ -171,7 +173,7 @@ int main(int argc, char **argv)
* stereo sources.
*/
has_angle_ext = alIsExtensionPresent("AL_EXT_STEREO_ANGLES");
- printf("AL_EXT_STEREO_ANGLES%s found\n", has_angle_ext?"":" not");
+ printf("AL_EXT_STEREO_ANGLES %sfound\n", has_angle_ext?"":"not ");
/* Check for user-preferred HRTF */
if(strcmp(argv[0], "-hrtf") == 0)
@@ -255,7 +257,7 @@ int main(int argc, char **argv)
alGenSources(1, &source);
alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE);
alSource3f(source, AL_POSITION, 0.0f, 0.0f, -1.0f);
- alSourcei(source, AL_BUFFER, buffer);
+ alSourcei(source, AL_BUFFER, (ALint)buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
@@ -264,6 +266,8 @@ int main(int argc, char **argv)
do {
al_nssleep(10000000);
+ alcSuspendContext(context);
+
/* Rotate the source around the listener by about 1/4 cycle per second,
* and keep it within -pi...+pi.
*/
@@ -282,6 +286,7 @@ int main(int argc, char **argv)
ALfloat angles[2] = { (ALfloat)(M_PI/6.0 - angle), (ALfloat)(-M_PI/6.0 - angle) };
alSourcefv(source, AL_STEREO_ANGLES, angles);
}
+ alcProcessContext(context);
alGetSourcei(source, AL_SOURCE_STATE, &state);
} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
diff --git a/examples/allatency.c b/examples/allatency.c
index 2bc76289..ad700cc1 100644
--- a/examples/allatency.c
+++ b/examples/allatency.c
@@ -115,7 +115,7 @@ static ALuint LoadSound(const char *filename)
* close the file. */
buffer = 0;
alGenBuffers(1, &buffer);
- alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate);
+ alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
Sound_FreeSample(sample);
/* Check if an error occured, and clean up if so. */
@@ -188,7 +188,7 @@ int main(int argc, char **argv)
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
- alSourcei(source, AL_BUFFER, buffer);
+ alSourcei(source, AL_BUFFER, (ALint)buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
diff --git a/examples/alloopback.c b/examples/alloopback.c
index 313b89d5..426a2af9 100644
--- a/examples/alloopback.c
+++ b/examples/alloopback.c
@@ -249,7 +249,7 @@ int main(int argc, char *argv[])
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
- alSourcei(source, AL_BUFFER, buffer);
+ alSourcei(source, AL_BUFFER, (ALint)buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
diff --git a/examples/almultireverb.c b/examples/almultireverb.c
index efd3bf16..b967a050 100644
--- a/examples/almultireverb.c
+++ b/examples/almultireverb.c
@@ -211,7 +211,7 @@ static ALuint LoadSound(const char *filename)
* close the file. */
buffer = 0;
alGenBuffers(1, &buffer);
- alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate);
+ alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
Sound_FreeSample(sample);
/* Check if an error occured, and clean up if so. */
@@ -443,8 +443,8 @@ static void UpdateListenerAndEffects(float timediff, const ALuint slots[2], cons
}
/* Finally, update the effect slots with the updated effect parameters. */
- alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, effects[0]);
- alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, effects[1]);
+ alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, (ALint)effects[0]);
+ alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, (ALint)effects[1]);
}
@@ -598,8 +598,8 @@ int main(int argc, char **argv)
* effect properties. Modifying or deleting the effect object afterward
* won't directly affect the effect slot until they're reapplied like this.
*/
- alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, effects[0]);
- alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, effects[1]);
+ alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, (ALint)effects[0]);
+ alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, (ALint)effects[1]);
assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
/* For the purposes of this example, prepare a filter that optionally
@@ -621,8 +621,8 @@ int main(int argc, char **argv)
alGenSources(1, &source);
alSourcei(source, AL_LOOPING, AL_TRUE);
alSource3f(source, AL_POSITION, -5.0f, 0.0f, -2.0f);
- alSourcei(source, AL_DIRECT_FILTER, direct_filter);
- alSourcei(source, AL_BUFFER, buffer);
+ alSourcei(source, AL_DIRECT_FILTER, (ALint)direct_filter);
+ alSourcei(source, AL_BUFFER, (ALint)buffer);
/* Connect the source to the effect slots. Here, we connect source send 0
* to Zone 0's slot, and send 1 to Zone 1's slot. Filters can be specified
@@ -631,8 +631,8 @@ int main(int argc, char **argv)
* can only see a zone through a window or thin wall may be attenuated for
* that zone.
*/
- alSource3i(source, AL_AUXILIARY_SEND_FILTER, slots[0], 0, AL_FILTER_NULL);
- alSource3i(source, AL_AUXILIARY_SEND_FILTER, slots[1], 1, AL_FILTER_NULL);
+ alSource3i(source, AL_AUXILIARY_SEND_FILTER, (ALint)slots[0], 0, AL_FILTER_NULL);
+ alSource3i(source, AL_AUXILIARY_SEND_FILTER, (ALint)slots[1], 1, AL_FILTER_NULL);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Get the current time as the base for timing in the main loop. */
diff --git a/examples/alplay.c b/examples/alplay.c
index 4ff8fb7f..09ad96b4 100644
--- a/examples/alplay.c
+++ b/examples/alplay.c
@@ -101,7 +101,7 @@ static ALuint LoadSound(const char *filename)
* close the file. */
buffer = 0;
alGenBuffers(1, &buffer);
- alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate);
+ alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
Sound_FreeSample(sample);
/* Check if an error occured, and clean up if so. */
@@ -151,7 +151,7 @@ int main(int argc, char **argv)
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
- alSourcei(source, AL_BUFFER, buffer);
+ alSourcei(source, AL_BUFFER, (ALint)buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
diff --git a/examples/alrecord.c b/examples/alrecord.c
index d65414c9..627f8540 100644
--- a/examples/alrecord.c
+++ b/examples/alrecord.c
@@ -73,9 +73,9 @@ typedef struct Recorder {
ALuint mDataSize;
float mRecTime;
- int mChannels;
- int mBits;
- int mSampleRate;
+ ALuint mChannels;
+ ALuint mBits;
+ ALuint mSampleRate;
ALuint mFrameSize;
ALbyte *mBuffer;
ALsizei mBufferSize;
@@ -139,7 +139,7 @@ int main(int argc, char **argv)
return 1;
}
- recorder.mChannels = strtol(argv[1], &end, 0);
+ recorder.mChannels = (ALuint)strtoul(argv[1], &end, 0);
if((recorder.mChannels != 1 && recorder.mChannels != 2) || (end && *end != '\0'))
{
fprintf(stderr, "Invalid channels: %s\n", argv[1]);
@@ -156,7 +156,7 @@ int main(int argc, char **argv)
return 1;
}
- recorder.mBits = strtol(argv[1], &end, 0);
+ recorder.mBits = (ALuint)strtoul(argv[1], &end, 0);
if((recorder.mBits != 8 && recorder.mBits != 16 && recorder.mBits != 32) ||
(end && *end != '\0'))
{
@@ -174,7 +174,7 @@ int main(int argc, char **argv)
return 1;
}
- recorder.mSampleRate = strtol(argv[1], &end, 0);
+ recorder.mSampleRate = (ALuint)strtoul(argv[1], &end, 0);
if(!(recorder.mSampleRate >= 8000 && recorder.mSampleRate <= 96000) || (end && *end != '\0'))
{
fprintf(stderr, "Invalid sample rate: %s\n", argv[1]);
@@ -285,15 +285,15 @@ int main(int argc, char **argv)
// 16-bit val, format type id (1 = integer PCM, 3 = float PCM)
fwrite16le((recorder.mBits == 32) ? 0x0003 : 0x0001, recorder.mFile);
// 16-bit val, channel count
- fwrite16le(recorder.mChannels, recorder.mFile);
+ fwrite16le((ALushort)recorder.mChannels, recorder.mFile);
// 32-bit val, frequency
fwrite32le(recorder.mSampleRate, recorder.mFile);
// 32-bit val, bytes per second
fwrite32le(recorder.mSampleRate * recorder.mFrameSize, recorder.mFile);
// 16-bit val, frame size
- fwrite16le(recorder.mFrameSize, recorder.mFile);
+ fwrite16le((ALushort)recorder.mFrameSize, recorder.mFile);
// 16-bit val, bits per sample
- fwrite16le(recorder.mBits, recorder.mFile);
+ fwrite16le((ALushort)recorder.mBits, recorder.mFile);
// 16-bit val, extra byte count
fwrite16le(0, recorder.mFile);
@@ -331,7 +331,7 @@ int main(int argc, char **argv)
}
if(count > recorder.mBufferSize)
{
- ALbyte *data = calloc(recorder.mFrameSize, count);
+ ALbyte *data = calloc(recorder.mFrameSize, (ALuint)count);
free(recorder.mBuffer);
recorder.mBuffer = data;
recorder.mBufferSize = count;
@@ -365,7 +365,7 @@ int main(int argc, char **argv)
}
}
#endif
- recorder.mDataSize += (ALuint)fwrite(recorder.mBuffer, recorder.mFrameSize, count,
+ recorder.mDataSize += (ALuint)fwrite(recorder.mBuffer, recorder.mFrameSize, (ALuint)count,
recorder.mFile);
}
alcCaptureStop(recorder.mDevice);
@@ -385,7 +385,7 @@ int main(int argc, char **argv)
{
fwrite32le(recorder.mDataSize*recorder.mFrameSize, recorder.mFile);
if(fseek(recorder.mFile, 4, SEEK_SET) == 0)
- fwrite32le(total_size - 8, recorder.mFile);
+ fwrite32le((ALuint)total_size - 8, recorder.mFile);
}
fclose(recorder.mFile);
diff --git a/examples/alreverb.c b/examples/alreverb.c
index e1d3c207..68f0269f 100644
--- a/examples/alreverb.c
+++ b/examples/alreverb.c
@@ -209,7 +209,7 @@ static ALuint LoadSound(const char *filename)
* close the file. */
buffer = 0;
alGenBuffers(1, &buffer);
- alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate);
+ alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
Sound_FreeSample(sample);
/* Check if an error occured, and clean up if so. */
@@ -309,18 +309,18 @@ int main(int argc, char **argv)
* effectively copies the effect properties. You can modify or delete the
* effect object afterward without affecting the effect slot.
*/
- alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, effect);
+ alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, (ALint)effect);
assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
- alSourcei(source, AL_BUFFER, buffer);
+ alSourcei(source, AL_BUFFER, (ALint)buffer);
/* Connect the source to the effect slot. This tells the source to use the
* effect slot 'slot', on send #0 with the AL_FILTER_NULL filter object.
*/
- alSource3i(source, AL_AUXILIARY_SEND_FILTER, slot, 0, AL_FILTER_NULL);
+ alSource3i(source, AL_AUXILIARY_SEND_FILTER, (ALint)slot, 0, AL_FILTER_NULL);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
diff --git a/examples/alstream.c b/examples/alstream.c
index cb447355..56505ddb 100644
--- a/examples/alstream.c
+++ b/examples/alstream.c
@@ -160,10 +160,10 @@ static int OpenPlayerFile(StreamPlayer *player, const char *filename)
fprintf(stderr, "Unsupported channel count: %d\n", player->sample->actual.channels);
goto error;
}
- player->srate = player->sample->actual.rate;
+ player->srate = (ALsizei)player->sample->actual.rate;
frame_size = player->sample->actual.channels *
- SDL_AUDIO_BITSIZE(player->sample->actual.format) / 8;
+ SDL_AUDIO_BITSIZE(player->sample->actual.format) / 8;
/* Set the buffer size, given the desired millisecond length. */
Sound_SetBufferSize(player->sample, (Uint32)((Uint64)player->srate*BUFFER_TIME_MS/1000) *
@@ -191,7 +191,7 @@ static void ClosePlayerFile(StreamPlayer *player)
/* Prebuffers some audio from the file, and starts playing the source */
static int StartPlayer(StreamPlayer *player)
{
- size_t i;
+ ALsizei i;
/* Rewind the source position and clear the buffer queue */
alSourceRewind(player->source);
@@ -204,8 +204,8 @@ static int StartPlayer(StreamPlayer *player)
Uint32 slen = Sound_Decode(player->sample);
if(slen == 0) break;
- alBufferData(player->buffers[i], player->format,
- player->sample->buffer, slen, player->srate);
+ alBufferData(player->buffers[i], player->format, player->sample->buffer, (ALsizei)slen,
+ player->srate);
}
if(alGetError() != AL_NO_ERROR)
{
@@ -255,8 +255,8 @@ static int UpdatePlayer(StreamPlayer *player)
slen = Sound_Decode(player->sample);
if(slen > 0)
{
- alBufferData(bufid, player->format, player->sample->buffer, slen,
- player->srate);
+ alBufferData(bufid, player->format, player->sample->buffer, (ALsizei)slen,
+ player->srate);
alSourceQueueBuffers(player->source, 1, &bufid);
}
if(alGetError() != AL_NO_ERROR)
@@ -323,8 +323,7 @@ int main(int argc, char **argv)
else
namepart = argv[i];
- printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
- player->srate);
+ printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format), player->srate);
fflush(stdout);
if(!StartPlayer(player))
diff --git a/examples/altonegen.c b/examples/altonegen.c
index 628e695d..aacc3496 100644
--- a/examples/altonegen.c
+++ b/examples/altonegen.c
@@ -91,7 +91,7 @@ static void ApplySin(ALfloat *data, ALdouble g, ALuint srate, ALuint freq)
static ALuint CreateWave(enum WaveType type, ALuint freq, ALuint srate)
{
ALuint seed = 22222;
- ALint data_size;
+ ALuint data_size;
ALfloat *data;
ALuint buffer;
ALenum err;
@@ -142,7 +142,7 @@ static ALuint CreateWave(enum WaveType type, ALuint freq, ALuint srate)
/* Buffer the audio data into a new buffer object. */
buffer = 0;
alGenBuffers(1, &buffer);
- alBufferData(buffer, AL_FORMAT_MONO_FLOAT32, data, data_size, srate);
+ alBufferData(buffer, AL_FORMAT_MONO_FLOAT32, data, (ALsizei)data_size, (ALsizei)srate);
free(data);
/* Check if an error occured, and clean up if so. */
@@ -257,7 +257,7 @@ int main(int argc, char *argv[])
srate = dev_rate;
/* Load the sound into a buffer. */
- buffer = CreateWave(wavetype, tone_freq, srate);
+ buffer = CreateWave(wavetype, (ALuint)tone_freq, (ALuint)srate);
if(!buffer)
{
CloseAL();
@@ -271,7 +271,7 @@ int main(int argc, char *argv[])
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
- alSourcei(source, AL_BUFFER, buffer);
+ alSourcei(source, AL_BUFFER, (ALint)buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound for a while. */
diff --git a/examples/common/alhelpers.c b/examples/common/alhelpers.c
index b387fd2d..730d2e13 100644
--- a/examples/common/alhelpers.c
+++ b/examples/common/alhelpers.c
@@ -159,12 +159,12 @@ int altime_get(void)
struct timespec ts;
int ret = clock_gettime(CLOCK_REALTIME, &ts);
if(ret != 0) return 0;
- cur_time = ts.tv_sec*1000 + ts.tv_nsec/1000000;
+ cur_time = (int)(ts.tv_sec*1000 + ts.tv_nsec/1000000);
#else /* _POSIX_TIMERS > 0 */
struct timeval tv;
int ret = gettimeofday(&tv, NULL);
if(ret != 0) return 0;
- cur_time = tv.tv_sec*1000 + tv.tv_usec/1000;
+ cur_time = (int)(tv.tv_sec*1000 + tv.tv_usec/1000);
#endif
if(!start_time)
diff --git a/utils/makemhr/loadsofa.cpp b/utils/makemhr/loadsofa.cpp
index e82376aa..02911e12 100644
--- a/utils/makemhr/loadsofa.cpp
+++ b/utils/makemhr/loadsofa.cpp
@@ -50,7 +50,7 @@ static const char *SofaErrorStr(int err)
* of other axes as necessary. The epsilons are used to constrain the
* equality of unique elements.
*/
-static uint GetUniquelySortedElems(const uint m, const float *triplets, const int axis,
+static uint GetUniquelySortedElems(const uint m, const float *triplets, const uint axis,
const double *const (&filters)[3], const double (&epsilons)[3], float *elems)
{
uint count{0u};
diff --git a/utils/openal-info.c b/utils/openal-info.c
index 12dc6311..cc628b6e 100644
--- a/utils/openal-info.c
+++ b/utils/openal-info.c
@@ -124,7 +124,7 @@ static void printList(const char *list, char separator)
next = strchr(list, separator);
if(next)
{
- len = next-list;
+ len = (size_t)(next-list);
do {
next++;
} while(*next == separator);
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-29 18:37:03 +0000