diff options
author | Chris Robinson <[email protected]> | 2019-09-12 03:14:01 -0700 |
---|---|---|
committer | Chris Robinson <[email protected]> | 2019-09-12 03:22:34 -0700 |
commit | 4c76f32ddac5145231609b1cb4f28028abed814b (patch) | |
tree | 139ec9836d367b014e17c6afc614d71338c15ff0 | |
parent | 474d478854ef2ec46bf7b0cb6148c91b7fb8cc2c (diff) |
Avoid implicit conversions with the examples and utils
-rw-r--r-- | examples/alffplay.cpp | 86 | ||||
-rw-r--r-- | examples/alhrtf.c | 13 | ||||
-rw-r--r-- | examples/allatency.c | 4 | ||||
-rw-r--r-- | examples/alloopback.c | 2 | ||||
-rw-r--r-- | examples/almultireverb.c | 18 | ||||
-rw-r--r-- | examples/alplay.c | 4 | ||||
-rw-r--r-- | examples/alrecord.c | 24 | ||||
-rw-r--r-- | examples/alreverb.c | 8 | ||||
-rw-r--r-- | examples/alstream.c | 17 | ||||
-rw-r--r-- | examples/altonegen.c | 8 | ||||
-rw-r--r-- | examples/common/alhelpers.c | 4 | ||||
-rw-r--r-- | utils/makemhr/loadsofa.cpp | 2 | ||||
-rw-r--r-- | utils/openal-info.c | 2 |
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); |