mirror of https://github.com/odrling/Aegisub
Eliminate some warnings
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parent
771525ae6f
commit
897f9d1254
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@ -31,7 +31,7 @@ void AudioProvider::GetAudioWithVolume(void *buf, int64_t start, int64_t count,
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auto buffer = static_cast<int16_t *>(buf);
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auto buffer = static_cast<int16_t *>(buf);
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for (size_t i = 0; i < (size_t)count; ++i)
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for (size_t i = 0; i < (size_t)count; ++i)
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buffer[i] = util::mid<int>(-0x8000, buffer[i] * volume + 0.5, 0x7FFF);
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buffer[i] = util::mid(-0x8000, static_cast<int>(buffer[i] * volume + 0.5), 0x7FFF);
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}
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}
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void AudioProvider::ZeroFill(void *buf, int64_t count) const {
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void AudioProvider::ZeroFill(void *buf, int64_t count) const {
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@ -39,7 +39,10 @@ public:
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bytes_per_sample = sizeof(Target);
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bytes_per_sample = sizeof(Target);
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}
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}
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void FillBuffer(void *buf, int64_t start, int64_t count) const override {
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void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
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auto count = static_cast<size_t>(count64);
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assert(count == count64);
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src_buf.resize(count * src_bytes_per_sample * channels);
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src_buf.resize(count * src_bytes_per_sample * channels);
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source->GetAudio(src_buf.data(), start, count);
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source->GetAudio(src_buf.data(), start, count);
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@ -60,9 +63,9 @@ public:
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}
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}
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if (static_cast<size_t>(src_bytes_per_sample) > sizeof(Target))
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if (static_cast<size_t>(src_bytes_per_sample) > sizeof(Target))
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sample /= 1 << (src_bytes_per_sample - sizeof(Target)) * 8;
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sample /= 1LL << (src_bytes_per_sample - sizeof(Target)) * 8;
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else if (static_cast<size_t>(src_bytes_per_sample) < sizeof(Target))
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else if (static_cast<size_t>(src_bytes_per_sample) < sizeof(Target))
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sample *= 1 << (sizeof(Target) - src_bytes_per_sample ) * 8;
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sample *= 1LL << (sizeof(Target) - src_bytes_per_sample ) * 8;
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dest[i] = static_cast<Target>(sample);
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dest[i] = static_cast<Target>(sample);
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}
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}
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@ -80,7 +83,10 @@ public:
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float_samples = false;
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float_samples = false;
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}
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}
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void FillBuffer(void *buf, int64_t start, int64_t count) const override {
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void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
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auto count = static_cast<size_t>(count64);
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assert(count == count64);
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src_buf.resize(count * channels);
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src_buf.resize(count * channels);
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source->GetAudio(&src_buf[0], start, count);
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source->GetAudio(&src_buf[0], start, count);
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@ -111,7 +117,10 @@ public:
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channels = 1;
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channels = 1;
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}
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}
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void FillBuffer(void *buf, int64_t start, int64_t count) const override {
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void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
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auto count = static_cast<size_t>(count64);
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assert(count == count64);
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src_buf.resize(count * src_channels);
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src_buf.resize(count * src_channels);
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source->GetAudio(&src_buf[0], start, count);
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source->GetAudio(&src_buf[0], start, count);
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@ -52,11 +52,11 @@ class DummyAudioProvider final : public AudioProvider {
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if (noise) {
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if (noise) {
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std::default_random_engine e;
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std::default_random_engine e;
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std::uniform_int_distribution<int16_t> uniform_dist(-5000, 5000);
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std::uniform_int_distribution<int16_t> uniform_dist(-5000, 5000);
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for (size_t i = 0; i < count; ++i)
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for (int64_t i = 0; i < count; ++i)
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static_cast<short *>(buf)[i] = uniform_dist(e);
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static_cast<short *>(buf)[i] = uniform_dist(e);
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}
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}
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else
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else
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memset(buf, 0, count * bytes_per_sample);
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memset(buf, 0, static_cast<size_t>(count) * bytes_per_sample);
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}
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}
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public:
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public:
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@ -1200,7 +1200,7 @@ static Instruction *fix_l (lua_State *L, int t) {
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}
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}
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lua_pushvalue(L, base + 2); /* get initial rule */
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lua_pushvalue(L, base + 2); /* get initial rule */
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lua_gettable(L, base + 1); /* get its position in postions table */
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lua_gettable(L, base + 1); /* get its position in postions table */
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i = lua_tonumber(L, -1); /* convert to number */
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i = lua_tointeger(L, -1); /* convert to number */
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lua_pop(L, 1);
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lua_pop(L, 1);
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if (i == 0) /* is it defined? */
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if (i == 0) /* is it defined? */
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luaL_error(L, "initial rule not defined in given grammar");
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luaL_error(L, "initial rule not defined in given grammar");
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