fix double promotions

src/ReplayGainGlobal.cxx

@@ -23,24 +23,23 @@
 #include "util/Math.hxx"
 
 #include <cassert>
+#include <cstdlib>
 #include <stdexcept>
 
-#include <stdlib.h>
-
 static float
 ParsePreamp(const char *s)
 {
 	assert(s != nullptr);
 
 	char *endptr;
-	float f = strtod(s, &endptr);
+	float f = std::strtof(s, &endptr);
 	if (endptr == s || *endptr != '\0')
 		throw std::invalid_argument("Not a numeric value");
 
-	if (f < -15 || f > 15)
+	if (f < -15.0f || f > 15.0f)
 		throw std::invalid_argument("Number must be between -15 and 15");
 
-	return pow(10, f / 20.0);
+	return std::pow(10.0f, f / 20.0f);
 }
 
 ReplayGainConfig
@@ -51,13 +50,13 @@ LoadReplayGainConfig(const ConfigData &config)
 	replay_gain_config.preamp = config.With(ConfigOption::REPLAYGAIN_PREAMP, [](const char *s){
 		return s != nullptr
 			? ParsePreamp(s)
-			: 1.0;
+			: 1.0f;
 	});
 
 	replay_gain_config.missing_preamp = config.With(ConfigOption::REPLAYGAIN_MISSING_PREAMP, [](const char *s){
 		return s != nullptr
 			? ParsePreamp(s)
-			: 1.0;
+			: 1.0f;
 	});
 
 	replay_gain_config.limit = config.GetBool(ConfigOption::REPLAYGAIN_LIMIT,

src/ReplayGainInfo.cxx

@@ -27,13 +27,13 @@ ReplayGainTuple::CalculateScale(const ReplayGainConfig &config) const noexcept
 	float scale;
 
 	if (IsDefined()) {
-		scale = pow(10.0, gain / 20.0);
+		scale = std::pow(10.0f, gain / 20.0f);
 		scale *= config.preamp;
-		if (scale > 15.0)
-			scale = 15.0;
+		if (scale > 15.0f)
+			scale = 15.0f;
 
-		if (config.limit && scale * peak > 1.0)
-			scale = 1.0 / peak;
+		if (config.limit && scale * peak > 1.0f)
+			scale = 1.0f / peak;
 	} else
 		scale = config.missing_preamp;
 

src/command/PlayerCommands.cxx

@@ -149,7 +149,7 @@ handle_status(Client &client, [[maybe_unused]] Request args, Response &r)
 		 partition.name.c_str(),
 		 (unsigned long)playlist.GetVersion(),
 		 playlist.GetLength(),
-		 pc.GetMixRampDb(),
+		 (double)pc.GetMixRampDb(),
 		 state);
 
 	if (pc.GetCrossFade() > FloatDuration::zero())

src/decoder/Bridge.cxx

@@ -611,7 +611,7 @@ DecoderBridge::SubmitReplayGain(const ReplayGainInfo *new_replay_gain_info) noex
 			const auto &tuple = new_replay_gain_info->Get(rgm);
 			const auto scale =
 				tuple.CalculateScale(dc.replay_gain_config);
-			dc.replay_gain_db = 20.0 * std::log10(scale);
+			dc.replay_gain_db = 20.0f * std::log10(scale);
 		}
 
 		replay_gain_info = *new_replay_gain_info;

src/decoder/plugins/MadDecoderPlugin.cxx

@@ -585,8 +585,8 @@ parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen) noexcept
 
 	mad_bit_skip(ptr, 16);
 
-	lame->peak = mad_f_todouble(mad_bit_read(ptr, 32) << 5); /* peak */
-	FormatDebug(mad_domain, "LAME peak found: %f", lame->peak);
+	lame->peak = MAD_F(mad_bit_read(ptr, 32) << 5); /* peak */
+	FormatDebug(mad_domain, "LAME peak found: %f", double(lame->peak));
 
 	lame->track_gain = 0;
 	unsigned name = mad_bit_read(ptr, 3); /* gain name */
@@ -594,9 +594,9 @@ parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen) noexcept
 	unsigned sign = mad_bit_read(ptr, 1); /* sign bit */
 	int gain = mad_bit_read(ptr, 9); /* gain*10 */
 	if (gain && name == 1 && orig != 0) {
-		lame->track_gain = ((sign ? -gain : gain) / 10.0) + adj;
+		lame->track_gain = ((sign ? -gain : gain) / 10.0f) + adj;
 		FormatDebug(mad_domain, "LAME track gain found: %f",
-			    lame->track_gain);
+			    double(lame->track_gain));
 	}
 
 	/* tmz reports that this isn't currently written by any version of lame
@@ -612,7 +612,7 @@ parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen) noexcept
 	if (gain && name == 2 && orig != 0) {
 		lame->album_gain = ((sign ? -gain : gain) / 10.0) + adj;
 		FormatDebug(mad_domain, "LAME album gain found: %f",
-			    lame->track_gain);
+			    double(lame->track_gain));
 	}
 #else
 	mad_bit_skip(ptr, 16);
@@ -746,7 +746,7 @@ MadDecoder::DecodeFirstFrame(Tag *tag) noexcept
 			/* Album gain isn't currently used.  See comment in
 			 * parse_lame() for details. -- jat */
 			if (client != nullptr && !found_replay_gain &&
-			    lame.track_gain) {
+			    lame.track_gain > 0.0f) {
 				ReplayGainInfo rgi;
 				rgi.Clear();
 				rgi.track.gain = lame.track_gain;

src/decoder/plugins/OpusTags.cxx

@@ -61,7 +61,7 @@ ScanOneOpusTag(StringView name, StringView value,
 		const char *endptr;
 		const auto l = ParseInt64(value, &endptr, 10);
 		if (endptr > value.begin() && endptr == value.end())
-			rgi->track.gain = double(l) / 256.;
+			rgi->track.gain = float(l) / 256.0f;
 	} else if (rgi != nullptr &&
 		   name.EqualsIgnoreCase("R128_ALBUM_GAIN")) {
 		/* R128_ALBUM_GAIN is a Q7.8 fixed point number in
@@ -70,7 +70,7 @@ ScanOneOpusTag(StringView name, StringView value,
 		const char *endptr;
 		const auto l = ParseInt64(value, &endptr, 10);
 		if (endptr > value.begin() && endptr == value.end())
-			rgi->album.gain = double(l) / 256.;
+			rgi->album.gain = float(l) / 256.0f;
 	}
 
 	handler.OnPair(name, value);

src/encoder/plugins/LameEncoderPlugin.cxx

@@ -76,9 +76,9 @@ PreparedLameEncoder::PreparedLameEncoder(const ConfigBlock &block)
 	if (value != nullptr) {
 		/* a quality was configured (VBR) */
 
-		quality = ParseDouble(value, &endptr);
+		quality = float(ParseDouble(value, &endptr));
 
-		if (*endptr != '\0' || quality < -1.0 || quality > 10.0)
+		if (*endptr != '\0' || quality < -1.0f || quality > 10.0f)
 			throw FormatRuntimeError("quality \"%s\" is not a number in the "
 						 "range -1 to 10",
 						 value);
@@ -110,13 +110,13 @@ static void
 lame_encoder_setup(lame_global_flags *gfp, float quality, int bitrate,
 		   const AudioFormat &audio_format)
 {
-	if (quality >= -1.0) {
+	if (quality >= -1.0f) {
 		/* a quality was configured (VBR) */
 
 		if (0 != lame_set_VBR(gfp, vbr_rh))
 			throw std::runtime_error("error setting lame VBR mode");
 
-		if (0 != lame_set_VBR_q(gfp, quality))
+		if (0 != lame_set_VBR_q(gfp, int(quality)))
 			throw std::runtime_error("error setting lame VBR quality");
 	} else {
 		/* a bit rate was configured */

src/encoder/plugins/TwolameEncoderPlugin.cxx

@@ -94,9 +94,9 @@ PreparedTwolameEncoder::PreparedTwolameEncoder(const ConfigBlock &block)
 	if (value != nullptr) {
 		/* a quality was configured (VBR) */
 
-		quality = ParseDouble(value, &endptr);
+		quality = float(ParseDouble(value, &endptr));
 
-		if (*endptr != '\0' || quality < -1.0 || quality > 10.0)
+		if (*endptr != '\0' || quality < -1.0f || quality > 10.0f)
 			throw FormatRuntimeError("quality \"%s\" is not a number in the "
 						 "range -1 to 10",
 						 value);
@@ -131,7 +131,7 @@ static void
 twolame_encoder_setup(twolame_options *options, float quality, int bitrate,
 		      const AudioFormat &audio_format)
 {
-	if (quality >= -1.0) {
+	if (quality >= -1.0f) {
 		/* a quality was configured (VBR) */
 
 		if (0 != twolame_set_VBR(options, true))

src/encoder/plugins/VorbisEncoderPlugin.cxx

@@ -84,7 +84,7 @@ PreparedVorbisEncoder::PreparedVorbisEncoder(const ConfigBlock &block)
 		char *endptr;
 		quality = ParseDouble(value, &endptr);
 
-		if (*endptr != '\0' || quality < -1.0 || quality > 10.0)
+		if (*endptr != '\0' || quality < -1.0f || quality > 10.0f)
 			throw FormatRuntimeError("quality \"%s\" is not a number in the "
 						 "range -1 to 10",
 						 value);
@@ -122,13 +122,13 @@ VorbisEncoder::VorbisEncoder(float quality, int bitrate,
 	_audio_format.format = SampleFormat::FLOAT;
 	audio_format = _audio_format;
 
-	if (quality >= -1.0) {
+	if (quality >= -1.0f) {
 		/* a quality was configured (VBR) */
 
 		if (0 != vorbis_encode_init_vbr(&vi,
 						audio_format.channels,
 						audio_format.sample_rate,
-						quality * 0.1)) {
+						quality * 0.1f)) {
 			vorbis_info_clear(&vi);
 			throw std::runtime_error("error initializing vorbis vbr");
 		}
@@ -138,7 +138,7 @@ VorbisEncoder::VorbisEncoder(float quality, int bitrate,
 		if (0 != vorbis_encode_init(&vi,
 					    audio_format.channels,
 					    audio_format.sample_rate, -1.0,
-					    bitrate * 1000, -1.0)) {
+					    bitrate * 1000, -1.0f)) {
 			vorbis_info_clear(&vi);
 			throw std::runtime_error("error initializing vorbis encoder");
 		}

src/mixer/plugins/PulseMixerPlugin.cxx

@@ -49,7 +49,7 @@ public:
 		   double _volume_scale_factor)
 		:Mixer(pulse_mixer_plugin, _listener),
 		 output(_output),
-		 volume_scale_factor(_volume_scale_factor)
+		 volume_scale_factor(float(_volume_scale_factor))
 	{
 	}
 
@@ -173,7 +173,7 @@ parse_volume_scale_factor(const char *value) {
 	char *endptr;
 	float factor = ParseFloat(value, &endptr);
 
-	if (endptr == value || *endptr != '\0' || factor < 0.5 || factor > 5.0)
+	if (endptr == value || *endptr != '\0' || factor < 0.5f || factor > 5.0f)
 		throw FormatRuntimeError("\"%s\" is not a number in the "
 					 "range 0.5 to 5.0",
 					 value);
@@ -188,7 +188,7 @@ pulse_mixer_init([[maybe_unused]] EventLoop &event_loop, AudioOutput &ao,
 {
 	auto &po = (PulseOutput &)ao;
 	float scale = parse_volume_scale_factor(block.GetBlockValue("scale_volume"));
-	auto *pm = new PulseMixer(po, listener, scale);
+	auto *pm = new PulseMixer(po, listener, (double)scale);
 
 	pulse_output_set_mixer(po, *pm);
 
@@ -214,7 +214,7 @@ PulseMixer::GetVolume()
 int
 PulseMixer::GetVolumeInternal()
 {
-	pa_volume_t max_pa_volume = volume_scale_factor * PA_VOLUME_NORM;
+	pa_volume_t max_pa_volume = pa_volume_t(volume_scale_factor * PA_VOLUME_NORM);
 	return online ?
 		(int)((100 * (pa_cvolume_avg(&volume) + 1)) / max_pa_volume)
 		: -1;
@@ -228,7 +228,7 @@ PulseMixer::SetVolume(unsigned new_volume)
 	if (!online)
 		throw std::runtime_error("disconnected");
 
-	pa_volume_t max_pa_volume = volume_scale_factor * PA_VOLUME_NORM;
+	pa_volume_t max_pa_volume = pa_volume_t(volume_scale_factor * PA_VOLUME_NORM);
 
 	struct pa_cvolume cvolume;
 	pa_cvolume_set(&cvolume, volume.channels,

src/mixer/plugins/SoftwareMixerPlugin.cxx

@@ -25,8 +25,6 @@
 #include <cassert>
 #include <cmath>
 
-#include <assert.h>
-
 class SoftwareMixer final : public Mixer {
 	Filter *filter = nullptr;
 
@@ -72,13 +70,14 @@ PercentVolumeToSoftwareVolume(unsigned volume) noexcept
 {
 	assert(volume <= 100);
 
-	if (volume >= 100)
+	if (volume == 100)
 		return PCM_VOLUME_1;
-	else if (volume > 0)
-		return pcm_float_to_volume((std::exp(volume / 25.0) - 1) /
+
+	if (volume > 0)
+		return pcm_float_to_volume((std::exp(volume / 25.0f) - 1) /
 					   (54.5981500331F - 1));
-	else
-		return 0;
+
+	return 0;
 }
 
 void

src/output/Source.cxx

@@ -195,7 +195,7 @@ AudioOutputSource::FilterChunk(const MusicChunk &chunk)
 			   only if the mix ratio is non-negative; a
 			   negative mix ratio is a MixRamp special
 			   case */
-			mix_ratio = 1.0 - mix_ratio;
+			mix_ratio = 1.0f - mix_ratio;
 
 		void *dest = cross_fade_buffer.Get(other_data.size);
 		memcpy(dest, other_data.data, other_data.size);

src/pcm/Dsd2Pcm.cxx

@@ -127,7 +127,7 @@ CalculateCtableValue(size_t t, int k, int e) noexcept
 		acc += (((e >> (7 - m)) & 1) * 2 - 1) * htaps[t * 8 + m];
 	}
 
-	return acc;
+	return float(acc);
 }
 
 /* this needs to be a struct because GCC 6 doesn't have constexpr
@@ -204,9 +204,9 @@ Dsd2Pcm::CalcOutputSample(size_t ffp) const noexcept
 	for (size_t i = 0; i < CTABLES; ++i) {
 		uint8_t bite1 = fifo[(ffp              -i) & FIFOMASK];
 		uint8_t bite2 = fifo[(ffp-(CTABLES*2-1)+i) & FIFOMASK];
-		acc += ctables[i][bite1] + ctables[i][bite2];
+		acc += double(ctables[i][bite1] + ctables[i][bite2]);
 	}
-	return acc;
+	return float(acc);
 }
 
 inline float

src/pcm/FloatConvert.hxx

@@ -54,7 +54,7 @@ struct IntegerToFloatSampleConvert {
 	typedef typename SrcTraits::value_type SV;
 	typedef typename DstTraits::value_type DV;
 
-	static constexpr DV factor = 1.0 / FloatToIntegerSampleConvert<F, Traits>::factor;
+	static constexpr DV factor = 1.0f / FloatToIntegerSampleConvert<F, Traits>::factor;
 	static_assert(factor > 0, "Wrong factor");
 
 	static constexpr DV Convert(SV src) noexcept {

src/pcm/Mix.cxx

@@ -221,7 +221,7 @@ pcm_mix(PcmDither &dither, void *buffer1, const void *buffer2, size_t size,
 	if (portion1 < 0)
 		return pcm_add(buffer1, buffer2, size, format);
 
-	s = sin(M_PI_2 * portion1);
+	s = std::sin((float)M_PI_2 * portion1);
 	s *= s;
 
 	int vol1 = lround(s * PCM_VOLUME_1S);

src/pcm/SoxrResampler.cxx

@@ -123,7 +123,7 @@ SoxrPcmResampler::Open(AudioFormat &af, unsigned new_sample_rate)
 	ratio = float(new_sample_rate) / float(af.sample_rate);
 	FormatDebug(soxr_domain,
 		    "samplerate conversion ratio to %.2lf",
-		    ratio);
+		    double(ratio));
 
 	/* libsoxr works with floating point samples */
 	af.format = SampleFormat::FLOAT;

src/pcm/Volume.hxx

@@ -48,7 +48,7 @@ static constexpr int PCM_VOLUME_1S = PCM_VOLUME_1;
 constexpr int
 pcm_float_to_volume(float volume) noexcept
 {
-	return volume * PCM_VOLUME_1 + 0.5;
+	return int(volume * PCM_VOLUME_1 + 0.5f);
 }
 
 constexpr float

src/queue/PlaylistState.cxx

@@ -92,7 +92,8 @@ playlist_state_save(BufferedOutputStream &os, const struct playlist &playlist,
 	os.Format(PLAYLIST_STATE_FILE_CONSUME "%i\n", playlist.queue.consume);
 	os.Format(PLAYLIST_STATE_FILE_CROSSFADE "%i\n",
 		  (int)pc.GetCrossFade().count());
-	os.Format(PLAYLIST_STATE_FILE_MIXRAMPDB "%f\n", pc.GetMixRampDb());
+	os.Format(PLAYLIST_STATE_FILE_MIXRAMPDB "%f\n",
+		  (double)pc.GetMixRampDb());
 	os.Format(PLAYLIST_STATE_FILE_MIXRAMPDELAY "%f\n",
 		  pc.GetMixRampDelay().count());
 	os.Write(PLAYLIST_STATE_FILE_PLAYLIST_BEGIN "\n");