From 7715311117972984459dfbc1a4d8be9a83205b8c Mon Sep 17 00:00:00 2001
From: Rosen Penev <rosenp@gmail.com>
Date: Sun, 15 Mar 2020 23:33:21 -0700
Subject: [PATCH] fix double promotions
Found with -Wdouble-promotion
Signed-off-by: Rosen Penev <rosenp@gmail.com>
---
src/ReplayGainGlobal.cxx | 12 ++++++------
src/ReplayGainInfo.cxx | 12 ++++++------
src/command/PlayerCommands.cxx | 2 +-
src/decoder/Bridge.cxx | 4 ++--
src/decoder/plugins/MadDecoderPlugin.cxx | 12 ++++++------
src/decoder/plugins/OpusTags.cxx | 4 ++--
src/encoder/plugins/LameEncoderPlugin.cxx | 8 ++++----
src/encoder/plugins/TwolameEncoderPlugin.cxx | 6 +++---
src/encoder/plugins/VorbisEncoderPlugin.cxx | 8 ++++----
src/mixer/plugins/PulseMixerPlugin.cxx | 10 +++++-----
src/mixer/plugins/SoftwareMixerPlugin.cxx | 12 ++++++------
src/output/Source.cxx | 2 +-
src/pcm/FloatConvert.hxx | 2 +-
src/pcm/PcmMix.cxx | 4 +++-
src/pcm/SoxrResampler.cxx | 2 +-
src/pcm/Volume.hxx | 2 +-
src/queue/PlaylistState.cxx | 3 ++-
17 files changed, 54 insertions(+), 51 deletions(-)
diff --git a/src/ReplayGainGlobal.cxx b/src/ReplayGainGlobal.cxx
index 56aaa7890..fc4306821 100644
--- a/src/ReplayGainGlobal.cxx
+++ b/src/ReplayGainGlobal.cxx
@@ -23,9 +23,9 @@
#include "config/Data.hxx"
#include "util/RuntimeError.hxx"
-#include <assert.h>
-#include <stdlib.h>
-#include <math.h>
+#include <cassert>
+#include <cstdlib>
+#include <cmath>
static float
ParsePreamp(const char *s)
@@ -33,14 +33,14 @@ 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);
}
static float
diff --git a/src/ReplayGainInfo.cxx b/src/ReplayGainInfo.cxx
index 18f40ec4b..7963ff7be 100644
--- a/src/ReplayGainInfo.cxx
+++ b/src/ReplayGainInfo.cxx
@@ -20,7 +20,7 @@
#include "ReplayGainInfo.hxx"
#include "ReplayGainConfig.hxx"
-#include <math.h>
+#include <cmath>
float
ReplayGainTuple::CalculateScale(const ReplayGainConfig &config) const noexcept
@@ -28,13 +28,13 @@ 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;
diff --git a/src/command/PlayerCommands.cxx b/src/command/PlayerCommands.cxx
index e8d88e6cf..cf9438620 100644
--- a/src/command/PlayerCommands.cxx
+++ b/src/command/PlayerCommands.cxx
@@ -148,7 +148,7 @@ handle_status(Client &client, gcc_unused Request args, Response &r)
playlist.GetConsume(),
(unsigned long)playlist.GetVersion(),
playlist.GetLength(),
- pc.GetMixRampDb(),
+ (double)pc.GetMixRampDb(),
state);
if (pc.GetCrossFade() > FloatDuration::zero())
diff --git a/src/decoder/Bridge.cxx b/src/decoder/Bridge.cxx
index e11a0a45a..9947bc3e5 100644
--- a/src/decoder/Bridge.cxx
+++ b/src/decoder/Bridge.cxx
@@ -33,11 +33,11 @@
#include "util/ConstBuffer.hxx"
#include "util/StringBuffer.hxx"
+#include <cmath>
#include <stdexcept>
#include <assert.h>
#include <string.h>
-#include <math.h>
DecoderBridge::~DecoderBridge()
{
@@ -597,7 +597,7 @@ DecoderBridge::SubmitReplayGain(const ReplayGainInfo *new_replay_gain_info)
const auto &tuple = new_replay_gain_info->Get(rgm);
const auto scale =
tuple.CalculateScale(dc.replay_gain_config);
- dc.replay_gain_db = 20.0 * log10f(scale);
+ dc.replay_gain_db = 20.0f * std::log10(scale);
}
replay_gain_info = *new_replay_gain_info;
diff --git a/src/decoder/plugins/MadDecoderPlugin.cxx b/src/decoder/plugins/MadDecoderPlugin.cxx
index b5a688bbf..a01a44001 100644
--- a/src/decoder/plugins/MadDecoderPlugin.cxx
+++ b/src/decoder/plugins/MadDecoderPlugin.cxx
@@ -617,8 +617,8 @@ 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 */
@@ -626,9 +626,9 @@ 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
@@ -644,7 +644,7 @@ 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);
@@ -778,7 +778,7 @@ 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;
diff --git a/src/decoder/plugins/OpusTags.cxx b/src/decoder/plugins/OpusTags.cxx
index 76ea63e6e..70e452860 100644
--- a/src/decoder/plugins/OpusTags.cxx
+++ b/src/decoder/plugins/OpusTags.cxx
@@ -53,7 +53,7 @@ ScanOneOpusTag(const char *name, const char *value,
char *endptr;
long l = strtol(value, &endptr, 10);
if (endptr > value && *endptr == 0)
- rgi->track.gain = double(l) / 256.;
+ rgi->track.gain = float(l) / 256.0f;
} else if (rgi != nullptr &&
StringEqualsCaseASCII(name, "R128_ALBUM_GAIN")) {
/* R128_ALBUM_GAIN is a Q7.8 fixed point number in
@@ -62,7 +62,7 @@ ScanOneOpusTag(const char *name, const char *value,
char *endptr;
long l = strtol(value, &endptr, 10);
if (endptr > value && *endptr == 0)
- rgi->album.gain = double(l) / 256.;
+ rgi->album.gain = float(l) / 256.0f;
}
handler.OnPair(name, value);
diff --git a/src/encoder/plugins/LameEncoderPlugin.cxx b/src/encoder/plugins/LameEncoderPlugin.cxx
index fe663c28c..7f6df6eb0 100644
--- a/src/encoder/plugins/LameEncoderPlugin.cxx
+++ b/src/encoder/plugins/LameEncoderPlugin.cxx
@@ -77,9 +77,9 @@ 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);
@@ -111,13 +111,13 @@ 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 */
diff --git a/src/encoder/plugins/TwolameEncoderPlugin.cxx b/src/encoder/plugins/TwolameEncoderPlugin.cxx
index e05f64764..debed7871 100644
--- a/src/encoder/plugins/TwolameEncoderPlugin.cxx
+++ b/src/encoder/plugins/TwolameEncoderPlugin.cxx
@@ -95,9 +95,9 @@ 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);
@@ -132,7 +132,7 @@ 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))
diff --git a/src/encoder/plugins/VorbisEncoderPlugin.cxx b/src/encoder/plugins/VorbisEncoderPlugin.cxx
index 7a17a77f3..90a8120c6 100644
--- a/src/encoder/plugins/VorbisEncoderPlugin.cxx
+++ b/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");
}
diff --git a/src/mixer/plugins/PulseMixerPlugin.cxx b/src/mixer/plugins/PulseMixerPlugin.cxx
index 44fcb56f7..f39044459 100644
--- a/src/mixer/plugins/PulseMixerPlugin.cxx
+++ b/src/mixer/plugins/PulseMixerPlugin.cxx
@@ -51,7 +51,7 @@ 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))
{
}
@@ -175,7 +175,7 @@ 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);
@@ -190,7 +190,7 @@ pulse_mixer_init(gcc_unused EventLoop &event_loop, AudioOutput &ao,
{
PulseOutput &po = (PulseOutput &)ao;
float scale = parse_volume_scale_factor(block.GetBlockValue("scale_volume"));
- PulseMixer *pm = new PulseMixer(po, listener, scale);
+ auto *pm = new PulseMixer(po, listener, (double)scale);
pulse_output_set_mixer(po, *pm);
@@ -216,7 +216,7 @@ 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;
@@ -230,7 +230,7 @@ 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,
diff --git a/src/mixer/plugins/SoftwareMixerPlugin.cxx b/src/mixer/plugins/SoftwareMixerPlugin.cxx
index 177c875d2..8105db537 100644
--- a/src/mixer/plugins/SoftwareMixerPlugin.cxx
+++ b/src/mixer/plugins/SoftwareMixerPlugin.cxx
@@ -22,8 +22,8 @@
#include "filter/plugins/VolumeFilterPlugin.hxx"
#include "pcm/Volume.hxx"
-#include <assert.h>
-#include <math.h>
+#include <cassert>
+#include <cmath>
class SoftwareMixer final : public Mixer {
Filter *filter = nullptr;
@@ -70,13 +70,13 @@ 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((exp(volume / 25.0) - 1) /
+ return pcm_float_to_volume((std::exp(volume / 25.0f) - 1) /
(54.5981500331F - 1));
- else
- return 0;
+
+ return 0;
}
void
diff --git a/src/output/Source.cxx b/src/output/Source.cxx
index 7c5098d8e..27c7c362b 100644
--- a/src/output/Source.cxx
+++ b/src/output/Source.cxx
@@ -187,7 +187,7 @@ 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);
diff --git a/src/pcm/FloatConvert.hxx b/src/pcm/FloatConvert.hxx
index 28f923c7a..17fcd27ce 100644
--- a/src/pcm/FloatConvert.hxx
+++ b/src/pcm/FloatConvert.hxx
@@ -53,7 +53,7 @@ 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 {
diff --git a/src/pcm/PcmMix.cxx b/src/pcm/PcmMix.cxx
index bbb26d1c4..11d201c3e 100644
--- a/src/pcm/PcmMix.cxx
+++ b/src/pcm/PcmMix.cxx
@@ -26,6 +26,8 @@
#include "PcmDither.cxx" // including the .cxx file to get inlined templates
+#include <cmath>
+
#include <assert.h>
template<SampleFormat F, class Traits=SampleTraits<F>>
@@ -221,7 +223,7 @@ 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);
diff --git a/src/pcm/SoxrResampler.cxx b/src/pcm/SoxrResampler.cxx
index 411e79b43..2dcb507e0 100644
--- a/src/pcm/SoxrResampler.cxx
+++ b/src/pcm/SoxrResampler.cxx
@@ -122,7 +122,7 @@ 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;
diff --git a/src/pcm/Volume.hxx b/src/pcm/Volume.hxx
index ddc6cb854..52a63d819 100644
--- a/src/pcm/Volume.hxx
+++ b/src/pcm/Volume.hxx
@@ -48,7 +48,7 @@ static constexpr int PCM_VOLUME_1S = PCM_VOLUME_1;
static constexpr inline int
pcm_float_to_volume(float volume) noexcept
{
- return volume * PCM_VOLUME_1 + 0.5;
+ return int(volume * PCM_VOLUME_1 + 0.5f);
}
static constexpr inline float
diff --git a/src/queue/PlaylistState.cxx b/src/queue/PlaylistState.cxx
index f57409de0..2e2ab3ec1 100644
--- a/src/queue/PlaylistState.cxx
+++ b/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");