diff --git a/src/ReplayGainGlobal.cxx b/src/ReplayGainGlobal.cxx index af82c2288..b5a8d1ae0 100644 --- a/src/ReplayGainGlobal.cxx +++ b/src/ReplayGainGlobal.cxx @@ -23,24 +23,23 @@ #include "util/Math.hxx" #include +#include #include -#include - 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, diff --git a/src/ReplayGainInfo.cxx b/src/ReplayGainInfo.cxx index 76713aded..de6ec0ab3 100644 --- a/src/ReplayGainInfo.cxx +++ b/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; diff --git a/src/command/PlayerCommands.cxx b/src/command/PlayerCommands.cxx index ecd2156e5..06cbca313 100644 --- a/src/command/PlayerCommands.cxx +++ b/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()) diff --git a/src/decoder/Bridge.cxx b/src/decoder/Bridge.cxx index b7f71c1b8..68a3df600 100644 --- a/src/decoder/Bridge.cxx +++ b/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; diff --git a/src/decoder/plugins/MadDecoderPlugin.cxx b/src/decoder/plugins/MadDecoderPlugin.cxx index c28f9be8a..95dc6f7ce 100644 --- a/src/decoder/plugins/MadDecoderPlugin.cxx +++ b/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; diff --git a/src/decoder/plugins/OpusTags.cxx b/src/decoder/plugins/OpusTags.cxx index eea1fbe9b..ee1d0cb22 100644 --- a/src/decoder/plugins/OpusTags.cxx +++ b/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); diff --git a/src/encoder/plugins/LameEncoderPlugin.cxx b/src/encoder/plugins/LameEncoderPlugin.cxx index ae120fdd8..dafaf4419 100644 --- a/src/encoder/plugins/LameEncoderPlugin.cxx +++ b/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 */ diff --git a/src/encoder/plugins/TwolameEncoderPlugin.cxx b/src/encoder/plugins/TwolameEncoderPlugin.cxx index 40f59a297..deecd62a4 100644 --- a/src/encoder/plugins/TwolameEncoderPlugin.cxx +++ b/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)) diff --git a/src/encoder/plugins/VorbisEncoderPlugin.cxx b/src/encoder/plugins/VorbisEncoderPlugin.cxx index 4e6f4ab8f..f83d44e69 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 0104f45b1..d4d2d3767 100644 --- a/src/mixer/plugins/PulseMixerPlugin.cxx +++ b/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, diff --git a/src/mixer/plugins/SoftwareMixerPlugin.cxx b/src/mixer/plugins/SoftwareMixerPlugin.cxx index f2801cf03..e3c10633b 100644 --- a/src/mixer/plugins/SoftwareMixerPlugin.cxx +++ b/src/mixer/plugins/SoftwareMixerPlugin.cxx @@ -25,8 +25,6 @@ #include #include -#include - 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 diff --git a/src/output/Source.cxx b/src/output/Source.cxx index f69a6091d..074a53b03 100644 --- a/src/output/Source.cxx +++ b/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); diff --git a/src/pcm/Dsd2Pcm.cxx b/src/pcm/Dsd2Pcm.cxx index 42f76d730..4ccbfa199 100644 --- a/src/pcm/Dsd2Pcm.cxx +++ b/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 diff --git a/src/pcm/FloatConvert.hxx b/src/pcm/FloatConvert.hxx index 8b01151e4..97bb09d18 100644 --- a/src/pcm/FloatConvert.hxx +++ b/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::factor; + static constexpr DV factor = 1.0f / FloatToIntegerSampleConvert::factor; static_assert(factor > 0, "Wrong factor"); static constexpr DV Convert(SV src) noexcept { diff --git a/src/pcm/Mix.cxx b/src/pcm/Mix.cxx index a39a85947..ed2b0f8db 100644 --- a/src/pcm/Mix.cxx +++ b/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); diff --git a/src/pcm/SoxrResampler.cxx b/src/pcm/SoxrResampler.cxx index 769445e58..e896bef7b 100644 --- a/src/pcm/SoxrResampler.cxx +++ b/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; diff --git a/src/pcm/Volume.hxx b/src/pcm/Volume.hxx index e38c85a71..b150a3e5d 100644 --- a/src/pcm/Volume.hxx +++ b/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 diff --git a/src/queue/PlaylistState.cxx b/src/queue/PlaylistState.cxx index f03164d39..97a0f701c 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");