fix double promotions
Found with -Wdouble-promotion Signed-off-by: Rosen Penev <rosenp@gmail.com>
This commit is contained in:
parent
fd71514068
commit
6d91b5c7b2
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@ -23,24 +23,23 @@
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#include "util/Math.hxx"
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#include <cassert>
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#include <cstdlib>
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#include <stdexcept>
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#include <stdlib.h>
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static float
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ParsePreamp(const char *s)
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{
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assert(s != nullptr);
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char *endptr;
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float f = strtod(s, &endptr);
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float f = std::strtof(s, &endptr);
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if (endptr == s || *endptr != '\0')
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throw std::invalid_argument("Not a numeric value");
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if (f < -15 || f > 15)
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if (f < -15.0f || f > 15.0f)
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throw std::invalid_argument("Number must be between -15 and 15");
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return pow(10, f / 20.0);
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return std::pow(10.0f, f / 20.0f);
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}
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ReplayGainConfig
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@ -51,13 +50,13 @@ LoadReplayGainConfig(const ConfigData &config)
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replay_gain_config.preamp = config.With(ConfigOption::REPLAYGAIN_PREAMP, [](const char *s){
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return s != nullptr
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? ParsePreamp(s)
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: 1.0;
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: 1.0f;
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});
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replay_gain_config.missing_preamp = config.With(ConfigOption::REPLAYGAIN_MISSING_PREAMP, [](const char *s){
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return s != nullptr
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? ParsePreamp(s)
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: 1.0;
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: 1.0f;
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});
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replay_gain_config.limit = config.GetBool(ConfigOption::REPLAYGAIN_LIMIT,
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@ -27,13 +27,13 @@ ReplayGainTuple::CalculateScale(const ReplayGainConfig &config) const noexcept
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float scale;
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if (IsDefined()) {
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scale = pow(10.0, gain / 20.0);
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scale = std::pow(10.0f, gain / 20.0f);
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scale *= config.preamp;
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if (scale > 15.0)
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scale = 15.0;
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if (scale > 15.0f)
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scale = 15.0f;
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if (config.limit && scale * peak > 1.0)
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scale = 1.0 / peak;
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if (config.limit && scale * peak > 1.0f)
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scale = 1.0f / peak;
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} else
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scale = config.missing_preamp;
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@ -149,7 +149,7 @@ handle_status(Client &client, [[maybe_unused]] Request args, Response &r)
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partition.name.c_str(),
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(unsigned long)playlist.GetVersion(),
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playlist.GetLength(),
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pc.GetMixRampDb(),
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(double)pc.GetMixRampDb(),
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state);
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if (pc.GetCrossFade() > FloatDuration::zero())
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@ -611,7 +611,7 @@ DecoderBridge::SubmitReplayGain(const ReplayGainInfo *new_replay_gain_info) noex
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const auto &tuple = new_replay_gain_info->Get(rgm);
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const auto scale =
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tuple.CalculateScale(dc.replay_gain_config);
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dc.replay_gain_db = 20.0 * std::log10(scale);
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dc.replay_gain_db = 20.0f * std::log10(scale);
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}
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replay_gain_info = *new_replay_gain_info;
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@ -585,8 +585,8 @@ parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen) noexcept
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mad_bit_skip(ptr, 16);
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lame->peak = mad_f_todouble(mad_bit_read(ptr, 32) << 5); /* peak */
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FormatDebug(mad_domain, "LAME peak found: %f", lame->peak);
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lame->peak = MAD_F(mad_bit_read(ptr, 32) << 5); /* peak */
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FormatDebug(mad_domain, "LAME peak found: %f", double(lame->peak));
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lame->track_gain = 0;
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unsigned name = mad_bit_read(ptr, 3); /* gain name */
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@ -594,9 +594,9 @@ parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen) noexcept
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unsigned sign = mad_bit_read(ptr, 1); /* sign bit */
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int gain = mad_bit_read(ptr, 9); /* gain*10 */
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if (gain && name == 1 && orig != 0) {
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lame->track_gain = ((sign ? -gain : gain) / 10.0) + adj;
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lame->track_gain = ((sign ? -gain : gain) / 10.0f) + adj;
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FormatDebug(mad_domain, "LAME track gain found: %f",
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lame->track_gain);
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double(lame->track_gain));
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}
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/* tmz reports that this isn't currently written by any version of lame
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@ -612,7 +612,7 @@ parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen) noexcept
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if (gain && name == 2 && orig != 0) {
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lame->album_gain = ((sign ? -gain : gain) / 10.0) + adj;
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FormatDebug(mad_domain, "LAME album gain found: %f",
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lame->track_gain);
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double(lame->track_gain));
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}
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#else
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mad_bit_skip(ptr, 16);
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@ -746,7 +746,7 @@ MadDecoder::DecodeFirstFrame(Tag *tag) noexcept
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/* Album gain isn't currently used. See comment in
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* parse_lame() for details. -- jat */
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if (client != nullptr && !found_replay_gain &&
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lame.track_gain) {
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lame.track_gain > 0.0f) {
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ReplayGainInfo rgi;
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rgi.Clear();
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rgi.track.gain = lame.track_gain;
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@ -61,7 +61,7 @@ ScanOneOpusTag(StringView name, StringView value,
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const char *endptr;
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const auto l = ParseInt64(value, &endptr, 10);
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if (endptr > value.begin() && endptr == value.end())
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rgi->track.gain = double(l) / 256.;
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rgi->track.gain = float(l) / 256.0f;
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} else if (rgi != nullptr &&
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name.EqualsIgnoreCase("R128_ALBUM_GAIN")) {
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/* R128_ALBUM_GAIN is a Q7.8 fixed point number in
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@ -70,7 +70,7 @@ ScanOneOpusTag(StringView name, StringView value,
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const char *endptr;
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const auto l = ParseInt64(value, &endptr, 10);
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if (endptr > value.begin() && endptr == value.end())
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rgi->album.gain = double(l) / 256.;
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rgi->album.gain = float(l) / 256.0f;
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}
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handler.OnPair(name, value);
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@ -76,9 +76,9 @@ PreparedLameEncoder::PreparedLameEncoder(const ConfigBlock &block)
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if (value != nullptr) {
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/* a quality was configured (VBR) */
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quality = ParseDouble(value, &endptr);
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quality = float(ParseDouble(value, &endptr));
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if (*endptr != '\0' || quality < -1.0 || quality > 10.0)
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if (*endptr != '\0' || quality < -1.0f || quality > 10.0f)
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throw FormatRuntimeError("quality \"%s\" is not a number in the "
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"range -1 to 10",
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value);
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@ -110,13 +110,13 @@ static void
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lame_encoder_setup(lame_global_flags *gfp, float quality, int bitrate,
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const AudioFormat &audio_format)
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{
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if (quality >= -1.0) {
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if (quality >= -1.0f) {
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/* a quality was configured (VBR) */
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if (0 != lame_set_VBR(gfp, vbr_rh))
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throw std::runtime_error("error setting lame VBR mode");
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if (0 != lame_set_VBR_q(gfp, quality))
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if (0 != lame_set_VBR_q(gfp, int(quality)))
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throw std::runtime_error("error setting lame VBR quality");
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} else {
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/* a bit rate was configured */
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@ -94,9 +94,9 @@ PreparedTwolameEncoder::PreparedTwolameEncoder(const ConfigBlock &block)
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if (value != nullptr) {
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/* a quality was configured (VBR) */
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quality = ParseDouble(value, &endptr);
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quality = float(ParseDouble(value, &endptr));
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if (*endptr != '\0' || quality < -1.0 || quality > 10.0)
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if (*endptr != '\0' || quality < -1.0f || quality > 10.0f)
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throw FormatRuntimeError("quality \"%s\" is not a number in the "
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"range -1 to 10",
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value);
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@ -131,7 +131,7 @@ static void
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twolame_encoder_setup(twolame_options *options, float quality, int bitrate,
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const AudioFormat &audio_format)
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{
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if (quality >= -1.0) {
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if (quality >= -1.0f) {
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/* a quality was configured (VBR) */
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if (0 != twolame_set_VBR(options, true))
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@ -84,7 +84,7 @@ PreparedVorbisEncoder::PreparedVorbisEncoder(const ConfigBlock &block)
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char *endptr;
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quality = ParseDouble(value, &endptr);
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if (*endptr != '\0' || quality < -1.0 || quality > 10.0)
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if (*endptr != '\0' || quality < -1.0f || quality > 10.0f)
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throw FormatRuntimeError("quality \"%s\" is not a number in the "
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"range -1 to 10",
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value);
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@ -122,13 +122,13 @@ VorbisEncoder::VorbisEncoder(float quality, int bitrate,
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_audio_format.format = SampleFormat::FLOAT;
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audio_format = _audio_format;
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if (quality >= -1.0) {
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if (quality >= -1.0f) {
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/* a quality was configured (VBR) */
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if (0 != vorbis_encode_init_vbr(&vi,
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audio_format.channels,
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audio_format.sample_rate,
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quality * 0.1)) {
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quality * 0.1f)) {
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vorbis_info_clear(&vi);
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throw std::runtime_error("error initializing vorbis vbr");
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}
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@ -138,7 +138,7 @@ VorbisEncoder::VorbisEncoder(float quality, int bitrate,
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if (0 != vorbis_encode_init(&vi,
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audio_format.channels,
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audio_format.sample_rate, -1.0,
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bitrate * 1000, -1.0)) {
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bitrate * 1000, -1.0f)) {
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vorbis_info_clear(&vi);
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throw std::runtime_error("error initializing vorbis encoder");
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}
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@ -49,7 +49,7 @@ public:
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double _volume_scale_factor)
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:Mixer(pulse_mixer_plugin, _listener),
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output(_output),
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volume_scale_factor(_volume_scale_factor)
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volume_scale_factor(float(_volume_scale_factor))
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{
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}
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@ -173,7 +173,7 @@ parse_volume_scale_factor(const char *value) {
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char *endptr;
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float factor = ParseFloat(value, &endptr);
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if (endptr == value || *endptr != '\0' || factor < 0.5 || factor > 5.0)
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if (endptr == value || *endptr != '\0' || factor < 0.5f || factor > 5.0f)
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throw FormatRuntimeError("\"%s\" is not a number in the "
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"range 0.5 to 5.0",
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value);
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@ -188,7 +188,7 @@ pulse_mixer_init([[maybe_unused]] EventLoop &event_loop, AudioOutput &ao,
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{
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auto &po = (PulseOutput &)ao;
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float scale = parse_volume_scale_factor(block.GetBlockValue("scale_volume"));
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auto *pm = new PulseMixer(po, listener, scale);
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auto *pm = new PulseMixer(po, listener, (double)scale);
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pulse_output_set_mixer(po, *pm);
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@ -214,7 +214,7 @@ PulseMixer::GetVolume()
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int
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PulseMixer::GetVolumeInternal()
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{
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pa_volume_t max_pa_volume = volume_scale_factor * PA_VOLUME_NORM;
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pa_volume_t max_pa_volume = pa_volume_t(volume_scale_factor * PA_VOLUME_NORM);
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return online ?
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(int)((100 * (pa_cvolume_avg(&volume) + 1)) / max_pa_volume)
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: -1;
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@ -228,7 +228,7 @@ PulseMixer::SetVolume(unsigned new_volume)
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if (!online)
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throw std::runtime_error("disconnected");
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pa_volume_t max_pa_volume = volume_scale_factor * PA_VOLUME_NORM;
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pa_volume_t max_pa_volume = pa_volume_t(volume_scale_factor * PA_VOLUME_NORM);
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struct pa_cvolume cvolume;
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pa_cvolume_set(&cvolume, volume.channels,
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@ -25,8 +25,6 @@
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#include <cassert>
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#include <cmath>
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#include <assert.h>
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class SoftwareMixer final : public Mixer {
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Filter *filter = nullptr;
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@ -72,13 +70,14 @@ PercentVolumeToSoftwareVolume(unsigned volume) noexcept
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{
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assert(volume <= 100);
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if (volume >= 100)
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if (volume == 100)
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return PCM_VOLUME_1;
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else if (volume > 0)
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return pcm_float_to_volume((std::exp(volume / 25.0) - 1) /
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if (volume > 0)
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return pcm_float_to_volume((std::exp(volume / 25.0f) - 1) /
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(54.5981500331F - 1));
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else
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return 0;
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return 0;
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}
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void
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@ -195,7 +195,7 @@ AudioOutputSource::FilterChunk(const MusicChunk &chunk)
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only if the mix ratio is non-negative; a
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negative mix ratio is a MixRamp special
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case */
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mix_ratio = 1.0 - mix_ratio;
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mix_ratio = 1.0f - mix_ratio;
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void *dest = cross_fade_buffer.Get(other_data.size);
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memcpy(dest, other_data.data, other_data.size);
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@ -127,7 +127,7 @@ CalculateCtableValue(size_t t, int k, int e) noexcept
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acc += (((e >> (7 - m)) & 1) * 2 - 1) * htaps[t * 8 + m];
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}
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return acc;
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return float(acc);
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}
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/* this needs to be a struct because GCC 6 doesn't have constexpr
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@ -204,9 +204,9 @@ Dsd2Pcm::CalcOutputSample(size_t ffp) const noexcept
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for (size_t i = 0; i < CTABLES; ++i) {
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uint8_t bite1 = fifo[(ffp -i) & FIFOMASK];
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uint8_t bite2 = fifo[(ffp-(CTABLES*2-1)+i) & FIFOMASK];
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acc += ctables[i][bite1] + ctables[i][bite2];
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acc += double(ctables[i][bite1] + ctables[i][bite2]);
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}
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return acc;
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return float(acc);
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}
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inline float
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@ -54,7 +54,7 @@ struct IntegerToFloatSampleConvert {
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typedef typename SrcTraits::value_type SV;
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typedef typename DstTraits::value_type DV;
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static constexpr DV factor = 1.0 / FloatToIntegerSampleConvert<F, Traits>::factor;
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static constexpr DV factor = 1.0f / FloatToIntegerSampleConvert<F, Traits>::factor;
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static_assert(factor > 0, "Wrong factor");
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static constexpr DV Convert(SV src) noexcept {
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@ -221,7 +221,7 @@ pcm_mix(PcmDither &dither, void *buffer1, const void *buffer2, size_t size,
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if (portion1 < 0)
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return pcm_add(buffer1, buffer2, size, format);
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s = sin(M_PI_2 * portion1);
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s = std::sin((float)M_PI_2 * portion1);
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s *= s;
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int vol1 = lround(s * PCM_VOLUME_1S);
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@ -123,7 +123,7 @@ SoxrPcmResampler::Open(AudioFormat &af, unsigned new_sample_rate)
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ratio = float(new_sample_rate) / float(af.sample_rate);
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FormatDebug(soxr_domain,
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"samplerate conversion ratio to %.2lf",
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ratio);
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double(ratio));
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/* libsoxr works with floating point samples */
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af.format = SampleFormat::FLOAT;
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@ -48,7 +48,7 @@ static constexpr int PCM_VOLUME_1S = PCM_VOLUME_1;
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constexpr int
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pcm_float_to_volume(float volume) noexcept
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{
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return volume * PCM_VOLUME_1 + 0.5;
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return int(volume * PCM_VOLUME_1 + 0.5f);
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}
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constexpr float
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@ -92,7 +92,8 @@ playlist_state_save(BufferedOutputStream &os, const struct playlist &playlist,
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os.Format(PLAYLIST_STATE_FILE_CONSUME "%i\n", playlist.queue.consume);
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os.Format(PLAYLIST_STATE_FILE_CROSSFADE "%i\n",
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(int)pc.GetCrossFade().count());
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os.Format(PLAYLIST_STATE_FILE_MIXRAMPDB "%f\n", pc.GetMixRampDb());
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os.Format(PLAYLIST_STATE_FILE_MIXRAMPDB "%f\n",
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(double)pc.GetMixRampDb());
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os.Format(PLAYLIST_STATE_FILE_MIXRAMPDELAY "%f\n",
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pc.GetMixRampDelay().count());
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os.Write(PLAYLIST_STATE_FILE_PLAYLIST_BEGIN "\n");
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