diff --git a/Makefile.am b/Makefile.am index 14cea53a9..5f0291d7e 100644 --- a/Makefile.am +++ b/Makefile.am @@ -51,7 +51,6 @@ src_mpd_LDADD = \ mpd_headers = \ src/check.h \ src/ack.h \ - src/audio_format.h \ src/filter_internal.h \ src/command.h \ src/conf.h \ @@ -97,7 +96,7 @@ src_mpd_SOURCES = \ src/notify.cxx src/notify.hxx \ src/AudioConfig.cxx src/AudioConfig.hxx \ src/CheckAudioFormat.cxx src/CheckAudioFormat.hxx \ - src/audio_format.c \ + src/AudioFormat.cxx src/AudioFormat.hxx \ src/AudioParser.cxx src/AudioParser.hxx \ src/protocol/ArgParser.cxx src/protocol/ArgParser.hxx \ src/protocol/Result.cxx src/protocol/Result.hxx \ @@ -1173,8 +1172,7 @@ test_run_decoder_SOURCES = test/run_decoder.cxx \ src/Tag.cxx src/TagNames.c src/TagPool.cxx src/TagHandler.cxx \ src/ReplayGainInfo.cxx \ src/fd_util.c \ - src/CheckAudioFormat.cxx \ - src/audio_format.c \ + src/AudioFormat.cxx src/CheckAudioFormat.cxx \ $(ARCHIVE_SRC) \ $(INPUT_SRC) \ $(TAG_SRC) \ @@ -1221,7 +1219,7 @@ test_run_filter_SOURCES = test/run_filter.cxx \ test/stdbin.h \ src/FilterPlugin.cxx src/FilterRegistry.cxx \ src/CheckAudioFormat.cxx \ - src/audio_format.c \ + src/AudioFormat.cxx \ src/AudioParser.cxx \ src/ReplayGainInfo.cxx \ src/AudioCompress/compress.c @@ -1240,7 +1238,7 @@ test_run_encoder_SOURCES = test/run_encoder.cxx \ test/stdbin.h \ src/Tag.cxx src/TagNames.c src/TagPool.cxx \ src/CheckAudioFormat.cxx \ - src/audio_format.c \ + src/AudioFormat.cxx \ src/AudioParser.cxx test_run_encoder_LDADD = \ $(ENCODER_LIBS) \ @@ -1258,7 +1256,7 @@ test_test_vorbis_encoder_SOURCES = test/test_vorbis_encoder.cxx \ test/stdbin.h \ src/Tag.cxx src/TagNames.c src/TagPool.cxx \ src/CheckAudioFormat.cxx \ - src/audio_format.c \ + src/AudioFormat.cxx \ src/AudioParser.cxx \ $(ENCODER_SRC) test_test_vorbis_encoder_CPPFLAGS = $(AM_CPPFLAGS) \ @@ -1289,7 +1287,7 @@ test_run_normalize_LDADD = \ $(GLIB_LIBS) test_run_convert_SOURCES = test/run_convert.cxx \ - src/audio_format.c \ + src/AudioFormat.cxx \ src/CheckAudioFormat.cxx \ src/AudioParser.cxx test_run_convert_LDADD = \ @@ -1313,7 +1311,7 @@ test_run_output_SOURCES = test/run_output.cxx \ test/stdbin.h \ src/IOThread.cxx \ src/CheckAudioFormat.cxx \ - src/audio_format.c \ + src/AudioFormat.cxx \ src/AudioParser.cxx \ src/Timer.cxx src/clock.c \ src/Tag.cxx src/TagNames.c src/TagPool.cxx \ diff --git a/src/AudioConfig.cxx b/src/AudioConfig.cxx index e546aed27..9ead61afe 100644 --- a/src/AudioConfig.cxx +++ b/src/AudioConfig.cxx @@ -19,18 +19,19 @@ #include "config.h" #include "AudioConfig.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "AudioParser.hxx" #include "conf.h" #include "mpd_error.h" -static struct audio_format configured_audio_format; +static AudioFormat configured_audio_format; -void getOutputAudioFormat(const struct audio_format *inAudioFormat, - struct audio_format *outAudioFormat) +AudioFormat +getOutputAudioFormat(AudioFormat inAudioFormat) { - *outAudioFormat = *inAudioFormat; - audio_format_mask_apply(outAudioFormat, &configured_audio_format); + AudioFormat out_audio_format = inAudioFormat; + out_audio_format.ApplyMask(configured_audio_format); + return out_audio_format; } void initAudioConfig(void) @@ -42,7 +43,7 @@ void initAudioConfig(void) if (param == NULL) return; - ret = audio_format_parse(&configured_audio_format, param->value, + ret = audio_format_parse(configured_audio_format, param->value, true, &error); if (!ret) MPD_ERROR("error parsing line %i: %s", diff --git a/src/AudioConfig.hxx b/src/AudioConfig.hxx index 717a8e2e7..ebe202974 100644 --- a/src/AudioConfig.hxx +++ b/src/AudioConfig.hxx @@ -20,10 +20,10 @@ #ifndef MPD_AUDIO_CONFIG_HXX #define MPD_AUDIO_CONFIG_HXX -struct audio_format; +struct AudioFormat; -void getOutputAudioFormat(const struct audio_format *inFormat, - struct audio_format *outFormat); +AudioFormat +getOutputAudioFormat(AudioFormat inFormat); /* make sure initPlayerData is called before this function!! */ void initAudioConfig(void); diff --git a/src/audio_format.c b/src/AudioFormat.cxx similarity index 56% rename from src/audio_format.c rename to src/AudioFormat.cxx index cfe3a4cae..04636c1e2 100644 --- a/src/audio_format.c +++ b/src/AudioFormat.cxx @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2011 The Music Player Daemon Project + * Copyright (C) 2003-2013 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,53 +17,52 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "audio_format.h" +#include "AudioFormat.hxx" #include #include void -audio_format_mask_apply(struct audio_format *af, - const struct audio_format *mask) +AudioFormat::ApplyMask(AudioFormat mask) { - assert(audio_format_valid(af)); - assert(audio_format_mask_valid(mask)); + assert(IsValid()); + assert(mask.IsMaskValid()); - if (mask->sample_rate != 0) - af->sample_rate = mask->sample_rate; + if (mask.sample_rate != 0) + sample_rate = mask.sample_rate; - if (mask->format != SAMPLE_FORMAT_UNDEFINED) - af->format = mask->format; + if (mask.format != SampleFormat::UNDEFINED) + format = mask.format; - if (mask->channels != 0) - af->channels = mask->channels; + if (mask.channels != 0) + channels = mask.channels; - assert(audio_format_valid(af)); + assert(IsValid()); } const char * -sample_format_to_string(enum sample_format format) +sample_format_to_string(SampleFormat format) { switch (format) { - case SAMPLE_FORMAT_UNDEFINED: + case SampleFormat::UNDEFINED: return "?"; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: return "8"; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: return "16"; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: return "24"; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: return "32"; - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: return "f"; - case SAMPLE_FORMAT_DSD: + case SampleFormat::DSD: return "dsd"; } @@ -73,15 +72,14 @@ sample_format_to_string(enum sample_format format) } const char * -audio_format_to_string(const struct audio_format *af, +audio_format_to_string(const AudioFormat af, struct audio_format_string *s) { - assert(af != NULL); - assert(s != NULL); + assert(s != nullptr); snprintf(s->buffer, sizeof(s->buffer), "%u:%s:%u", - af->sample_rate, sample_format_to_string(af->format), - af->channels); + af.sample_rate, sample_format_to_string(af.format), + af.channels); return s->buffer; } diff --git a/src/AudioFormat.hxx b/src/AudioFormat.hxx new file mode 100644 index 000000000..eb3f9b062 --- /dev/null +++ b/src/AudioFormat.hxx @@ -0,0 +1,315 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_AUDIO_FORMAT_HXX +#define MPD_AUDIO_FORMAT_HXX + +#include "gcc.h" + +#include +#include + +#if defined(WIN32) && GCC_CHECK_VERSION(4,6) +/* on WIN32, "FLOAT" is already defined, and this triggers -Wshadow */ +#pragma GCC diagnostic push +#pragma GCC diagnostic ignored "-Wshadow" +#endif + +enum class SampleFormat : uint8_t { + UNDEFINED = 0, + + S8, + S16, + + /** + * Signed 24 bit integer samples, packed in 32 bit integers + * (the most significant byte is filled with the sign bit). + */ + S24_P32, + + S32, + + /** + * 32 bit floating point samples in the host's format. The + * range is -1.0f to +1.0f. + */ + FLOAT, + + /** + * Direct Stream Digital. 1-bit samples; each frame has one + * byte (8 samples) per channel. + */ + DSD, +}; + +#if defined(WIN32) && GCC_CHECK_VERSION(4,6) +#pragma GCC diagnostic pop +#endif + +static constexpr unsigned MAX_CHANNELS = 8; + +/** + * This structure describes the format of a raw PCM stream. + */ +struct AudioFormat { + /** + * The sample rate in Hz. A better name for this attribute is + * "frame rate", because technically, you have two samples per + * frame in stereo sound. + */ + uint32_t sample_rate; + + /** + * The format samples are stored in. See the #sample_format + * enum for valid values. + */ + SampleFormat format; + + /** + * The number of channels. Only mono (1) and stereo (2) are + * fully supported currently. + */ + uint8_t channels; + + AudioFormat() = default; + + constexpr AudioFormat(uint32_t _sample_rate, + SampleFormat _format, uint8_t _channels) + :sample_rate(_sample_rate), + format(_format), channels(_channels) {} + + static constexpr AudioFormat Undefined() { + return AudioFormat(0, SampleFormat::UNDEFINED,0); + } + + /** + * Clears the #audio_format object, i.e. sets all attributes to an + * undefined (invalid) value. + */ + void Clear() { + sample_rate = 0; + format = SampleFormat::UNDEFINED; + channels = 0; + } + + /** + * Checks whether the object has a defined value. + */ + constexpr bool IsDefined() const { + return sample_rate != 0; + } + + /** + * Checks whether the object is full, i.e. all attributes are + * defined. This is more complete than IsDefined(), but + * slower. + */ + constexpr bool IsFullyDefined() const { + return sample_rate != 0 && format != SampleFormat::UNDEFINED && + channels != 0; + } + + /** + * Checks whether the object has at least one defined value. + */ + constexpr bool IsMaskDefined() const { + return sample_rate != 0 || format != SampleFormat::UNDEFINED || + channels != 0; + } + + bool IsValid() const; + bool IsMaskValid() const; + + constexpr bool operator==(const AudioFormat other) const { + return sample_rate == other.sample_rate && + format == other.format && + channels == other.channels; + } + + constexpr bool operator!=(const AudioFormat other) const { + return !(*this == other); + } + + void ApplyMask(AudioFormat mask); + + /** + * Returns the size of each (mono) sample in bytes. + */ + unsigned GetSampleSize() const; + + /** + * Returns the size of each full frame in bytes. + */ + unsigned GetFrameSize() const; + + /** + * Returns the floating point factor which converts a time + * span to a storage size in bytes. + */ + double GetTimeToSize() const; +}; + +/** + * Buffer for audio_format_string(). + */ +struct audio_format_string { + char buffer[24]; +}; + +/** + * Checks whether the sample rate is valid. + * + * @param sample_rate the sample rate in Hz + */ +static constexpr inline bool +audio_valid_sample_rate(unsigned sample_rate) +{ + return sample_rate > 0 && sample_rate < (1 << 30); +} + +/** + * Checks whether the sample format is valid. + * + * @param bits the number of significant bits per sample + */ +static inline bool +audio_valid_sample_format(SampleFormat format) +{ + switch (format) { + case SampleFormat::S8: + case SampleFormat::S16: + case SampleFormat::S24_P32: + case SampleFormat::S32: + case SampleFormat::FLOAT: + case SampleFormat::DSD: + return true; + + case SampleFormat::UNDEFINED: + break; + } + + return false; +} + +/** + * Checks whether the number of channels is valid. + */ +static constexpr inline bool +audio_valid_channel_count(unsigned channels) +{ + return channels >= 1 && channels <= MAX_CHANNELS; +} + +/** + * Returns false if the format is not valid for playback with MPD. + * This function performs some basic validity checks. + */ +inline bool +AudioFormat::IsValid() const +{ + return audio_valid_sample_rate(sample_rate) && + audio_valid_sample_format(format) && + audio_valid_channel_count(channels); +} + +/** + * Returns false if the format mask is not valid for playback with + * MPD. This function performs some basic validity checks. + */ +inline bool +AudioFormat::IsMaskValid() const +{ + return (sample_rate == 0 || + audio_valid_sample_rate(sample_rate)) && + (format == SampleFormat::UNDEFINED || + audio_valid_sample_format(format)) && + (channels == 0 || audio_valid_channel_count(channels)); +} + +gcc_const +static inline unsigned +sample_format_size(SampleFormat format) +{ + switch (format) { + case SampleFormat::S8: + return 1; + + case SampleFormat::S16: + return 2; + + case SampleFormat::S24_P32: + case SampleFormat::S32: + case SampleFormat::FLOAT: + return 4; + + case SampleFormat::DSD: + /* each frame has 8 samples per channel */ + return 1; + + case SampleFormat::UNDEFINED: + return 0; + } + + assert(false); + gcc_unreachable(); +} + +inline unsigned +AudioFormat::GetSampleSize() const +{ + return sample_format_size(format); +} + +inline unsigned +AudioFormat::GetFrameSize() const +{ + return GetSampleSize() * channels; +} + +inline double +AudioFormat::GetTimeToSize() const +{ + return sample_rate * GetFrameSize(); +} + +/** + * Renders a #sample_format enum into a string, e.g. for printing it + * in a log file. + * + * @param format a #sample_format enum value + * @return the string + */ +gcc_pure gcc_malloc +const char * +sample_format_to_string(SampleFormat format); + +/** + * Renders the #audio_format object into a string, e.g. for printing + * it in a log file. + * + * @param af the #audio_format object + * @param s a buffer to print into + * @return the string, or NULL if the #audio_format object is invalid + */ +gcc_pure gcc_malloc +const char * +audio_format_to_string(AudioFormat af, + struct audio_format_string *s); + +#endif diff --git a/src/AudioParser.cxx b/src/AudioParser.cxx index 297d99f12..4c345ca33 100644 --- a/src/AudioParser.cxx +++ b/src/AudioParser.cxx @@ -24,7 +24,7 @@ #include "config.h" #include "AudioParser.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "CheckAudioFormat.hxx" #include "gcc.h" @@ -69,27 +69,27 @@ parse_sample_rate(const char *src, bool mask, uint32_t *sample_rate_r, static bool parse_sample_format(const char *src, bool mask, - enum sample_format *sample_format_r, + SampleFormat *sample_format_r, const char **endptr_r, GError **error_r) { unsigned long value; char *endptr; - enum sample_format sample_format; + SampleFormat sample_format; if (mask && *src == '*') { - *sample_format_r = SAMPLE_FORMAT_UNDEFINED; + *sample_format_r = SampleFormat::UNDEFINED; *endptr_r = src + 1; return true; } if (*src == 'f') { - *sample_format_r = SAMPLE_FORMAT_FLOAT; + *sample_format_r = SampleFormat::FLOAT; *endptr_r = src + 1; return true; } if (memcmp(src, "dsd", 3) == 0) { - *sample_format_r = SAMPLE_FORMAT_DSD; + *sample_format_r = SampleFormat::DSD; *endptr_r = src + 3; return true; } @@ -103,11 +103,11 @@ parse_sample_format(const char *src, bool mask, switch (value) { case 8: - sample_format = SAMPLE_FORMAT_S8; + sample_format = SampleFormat::S8; break; case 16: - sample_format = SAMPLE_FORMAT_S16; + sample_format = SampleFormat::S16; break; case 24: @@ -115,11 +115,11 @@ parse_sample_format(const char *src, bool mask, /* for backwards compatibility */ endptr += 2; - sample_format = SAMPLE_FORMAT_S24_P32; + sample_format = SampleFormat::S24_P32; break; case 32: - sample_format = SAMPLE_FORMAT_S32; + sample_format = SampleFormat::S32; break; default: @@ -162,14 +162,14 @@ parse_channel_count(const char *src, bool mask, uint8_t *channels_r, } bool -audio_format_parse(struct audio_format *dest, const char *src, +audio_format_parse(AudioFormat &dest, const char *src, bool mask, GError **error_r) { uint32_t rate; - enum sample_format sample_format; + SampleFormat sample_format; uint8_t channels; - audio_format_clear(dest); + dest.Clear(); /* parse sample rate */ @@ -191,7 +191,7 @@ audio_format_parse(struct audio_format *dest, const char *src, #if GCC_CHECK_VERSION(4,7) /* workaround -Wmaybe-uninitialized false positive */ - sample_format = SAMPLE_FORMAT_UNDEFINED; + sample_format = SampleFormat::UNDEFINED; #endif if (!parse_sample_format(src, mask, &sample_format, &src, error_r)) @@ -214,9 +214,10 @@ audio_format_parse(struct audio_format *dest, const char *src, return false; } - audio_format_init(dest, rate, sample_format, channels); - assert(mask ? audio_format_mask_valid(dest) - : audio_format_valid(dest)); + dest = AudioFormat(rate, sample_format, channels); + assert(mask + ? dest.IsMaskValid() + : dest.IsValid()); return true; } diff --git a/src/AudioParser.hxx b/src/AudioParser.hxx index f7855e8e3..14c26392c 100644 --- a/src/AudioParser.hxx +++ b/src/AudioParser.hxx @@ -27,7 +27,7 @@ #include "gerror.h" -struct audio_format; +struct AudioFormat; /** * Parses a string in the form "SAMPLE_RATE:BITS:CHANNELS" into an @@ -41,7 +41,7 @@ struct audio_format; * @return true on success */ bool -audio_format_parse(struct audio_format *dest, const char *src, +audio_format_parse(AudioFormat &dest, const char *src, bool mask, GError **error_r); #endif diff --git a/src/CheckAudioFormat.cxx b/src/CheckAudioFormat.cxx index 6d484be19..fabffd56c 100644 --- a/src/CheckAudioFormat.cxx +++ b/src/CheckAudioFormat.cxx @@ -18,7 +18,7 @@ */ #include "CheckAudioFormat.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -35,11 +35,12 @@ audio_check_sample_rate(unsigned long sample_rate, GError **error_r) } bool -audio_check_sample_format(enum sample_format sample_format, GError **error_r) +audio_check_sample_format(SampleFormat sample_format, GError **error_r) { if (!audio_valid_sample_format(sample_format)) { g_set_error(error_r, audio_format_quark(), 0, - "Invalid sample format: %u", sample_format); + "Invalid sample format: %u", + unsigned(sample_format)); return false; } @@ -59,15 +60,15 @@ audio_check_channel_count(unsigned channels, GError **error_r) } bool -audio_format_init_checked(struct audio_format *af, unsigned long sample_rate, - enum sample_format sample_format, unsigned channels, +audio_format_init_checked(AudioFormat &af, unsigned long sample_rate, + SampleFormat sample_format, unsigned channels, GError **error_r) { if (audio_check_sample_rate(sample_rate, error_r) && audio_check_sample_format(sample_format, error_r) && audio_check_channel_count(channels, error_r)) { - audio_format_init(af, sample_rate, sample_format, channels); - assert(audio_format_valid(af)); + af = AudioFormat(sample_rate, sample_format, channels); + assert(af.IsValid()); return true; } else return false; diff --git a/src/CheckAudioFormat.hxx b/src/CheckAudioFormat.hxx index cb3e43c22..7fbce7f98 100644 --- a/src/CheckAudioFormat.hxx +++ b/src/CheckAudioFormat.hxx @@ -20,7 +20,7 @@ #ifndef MPD_CHECK_AUDIO_FORMAT_HXX #define MPD_CHECK_AUDIO_FORMAT_HXX -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -38,7 +38,7 @@ bool audio_check_sample_rate(unsigned long sample_rate, GError **error_r); bool -audio_check_sample_format(enum sample_format, GError **error_r); +audio_check_sample_format(SampleFormat sample_format, GError **error_r); bool audio_check_channel_count(unsigned sample_format, GError **error_r); @@ -47,8 +47,8 @@ audio_check_channel_count(unsigned sample_format, GError **error_r); * Wrapper for audio_format_init(), which checks all attributes. */ bool -audio_format_init_checked(struct audio_format *af, unsigned long sample_rate, - enum sample_format sample_format, unsigned channels, +audio_format_init_checked(AudioFormat &af, unsigned long sample_rate, + SampleFormat sample_format, unsigned channels, GError **error_r); #endif diff --git a/src/CrossFade.cxx b/src/CrossFade.cxx index 253038b26..4f5ff32ca 100644 --- a/src/CrossFade.cxx +++ b/src/CrossFade.cxx @@ -20,7 +20,7 @@ #include "config.h" #include "CrossFade.hxx" #include "MusicChunk.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "Tag.hxx" #include @@ -97,8 +97,8 @@ unsigned cross_fade_calc(float duration, float total_time, float mixramp_db, float mixramp_delay, float replay_gain_db, float replay_gain_prev_db, char *mixramp_start, char *mixramp_prev_end, - const struct audio_format *af, - const struct audio_format *old_format, + const AudioFormat af, + const AudioFormat old_format, unsigned max_chunks) { unsigned int chunks = 0; @@ -107,13 +107,13 @@ unsigned cross_fade_calc(float duration, float total_time, if (duration < 0 || duration >= total_time || /* we can't crossfade when the audio formats are different */ - !audio_format_equals(af, old_format)) + af != old_format) return 0; assert(duration >= 0); - assert(audio_format_valid(af)); + assert(af.IsValid()); - chunks_f = (float)audio_format_time_to_size(af) / (float)CHUNK_SIZE; + chunks_f = (float)af.GetTimeToSize() / (float)CHUNK_SIZE; if (std::isnan(mixramp_delay) || !mixramp_start || !mixramp_prev_end) { chunks = (chunks_f * duration + 0.5); diff --git a/src/CrossFade.hxx b/src/CrossFade.hxx index 1c4670758..f285c0e9a 100644 --- a/src/CrossFade.hxx +++ b/src/CrossFade.hxx @@ -20,7 +20,7 @@ #ifndef MPD_CROSSFADE_HXX #define MPD_CROSSFADE_HXX -struct audio_format; +struct AudioFormat; struct music_chunk; /** @@ -44,8 +44,7 @@ unsigned cross_fade_calc(float duration, float total_time, float mixramp_db, float mixramp_delay, float replay_gain_db, float replay_gain_prev_db, char *mixramp_start, char *mixramp_prev_end, - const struct audio_format *af, - const struct audio_format *old_format, + AudioFormat af, AudioFormat old_format, unsigned max_chunks); #endif diff --git a/src/DecoderAPI.cxx b/src/DecoderAPI.cxx index a4bdf37c4..03a75b2f9 100644 --- a/src/DecoderAPI.cxx +++ b/src/DecoderAPI.cxx @@ -40,7 +40,7 @@ void decoder_initialized(struct decoder *decoder, - const struct audio_format *audio_format, + const AudioFormat audio_format, bool seekable, float total_time) { struct decoder_control *dc = decoder->dc; @@ -52,12 +52,11 @@ decoder_initialized(struct decoder *decoder, assert(decoder->stream_tag == NULL); assert(decoder->decoder_tag == NULL); assert(!decoder->seeking); - assert(audio_format != NULL); - assert(audio_format_defined(audio_format)); - assert(audio_format_valid(audio_format)); + assert(audio_format.IsDefined()); + assert(audio_format.IsValid()); - dc->in_audio_format = *audio_format; - getOutputAudioFormat(audio_format, &dc->out_audio_format); + dc->in_audio_format = audio_format; + dc->out_audio_format = getOutputAudioFormat(audio_format); dc->seekable = seekable; dc->total_time = total_time; @@ -68,13 +67,12 @@ decoder_initialized(struct decoder *decoder, dc->Unlock(); g_debug("audio_format=%s, seekable=%s", - audio_format_to_string(&dc->in_audio_format, &af_string), + audio_format_to_string(dc->in_audio_format, &af_string), seekable ? "true" : "false"); - if (!audio_format_equals(&dc->in_audio_format, - &dc->out_audio_format)) + if (dc->in_audio_format != dc->out_audio_format) g_debug("converting to %s", - audio_format_to_string(&dc->out_audio_format, + audio_format_to_string(dc->out_audio_format, &af_string)); } @@ -371,7 +369,7 @@ decoder_data(struct decoder *decoder, assert(dc->state == DECODE_STATE_DECODE); assert(dc->pipe != NULL); - assert(length % audio_format_frame_size(&dc->in_audio_format) == 0); + assert(length % dc->in_audio_format.GetFrameSize() == 0); dc->Lock(); cmd = decoder_get_virtual_command(decoder); @@ -398,10 +396,10 @@ decoder_data(struct decoder *decoder, return cmd; } - if (!audio_format_equals(&dc->in_audio_format, &dc->out_audio_format)) { - data = decoder->conv_state.Convert(&dc->in_audio_format, + if (dc->in_audio_format != dc->out_audio_format) { + data = decoder->conv_state.Convert(dc->in_audio_format, data, length, - &dc->out_audio_format, + dc->out_audio_format, &length, &error); if (data == NULL) { @@ -457,7 +455,7 @@ decoder_data(struct decoder *decoder, length -= nbytes; decoder->timestamp += (double)nbytes / - audio_format_time_to_size(&dc->out_audio_format); + dc->out_audio_format.GetTimeToSize(); if (dc->end_ms > 0 && decoder->timestamp >= dc->end_ms / 1000.0) diff --git a/src/DecoderAPI.hxx b/src/DecoderAPI.hxx index c70365bff..a5eb2f236 100644 --- a/src/DecoderAPI.hxx +++ b/src/DecoderAPI.hxx @@ -33,7 +33,7 @@ #include "input_stream.h" #include "replay_gain_info.h" #include "Tag.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "conf.h" /** @@ -48,7 +48,7 @@ */ void decoder_initialized(struct decoder *decoder, - const struct audio_format *audio_format, + AudioFormat audio_format, bool seekable, float total_time); /** diff --git a/src/DecoderControl.hxx b/src/DecoderControl.hxx index 69de4800c..31c72657b 100644 --- a/src/DecoderControl.hxx +++ b/src/DecoderControl.hxx @@ -21,7 +21,7 @@ #define MPD_DECODER_CONTROL_HXX #include "DecoderCommand.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "thread/Mutex.hxx" #include "thread/Cond.hxx" @@ -85,10 +85,10 @@ struct decoder_control { double seek_where; /** the format of the song file */ - struct audio_format in_audio_format; + AudioFormat in_audio_format; /** the format being sent to the music pipe */ - struct audio_format out_audio_format; + AudioFormat out_audio_format; /** * The song currently being decoded. This attribute is set by diff --git a/src/EncoderAPI.hxx b/src/EncoderAPI.hxx index d35955f1b..d430214d6 100644 --- a/src/EncoderAPI.hxx +++ b/src/EncoderAPI.hxx @@ -26,7 +26,7 @@ #define MPD_ENCODER_API_HXX #include "EncoderPlugin.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "Tag.hxx" #include "conf.h" diff --git a/src/EncoderPlugin.hxx b/src/EncoderPlugin.hxx index 868a9998c..70bdabe1a 100644 --- a/src/EncoderPlugin.hxx +++ b/src/EncoderPlugin.hxx @@ -27,7 +27,7 @@ #include struct EncoderPlugin; -struct audio_format; +struct AudioFormat; struct config_param; struct Tag; @@ -55,7 +55,7 @@ struct EncoderPlugin { void (*finish)(Encoder *encoder); bool (*open)(Encoder *encoder, - struct audio_format *audio_format, + AudioFormat &audio_format, GError **error); void (*close)(Encoder *encoder); @@ -122,7 +122,7 @@ encoder_finish(Encoder *encoder) * @return true on success */ static inline bool -encoder_open(Encoder *encoder, struct audio_format *audio_format, +encoder_open(Encoder *encoder, AudioFormat &audio_format, GError **error) { assert(!encoder->open); diff --git a/src/FilterInternal.hxx b/src/FilterInternal.hxx index cdc2d0ea1..103687b61 100644 --- a/src/FilterInternal.hxx +++ b/src/FilterInternal.hxx @@ -25,7 +25,7 @@ #ifndef MPD_FILTER_INTERNAL_HXX #define MPD_FILTER_INTERNAL_HXX -struct audio_format; +struct AudioFormat; class Filter { public: @@ -40,10 +40,10 @@ public: * format * @param error location to store the error occurring, or NULL * to ignore errors. - * @return the format of outgoing data + * @return the format of outgoing data or + * AudioFormat::Undefined() on error */ - virtual const audio_format *Open(audio_format &af, - GError **error_r) = 0; + virtual AudioFormat Open(AudioFormat &af, GError **error_r) = 0; /** * Closes the filter. After that, you may call Open() again. diff --git a/src/MusicChunk.cxx b/src/MusicChunk.cxx index 55d2f7f30..fb9019988 100644 --- a/src/MusicChunk.cxx +++ b/src/MusicChunk.cxx @@ -19,7 +19,7 @@ #include "config.h" #include "MusicChunk.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "Tag.hxx" #include @@ -31,22 +31,21 @@ music_chunk::~music_chunk() #ifndef NDEBUG bool -music_chunk::CheckFormat(const struct audio_format &other_format) const +music_chunk::CheckFormat(const AudioFormat other_format) const { - assert(audio_format_valid(&other_format)); + assert(other_format.IsValid()); - return length == 0 || - audio_format_equals(&audio_format, &other_format); + return length == 0 || audio_format == other_format; } #endif void * -music_chunk::Write(const struct audio_format &af, +music_chunk::Write(const AudioFormat af, float data_time, uint16_t _bit_rate, size_t *max_length_r) { assert(CheckFormat(af)); - assert(length == 0 || audio_format_valid(&audio_format)); + assert(length == 0 || audio_format.IsValid()); if (length == 0) { /* if the chunk is empty, nobody has set bitRate and @@ -56,7 +55,7 @@ music_chunk::Write(const struct audio_format &af, times = data_time; } - const size_t frame_size = audio_format_frame_size(&af); + const size_t frame_size = af.GetFrameSize(); size_t num_frames = (sizeof(data) - length) / frame_size; if (num_frames == 0) return NULL; @@ -70,12 +69,12 @@ music_chunk::Write(const struct audio_format &af, } bool -music_chunk::Expand(const struct audio_format &af, size_t _length) +music_chunk::Expand(const AudioFormat af, size_t _length) { - const size_t frame_size = audio_format_frame_size(&af); + const size_t frame_size = af.GetFrameSize(); assert(length + _length <= sizeof(data)); - assert(audio_format_equals(&audio_format, &af)); + assert(audio_format == af); length += _length; diff --git a/src/MusicChunk.hxx b/src/MusicChunk.hxx index a15b3a702..23c3b7423 100644 --- a/src/MusicChunk.hxx +++ b/src/MusicChunk.hxx @@ -23,7 +23,7 @@ #include "replay_gain_info.h" #ifndef NDEBUG -#include "audio_format.h" +#include "AudioFormat.hxx" #endif #include @@ -33,7 +33,7 @@ enum { CHUNK_SIZE = 4096, }; -struct audio_format; +struct AudioFormat; struct Tag; /** @@ -90,7 +90,7 @@ struct music_chunk { char data[CHUNK_SIZE]; #ifndef NDEBUG - struct audio_format audio_format; + AudioFormat audio_format; #endif music_chunk() @@ -111,7 +111,7 @@ struct music_chunk { * specified audio_format. */ gcc_pure - bool CheckFormat(const struct audio_format &audio_format) const; + bool CheckFormat(AudioFormat audio_format) const; #endif /** @@ -128,7 +128,7 @@ struct music_chunk { * here * @return a writable buffer, or NULL if the chunk is full */ - void *Write(const struct audio_format &af, + void *Write(AudioFormat af, float data_time, uint16_t bit_rate, size_t *max_length_r); @@ -142,7 +142,7 @@ struct music_chunk { * @param length the number of bytes which were appended * @return true if the chunk is full */ - bool Expand(const struct audio_format &af, size_t length); + bool Expand(AudioFormat af, size_t length); }; void diff --git a/src/MusicPipe.cxx b/src/MusicPipe.cxx index 6f25eff82..5da56cd0c 100644 --- a/src/MusicPipe.cxx +++ b/src/MusicPipe.cxx @@ -41,13 +41,13 @@ struct music_pipe { mutable Mutex mutex; #ifndef NDEBUG - struct audio_format audio_format; + AudioFormat audio_format; #endif music_pipe() :head(nullptr), tail_r(&head), size(0) { #ifndef NDEBUG - audio_format_clear(&audio_format); + audio_format.Clear(); #endif } @@ -73,13 +73,12 @@ music_pipe_free(struct music_pipe *mp) bool music_pipe_check_format(const struct music_pipe *pipe, - const struct audio_format *audio_format) + const AudioFormat audio_format) { assert(pipe != NULL); - assert(audio_format != NULL); - return !audio_format_defined(&pipe->audio_format) || - audio_format_equals(&pipe->audio_format, audio_format); + return !pipe->audio_format.IsDefined() || + pipe->audio_format == audio_format; } bool @@ -131,7 +130,7 @@ music_pipe_shift(struct music_pipe *mp) chunk->next = (struct music_chunk *)(void *)0x01010101; if (mp->size == 0) - audio_format_clear(&mp->audio_format); + mp->audio_format.Clear(); #endif } @@ -151,16 +150,16 @@ void music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk) { assert(!chunk->IsEmpty()); - assert(chunk->length == 0 || audio_format_valid(&chunk->audio_format)); + assert(chunk->length == 0 || chunk->audio_format.IsValid()); const ScopeLock protect(mp->mutex); - assert(mp->size > 0 || !audio_format_defined(&mp->audio_format)); - assert(!audio_format_defined(&mp->audio_format) || + assert(mp->size > 0 || !mp->audio_format.IsDefined()); + assert(!mp->audio_format.IsDefined() || chunk->CheckFormat(mp->audio_format)); #ifndef NDEBUG - if (!audio_format_defined(&mp->audio_format) && chunk->length > 0) + if (!mp->audio_format.IsDefined() && chunk->length > 0) mp->audio_format = chunk->audio_format; #endif diff --git a/src/MusicPipe.hxx b/src/MusicPipe.hxx index 99561ca62..39ad8e981 100644 --- a/src/MusicPipe.hxx +++ b/src/MusicPipe.hxx @@ -23,7 +23,7 @@ #include "gcc.h" #ifndef NDEBUG -struct audio_format; +struct AudioFormat; #endif struct music_chunk; @@ -56,7 +56,7 @@ music_pipe_free(struct music_pipe *mp); */ bool music_pipe_check_format(const struct music_pipe *pipe, - const struct audio_format *audio_format); + AudioFormat audio_format); /** * Checks if the specified chunk is enqueued in the music pipe. diff --git a/src/OutputAPI.hxx b/src/OutputAPI.hxx index 951d723ea..9c79d9b81 100644 --- a/src/OutputAPI.hxx +++ b/src/OutputAPI.hxx @@ -22,7 +22,7 @@ #include "OutputPlugin.hxx" #include "OutputInternal.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "Tag.hxx" #include "conf.h" diff --git a/src/OutputAll.cxx b/src/OutputAll.cxx index cdbdcfbfc..a73afbbb6 100644 --- a/src/OutputAll.cxx +++ b/src/OutputAll.cxx @@ -36,7 +36,7 @@ #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "output" -static struct audio_format input_audio_format; +static AudioFormat input_audio_format; static struct audio_output **audio_outputs; static unsigned int num_audio_outputs; @@ -246,12 +246,12 @@ audio_output_all_update(void) unsigned int i; bool ret = false; - if (!audio_format_defined(&input_audio_format)) + if (!input_audio_format.IsDefined()) return false; for (i = 0; i < num_audio_outputs; ++i) ret = audio_output_update(audio_outputs[i], - &input_audio_format, g_mp) || ret; + input_audio_format, g_mp) || ret; return ret; } @@ -291,14 +291,13 @@ audio_output_all_play(struct music_chunk *chunk, GError **error_r) } bool -audio_output_all_open(const struct audio_format *audio_format, +audio_output_all_open(const AudioFormat audio_format, struct music_buffer *buffer, GError **error_r) { bool ret = false, enabled = false; unsigned int i; - assert(audio_format != NULL); assert(buffer != NULL); assert(g_music_buffer == NULL || g_music_buffer == buffer); assert((g_mp == NULL) == (g_music_buffer == NULL)); @@ -315,10 +314,9 @@ audio_output_all_open(const struct audio_format *audio_format, /* if the pipe hasn't been cleared, the the audio format must not have changed */ assert(music_pipe_empty(g_mp) || - audio_format_equals(audio_format, - &input_audio_format)); + audio_format == input_audio_format); - input_audio_format = *audio_format; + input_audio_format = audio_format; audio_output_all_reset_reopen(); audio_output_all_enable_disable(); @@ -547,7 +545,7 @@ audio_output_all_close(void) g_music_buffer = NULL; - audio_format_clear(&input_audio_format); + input_audio_format.Clear(); audio_output_all_elapsed_time = -1.0; } @@ -570,7 +568,7 @@ audio_output_all_release(void) g_music_buffer = NULL; - audio_format_clear(&input_audio_format); + input_audio_format.Clear(); audio_output_all_elapsed_time = -1.0; } diff --git a/src/OutputAll.hxx b/src/OutputAll.hxx index 0bb9f0763..10f8196aa 100644 --- a/src/OutputAll.hxx +++ b/src/OutputAll.hxx @@ -29,7 +29,7 @@ #include "replay_gain_info.h" #include "gerror.h" -struct audio_format; +struct AudioFormat; struct music_buffer; struct music_chunk; struct player_control; @@ -76,14 +76,13 @@ audio_output_all_enable_disable(void); /** * Opens all audio outputs which are not disabled. * - * @param audio_format the preferred audio format, or NULL to reuse - * the previous format + * @param audio_format the preferred audio format * @param buffer the #music_buffer where consumed #music_chunk objects * should be returned * @return true on success, false on failure */ bool -audio_output_all_open(const struct audio_format *audio_format, +audio_output_all_open(AudioFormat audio_format, struct music_buffer *buffer, GError **error_r); diff --git a/src/OutputControl.cxx b/src/OutputControl.cxx index a11b66116..b8f3a3ea4 100644 --- a/src/OutputControl.cxx +++ b/src/OutputControl.cxx @@ -139,14 +139,14 @@ audio_output_disable(struct audio_output *ao) */ static bool audio_output_open(struct audio_output *ao, - const struct audio_format *audio_format, + const AudioFormat audio_format, const struct music_pipe *mp) { bool open; assert(ao != NULL); assert(ao->allow_play); - assert(audio_format_valid(audio_format)); + assert(audio_format.IsValid()); assert(mp != NULL); if (ao->fail_timer != NULL) { @@ -154,8 +154,7 @@ audio_output_open(struct audio_output *ao, ao->fail_timer = NULL; } - if (ao->open && - audio_format_equals(audio_format, &ao->in_audio_format)) { + if (ao->open && audio_format == ao->in_audio_format) { assert(ao->pipe == mp || (ao->always_on && ao->pause)); @@ -176,7 +175,7 @@ audio_output_open(struct audio_output *ao, return true; } - ao->in_audio_format = *audio_format; + ao->in_audio_format = audio_format; ao->chunk = NULL; ao->pipe = mp; @@ -225,7 +224,7 @@ audio_output_close_locked(struct audio_output *ao) bool audio_output_update(struct audio_output *ao, - const struct audio_format *audio_format, + const AudioFormat audio_format, const struct music_pipe *mp) { assert(mp != NULL); diff --git a/src/OutputControl.hxx b/src/OutputControl.hxx index 2ff095398..373b1849c 100644 --- a/src/OutputControl.hxx +++ b/src/OutputControl.hxx @@ -25,7 +25,7 @@ #include struct audio_output; -struct audio_format; +struct AudioFormat; struct config_param; struct music_pipe; struct player_control; @@ -53,7 +53,7 @@ audio_output_disable(struct audio_output *ao); */ bool audio_output_update(struct audio_output *ao, - const struct audio_format *audio_format, + AudioFormat audio_format, const struct music_pipe *mp); void diff --git a/src/OutputInit.cxx b/src/OutputInit.cxx index 6dd40cd67..cef4755c6 100644 --- a/src/OutputInit.cxx +++ b/src/OutputInit.cxx @@ -155,16 +155,16 @@ ao_base_init(struct audio_output *ao, NULL); if (p != NULL) { bool success = - audio_format_parse(&ao->config_audio_format, + audio_format_parse(ao->config_audio_format, p, true, error_r); if (!success) return false; } else - audio_format_clear(&ao->config_audio_format); + ao->config_audio_format.Clear(); } else { ao->name = "default detected output"; - audio_format_clear(&ao->config_audio_format); + ao->config_audio_format.Clear(); } ao->plugin = plugin; diff --git a/src/OutputInternal.hxx b/src/OutputInternal.hxx index a0d376827..b23ec95b6 100644 --- a/src/OutputInternal.hxx +++ b/src/OutputInternal.hxx @@ -20,7 +20,7 @@ #ifndef MPD_OUTPUT_INTERNAL_HXX #define MPD_OUTPUT_INTERNAL_HXX -#include "audio_format.h" +#include "AudioFormat.hxx" #include "pcm/PcmBuffer.hxx" #include "thread/Mutex.hxx" #include "thread/Cond.hxx" @@ -134,13 +134,13 @@ struct audio_output { /** * The configured audio format. */ - struct audio_format config_audio_format; + AudioFormat config_audio_format; /** * The audio_format in which audio data is received from the * player thread (which in turn receives it from the decoder). */ - struct audio_format in_audio_format; + AudioFormat in_audio_format; /** * The audio_format which is really sent to the device. This @@ -148,7 +148,7 @@ struct audio_output { * in_audio_format, but may have been modified by * plugin->open(). */ - struct audio_format out_audio_format; + AudioFormat out_audio_format; /** * The buffer used to allocate the cross-fading result. diff --git a/src/OutputPlugin.cxx b/src/OutputPlugin.cxx index 1e49ddbaa..00901aad2 100644 --- a/src/OutputPlugin.cxx +++ b/src/OutputPlugin.cxx @@ -54,7 +54,7 @@ ao_plugin_disable(struct audio_output *ao) } bool -ao_plugin_open(struct audio_output *ao, struct audio_format *audio_format, +ao_plugin_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { return ao->plugin->open(ao, audio_format, error); diff --git a/src/OutputPlugin.hxx b/src/OutputPlugin.hxx index 8eb125029..4e19ffcfb 100644 --- a/src/OutputPlugin.hxx +++ b/src/OutputPlugin.hxx @@ -26,7 +26,7 @@ #include struct config_param; -struct audio_format; +struct AudioFormat; struct Tag; /** @@ -89,7 +89,7 @@ struct audio_output_plugin { * @param error location to store the error occurring, or NULL * to ignore errors */ - bool (*open)(struct audio_output *data, struct audio_format *audio_format, + bool (*open)(struct audio_output *data, AudioFormat &audio_format, GError **error); /** @@ -181,7 +181,7 @@ void ao_plugin_disable(struct audio_output *ao); bool -ao_plugin_open(struct audio_output *ao, struct audio_format *audio_format, +ao_plugin_open(struct audio_output *ao, AudioFormat &audio_format, GError **error); void diff --git a/src/OutputThread.cxx b/src/OutputThread.cxx index a57fe7f60..483d35680 100644 --- a/src/OutputThread.cxx +++ b/src/OutputThread.cxx @@ -94,11 +94,11 @@ ao_disable(struct audio_output *ao) } } -static const struct audio_format * -ao_filter_open(struct audio_output *ao, audio_format &format, +static AudioFormat +ao_filter_open(struct audio_output *ao, AudioFormat &format, GError **error_r) { - assert(audio_format_valid(&format)); + assert(format.IsValid()); /* the replay_gain filter cannot fail here */ if (ao->replay_gain_filter != NULL) @@ -106,9 +106,8 @@ ao_filter_open(struct audio_output *ao, audio_format &format, if (ao->other_replay_gain_filter != NULL) ao->other_replay_gain_filter->Open(format, error_r); - const struct audio_format *af - = ao->filter->Open(format, error_r); - if (af == NULL) { + const AudioFormat af = ao->filter->Open(format, error_r); + if (!af.IsDefined()) { if (ao->replay_gain_filter != NULL) ao->replay_gain_filter->Close(); if (ao->other_replay_gain_filter != NULL) @@ -139,7 +138,7 @@ ao_open(struct audio_output *ao) assert(!ao->open); assert(ao->pipe != NULL); assert(ao->chunk == NULL); - assert(audio_format_valid(&ao->in_audio_format)); + assert(ao->in_audio_format.IsValid()); if (ao->fail_timer != NULL) { /* this can only happen when this @@ -158,9 +157,9 @@ ao_open(struct audio_output *ao) /* open the filter */ - const audio_format *filter_audio_format = + const AudioFormat filter_audio_format = ao_filter_open(ao, ao->in_audio_format, &error); - if (filter_audio_format == NULL) { + if (!filter_audio_format.IsDefined()) { g_warning("Failed to open filter for \"%s\" [%s]: %s", ao->name, ao->plugin->name, error->message); g_error_free(error); @@ -169,14 +168,13 @@ ao_open(struct audio_output *ao) return; } - assert(audio_format_valid(filter_audio_format)); + assert(filter_audio_format.IsValid()); - ao->out_audio_format = *filter_audio_format; - audio_format_mask_apply(&ao->out_audio_format, - &ao->config_audio_format); + ao->out_audio_format = filter_audio_format; + ao->out_audio_format.ApplyMask(ao->config_audio_format); ao->mutex.unlock(); - success = ao_plugin_open(ao, &ao->out_audio_format, &error); + success = ao_plugin_open(ao, ao->out_audio_format, &error); ao->mutex.lock(); assert(!ao->open); @@ -198,12 +196,11 @@ ao_open(struct audio_output *ao) g_debug("opened plugin=%s name=\"%s\" " "audio_format=%s", ao->plugin->name, ao->name, - audio_format_to_string(&ao->out_audio_format, &af_string)); + audio_format_to_string(ao->out_audio_format, &af_string)); - if (!audio_format_equals(&ao->in_audio_format, - &ao->out_audio_format)) + if (ao->in_audio_format != ao->out_audio_format) g_debug("converting from %s", - audio_format_to_string(&ao->in_audio_format, + audio_format_to_string(ao->in_audio_format, &af_string)); } @@ -235,12 +232,12 @@ ao_close(struct audio_output *ao, bool drain) static void ao_reopen_filter(struct audio_output *ao) { - const struct audio_format *filter_audio_format; GError *error = NULL; ao_filter_close(ao); - filter_audio_format = ao_filter_open(ao, ao->in_audio_format, &error); - if (filter_audio_format == NULL) { + const AudioFormat filter_audio_format = + ao_filter_open(ao, ao->in_audio_format, &error); + if (!filter_audio_format.IsDefined()) { g_warning("Failed to open filter for \"%s\" [%s]: %s", ao->name, ao->plugin->name, error->message); g_error_free(error); @@ -268,7 +265,7 @@ ao_reopen_filter(struct audio_output *ao) static void ao_reopen(struct audio_output *ao) { - if (!audio_format_fully_defined(&ao->config_audio_format)) { + if (!ao->config_audio_format.IsFullyDefined()) { if (ao->open) { const struct music_pipe *mp = ao->pipe; ao_close(ao, true); @@ -279,8 +276,7 @@ ao_reopen(struct audio_output *ao) the output's open() method determine the effective out_audio_format */ ao->out_audio_format = ao->in_audio_format; - audio_format_mask_apply(&ao->out_audio_format, - &ao->config_audio_format); + ao->out_audio_format.ApplyMask(ao->config_audio_format); } if (ao->open) @@ -327,7 +323,7 @@ ao_chunk_data(struct audio_output *ao, const struct music_chunk *chunk, (void)ao; - assert(length % audio_format_frame_size(&ao->in_audio_format) == 0); + assert(length % ao->in_audio_format.GetFrameSize() == 0); if (length > 0 && replay_gain_filter != NULL) { if (chunk->replay_gain_serial != *replay_gain_serial_p) { @@ -399,10 +395,10 @@ ao_filter_chunk(struct audio_output *ao, const struct music_chunk *chunk, void *dest = ao->cross_fade_buffer.Get(other_length); memcpy(dest, other_data, other_length); if (!pcm_mix(dest, data, length, - sample_format(ao->in_audio_format.format), + ao->in_audio_format.format, 1.0 - chunk->mix_ratio)) { g_warning("Cannot cross-fade format %s", - sample_format_to_string(sample_format(ao->in_audio_format.format))); + sample_format_to_string(ao->in_audio_format.format)); return NULL; } @@ -479,7 +475,7 @@ ao_play_chunk(struct audio_output *ao, const struct music_chunk *chunk) } assert(nbytes <= size); - assert(nbytes % audio_format_frame_size(&ao->out_audio_format) == 0); + assert(nbytes % ao->out_audio_format.GetFrameSize() == 0); data += nbytes; size -= nbytes; diff --git a/src/PlayerCommands.cxx b/src/PlayerCommands.cxx index 1c1c346b7..0d4089940 100644 --- a/src/PlayerCommands.cxx +++ b/src/PlayerCommands.cxx @@ -31,7 +31,7 @@ #include "protocol/ArgParser.hxx" extern "C" { -#include "audio_format.h" +#include "AudioFormat.hxx" } #include "replay_gain_config.h" @@ -178,12 +178,12 @@ handle_status(Client *client, player_status.elapsed_time, player_status.bit_rate); - if (audio_format_defined(&player_status.audio_format)) { + if (player_status.audio_format.IsDefined()) { struct audio_format_string af_string; client_printf(client, COMMAND_STATUS_AUDIO ": %s\n", - audio_format_to_string(&player_status.audio_format, + audio_format_to_string(player_status.audio_format, &af_string)); } } diff --git a/src/PlayerControl.hxx b/src/PlayerControl.hxx index ab87f6092..bea2e05eb 100644 --- a/src/PlayerControl.hxx +++ b/src/PlayerControl.hxx @@ -20,7 +20,7 @@ #ifndef MPD_PLAYER_H #define MPD_PLAYER_H -#include "audio_format.h" +#include "AudioFormat.hxx" #include "thread/Mutex.hxx" #include "thread/Cond.hxx" @@ -85,7 +85,7 @@ enum player_error { struct player_status { enum player_state state; uint16_t bit_rate; - struct audio_format audio_format; + AudioFormat audio_format; float total_time; float elapsed_time; }; @@ -130,7 +130,7 @@ struct player_control { GError *error; uint16_t bit_rate; - struct audio_format audio_format; + AudioFormat audio_format; float total_time; float elapsed_time; diff --git a/src/PlayerThread.cxx b/src/PlayerThread.cxx index d838d4ba9..68fd6390f 100644 --- a/src/PlayerThread.cxx +++ b/src/PlayerThread.cxx @@ -115,7 +115,7 @@ struct player { /** * The current audio format for the audio outputs. */ - struct audio_format play_audio_format; + AudioFormat play_audio_format; /** * The time stamp of the chunk most recently sent to the @@ -279,7 +279,7 @@ player_wait_for_decoder(struct player *player) /* update player_control's song information */ pc->total_time = pc->next_song->GetDuration(); pc->bit_rate = 0; - audio_format_clear(&pc->audio_format); + pc->audio_format.Clear(); /* clear the queued song */ pc->next_song = NULL; @@ -323,12 +323,12 @@ player_open_output(struct player *player) { struct player_control *pc = player->pc; - assert(audio_format_defined(&player->play_audio_format)); + assert(player->play_audio_format.IsDefined()); assert(pc->state == PLAYER_STATE_PLAY || pc->state == PLAYER_STATE_PAUSE); GError *error = NULL; - if (audio_output_all_open(&player->play_audio_format, player_buffer, + if (audio_output_all_open(player->play_audio_format, player_buffer, &error)) { player->output_open = true; player->paused = false; @@ -439,7 +439,7 @@ static bool player_send_silence(struct player *player) { assert(player->output_open); - assert(audio_format_defined(&player->play_audio_format)); + assert(player->play_audio_format.IsDefined()); struct music_chunk *chunk = music_buffer_allocate(player_buffer); if (chunk == NULL) { @@ -451,8 +451,7 @@ player_send_silence(struct player *player) chunk->audio_format = player->play_audio_format; #endif - size_t frame_size = - audio_format_frame_size(&player->play_audio_format); + const size_t frame_size = player->play_audio_format.GetFrameSize(); /* this formula ensures that we don't send partial frames */ unsigned num_frames = sizeof(chunk->data) / frame_size; @@ -597,7 +596,7 @@ static void player_process_command(struct player *player) pc->Lock(); pc->state = PLAYER_STATE_PAUSE; - } else if (!audio_format_defined(&player->play_audio_format)) { + } else if (!player->play_audio_format.IsDefined()) { /* the decoder hasn't provided an audio format yet - don't open the audio device yet */ pc->Lock(); @@ -689,10 +688,10 @@ update_song_tag(Song *song, const Tag &new_tag) static bool play_chunk(struct player_control *pc, Song *song, struct music_chunk *chunk, - const struct audio_format *format, + const AudioFormat format, GError **error_r) { - assert(chunk->CheckFormat(*format)); + assert(chunk->CheckFormat(format)); if (chunk->tag != NULL) update_song_tag(song, *chunk->tag); @@ -712,7 +711,7 @@ play_chunk(struct player_control *pc, return false; pc->total_play_time += (double)chunk->length / - audio_format_time_to_size(format); + format.GetTimeToSize(); return true; } @@ -825,7 +824,7 @@ play_next_chunk(struct player *player) GError *error = NULL; if (!play_chunk(player->pc, player->song, chunk, - &player->play_audio_format, &error)) { + player->play_audio_format, &error)) { g_warning("%s", error->message); music_buffer_return(player_buffer, chunk); @@ -1019,8 +1018,8 @@ static void do_play(struct player_control *pc, struct decoder_control *dc) dc->replay_gain_prev_db, dc->mixramp_start, dc->mixramp_prev_end, - &dc->out_audio_format, - &player.play_audio_format, + dc->out_audio_format, + player.play_audio_format, music_buffer_size(player_buffer) - pc->buffered_before_play); if (player.cross_fade_chunks > 0) { diff --git a/src/Timer.cxx b/src/Timer.cxx index 7ddbda3da..18ccffbcb 100644 --- a/src/Timer.cxx +++ b/src/Timer.cxx @@ -19,7 +19,7 @@ #include "config.h" #include "Timer.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "clock.h" #include @@ -28,10 +28,10 @@ #include #include -Timer::Timer(const struct audio_format &af) +Timer::Timer(const AudioFormat af) : time(0), started(false), - rate(af.sample_rate * audio_format_frame_size(&af)) + rate(af.sample_rate * af.GetFrameSize()) { } diff --git a/src/Timer.hxx b/src/Timer.hxx index 96446a988..ccc29b873 100644 --- a/src/Timer.hxx +++ b/src/Timer.hxx @@ -22,14 +22,14 @@ #include -struct audio_format; +struct AudioFormat; class Timer { uint64_t time; bool started; const int rate; public: - explicit Timer(const struct audio_format& af); + explicit Timer(AudioFormat af); bool IsStarted() const { return started; } diff --git a/src/audio_format.h b/src/audio_format.h deleted file mode 100644 index 642eaa52a..000000000 --- a/src/audio_format.h +++ /dev/null @@ -1,316 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_AUDIO_FORMAT_H -#define MPD_AUDIO_FORMAT_H - -#include "gcc.h" - -#include -#include -#include - -enum sample_format { - SAMPLE_FORMAT_UNDEFINED = 0, - - SAMPLE_FORMAT_S8, - SAMPLE_FORMAT_S16, - - /** - * Signed 24 bit integer samples, packed in 32 bit integers - * (the most significant byte is filled with the sign bit). - */ - SAMPLE_FORMAT_S24_P32, - - SAMPLE_FORMAT_S32, - - /** - * 32 bit floating point samples in the host's format. The - * range is -1.0f to +1.0f. - */ - SAMPLE_FORMAT_FLOAT, - - /** - * Direct Stream Digital. 1-bit samples; each frame has one - * byte (8 samples) per channel. - */ - SAMPLE_FORMAT_DSD, -}; - -static const unsigned MAX_CHANNELS = 8; - -/** - * This structure describes the format of a raw PCM stream. - */ -struct audio_format { - /** - * The sample rate in Hz. A better name for this attribute is - * "frame rate", because technically, you have two samples per - * frame in stereo sound. - */ - uint32_t sample_rate; - - /** - * The format samples are stored in. See the #sample_format - * enum for valid values. - */ - uint8_t format; - - /** - * The number of channels. Only mono (1) and stereo (2) are - * fully supported currently. - */ - uint8_t channels; -}; - -/** - * Buffer for audio_format_string(). - */ -struct audio_format_string { - char buffer[24]; -}; - -/** - * Clears the #audio_format object, i.e. sets all attributes to an - * undefined (invalid) value. - */ -static inline void audio_format_clear(struct audio_format *af) -{ - af->sample_rate = 0; - af->format = SAMPLE_FORMAT_UNDEFINED; - af->channels = 0; -} - -/** - * Initializes an #audio_format object, i.e. sets all - * attributes to valid values. - */ -static inline void audio_format_init(struct audio_format *af, - uint32_t sample_rate, - enum sample_format format, uint8_t channels) -{ - af->sample_rate = sample_rate; - af->format = (uint8_t)format; - af->channels = channels; -} - -/** - * Checks whether the specified #audio_format object has a defined - * value. - */ -static inline bool audio_format_defined(const struct audio_format *af) -{ - return af->sample_rate != 0; -} - -/** - * Checks whether the specified #audio_format object is full, i.e. all - * attributes are defined. This is more complete than - * audio_format_defined(), but slower. - */ -static inline bool -audio_format_fully_defined(const struct audio_format *af) -{ - return af->sample_rate != 0 && af->format != SAMPLE_FORMAT_UNDEFINED && - af->channels != 0; -} - -/** - * Checks whether the specified #audio_format object has at least one - * defined value. - */ -static inline bool -audio_format_mask_defined(const struct audio_format *af) -{ - return af->sample_rate != 0 || af->format != SAMPLE_FORMAT_UNDEFINED || - af->channels != 0; -} - -/** - * Checks whether the sample rate is valid. - * - * @param sample_rate the sample rate in Hz - */ -static inline bool -audio_valid_sample_rate(unsigned sample_rate) -{ - return sample_rate > 0 && sample_rate < (1 << 30); -} - -/** - * Checks whether the sample format is valid. - * - * @param bits the number of significant bits per sample - */ -static inline bool -audio_valid_sample_format(enum sample_format format) -{ - switch (format) { - case SAMPLE_FORMAT_S8: - case SAMPLE_FORMAT_S16: - case SAMPLE_FORMAT_S24_P32: - case SAMPLE_FORMAT_S32: - case SAMPLE_FORMAT_FLOAT: - case SAMPLE_FORMAT_DSD: - return true; - - case SAMPLE_FORMAT_UNDEFINED: - break; - } - - return false; -} - -/** - * Checks whether the number of channels is valid. - */ -static inline bool -audio_valid_channel_count(unsigned channels) -{ - return channels >= 1 && channels <= MAX_CHANNELS; -} - -/** - * Returns false if the format is not valid for playback with MPD. - * This function performs some basic validity checks. - */ -gcc_pure -static inline bool audio_format_valid(const struct audio_format *af) -{ - return audio_valid_sample_rate(af->sample_rate) && - audio_valid_sample_format((enum sample_format)af->format) && - audio_valid_channel_count(af->channels); -} - -/** - * Returns false if the format mask is not valid for playback with - * MPD. This function performs some basic validity checks. - */ -gcc_pure -static inline bool audio_format_mask_valid(const struct audio_format *af) -{ - return (af->sample_rate == 0 || - audio_valid_sample_rate(af->sample_rate)) && - (af->format == SAMPLE_FORMAT_UNDEFINED || - audio_valid_sample_format((enum sample_format)af->format)) && - (af->channels == 0 || audio_valid_channel_count(af->channels)); -} - -static inline bool audio_format_equals(const struct audio_format *a, - const struct audio_format *b) -{ - return a->sample_rate == b->sample_rate && - a->format == b->format && - a->channels == b->channels; -} - -#ifdef __cplusplus -extern "C" { -#endif - -void -audio_format_mask_apply(struct audio_format *af, - const struct audio_format *mask); - -gcc_const -static inline unsigned -sample_format_size(enum sample_format format) -{ - switch (format) { - case SAMPLE_FORMAT_S8: - return 1; - - case SAMPLE_FORMAT_S16: - return 2; - - case SAMPLE_FORMAT_S24_P32: - case SAMPLE_FORMAT_S32: - case SAMPLE_FORMAT_FLOAT: - return 4; - - case SAMPLE_FORMAT_DSD: - /* each frame has 8 samples per channel */ - return 1; - - case SAMPLE_FORMAT_UNDEFINED: - return 0; - } - - assert(false); - gcc_unreachable(); -} - -/** - * Returns the size of each (mono) sample in bytes. - */ -gcc_pure -static inline unsigned audio_format_sample_size(const struct audio_format *af) -{ - return sample_format_size((enum sample_format)af->format); -} - -/** - * Returns the size of each full frame in bytes. - */ -gcc_pure -static inline unsigned -audio_format_frame_size(const struct audio_format *af) -{ - return audio_format_sample_size(af) * af->channels; -} - -/** - * Returns the floating point factor which converts a time span to a - * storage size in bytes. - */ -gcc_pure -static inline double audio_format_time_to_size(const struct audio_format *af) -{ - return af->sample_rate * audio_format_frame_size(af); -} - -/** - * Renders a #sample_format enum into a string, e.g. for printing it - * in a log file. - * - * @param format a #sample_format enum value - * @return the string - */ -gcc_pure gcc_malloc -const char * -sample_format_to_string(enum sample_format format); - -/** - * Renders the #audio_format object into a string, e.g. for printing - * it in a log file. - * - * @param af the #audio_format object - * @param s a buffer to print into - * @return the string, or NULL if the #audio_format object is invalid - */ -gcc_pure gcc_malloc -const char * -audio_format_to_string(const struct audio_format *af, - struct audio_format_string *s); - -#ifdef __cplusplus -} -#endif - -#endif diff --git a/src/decoder/AdPlugDecoderPlugin.cxx b/src/decoder/AdPlugDecoderPlugin.cxx index a9a90c283..37a95ce5d 100644 --- a/src/decoder/AdPlugDecoderPlugin.cxx +++ b/src/decoder/AdPlugDecoderPlugin.cxx @@ -60,11 +60,10 @@ adplug_file_decode(struct decoder *decoder, const char *path_fs) if (player == nullptr) return; - struct audio_format audio_format; - audio_format_init(&audio_format, sample_rate, SAMPLE_FORMAT_S16, 2); - assert(audio_format_valid(&audio_format)); + const AudioFormat audio_format(sample_rate, SampleFormat::S16, 2); + assert(audio_format.IsValid()); - decoder_initialized(decoder, &audio_format, false, + decoder_initialized(decoder, audio_format, false, player->songlength() / 1000.); int16_t buffer[2048]; diff --git a/src/decoder/AudiofileDecoderPlugin.cxx b/src/decoder/AudiofileDecoderPlugin.cxx index b77c41d02..9c00b20ce 100644 --- a/src/decoder/AudiofileDecoderPlugin.cxx +++ b/src/decoder/AudiofileDecoderPlugin.cxx @@ -114,27 +114,27 @@ setup_virtual_fops(struct input_stream *stream) return vf; } -static enum sample_format +static SampleFormat audiofile_bits_to_sample_format(int bits) { switch (bits) { case 8: - return SAMPLE_FORMAT_S8; + return SampleFormat::S8; case 16: - return SAMPLE_FORMAT_S16; + return SampleFormat::S16; case 24: - return SAMPLE_FORMAT_S24_P32; + return SampleFormat::S24_P32; case 32: - return SAMPLE_FORMAT_S32; + return SampleFormat::S32; } - return SAMPLE_FORMAT_UNDEFINED; + return SampleFormat::UNDEFINED; } -static enum sample_format +static SampleFormat audiofile_setup_sample_format(AFfilehandle af_fp) { int fs, bits; @@ -160,7 +160,7 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) AFvirtualfile *vf; int fs, frame_count; AFfilehandle af_fp; - struct audio_format audio_format; + AudioFormat audio_format; float total_time; uint16_t bit_rate; int ret; @@ -180,7 +180,7 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) return; } - if (!audio_format_init_checked(&audio_format, + if (!audio_format_init_checked(audio_format, afGetRate(af_fp, AF_DEFAULT_TRACK), audiofile_setup_sample_format(af_fp), afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK), @@ -199,7 +199,7 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); - decoder_initialized(decoder, &audio_format, true, total_time); + decoder_initialized(decoder, audio_format, true, total_time); do { ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk, diff --git a/src/decoder/DsdiffDecoderPlugin.cxx b/src/decoder/DsdiffDecoderPlugin.cxx index 4b9a59a7a..10b31a204 100644 --- a/src/decoder/DsdiffDecoderPlugin.cxx +++ b/src/decoder/DsdiffDecoderPlugin.cxx @@ -433,9 +433,9 @@ dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is) return; GError *error = nullptr; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8, - SAMPLE_FORMAT_DSD, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8, + SampleFormat::DSD, metadata.channels, &error)) { g_warning("%s", error->message); g_error_free(error); @@ -448,7 +448,7 @@ dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is) (float) metadata.sample_rate; /* success: file was recognized */ - decoder_initialized(decoder, &audio_format, false, songtime); + decoder_initialized(decoder, audio_format, false, songtime); /* every iteration of the following loop decodes one "DSD" chunk from a DFF file */ @@ -487,9 +487,9 @@ dsdiff_scan_stream(struct input_stream *is, if (!dsdiff_read_metadata(nullptr, is, &metadata, &chunk_header)) return false; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8, - SAMPLE_FORMAT_DSD, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8, + SampleFormat::DSD, metadata.channels, nullptr)) /* refuse to parse files which we cannot play anyway */ return false; diff --git a/src/decoder/DsfDecoderPlugin.cxx b/src/decoder/DsfDecoderPlugin.cxx index 9661d70e6..ad1323d88 100644 --- a/src/decoder/DsfDecoderPlugin.cxx +++ b/src/decoder/DsfDecoderPlugin.cxx @@ -285,9 +285,9 @@ dsf_stream_decode(struct decoder *decoder, struct input_stream *is) return; GError *error = NULL; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8, - SAMPLE_FORMAT_DSD, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8, + SampleFormat::DSD, metadata.channels, &error)) { g_warning("%s", error->message); g_error_free(error); @@ -299,7 +299,7 @@ dsf_stream_decode(struct decoder *decoder, struct input_stream *is) (float) metadata.sample_rate; /* success: file was recognized */ - decoder_initialized(decoder, &audio_format, false, songtime); + decoder_initialized(decoder, audio_format, false, songtime); if (!dsf_decode_chunk(decoder, is, metadata.channels, chunk_size, @@ -317,9 +317,9 @@ dsf_scan_stream(struct input_stream *is, if (!dsf_read_metadata(NULL, is, &metadata)) return false; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8, - SAMPLE_FORMAT_DSD, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8, + SampleFormat::DSD, metadata.channels, NULL)) /* refuse to parse files which we cannot play anyway */ return false; diff --git a/src/decoder/FaadDecoderPlugin.cxx b/src/decoder/FaadDecoderPlugin.cxx index 1b7edb49f..547ba24e0 100644 --- a/src/decoder/FaadDecoderPlugin.cxx +++ b/src/decoder/FaadDecoderPlugin.cxx @@ -248,7 +248,7 @@ faad_song_duration(DecoderBuffer *buffer, struct input_stream *is) */ static bool faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer, - struct audio_format *audio_format, GError **error_r) + AudioFormat &audio_format, GError **error_r) { int32_t nbytes; uint32_t sample_rate; @@ -285,7 +285,7 @@ faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer, decoder_buffer_consume(buffer, nbytes); return audio_format_init_checked(audio_format, sample_rate, - SAMPLE_FORMAT_S16, channels, error_r); + SampleFormat::S16, channels, error_r); } /** @@ -325,7 +325,7 @@ faad_get_file_time_float(struct input_stream *is) if (length < 0) { bool ret; - struct audio_format audio_format; + AudioFormat audio_format; NeAACDecHandle decoder = NeAACDecOpen(); @@ -336,7 +336,7 @@ faad_get_file_time_float(struct input_stream *is) decoder_buffer_fill(buffer); - ret = faad_decoder_init(decoder, buffer, &audio_format, nullptr); + ret = faad_decoder_init(decoder, buffer, audio_format, nullptr); if (ret) length = 0; @@ -370,7 +370,7 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) { GError *error = nullptr; float total_time = 0; - struct audio_format audio_format; + AudioFormat audio_format; bool ret; uint16_t bit_rate = 0; DecoderBuffer *buffer; @@ -400,7 +400,7 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) /* initialize it */ - ret = faad_decoder_init(decoder, buffer, &audio_format, &error); + ret = faad_decoder_init(decoder, buffer, audio_format, &error); if (!ret) { g_warning("%s", error->message); g_error_free(error); @@ -410,7 +410,7 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) /* initialize the MPD core */ - decoder_initialized(mpd_decoder, &audio_format, false, total_time); + decoder_initialized(mpd_decoder, audio_format, false, total_time); /* the decoder loop */ diff --git a/src/decoder/FfmpegDecoderPlugin.cxx b/src/decoder/FfmpegDecoderPlugin.cxx index b4aa947c9..e4330f4d6 100644 --- a/src/decoder/FfmpegDecoderPlugin.cxx +++ b/src/decoder/FfmpegDecoderPlugin.cxx @@ -52,6 +52,11 @@ extern "C" { #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "ffmpeg" +/* suppress the ffmpeg compatibility macro */ +#ifdef SampleFormat +#undef SampleFormat +#endif + static GLogLevelFlags level_ffmpeg_to_glib(int level) { @@ -297,20 +302,20 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is, } G_GNUC_CONST -static enum sample_format +static SampleFormat ffmpeg_sample_format(enum AVSampleFormat sample_fmt) { switch (sample_fmt) { case AV_SAMPLE_FMT_S16: case AV_SAMPLE_FMT_S16P: - return SAMPLE_FORMAT_S16; + return SampleFormat::S16; case AV_SAMPLE_FMT_S32: case AV_SAMPLE_FMT_S32P: - return SAMPLE_FORMAT_S32; + return SampleFormat::S32; case AV_SAMPLE_FMT_FLTP: - return SAMPLE_FORMAT_FLOAT; + return SampleFormat::FLOAT; default: break; @@ -325,7 +330,7 @@ ffmpeg_sample_format(enum AVSampleFormat sample_fmt) else g_warning("Unsupported libavcodec SampleFormat value: %d", sample_fmt); - return SAMPLE_FORMAT_UNDEFINED; + return SampleFormat::UNDEFINED; } static AVInputFormat * @@ -420,14 +425,14 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input) return; } - const enum sample_format sample_format = + const SampleFormat sample_format = ffmpeg_sample_format(codec_context->sample_fmt); - if (sample_format == SAMPLE_FORMAT_UNDEFINED) + if (sample_format == SampleFormat::UNDEFINED) return; GError *error = NULL; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, codec_context->sample_rate, sample_format, codec_context->channels, &error)) { @@ -455,7 +460,7 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input) ? format_context->duration / AV_TIME_BASE : 0; - decoder_initialized(decoder, &audio_format, + decoder_initialized(decoder, audio_format, input->seekable, total_time); AVFrame *frame = avcodec_alloc_frame(); diff --git a/src/decoder/FfmpegMetaData.hxx b/src/decoder/FfmpegMetaData.hxx index 466d2cb1d..0fd73df04 100644 --- a/src/decoder/FfmpegMetaData.hxx +++ b/src/decoder/FfmpegMetaData.hxx @@ -26,6 +26,11 @@ extern "C" { #include } +/* suppress the ffmpeg compatibility macro */ +#ifdef SampleFormat +#undef SampleFormat +#endif + struct tag_handler; void diff --git a/src/decoder/FlacCommon.cxx b/src/decoder/FlacCommon.cxx index 32917ed86..5bcc20b97 100644 --- a/src/decoder/FlacCommon.cxx +++ b/src/decoder/FlacCommon.cxx @@ -40,24 +40,24 @@ flac_data::flac_data(struct decoder *_decoder, { } -static enum sample_format +static SampleFormat flac_sample_format(unsigned bits_per_sample) { switch (bits_per_sample) { case 8: - return SAMPLE_FORMAT_S8; + return SampleFormat::S8; case 16: - return SAMPLE_FORMAT_S16; + return SampleFormat::S16; case 24: - return SAMPLE_FORMAT_S24_P32; + return SampleFormat::S24_P32; case 32: - return SAMPLE_FORMAT_S32; + return SampleFormat::S32; default: - return SAMPLE_FORMAT_UNDEFINED; + return SampleFormat::UNDEFINED; } } @@ -69,7 +69,7 @@ flac_got_stream_info(struct flac_data *data, return; GError *error = nullptr; - if (!audio_format_init_checked(&data->audio_format, + if (!audio_format_init_checked(data->audio_format, stream_info->sample_rate, flac_sample_format(stream_info->bits_per_sample), stream_info->channels, &error)) { @@ -79,7 +79,7 @@ flac_got_stream_info(struct flac_data *data, return; } - data->frame_size = audio_format_frame_size(&data->audio_format); + data->frame_size = data->audio_format.GetFrameSize(); if (data->total_frames == 0) data->total_frames = stream_info->total_samples; @@ -132,7 +132,7 @@ flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header) return false; GError *error = nullptr; - if (!audio_format_init_checked(&data->audio_format, + if (!audio_format_init_checked(data->audio_format, header->sample_rate, flac_sample_format(header->bits_per_sample), header->channels, &error)) { @@ -142,9 +142,9 @@ flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header) return false; } - data->frame_size = audio_format_frame_size(&data->audio_format); + data->frame_size = data->audio_format.GetFrameSize(); - decoder_initialized(data->decoder, &data->audio_format, + decoder_initialized(data->decoder, data->audio_format, data->input_stream->seekable, (float)data->total_frames / (float)data->audio_format.sample_rate); @@ -170,7 +170,7 @@ flac_common_write(struct flac_data *data, const FLAC__Frame * frame, buffer = data->buffer.Get(buffer_size); flac_convert(buffer, frame->header.channels, - (enum sample_format)data->audio_format.format, buf, + data->audio_format.format, buf, 0, frame->header.blocksize); if (nbytes > 0) diff --git a/src/decoder/FlacCommon.hxx b/src/decoder/FlacCommon.hxx index d2e240d81..f9fade6fc 100644 --- a/src/decoder/FlacCommon.hxx +++ b/src/decoder/FlacCommon.hxx @@ -56,7 +56,7 @@ struct flac_data : public FlacInput { * The validated audio format of the FLAC file. This * attribute is defined if "initialized" is true. */ - struct audio_format audio_format; + AudioFormat audio_format; /** * The total number of frames in this song. The decoder diff --git a/src/decoder/FlacDecoderPlugin.cxx b/src/decoder/FlacDecoderPlugin.cxx index fc0925610..7becf73e5 100644 --- a/src/decoder/FlacDecoderPlugin.cxx +++ b/src/decoder/FlacDecoderPlugin.cxx @@ -144,7 +144,7 @@ flac_decoder_initialize(struct flac_data *data, FLAC__StreamDecoder *sd, if (data->initialized) { /* done */ - decoder_initialized(data->decoder, &data->audio_format, + decoder_initialized(data->decoder, data->audio_format, data->input_stream->seekable, (float)data->total_frames / (float)data->audio_format.sample_rate); diff --git a/src/decoder/FlacPcm.cxx b/src/decoder/FlacPcm.cxx index 17a13edda..ff855fa70 100644 --- a/src/decoder/FlacPcm.cxx +++ b/src/decoder/FlacPcm.cxx @@ -76,12 +76,12 @@ flac_convert_8(int8_t *dest, void flac_convert(void *dest, - unsigned int num_channels, enum sample_format sample_format, + unsigned int num_channels, SampleFormat sample_format, const FLAC__int32 *const buf[], unsigned int position, unsigned int end) { switch (sample_format) { - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: if (num_channels == 2) flac_convert_stereo16((int16_t*)dest, buf, position, end); @@ -90,20 +90,20 @@ flac_convert(void *dest, position, end); break; - case SAMPLE_FORMAT_S24_P32: - case SAMPLE_FORMAT_S32: + case SampleFormat::S24_P32: + case SampleFormat::S32: flac_convert_32((int32_t*)dest, num_channels, buf, position, end); break; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: flac_convert_8((int8_t*)dest, num_channels, buf, position, end); break; - case SAMPLE_FORMAT_FLOAT: - case SAMPLE_FORMAT_DSD: - case SAMPLE_FORMAT_UNDEFINED: + case SampleFormat::FLOAT: + case SampleFormat::DSD: + case SampleFormat::UNDEFINED: assert(false); gcc_unreachable(); } diff --git a/src/decoder/FlacPcm.hxx b/src/decoder/FlacPcm.hxx index 97d214c17..fa85f65dd 100644 --- a/src/decoder/FlacPcm.hxx +++ b/src/decoder/FlacPcm.hxx @@ -20,13 +20,13 @@ #ifndef MPD_FLAC_PCM_HXX #define MPD_FLAC_PCM_HXX -#include "audio_format.h" +#include "AudioFormat.hxx" #include void flac_convert(void *dest, - unsigned int num_channels, enum sample_format sample_format, + unsigned int num_channels, SampleFormat sample_format, const FLAC__int32 *const buf[], unsigned int position, unsigned int end); diff --git a/src/decoder/FluidsynthDecoderPlugin.cxx b/src/decoder/FluidsynthDecoderPlugin.cxx index 15d2f5e6b..e559ad45e 100644 --- a/src/decoder/FluidsynthDecoderPlugin.cxx +++ b/src/decoder/FluidsynthDecoderPlugin.cxx @@ -166,9 +166,8 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs) /* initialization complete - announce the audio format to the MPD core */ - struct audio_format audio_format; - audio_format_init(&audio_format, sample_rate, SAMPLE_FORMAT_S16, 2); - decoder_initialized(decoder, &audio_format, false, -1); + const AudioFormat audio_format(sample_rate, SampleFormat::S16, 2); + decoder_initialized(decoder, audio_format, false, -1); while (fluid_player_get_status(player) == FLUID_PLAYER_PLAYING) { int16_t buffer[2048]; diff --git a/src/decoder/GmeDecoderPlugin.cxx b/src/decoder/GmeDecoderPlugin.cxx index 8158ab553..d8edbe4cb 100644 --- a/src/decoder/GmeDecoderPlugin.cxx +++ b/src/decoder/GmeDecoderPlugin.cxx @@ -153,9 +153,9 @@ gme_file_decode(struct decoder *decoder, const char *path_fs) /* initialize the MPD decoder */ GError *error = nullptr; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, GME_SAMPLE_RATE, - SAMPLE_FORMAT_S16, GME_CHANNELS, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, GME_SAMPLE_RATE, + SampleFormat::S16, GME_CHANNELS, &error)) { g_warning("%s", error->message); g_error_free(error); @@ -164,7 +164,7 @@ gme_file_decode(struct decoder *decoder, const char *path_fs) return; } - decoder_initialized(decoder, &audio_format, true, song_len); + decoder_initialized(decoder, audio_format, true, song_len); gme_err = gme_start_track(emu, song_num); if (gme_err != nullptr) diff --git a/src/decoder/MadDecoderPlugin.cxx b/src/decoder/MadDecoderPlugin.cxx index b75e12343..04d171b9b 100644 --- a/src/decoder/MadDecoderPlugin.cxx +++ b/src/decoder/MadDecoderPlugin.cxx @@ -1124,11 +1124,11 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream) return; } - struct audio_format audio_format; + AudioFormat audio_format; GError *error = nullptr; - if (!audio_format_init_checked(&audio_format, + if (!audio_format_init_checked(audio_format, data.frame.header.samplerate, - SAMPLE_FORMAT_S24_P32, + SampleFormat::S24_P32, MAD_NCHANNELS(&data.frame.header), &error)) { g_warning("%s", error->message); @@ -1138,7 +1138,7 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream) return; } - decoder_initialized(decoder, &audio_format, + decoder_initialized(decoder, audio_format, input_stream_is_seekable(input_stream), data.total_time); diff --git a/src/decoder/MikmodDecoderPlugin.cxx b/src/decoder/MikmodDecoderPlugin.cxx index d332664ee..3aa8a68ed 100644 --- a/src/decoder/MikmodDecoderPlugin.cxx +++ b/src/decoder/MikmodDecoderPlugin.cxx @@ -147,7 +147,6 @@ mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs) { char *path2; MODULE *handle; - struct audio_format audio_format; int ret; SBYTE buffer[MIKMOD_FRAME_SIZE]; enum decoder_command cmd = DECODE_COMMAND_NONE; @@ -164,10 +163,10 @@ mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs) /* Prevent module from looping forever */ handle->loop = 0; - audio_format_init(&audio_format, mikmod_sample_rate, SAMPLE_FORMAT_S16, 2); - assert(audio_format_valid(&audio_format)); + const AudioFormat audio_format(mikmod_sample_rate, SampleFormat::S16, 2); + assert(audio_format.IsValid()); - decoder_initialized(decoder, &audio_format, false, 0); + decoder_initialized(decoder, audio_format, false, 0); Player_Start(handle); while (cmd == DECODE_COMMAND_NONE && Player_Active()) { diff --git a/src/decoder/ModplugDecoderPlugin.cxx b/src/decoder/ModplugDecoderPlugin.cxx index 2ba3b0f49..b95736bf8 100644 --- a/src/decoder/ModplugDecoderPlugin.cxx +++ b/src/decoder/ModplugDecoderPlugin.cxx @@ -94,7 +94,6 @@ mod_decode(struct decoder *decoder, struct input_stream *is) ModPlugFile *f; ModPlug_Settings settings; GByteArray *bdatas; - struct audio_format audio_format; int ret; char audio_buffer[MODPLUG_FRAME_SIZE]; enum decoder_command cmd = DECODE_COMMAND_NONE; @@ -122,10 +121,10 @@ mod_decode(struct decoder *decoder, struct input_stream *is) return; } - audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2); - assert(audio_format_valid(&audio_format)); + static constexpr AudioFormat audio_format(44100, SampleFormat::S16, 2); + assert(audio_format.IsValid()); - decoder_initialized(decoder, &audio_format, + decoder_initialized(decoder, audio_format, input_stream_is_seekable(is), ModPlug_GetLength(f) / 1000.0); diff --git a/src/decoder/MpcdecDecoderPlugin.cxx b/src/decoder/MpcdecDecoderPlugin.cxx index 921d7d923..cfb9c034b 100644 --- a/src/decoder/MpcdecDecoderPlugin.cxx +++ b/src/decoder/MpcdecDecoderPlugin.cxx @@ -154,9 +154,9 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) mpc_demux_get_info(demux, &info); GError *error = nullptr; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, info.sample_freq, - SAMPLE_FORMAT_S24_P32, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, info.sample_freq, + SampleFormat::S24_P32, info.channels, &error)) { g_warning("%s", error->message); g_error_free(error); @@ -173,7 +173,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) decoder_replay_gain(mpd_decoder, &replay_gain_info); - decoder_initialized(mpd_decoder, &audio_format, + decoder_initialized(mpd_decoder, audio_format, input_stream_is_seekable(is), mpc_streaminfo_get_length(&info)); diff --git a/src/decoder/Mpg123DecoderPlugin.cxx b/src/decoder/Mpg123DecoderPlugin.cxx index 73f94ea44..1aac825a2 100644 --- a/src/decoder/Mpg123DecoderPlugin.cxx +++ b/src/decoder/Mpg123DecoderPlugin.cxx @@ -56,7 +56,7 @@ mpd_mpg123_finish(void) */ static bool mpd_mpg123_open(mpg123_handle *handle, const char *path_fs, - struct audio_format *audio_format) + AudioFormat &audio_format) { GError *gerror = nullptr; char *path_dup; @@ -90,7 +90,7 @@ mpd_mpg123_open(mpg123_handle *handle, const char *path_fs, return false; } - if (!audio_format_init_checked(audio_format, rate, SAMPLE_FORMAT_S16, + if (!audio_format_init_checked(audio_format, rate, SampleFormat::S16, channels, &gerror)) { g_warning("%s", gerror->message); g_error_free(gerror); @@ -103,7 +103,6 @@ mpd_mpg123_open(mpg123_handle *handle, const char *path_fs, static void mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs) { - struct audio_format audio_format; mpg123_handle *handle; int error; off_t num_samples; @@ -119,7 +118,8 @@ mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs) return; } - if (!mpd_mpg123_open(handle, path_fs, &audio_format)) { + AudioFormat audio_format; + if (!mpd_mpg123_open(handle, path_fs, audio_format)) { mpg123_delete(handle); return; } @@ -128,7 +128,7 @@ mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs) /* tell MPD core we're ready */ - decoder_initialized(decoder, &audio_format, true, + decoder_initialized(decoder, audio_format, true, (float)num_samples / (float)audio_format.sample_rate); @@ -198,7 +198,6 @@ static bool mpd_mpg123_scan_file(const char *path_fs, const struct tag_handler *handler, void *handler_ctx) { - struct audio_format audio_format; mpg123_handle *handle; int error; off_t num_samples; @@ -210,7 +209,8 @@ mpd_mpg123_scan_file(const char *path_fs, return false; } - if (!mpd_mpg123_open(handle, path_fs, &audio_format)) { + AudioFormat audio_format; + if (!mpd_mpg123_open(handle, path_fs, audio_format)) { mpg123_delete(handle); return false; } diff --git a/src/decoder/OpusDecoderPlugin.cxx b/src/decoder/OpusDecoderPlugin.cxx index 08c67b570..94c687317 100644 --- a/src/decoder/OpusDecoderPlugin.cxx +++ b/src/decoder/OpusDecoderPlugin.cxx @@ -202,11 +202,10 @@ MPDOpusDecoder::HandleBOS(const ogg_packet &packet) return DECODE_COMMAND_STOP; } - struct audio_format audio_format; - audio_format_init(&audio_format, opus_sample_rate, - SAMPLE_FORMAT_S16, channels); - decoder_initialized(decoder, &audio_format, false, -1); - frame_size = audio_format_frame_size(&audio_format); + const AudioFormat audio_format(opus_sample_rate, + SampleFormat::S16, channels); + decoder_initialized(decoder, audio_format, false, -1); + frame_size = audio_format.GetFrameSize(); /* allocate an output buffer for 16 bit PCM samples big enough to hold a quarter second, larger than 120ms required by diff --git a/src/decoder/PcmDecoderPlugin.cxx b/src/decoder/PcmDecoderPlugin.cxx index f64357e68..8976f511f 100644 --- a/src/decoder/PcmDecoderPlugin.cxx +++ b/src/decoder/PcmDecoderPlugin.cxx @@ -36,9 +36,9 @@ extern "C" { static void pcm_stream_decode(struct decoder *decoder, struct input_stream *is) { - static constexpr struct audio_format audio_format = { + static constexpr AudioFormat audio_format = { 44100, - SAMPLE_FORMAT_S16, + SampleFormat::S16, 2, }; @@ -49,14 +49,14 @@ pcm_stream_decode(struct decoder *decoder, struct input_stream *is) GError *error = nullptr; enum decoder_command cmd; - double time_to_size = audio_format_time_to_size(&audio_format); + const double time_to_size = audio_format.GetTimeToSize(); float total_time = -1; const goffset size = input_stream_get_size(is); if (size >= 0) total_time = size / time_to_size; - decoder_initialized(decoder, &audio_format, + decoder_initialized(decoder, audio_format, input_stream_is_seekable(is), total_time); do { diff --git a/src/decoder/SndfileDecoderPlugin.cxx b/src/decoder/SndfileDecoderPlugin.cxx index b1bb97538..63401a47b 100644 --- a/src/decoder/SndfileDecoderPlugin.cxx +++ b/src/decoder/SndfileDecoderPlugin.cxx @@ -99,7 +99,7 @@ static SF_VIRTUAL_IO vio = { * Converts a frame number to a timestamp (in seconds). */ static float -frame_to_time(sf_count_t frame, const struct audio_format *audio_format) +frame_to_time(sf_count_t frame, const AudioFormat *audio_format) { return (float)frame / (float)audio_format->sample_rate; } @@ -108,7 +108,7 @@ frame_to_time(sf_count_t frame, const struct audio_format *audio_format) * Converts a timestamp (in seconds) to a frame number. */ static sf_count_t -time_to_frame(float t, const struct audio_format *audio_format) +time_to_frame(float t, const AudioFormat *audio_format) { return (sf_count_t)(t * audio_format->sample_rate); } @@ -119,7 +119,6 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is) GError *error = nullptr; SNDFILE *sf; SF_INFO info; - struct audio_format audio_format; size_t frame_size; sf_count_t read_frames, num_frames; int buffer[4096]; @@ -136,18 +135,19 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is) /* for now, always read 32 bit samples. Later, we could lower MPD's CPU usage by reading 16 bit samples with sf_readf_short() on low-quality source files. */ - if (!audio_format_init_checked(&audio_format, info.samplerate, - SAMPLE_FORMAT_S32, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, info.samplerate, + SampleFormat::S32, info.channels, &error)) { g_warning("%s", error->message); g_error_free(error); return; } - decoder_initialized(decoder, &audio_format, info.seekable, + decoder_initialized(decoder, audio_format, info.seekable, frame_to_time(info.frames, &audio_format)); - frame_size = audio_format_frame_size(&audio_format); + frame_size = audio_format.GetFrameSize(); read_frames = sizeof(buffer) / frame_size; do { diff --git a/src/decoder/VorbisDecoderPlugin.cxx b/src/decoder/VorbisDecoderPlugin.cxx index 68d5a21f0..f51480d71 100644 --- a/src/decoder/VorbisDecoderPlugin.cxx +++ b/src/decoder/VorbisDecoderPlugin.cxx @@ -202,12 +202,12 @@ vorbis_stream_decode(struct decoder *decoder, return; } - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, vi->rate, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, vi->rate, #ifdef HAVE_TREMOR - SAMPLE_FORMAT_S16, + SampleFormat::S16, #else - SAMPLE_FORMAT_FLOAT, + SampleFormat::FLOAT, #endif vi->channels, &error)) { g_warning("%s", error->message); @@ -219,7 +219,7 @@ vorbis_stream_decode(struct decoder *decoder, if (total_time < 0) total_time = 0; - decoder_initialized(decoder, &audio_format, vis.seekable, total_time); + decoder_initialized(decoder, audio_format, vis.seekable, total_time); enum decoder_command cmd = decoder_get_command(decoder); diff --git a/src/decoder/WavpackDecoderPlugin.cxx b/src/decoder/WavpackDecoderPlugin.cxx index 1a31b7aac..aa62a0f67 100644 --- a/src/decoder/WavpackDecoderPlugin.cxx +++ b/src/decoder/WavpackDecoderPlugin.cxx @@ -106,27 +106,27 @@ format_samples_float(G_GNUC_UNUSED int bytes_per_sample, void *buffer, /** * Choose a MPD sample format from libwavpacks' number of bits. */ -static enum sample_format +static SampleFormat wavpack_bits_to_sample_format(bool is_float, int bytes_per_sample) { if (is_float) - return SAMPLE_FORMAT_FLOAT; + return SampleFormat::FLOAT; switch (bytes_per_sample) { case 1: - return SAMPLE_FORMAT_S8; + return SampleFormat::S8; case 2: - return SAMPLE_FORMAT_S16; + return SampleFormat::S16; case 3: - return SAMPLE_FORMAT_S24_P32; + return SampleFormat::S24_P32; case 4: - return SAMPLE_FORMAT_S32; + return SampleFormat::S32; default: - return SAMPLE_FORMAT_UNDEFINED; + return SampleFormat::UNDEFINED; } } @@ -139,8 +139,8 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek) { GError *error = NULL; bool is_float; - enum sample_format sample_format; - struct audio_format audio_format; + SampleFormat sample_format; + AudioFormat audio_format; format_samples_t format_samples; float total_time; int bytes_per_sample, output_sample_size; @@ -150,7 +150,7 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek) wavpack_bits_to_sample_format(is_float, WavpackGetBytesPerSample(wpc)); - if (!audio_format_init_checked(&audio_format, + if (!audio_format_init_checked(audio_format, WavpackGetSampleRate(wpc), sample_format, WavpackGetNumChannels(wpc), &error)) { @@ -168,14 +168,14 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek) total_time = WavpackGetNumSamples(wpc); total_time /= audio_format.sample_rate; bytes_per_sample = WavpackGetBytesPerSample(wpc); - output_sample_size = audio_format_frame_size(&audio_format); + output_sample_size = audio_format.GetFrameSize(); /* wavpack gives us all kind of samples in a 32-bit space */ int32_t chunk[1024]; const uint32_t samples_requested = G_N_ELEMENTS(chunk) / audio_format.channels; - decoder_initialized(decoder, &audio_format, can_seek, total_time); + decoder_initialized(decoder, audio_format, can_seek, total_time); enum decoder_command cmd = decoder_get_command(decoder); while (cmd != DECODE_COMMAND_STOP) { diff --git a/src/decoder/WildmidiDecoderPlugin.cxx b/src/decoder/WildmidiDecoderPlugin.cxx index 721229f87..832cabe76 100644 --- a/src/decoder/WildmidiDecoderPlugin.cxx +++ b/src/decoder/WildmidiDecoderPlugin.cxx @@ -60,9 +60,9 @@ wildmidi_finish(void) static void wildmidi_file_decode(struct decoder *decoder, const char *path_fs) { - static const struct audio_format audio_format = { + static constexpr AudioFormat audio_format = { WILDMIDI_SAMPLE_RATE, - SAMPLE_FORMAT_S16, + SampleFormat::S16, 2, }; midi *wm; @@ -79,7 +79,7 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs) return; } - decoder_initialized(decoder, &audio_format, true, + decoder_initialized(decoder, audio_format, true, info->approx_total_samples / WILDMIDI_SAMPLE_RATE); do { diff --git a/src/decoder/sidplay_decoder_plugin.cxx b/src/decoder/sidplay_decoder_plugin.cxx index cfe82cf57..d63dca6af 100644 --- a/src/decoder/sidplay_decoder_plugin.cxx +++ b/src/decoder/sidplay_decoder_plugin.cxx @@ -285,11 +285,10 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) /* initialize the MPD decoder */ - struct audio_format audio_format; - audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, channels); - assert(audio_format_valid(&audio_format)); + const AudioFormat audio_format(48000, SampleFormat::S16, channels); + assert(audio_format.IsValid()); - decoder_initialized(decoder, &audio_format, true, (float)song_len); + decoder_initialized(decoder, audio_format, true, (float)song_len); /* .. and play */ diff --git a/src/encoder/FlacEncoderPlugin.cxx b/src/encoder/FlacEncoderPlugin.cxx index 3aeb96cf7..5e7e7f526 100644 --- a/src/encoder/FlacEncoderPlugin.cxx +++ b/src/encoder/FlacEncoderPlugin.cxx @@ -20,7 +20,7 @@ #include "config.h" #include "FlacEncoderPlugin.hxx" #include "EncoderAPI.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "pcm/PcmBuffer.hxx" #include "util/fifo_buffer.h" @@ -40,7 +40,7 @@ extern "C" { struct flac_encoder { Encoder encoder; - struct audio_format audio_format; + AudioFormat audio_format; unsigned compression; FLAC__StreamEncoder *fse; @@ -160,31 +160,31 @@ flac_encoder_close(Encoder *_encoder) } static bool -flac_encoder_open(Encoder *_encoder, struct audio_format *audio_format, +flac_encoder_open(Encoder *_encoder, AudioFormat &audio_format, GError **error) { struct flac_encoder *encoder = (struct flac_encoder *)_encoder; unsigned bits_per_sample; - encoder->audio_format = *audio_format; + encoder->audio_format = audio_format; /* FIXME: flac should support 32bit as well */ - switch (audio_format->format) { - case SAMPLE_FORMAT_S8: + switch (audio_format.format) { + case SampleFormat::S8: bits_per_sample = 8; break; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: bits_per_sample = 16; break; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: bits_per_sample = 24; break; default: bits_per_sample = 24; - audio_format->format = SAMPLE_FORMAT_S24_P32; + audio_format.format = SampleFormat::S24_P32; } /* allocate the encoder */ @@ -263,30 +263,33 @@ flac_encoder_write(Encoder *_encoder, /* format conversion */ - num_frames = length / audio_format_frame_size(&encoder->audio_format); + num_frames = length / encoder->audio_format.GetFrameSize(); num_samples = num_frames * encoder->audio_format.channels; switch (encoder->audio_format.format) { - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: exbuffer = encoder->expand_buffer.Get(length * 4); pcm8_to_flac((int32_t *)exbuffer, (const int8_t *)data, num_samples); buffer = exbuffer; break; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: exbuffer = encoder->expand_buffer.Get(length * 2); pcm16_to_flac((int32_t *)exbuffer, (const int16_t *)data, num_samples); buffer = exbuffer; break; - case SAMPLE_FORMAT_S24_P32: - case SAMPLE_FORMAT_S32: + case SampleFormat::S24_P32: + case SampleFormat::S32: /* nothing need to be done; format is the same for both mpd and libFLAC */ buffer = data; break; + + default: + gcc_unreachable(); } /* feed samples to encoder */ diff --git a/src/encoder/LameEncoderPlugin.cxx b/src/encoder/LameEncoderPlugin.cxx index 933fa3ff2..ea0b91380 100644 --- a/src/encoder/LameEncoderPlugin.cxx +++ b/src/encoder/LameEncoderPlugin.cxx @@ -20,7 +20,7 @@ #include "config.h" #include "LameEncoderPlugin.hxx" #include "EncoderAPI.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -32,7 +32,7 @@ struct LameEncoder final { Encoder encoder; - struct audio_format audio_format; + AudioFormat audio_format; float quality; int bitrate; @@ -187,15 +187,15 @@ lame_encoder_setup(LameEncoder *encoder, GError **error) } static bool -lame_encoder_open(Encoder *_encoder, struct audio_format *audio_format, +lame_encoder_open(Encoder *_encoder, AudioFormat &audio_format, GError **error) { LameEncoder *encoder = (LameEncoder *)_encoder; - audio_format->format = SAMPLE_FORMAT_S16; - audio_format->channels = 2; + audio_format.format = SampleFormat::S16; + audio_format.channels = 2; - encoder->audio_format = *audio_format; + encoder->audio_format = audio_format; encoder->gfp = lame_init(); if (encoder->gfp == nullptr) { @@ -233,7 +233,7 @@ lame_encoder_write(Encoder *_encoder, assert(encoder->buffer_length == 0); const unsigned num_frames = - length / audio_format_frame_size(&encoder->audio_format); + length / encoder->audio_format.GetFrameSize(); float *left = g_new(float, num_frames); float *right = g_new(float, num_frames); diff --git a/src/encoder/NullEncoderPlugin.cxx b/src/encoder/NullEncoderPlugin.cxx index 39e391063..206d55c2f 100644 --- a/src/encoder/NullEncoderPlugin.cxx +++ b/src/encoder/NullEncoderPlugin.cxx @@ -66,7 +66,7 @@ null_encoder_close(Encoder *_encoder) static bool null_encoder_open(Encoder *_encoder, - gcc_unused struct audio_format *audio_format, + gcc_unused AudioFormat &audio_format, gcc_unused GError **error) { NullEncoder *encoder = (NullEncoder *)_encoder; diff --git a/src/encoder/OpusEncoderPlugin.cxx b/src/encoder/OpusEncoderPlugin.cxx index a5947e4b8..a6f36f7d5 100644 --- a/src/encoder/OpusEncoderPlugin.cxx +++ b/src/encoder/OpusEncoderPlugin.cxx @@ -21,7 +21,7 @@ #include "OpusEncoderPlugin.hxx" #include "OggStream.hxx" #include "EncoderAPI.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "mpd_error.h" #include @@ -44,7 +44,7 @@ struct opus_encoder { /* runtime information */ - struct audio_format audio_format; + AudioFormat audio_format; size_t frame_size; @@ -144,37 +144,37 @@ opus_encoder_finish(Encoder *_encoder) static bool opus_encoder_open(Encoder *_encoder, - struct audio_format *audio_format, + AudioFormat &audio_format, GError **error_r) { struct opus_encoder *encoder = (struct opus_encoder *)_encoder; /* libopus supports only 48 kHz */ - audio_format->sample_rate = 48000; + audio_format.sample_rate = 48000; - if (audio_format->channels > 2) - audio_format->channels = 1; + if (audio_format.channels > 2) + audio_format.channels = 1; - switch ((enum sample_format)audio_format->format) { - case SAMPLE_FORMAT_S16: - case SAMPLE_FORMAT_FLOAT: + switch (audio_format.format) { + case SampleFormat::S16: + case SampleFormat::FLOAT: break; - case SAMPLE_FORMAT_S8: - audio_format->format = SAMPLE_FORMAT_S16; + case SampleFormat::S8: + audio_format.format = SampleFormat::S16; break; default: - audio_format->format = SAMPLE_FORMAT_FLOAT; + audio_format.format = SampleFormat::FLOAT; break; } - encoder->audio_format = *audio_format; - encoder->frame_size = audio_format_frame_size(audio_format); + encoder->audio_format = audio_format; + encoder->frame_size = audio_format.GetFrameSize(); int error; - encoder->enc = opus_encoder_create(audio_format->sample_rate, - audio_format->channels, + encoder->enc = opus_encoder_create(audio_format.sample_rate, + audio_format.channels, OPUS_APPLICATION_AUDIO, &error); if (encoder->enc == nullptr) { @@ -190,7 +190,7 @@ opus_encoder_open(Encoder *_encoder, opus_encoder_ctl(encoder->enc, OPUS_GET_LOOKAHEAD(&encoder->lookahead)); - encoder->buffer_frames = audio_format->sample_rate / 50; + encoder->buffer_frames = audio_format.sample_rate / 50; encoder->buffer_size = encoder->frame_size * encoder->buffer_frames; encoder->buffer_position = 0; encoder->buffer = (unsigned char *)g_malloc(encoder->buffer_size); @@ -218,7 +218,7 @@ opus_encoder_do_encode(struct opus_encoder *encoder, bool eos, assert(encoder->buffer_position == encoder->buffer_size); opus_int32 result = - encoder->audio_format.format == SAMPLE_FORMAT_S16 + encoder->audio_format.format == SampleFormat::S16 ? opus_encode(encoder->enc, (const opus_int16 *)encoder->buffer, encoder->buffer_frames, diff --git a/src/encoder/TwolameEncoderPlugin.cxx b/src/encoder/TwolameEncoderPlugin.cxx index c307b7b4f..4e2d47b63 100644 --- a/src/encoder/TwolameEncoderPlugin.cxx +++ b/src/encoder/TwolameEncoderPlugin.cxx @@ -20,7 +20,7 @@ #include "config.h" #include "TwolameEncoderPlugin.hxx" #include "EncoderAPI.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -32,7 +32,7 @@ struct TwolameEncoder final { Encoder encoder; - struct audio_format audio_format; + AudioFormat audio_format; float quality; int bitrate; @@ -187,15 +187,15 @@ twolame_encoder_setup(TwolameEncoder *encoder, GError **error) } static bool -twolame_encoder_open(Encoder *_encoder, struct audio_format *audio_format, +twolame_encoder_open(Encoder *_encoder, AudioFormat &audio_format, GError **error) { TwolameEncoder *encoder = (TwolameEncoder *)_encoder; - audio_format->format = SAMPLE_FORMAT_S16; - audio_format->channels = 2; + audio_format.format = SampleFormat::S16; + audio_format.channels = 2; - encoder->audio_format = *audio_format; + encoder->audio_format = audio_format; encoder->options = twolame_init(); if (encoder->options == nullptr) { @@ -243,7 +243,7 @@ twolame_encoder_write(Encoder *_encoder, assert(encoder->buffer_length == 0); const unsigned num_frames = - length / audio_format_frame_size(&encoder->audio_format); + length / encoder->audio_format.GetFrameSize(); int bytes_out = twolame_encode_buffer_interleaved(encoder->options, src, num_frames, diff --git a/src/encoder/VorbisEncoderPlugin.cxx b/src/encoder/VorbisEncoderPlugin.cxx index 1fc9bde67..bc43ffa43 100644 --- a/src/encoder/VorbisEncoderPlugin.cxx +++ b/src/encoder/VorbisEncoderPlugin.cxx @@ -22,7 +22,7 @@ #include "OggStream.hxx" #include "EncoderAPI.hxx" #include "Tag.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "mpd_error.h" #include @@ -43,7 +43,7 @@ struct vorbis_encoder { /* runtime information */ - struct audio_format audio_format; + AudioFormat audio_format; vorbis_dsp_state vd; vorbis_block vb; @@ -202,14 +202,14 @@ vorbis_encoder_send_header(struct vorbis_encoder *encoder) static bool vorbis_encoder_open(Encoder *_encoder, - struct audio_format *audio_format, + AudioFormat &audio_format, GError **error) { struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder; - audio_format->format = SAMPLE_FORMAT_FLOAT; + audio_format.format = SampleFormat::FLOAT; - encoder->audio_format = *audio_format; + encoder->audio_format = audio_format; if (!vorbis_encoder_reinit(encoder, error)) return false; @@ -328,8 +328,7 @@ vorbis_encoder_write(Encoder *_encoder, { struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder; - unsigned num_frames = length - / audio_format_frame_size(&encoder->audio_format); + unsigned num_frames = length / encoder->audio_format.GetFrameSize(); /* this is for only 16-bit audio */ diff --git a/src/encoder/WaveEncoderPlugin.cxx b/src/encoder/WaveEncoderPlugin.cxx index 939012d89..cc94247cb 100644 --- a/src/encoder/WaveEncoderPlugin.cxx +++ b/src/encoder/WaveEncoderPlugin.cxx @@ -100,32 +100,32 @@ wave_encoder_finish(Encoder *_encoder) static bool wave_encoder_open(Encoder *_encoder, - gcc_unused struct audio_format *audio_format, + AudioFormat &audio_format, gcc_unused GError **error) { WaveEncoder *encoder = (WaveEncoder *)_encoder; - assert(audio_format_valid(audio_format)); + assert(audio_format.IsValid()); - switch (audio_format->format) { - case SAMPLE_FORMAT_S8: + switch (audio_format.format) { + case SampleFormat::S8: encoder->bits = 8; break; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: encoder->bits = 16; break; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: encoder->bits = 24; break; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: encoder->bits = 32; break; default: - audio_format->format = SAMPLE_FORMAT_S16; + audio_format.format = SampleFormat::S16; encoder->bits = 16; break; } @@ -136,10 +136,10 @@ wave_encoder_open(Encoder *_encoder, /* create PCM wave header in initial buffer */ fill_wave_header(header, - audio_format->channels, + audio_format.channels, encoder->bits, - audio_format->sample_rate, - (encoder->bits / 8) * audio_format->channels ); + audio_format.sample_rate, + (encoder->bits / 8) * audio_format.channels); fifo_buffer_append(encoder->buffer, sizeof(*header)); return true; diff --git a/src/filter/AutoConvertFilterPlugin.cxx b/src/filter/AutoConvertFilterPlugin.cxx index 55ee46948..1ad42c2ab 100644 --- a/src/filter/AutoConvertFilterPlugin.cxx +++ b/src/filter/AutoConvertFilterPlugin.cxx @@ -23,18 +23,11 @@ #include "FilterPlugin.hxx" #include "FilterInternal.hxx" #include "FilterRegistry.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include class AutoConvertFilter final : public Filter { - /** - * The audio format being fed to the underlying filter. This - * plugin actually doesn't need this variable, we have it here - * just so our open() method doesn't return a stack pointer. - */ - audio_format child_audio_format; - /** * The underlying filter. */ @@ -52,46 +45,45 @@ public: delete filter; } - virtual const audio_format *Open(audio_format &af, GError **error_r); + virtual AudioFormat Open(AudioFormat &af, GError **error_r); virtual void Close(); virtual const void *FilterPCM(const void *src, size_t src_size, size_t *dest_size_r, GError **error_r); }; -const struct audio_format * -AutoConvertFilter::Open(audio_format &in_audio_format, GError **error_r) +AudioFormat +AutoConvertFilter::Open(AudioFormat &in_audio_format, GError **error_r) { - assert(audio_format_valid(&in_audio_format)); + assert(in_audio_format.IsValid()); /* open the "real" filter */ - child_audio_format = in_audio_format; - const audio_format *out_audio_format = - filter->Open(child_audio_format, error_r); - if (out_audio_format == nullptr) - return nullptr; + const AudioFormat child_audio_format = in_audio_format; + AudioFormat out_audio_format = filter->Open(in_audio_format, error_r); + if (!out_audio_format.IsDefined()) + return out_audio_format; /* need to convert? */ - if (!audio_format_equals(&child_audio_format, &in_audio_format)) { + if (in_audio_format != child_audio_format) { /* yes - create a convert_filter */ convert = filter_new(&convert_filter_plugin, nullptr, error_r); if (convert == nullptr) { filter->Close(); - return nullptr; + return AudioFormat::Undefined(); } - audio_format audio_format2 = in_audio_format; - const audio_format *audio_format3 = + AudioFormat audio_format2 = in_audio_format; + AudioFormat audio_format3 = convert->Open(audio_format2, error_r); - if (audio_format3 == nullptr) { + if (!audio_format3.IsDefined()) { delete convert; filter->Close(); - return nullptr; + return AudioFormat::Undefined(); } - assert(audio_format_equals(&audio_format2, &in_audio_format)); + assert(audio_format2 == in_audio_format); convert_filter_set(convert, child_audio_format); } else diff --git a/src/filter/ChainFilterPlugin.cxx b/src/filter/ChainFilterPlugin.cxx index ac0f10980..2d7fdf938 100644 --- a/src/filter/ChainFilterPlugin.cxx +++ b/src/filter/ChainFilterPlugin.cxx @@ -23,7 +23,7 @@ #include "FilterPlugin.hxx" #include "FilterInternal.hxx" #include "FilterRegistry.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -53,7 +53,7 @@ public: children.emplace_back(name, filter); } - virtual const audio_format *Open(audio_format &af, GError **error_r); + virtual AudioFormat Open(AudioFormat &af, GError **error_r) override; virtual void Close(); virtual const void *FilterPCM(const void *src, size_t src_size, size_t *dest_size_r, GError **error_r); @@ -96,43 +96,43 @@ ChainFilter::CloseUntil(const Filter *until) gcc_unreachable(); } -static const struct audio_format * +static AudioFormat chain_open_child(const char *name, Filter *filter, - const audio_format &prev_audio_format, + const AudioFormat &prev_audio_format, GError **error_r) { - audio_format conv_audio_format = prev_audio_format; - const audio_format *next_audio_format = + AudioFormat conv_audio_format = prev_audio_format; + const AudioFormat next_audio_format = filter->Open(conv_audio_format, error_r); - if (next_audio_format == NULL) - return NULL; + if (!next_audio_format.IsDefined()) + return next_audio_format; - if (!audio_format_equals(&conv_audio_format, &prev_audio_format)) { + if (conv_audio_format != prev_audio_format) { struct audio_format_string s; filter->Close(); g_set_error(error_r, filter_quark(), 0, "Audio format not supported by filter '%s': %s", name, - audio_format_to_string(&prev_audio_format, &s)); - return NULL; + audio_format_to_string(prev_audio_format, &s)); + return AudioFormat::Undefined(); } return next_audio_format; } -const audio_format * -ChainFilter::Open(audio_format &in_audio_format, GError **error_r) +AudioFormat +ChainFilter::Open(AudioFormat &in_audio_format, GError **error_r) { - const audio_format *audio_format = &in_audio_format; + AudioFormat audio_format = in_audio_format; for (auto &child : children) { audio_format = chain_open_child(child.name, child.filter, - *audio_format, error_r); - if (audio_format == NULL) { + audio_format, error_r); + if (!audio_format.IsDefined()) { /* rollback, close all children */ CloseUntil(child.filter); - return NULL; + break; } } diff --git a/src/filter/ConvertFilterPlugin.cxx b/src/filter/ConvertFilterPlugin.cxx index 09a2c9848..f98184489 100644 --- a/src/filter/ConvertFilterPlugin.cxx +++ b/src/filter/ConvertFilterPlugin.cxx @@ -25,7 +25,7 @@ #include "conf.h" #include "pcm/PcmConvert.hxx" #include "util/Manual.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "poison.h" #include @@ -36,27 +36,27 @@ class ConvertFilter final : public Filter { * The input audio format; PCM data is passed to the filter() * method in this format. */ - audio_format in_audio_format; + AudioFormat in_audio_format; /** * The output audio format; the consumer of this plugin * expects PCM data in this format. This defaults to * #in_audio_format, and can be set with convert_filter_set(). */ - audio_format out_audio_format; + AudioFormat out_audio_format; Manual state; public: - void Set(const audio_format &_out_audio_format) { - assert(audio_format_valid(&in_audio_format)); - assert(audio_format_valid(&out_audio_format)); - assert(audio_format_valid(&_out_audio_format)); + void Set(const AudioFormat &_out_audio_format) { + assert(in_audio_format.IsValid()); + assert(out_audio_format.IsValid()); + assert(_out_audio_format.IsValid()); out_audio_format = _out_audio_format; } - virtual const audio_format *Open(audio_format &af, GError **error_r); + virtual AudioFormat Open(AudioFormat &af, GError **error_r) override; virtual void Close(); virtual const void *FilterPCM(const void *src, size_t src_size, size_t *dest_size_r, GError **error_r); @@ -69,15 +69,15 @@ convert_filter_init(gcc_unused const struct config_param *param, return new ConvertFilter(); } -const struct audio_format * -ConvertFilter::Open(audio_format &audio_format, gcc_unused GError **error_r) +AudioFormat +ConvertFilter::Open(AudioFormat &audio_format, gcc_unused GError **error_r) { - assert(audio_format_valid(&audio_format)); + assert(audio_format.IsValid()); in_audio_format = out_audio_format = audio_format; state.Construct(); - return &in_audio_format; + return in_audio_format; } void @@ -93,15 +93,15 @@ const void * ConvertFilter::FilterPCM(const void *src, size_t src_size, size_t *dest_size_r, GError **error_r) { - if (audio_format_equals(&in_audio_format, &out_audio_format)) { + if (in_audio_format == out_audio_format) { /* optimized special case: no-op */ *dest_size_r = src_size; return src; } - return state->Convert(&in_audio_format, + return state->Convert(in_audio_format, src, src_size, - &out_audio_format, dest_size_r, + out_audio_format, dest_size_r, error_r); } @@ -111,7 +111,7 @@ const struct filter_plugin convert_filter_plugin = { }; void -convert_filter_set(Filter *_filter, const audio_format &out_audio_format) +convert_filter_set(Filter *_filter, const AudioFormat out_audio_format) { ConvertFilter *filter = (ConvertFilter *)_filter; diff --git a/src/filter/ConvertFilterPlugin.hxx b/src/filter/ConvertFilterPlugin.hxx index 840bf496f..c814aaf49 100644 --- a/src/filter/ConvertFilterPlugin.hxx +++ b/src/filter/ConvertFilterPlugin.hxx @@ -21,7 +21,7 @@ #define MPD_CONVERT_FILTER_PLUGIN_HXX class Filter; -struct audio_format; +struct AudioFormat; /** * Sets the output audio format for the specified filter. You must @@ -30,6 +30,6 @@ struct audio_format; * the last in a chain. */ void -convert_filter_set(Filter *filter, const audio_format &out_audio_format); +convert_filter_set(Filter *filter, AudioFormat out_audio_format); #endif diff --git a/src/filter/NormalizeFilterPlugin.cxx b/src/filter/NormalizeFilterPlugin.cxx index f4e2963cc..31bcabd36 100644 --- a/src/filter/NormalizeFilterPlugin.cxx +++ b/src/filter/NormalizeFilterPlugin.cxx @@ -22,7 +22,7 @@ #include "FilterInternal.hxx" #include "FilterRegistry.hxx" #include "pcm/PcmBuffer.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "AudioCompress/compress.h" #include @@ -34,7 +34,7 @@ class NormalizeFilter final : public Filter { PcmBuffer buffer; public: - virtual const audio_format *Open(audio_format &af, GError **error_r); + virtual AudioFormat Open(AudioFormat &af, GError **error_r) override; virtual void Close(); virtual const void *FilterPCM(const void *src, size_t src_size, size_t *dest_size_r, GError **error_r); @@ -47,14 +47,14 @@ normalize_filter_init(gcc_unused const struct config_param *param, return new NormalizeFilter(); } -const struct audio_format * -NormalizeFilter::Open(audio_format &audio_format, gcc_unused GError **error_r) +AudioFormat +NormalizeFilter::Open(AudioFormat &audio_format, gcc_unused GError **error_r) { - audio_format.format = SAMPLE_FORMAT_S16; + audio_format.format = SampleFormat::S16; compressor = Compressor_new(0); - return &audio_format; + return audio_format; } void diff --git a/src/filter/NullFilterPlugin.cxx b/src/filter/NullFilterPlugin.cxx index d68065a39..3ff78aa6f 100644 --- a/src/filter/NullFilterPlugin.cxx +++ b/src/filter/NullFilterPlugin.cxx @@ -28,13 +28,14 @@ #include "FilterPlugin.hxx" #include "FilterInternal.hxx" #include "FilterRegistry.hxx" +#include "AudioFormat.hxx" #include "gcc.h" class NullFilter final : public Filter { public: - virtual const audio_format *Open(audio_format &af, - gcc_unused GError **error_r) { - return ⁡ + virtual AudioFormat Open(AudioFormat &af, + gcc_unused GError **error_r) { + return af; } virtual void Close() {} diff --git a/src/filter/ReplayGainFilterPlugin.cxx b/src/filter/ReplayGainFilterPlugin.cxx index d736c910f..be6c7ad2b 100644 --- a/src/filter/ReplayGainFilterPlugin.cxx +++ b/src/filter/ReplayGainFilterPlugin.cxx @@ -22,7 +22,7 @@ #include "FilterPlugin.hxx" #include "FilterInternal.hxx" #include "FilterRegistry.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "replay_gain_info.h" #include "replay_gain_config.h" #include "MixerControl.hxx" @@ -66,7 +66,7 @@ class ReplayGainFilter final : public Filter { */ unsigned volume; - struct audio_format format; + AudioFormat format; PcmBuffer buffer; @@ -112,7 +112,7 @@ public: */ void Update(); - virtual const audio_format *Open(audio_format &af, GError **error_r); + virtual AudioFormat Open(AudioFormat &af, GError **error_r) override; virtual void Close(); virtual const void *FilterPCM(const void *src, size_t src_size, size_t *dest_size_r, GError **error_r); @@ -159,12 +159,12 @@ replay_gain_filter_init(gcc_unused const struct config_param *param, return new ReplayGainFilter(); } -const audio_format * -ReplayGainFilter::Open(audio_format &af, gcc_unused GError **error_r) +AudioFormat +ReplayGainFilter::Open(AudioFormat &af, gcc_unused GError **error_r) { format = af; - return &format; + return format; } void @@ -196,7 +196,7 @@ ReplayGainFilter::FilterPCM(const void *src, size_t src_size, memcpy(dest, src, src_size); bool success = pcm_volume(dest, src_size, - sample_format(format.format), + format.format, volume); if (!success) { g_set_error(error_r, replay_gain_quark(), 0, diff --git a/src/filter/RouteFilterPlugin.cxx b/src/filter/RouteFilterPlugin.cxx index 3dc0991f9..4f4ceacde 100644 --- a/src/filter/RouteFilterPlugin.cxx +++ b/src/filter/RouteFilterPlugin.cxx @@ -42,7 +42,7 @@ #include "config.h" #include "conf.h" #include "ConfigQuark.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "CheckAudioFormat.hxx" #include "FilterPlugin.hxx" #include "FilterInternal.hxx" @@ -79,12 +79,12 @@ class RouteFilter final : public Filter { /** * The actual input format of our signal, once opened */ - struct audio_format input_format; + AudioFormat input_format; /** * The decided upon output format, once opened */ - struct audio_format output_format; + AudioFormat output_format; /** * The size, in bytes, of each multichannel frame in the @@ -120,7 +120,7 @@ public: */ bool Configure(const config_param *param, GError **error_r); - virtual const audio_format *Open(audio_format &af, GError **error_r); + virtual AudioFormat Open(AudioFormat &af, GError **error_r) override; virtual void Close(); virtual const void *FilterPCM(const void *src, size_t src_size, size_t *dest_size_r, GError **error_r); @@ -241,12 +241,12 @@ route_filter_init(const config_param *param, GError **error_r) return filter; } -const struct audio_format * -RouteFilter::Open(audio_format &audio_format, gcc_unused GError **error_r) +AudioFormat +RouteFilter::Open(AudioFormat &audio_format, gcc_unused GError **error_r) { // Copy the input format for later reference input_format = audio_format; - input_frame_size = audio_format_frame_size(&input_format); + input_frame_size = input_format.GetFrameSize(); // Decide on an output format which has enough channels, // and is otherwise identical @@ -254,9 +254,9 @@ RouteFilter::Open(audio_format &audio_format, gcc_unused GError **error_r) output_format.channels = min_output_channels; // Precalculate this simple value, to speed up allocation later - output_frame_size = audio_format_frame_size(&output_format); + output_frame_size = output_format.GetFrameSize(); - return &output_format; + return output_format; } void @@ -271,8 +271,7 @@ RouteFilter::FilterPCM(const void *src, size_t src_size, { size_t number_of_frames = src_size / input_frame_size; - size_t bytes_per_frame_per_channel = - audio_format_sample_size(&input_format); + const size_t bytes_per_frame_per_channel = input_format.GetSampleSize(); // A moving pointer that always refers to channel 0 in the input, at the currently handled frame const uint8_t *base_source = (const uint8_t *)src; diff --git a/src/filter/VolumeFilterPlugin.cxx b/src/filter/VolumeFilterPlugin.cxx index 824ad0ab4..3888cefe4 100644 --- a/src/filter/VolumeFilterPlugin.cxx +++ b/src/filter/VolumeFilterPlugin.cxx @@ -25,7 +25,7 @@ #include "conf.h" #include "pcm/PcmVolume.hxx" #include "pcm/PcmBuffer.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include #include @@ -36,7 +36,7 @@ class VolumeFilter final : public Filter { */ unsigned volume; - struct audio_format format; + AudioFormat format; PcmBuffer buffer; @@ -56,7 +56,7 @@ public: volume = _volume; } - virtual const audio_format *Open(audio_format &af, GError **error_r); + virtual AudioFormat Open(AudioFormat &af, GError **error_r) override; virtual void Close(); virtual const void *FilterPCM(const void *src, size_t src_size, size_t *dest_size_r, GError **error_r); @@ -75,12 +75,12 @@ volume_filter_init(gcc_unused const struct config_param *param, return new VolumeFilter(); } -const struct audio_format * -VolumeFilter::Open(audio_format &audio_format, gcc_unused GError **error_r) +AudioFormat +VolumeFilter::Open(AudioFormat &audio_format, gcc_unused GError **error_r) { format = audio_format; - return &format; + return format; } void @@ -112,7 +112,7 @@ VolumeFilter::FilterPCM(const void *src, size_t src_size, memcpy(dest, src, src_size); bool success = pcm_volume(dest, src_size, - sample_format(format.format), + format.format, volume); if (!success) { g_set_error(error_r, volume_quark(), 0, diff --git a/src/output/AlsaOutputPlugin.cxx b/src/output/AlsaOutputPlugin.cxx index 7d0e9e09c..0e70babb6 100644 --- a/src/output/AlsaOutputPlugin.cxx +++ b/src/output/AlsaOutputPlugin.cxx @@ -233,26 +233,26 @@ alsa_test_default_device(void) } static snd_pcm_format_t -get_bitformat(enum sample_format sample_format) +get_bitformat(SampleFormat sample_format) { switch (sample_format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: return SND_PCM_FORMAT_UNKNOWN; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: return SND_PCM_FORMAT_S8; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: return SND_PCM_FORMAT_S16; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: return SND_PCM_FORMAT_S24; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: return SND_PCM_FORMAT_S32; - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: return SND_PCM_FORMAT_FLOAT; } @@ -324,7 +324,7 @@ alsa_try_format_or_packed(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, */ static int alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, - enum sample_format sample_format, + SampleFormat sample_format, bool *packed_r, bool *reverse_endian_r) { snd_pcm_format_t alsa_format = get_bitformat(sample_format); @@ -355,36 +355,36 @@ alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, */ static int alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, - struct audio_format *audio_format, + AudioFormat &audio_format, bool *packed_r, bool *reverse_endian_r) { /* try the input format first */ int err = alsa_output_try_format(pcm, hwparams, - sample_format(audio_format->format), + audio_format.format, packed_r, reverse_endian_r); /* if unsupported by the hardware, try other formats */ - static const enum sample_format probe_formats[] = { - SAMPLE_FORMAT_S24_P32, - SAMPLE_FORMAT_S32, - SAMPLE_FORMAT_S16, - SAMPLE_FORMAT_S8, - SAMPLE_FORMAT_UNDEFINED, + static const SampleFormat probe_formats[] = { + SampleFormat::S24_P32, + SampleFormat::S32, + SampleFormat::S16, + SampleFormat::S8, + SampleFormat::UNDEFINED, }; for (unsigned i = 0; - err == -EINVAL && probe_formats[i] != SAMPLE_FORMAT_UNDEFINED; + err == -EINVAL && probe_formats[i] != SampleFormat::UNDEFINED; ++i) { - const enum sample_format mpd_format = probe_formats[i]; - if (mpd_format == audio_format->format) + const SampleFormat mpd_format = probe_formats[i]; + if (mpd_format == audio_format.format) continue; err = alsa_output_try_format(pcm, hwparams, mpd_format, packed_r, reverse_endian_r); if (err == 0) - audio_format->format = mpd_format; + audio_format.format = mpd_format; } return err; @@ -395,11 +395,11 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, * the configured settings and the audio format. */ static bool -alsa_setup(AlsaOutput *ad, struct audio_format *audio_format, +alsa_setup(AlsaOutput *ad, AudioFormat &audio_format, bool *packed_r, bool *reverse_endian_r, GError **error) { - unsigned int sample_rate = audio_format->sample_rate; - unsigned int channels = audio_format->channels; + unsigned int sample_rate = audio_format.sample_rate; + unsigned int channels = audio_format.channels; int err; const char *cmd = NULL; int retry = MPD_ALSA_RETRY_NR; @@ -443,7 +443,7 @@ configure_hw: g_set_error(error, alsa_output_quark(), err, "ALSA device \"%s\" does not support format %s: %s", alsa_device(ad), - sample_format_to_string(sample_format(audio_format->format)), + sample_format_to_string(audio_format.format), snd_strerror(-err)); return false; } @@ -458,21 +458,21 @@ configure_hw: if (err < 0) { g_set_error(error, alsa_output_quark(), err, "ALSA device \"%s\" does not support %i channels: %s", - alsa_device(ad), (int)audio_format->channels, + alsa_device(ad), (int)audio_format.channels, snd_strerror(-err)); return false; } - audio_format->channels = (int8_t)channels; + audio_format.channels = (int8_t)channels; err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams, &sample_rate, NULL); if (err < 0 || sample_rate == 0) { g_set_error(error, alsa_output_quark(), err, "ALSA device \"%s\" does not support %u Hz audio", - alsa_device(ad), audio_format->sample_rate); + alsa_device(ad), audio_format.sample_rate); return false; } - audio_format->sample_rate = sample_rate; + audio_format.sample_rate = sample_rate; snd_pcm_uframes_t buffer_size_min, buffer_size_max; snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min); @@ -603,22 +603,22 @@ error: } static bool -alsa_setup_dsd(AlsaOutput *ad, struct audio_format *audio_format, +alsa_setup_dsd(AlsaOutput *ad, const AudioFormat audio_format, bool *shift8_r, bool *packed_r, bool *reverse_endian_r, GError **error_r) { assert(ad->dsd_usb); - assert(audio_format->format == SAMPLE_FORMAT_DSD); + assert(audio_format.format == SampleFormat::DSD); /* pass 24 bit to alsa_setup() */ - struct audio_format usb_format = *audio_format; - usb_format.format = SAMPLE_FORMAT_S24_P32; + AudioFormat usb_format = audio_format; + usb_format.format = SampleFormat::S24_P32; usb_format.sample_rate /= 2; - const struct audio_format check = usb_format; + const AudioFormat check = usb_format; - if (!alsa_setup(ad, &usb_format, packed_r, reverse_endian_r, error_r)) + if (!alsa_setup(ad, usb_format, packed_r, reverse_endian_r, error_r)) return false; /* if the device allows only 32 bit, shift all DSD-over-USB @@ -626,11 +626,11 @@ alsa_setup_dsd(AlsaOutput *ad, struct audio_format *audio_format, the DSD-over-USB documentation does not specify whether this is legal, but there is anecdotical evidence that this is possible (and the only option for some devices) */ - *shift8_r = usb_format.format == SAMPLE_FORMAT_S32; - if (usb_format.format == SAMPLE_FORMAT_S32) - usb_format.format = SAMPLE_FORMAT_S24_P32; + *shift8_r = usb_format.format == SampleFormat::S32; + if (usb_format.format == SampleFormat::S32) + usb_format.format = SampleFormat::S24_P32; - if (!audio_format_equals(&usb_format, &check)) { + if (usb_format != check) { /* no bit-perfect playback, which is required for DSD over USB */ g_set_error(error_r, alsa_output_quark(), 0, @@ -644,13 +644,13 @@ alsa_setup_dsd(AlsaOutput *ad, struct audio_format *audio_format, } static bool -alsa_setup_or_dsd(AlsaOutput *ad, struct audio_format *audio_format, +alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format, GError **error_r) { bool shift8 = false, packed, reverse_endian; const bool dsd_usb = ad->dsd_usb && - audio_format->format == SAMPLE_FORMAT_DSD; + audio_format.format == SampleFormat::DSD; const bool success = dsd_usb ? alsa_setup_dsd(ad, audio_format, &shift8, &packed, &reverse_endian, @@ -660,14 +660,14 @@ alsa_setup_or_dsd(AlsaOutput *ad, struct audio_format *audio_format, if (!success) return false; - ad->pcm_export->Open(sample_format(audio_format->format), - audio_format->channels, + ad->pcm_export->Open(audio_format.format, + audio_format.channels, dsd_usb, shift8, packed, reverse_endian); return true; } static bool -alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error) +alsa_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { AlsaOutput *ad = (AlsaOutput *)ao; @@ -688,8 +688,8 @@ alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **e return false; } - ad->in_frame_size = audio_format_frame_size(audio_format); - ad->out_frame_size = ad->pcm_export->GetFrameSize(*audio_format); + ad->in_frame_size = audio_format.GetFrameSize(); + ad->out_frame_size = ad->pcm_export->GetFrameSize(audio_format); return true; } diff --git a/src/output/AoOutputPlugin.cxx b/src/output/AoOutputPlugin.cxx index b534ddf08..d25a30973 100644 --- a/src/output/AoOutputPlugin.cxx +++ b/src/output/AoOutputPlugin.cxx @@ -200,18 +200,18 @@ ao_output_close(struct audio_output *ao) } static bool -ao_output_open(struct audio_output *ao, struct audio_format *audio_format, +ao_output_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { ao_sample_format format = OUR_AO_FORMAT_INITIALIZER; AoOutput *ad = (AoOutput *)ao; - switch (audio_format->format) { - case SAMPLE_FORMAT_S8: + switch (audio_format.format) { + case SampleFormat::S8: format.bits = 8; break; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: format.bits = 16; break; @@ -219,14 +219,14 @@ ao_output_open(struct audio_output *ao, struct audio_format *audio_format, /* support for 24 bit samples in libao is currently dubious, and until we have sorted that out, convert everything to 16 bit */ - audio_format->format = SAMPLE_FORMAT_S16; + audio_format.format = SampleFormat::S16; format.bits = 16; break; } - format.rate = audio_format->sample_rate; + format.rate = audio_format.sample_rate; format.byte_format = AO_FMT_NATIVE; - format.channels = audio_format->channels; + format.channels = audio_format.channels; ad->device = ao_open_live(ad->driver, &format, ad->options); diff --git a/src/output/FifoOutputPlugin.cxx b/src/output/FifoOutputPlugin.cxx index 1151d9f3a..a0d07c702 100644 --- a/src/output/FifoOutputPlugin.cxx +++ b/src/output/FifoOutputPlugin.cxx @@ -227,12 +227,12 @@ fifo_output_finish(struct audio_output *ao) } static bool -fifo_output_open(struct audio_output *ao, struct audio_format *audio_format, +fifo_output_open(struct audio_output *ao, AudioFormat &audio_format, G_GNUC_UNUSED GError **error) { FifoOutput *fd = (FifoOutput *)ao; - fd->timer = new Timer(*audio_format); + fd->timer = new Timer(audio_format); return true; } diff --git a/src/output/HttpdInternal.hxx b/src/output/HttpdInternal.hxx index c2ac96be5..8bc0c130e 100644 --- a/src/output/HttpdInternal.hxx +++ b/src/output/HttpdInternal.hxx @@ -131,13 +131,13 @@ struct HttpdOutput final : private ServerSocket { /** * Caller must lock the mutex. */ - bool OpenEncoder(struct audio_format *audio_format, + bool OpenEncoder(AudioFormat &audio_format, GError **error_r); /** * Caller must lock the mutex. */ - bool Open(struct audio_format *audio_format, GError **error_r); + bool Open(AudioFormat &audio_format, GError **error_r); /** * Caller must lock the mutex. diff --git a/src/output/HttpdOutputPlugin.cxx b/src/output/HttpdOutputPlugin.cxx index 36bd1aee1..462dd3429 100644 --- a/src/output/HttpdOutputPlugin.cxx +++ b/src/output/HttpdOutputPlugin.cxx @@ -291,7 +291,7 @@ httpd_output_disable(struct audio_output *ao) } inline bool -HttpdOutput::OpenEncoder(struct audio_format *audio_format, GError **error) +HttpdOutput::OpenEncoder(AudioFormat &audio_format, GError **error) { if (!encoder_open(encoder, audio_format, error)) return false; @@ -307,7 +307,7 @@ HttpdOutput::OpenEncoder(struct audio_format *audio_format, GError **error) } inline bool -HttpdOutput::Open(struct audio_format *audio_format, GError **error_r) +HttpdOutput::Open(AudioFormat &audio_format, GError **error_r) { assert(!open); assert(clients.empty()); @@ -320,7 +320,7 @@ HttpdOutput::Open(struct audio_format *audio_format, GError **error_r) /* initialize other attributes */ clients_cnt = 0; - timer = new Timer(*audio_format); + timer = new Timer(audio_format); open = true; @@ -328,7 +328,7 @@ HttpdOutput::Open(struct audio_format *audio_format, GError **error_r) } static bool -httpd_output_open(struct audio_output *ao, struct audio_format *audio_format, +httpd_output_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { HttpdOutput *httpd = Cast(ao); diff --git a/src/output/JackOutputPlugin.cxx b/src/output/JackOutputPlugin.cxx index 4f21b36f8..6c9ee9335 100644 --- a/src/output/JackOutputPlugin.cxx +++ b/src/output/JackOutputPlugin.cxx @@ -67,7 +67,7 @@ struct JackOutput { size_t ringbuffer_size; /* the current audio format */ - struct audio_format audio_format; + AudioFormat audio_format; /* jack library stuff */ jack_port_t *ports[MAX_PORTS]; @@ -203,18 +203,18 @@ mpd_jack_shutdown(void *arg) } static void -set_audioformat(JackOutput *jd, struct audio_format *audio_format) +set_audioformat(JackOutput *jd, AudioFormat &audio_format) { - audio_format->sample_rate = jack_get_sample_rate(jd->client); + audio_format.sample_rate = jack_get_sample_rate(jd->client); if (jd->num_source_ports == 1) - audio_format->channels = 1; - else if (audio_format->channels > jd->num_source_ports) - audio_format->channels = 2; + audio_format.channels = 1; + else if (audio_format.channels > jd->num_source_ports) + audio_format.channels = 2; - if (audio_format->format != SAMPLE_FORMAT_S16 && - audio_format->format != SAMPLE_FORMAT_S24_P32) - audio_format->format = SAMPLE_FORMAT_S24_P32; + if (audio_format.format != SampleFormat::S16 && + audio_format.format != SampleFormat::S24_P32) + audio_format.format = SampleFormat::S24_P32; } static void @@ -591,7 +591,7 @@ mpd_jack_start(JackOutput *jd, GError **error_r) } static bool -mpd_jack_open(struct audio_output *ao, struct audio_format *audio_format, +mpd_jack_open(struct audio_output *ao, AudioFormat &audio_format, GError **error_r) { JackOutput *jd = (JackOutput *)ao; @@ -607,7 +607,7 @@ mpd_jack_open(struct audio_output *ao, struct audio_format *audio_format, return false; set_audioformat(jd, audio_format); - jd->audio_format = *audio_format; + jd->audio_format = audio_format; if (!mpd_jack_start(jd, error_r)) return false; @@ -684,12 +684,12 @@ mpd_jack_write_samples(JackOutput *jd, const void *src, unsigned num_samples) { switch (jd->audio_format.format) { - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: mpd_jack_write_samples_16(jd, (const int16_t*)src, num_samples); break; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: mpd_jack_write_samples_24(jd, (const int32_t*)src, num_samples); break; @@ -705,7 +705,7 @@ mpd_jack_play(struct audio_output *ao, const void *chunk, size_t size, GError **error_r) { JackOutput *jd = (JackOutput *)ao; - const size_t frame_size = audio_format_frame_size(&jd->audio_format); + const size_t frame_size = jd->audio_format.GetFrameSize(); size_t space = 0, space1; jd->pause = false; diff --git a/src/output/NullOutputPlugin.cxx b/src/output/NullOutputPlugin.cxx index 07452bbdb..814a9d2e2 100644 --- a/src/output/NullOutputPlugin.cxx +++ b/src/output/NullOutputPlugin.cxx @@ -66,13 +66,13 @@ null_finish(struct audio_output *ao) } static bool -null_open(struct audio_output *ao, struct audio_format *audio_format, +null_open(struct audio_output *ao, AudioFormat &audio_format, gcc_unused GError **error) { NullOutput *nd = (NullOutput *)ao; if (nd->sync) - nd->timer = new Timer(*audio_format); + nd->timer = new Timer(audio_format); return true; } diff --git a/src/output/OSXOutputPlugin.cxx b/src/output/OSXOutputPlugin.cxx index cbd27413f..afe8e064d 100644 --- a/src/output/OSXOutputPlugin.cxx +++ b/src/output/OSXOutputPlugin.cxx @@ -316,30 +316,30 @@ osx_output_close(struct audio_output *ao) } static bool -osx_output_open(struct audio_output *ao, struct audio_format *audio_format, GError **error) +osx_output_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { OSXOutput *od = (OSXOutput *)ao; AudioStreamBasicDescription stream_description; - stream_description.mSampleRate = audio_format->sample_rate; + stream_description.mSampleRate = audio_format.sample_rate; stream_description.mFormatID = kAudioFormatLinearPCM; stream_description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger; - switch (audio_format->format) { - case SAMPLE_FORMAT_S8: + switch (audio_format.format) { + case SampleFormat::S8: stream_description.mBitsPerChannel = 8; break; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: stream_description.mBitsPerChannel = 16; break; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: stream_description.mBitsPerChannel = 32; break; default: - audio_format->format = SAMPLE_FORMAT_S32; + audio_format.format = SampleFormat::S32; stream_description.mBitsPerChannel = 32; break; } @@ -348,11 +348,10 @@ osx_output_open(struct audio_output *ao, struct audio_format *audio_format, GErr stream_description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; #endif - stream_description.mBytesPerPacket = - audio_format_frame_size(audio_format); + stream_description.mBytesPerPacket = audio_format.GetFrameSize(); stream_description.mFramesPerPacket = 1; stream_description.mBytesPerFrame = stream_description.mBytesPerPacket; - stream_description.mChannelsPerFrame = audio_format->channels; + stream_description.mChannelsPerFrame = audio_format.channels; ComponentResult result = AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat, @@ -374,8 +373,8 @@ osx_output_open(struct audio_output *ao, struct audio_format *audio_format, GErr } /* create a buffer of 1s */ - od->buffer = fifo_buffer_new(audio_format->sample_rate * - audio_format_frame_size(audio_format)); + od->buffer = fifo_buffer_new(audio_format.sample_rate * + audio_format.GetFrameSize()); status = AudioOutputUnitStart(od->au); if (status != 0) { diff --git a/src/output/OpenALOutputPlugin.cxx b/src/output/OpenALOutputPlugin.cxx index e68032d75..81c75f6c4 100644 --- a/src/output/OpenALOutputPlugin.cxx +++ b/src/output/OpenALOutputPlugin.cxx @@ -66,26 +66,26 @@ openal_output_quark(void) } static ALenum -openal_audio_format(struct audio_format *audio_format) +openal_audio_format(AudioFormat &audio_format) { - /* note: cannot map SAMPLE_FORMAT_S8 to AL_FORMAT_STEREO8 or + /* note: cannot map SampleFormat::S8 to AL_FORMAT_STEREO8 or AL_FORMAT_MONO8 since OpenAL expects unsigned 8 bit samples, while MPD uses signed samples */ - switch (audio_format->format) { - case SAMPLE_FORMAT_S16: - if (audio_format->channels == 2) + switch (audio_format.format) { + case SampleFormat::S16: + if (audio_format.channels == 2) return AL_FORMAT_STEREO16; - if (audio_format->channels == 1) + if (audio_format.channels == 1) return AL_FORMAT_MONO16; /* fall back to mono */ - audio_format->channels = 1; + audio_format.channels = 1; return openal_audio_format(audio_format); default: /* fall back to 16 bit */ - audio_format->format = SAMPLE_FORMAT_S16; + audio_format.format = SampleFormat::S16; return openal_audio_format(audio_format); } } @@ -169,7 +169,7 @@ openal_finish(struct audio_output *ao) } static bool -openal_open(struct audio_output *ao, struct audio_format *audio_format, +openal_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { OpenALOutput *od = (OpenALOutput *)ao; @@ -199,7 +199,7 @@ openal_open(struct audio_output *ao, struct audio_format *audio_format, } od->filled = 0; - od->frequency = audio_format->sample_rate; + od->frequency = audio_format.sample_rate; return true; } diff --git a/src/output/OssOutputPlugin.cxx b/src/output/OssOutputPlugin.cxx index 70ffd8bfc..49b02386b 100644 --- a/src/output/OssOutputPlugin.cxx +++ b/src/output/OssOutputPlugin.cxx @@ -70,7 +70,7 @@ struct OssOutput { * The current input audio format. This is needed to reopen * the device after cancel(). */ - struct audio_format audio_format; + AudioFormat audio_format; /** * The current OSS audio format. This is needed to reopen the @@ -308,10 +308,10 @@ oss_try_ioctl(int fd, unsigned long request, int value, * specified number is not supported. */ static bool -oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r) +oss_setup_channels(int fd, AudioFormat &audio_format, GError **error_r) { const char *const msg = "Failed to set channel count"; - int channels = audio_format->channels; + int channels = audio_format.channels; enum oss_setup_result result = oss_try_ioctl_r(fd, SNDCTL_DSP_CHANNELS, &channels, msg, error_r); switch (result) { @@ -319,7 +319,7 @@ oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r) if (!audio_valid_channel_count(channels)) break; - audio_format->channels = channels; + audio_format.channels = channels; return true; case ERROR: @@ -330,7 +330,7 @@ oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r) } for (unsigned i = 1; i < 2; ++i) { - if (i == audio_format->channels) + if (i == audio_format.channels) /* don't try that again */ continue; @@ -342,7 +342,7 @@ oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r) if (!audio_valid_channel_count(channels)) break; - audio_format->channels = channels; + audio_format.channels = channels; return true; case ERROR: @@ -362,11 +362,11 @@ oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r) * specified sample rate is not supported. */ static bool -oss_setup_sample_rate(int fd, struct audio_format *audio_format, +oss_setup_sample_rate(int fd, AudioFormat &audio_format, GError **error_r) { const char *const msg = "Failed to set sample rate"; - int sample_rate = audio_format->sample_rate; + int sample_rate = audio_format.sample_rate; enum oss_setup_result result = oss_try_ioctl_r(fd, SNDCTL_DSP_SPEED, &sample_rate, msg, error_r); @@ -375,7 +375,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format, if (!audio_valid_sample_rate(sample_rate)) break; - audio_format->sample_rate = sample_rate; + audio_format.sample_rate = sample_rate; return true; case ERROR: @@ -388,7 +388,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format, static const int sample_rates[] = { 48000, 44100, 0 }; for (unsigned i = 0; sample_rates[i] != 0; ++i) { sample_rate = sample_rates[i]; - if (sample_rate == (int)audio_format->sample_rate) + if (sample_rate == (int)audio_format.sample_rate) continue; result = oss_try_ioctl_r(fd, SNDCTL_DSP_SPEED, &sample_rate, @@ -398,7 +398,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format, if (!audio_valid_sample_rate(sample_rate)) break; - audio_format->sample_rate = sample_rate; + audio_format.sample_rate = sample_rate; return true; case ERROR: @@ -418,28 +418,28 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format, * AFMT_QUERY if there is no direct counterpart. */ static int -sample_format_to_oss(enum sample_format format) +sample_format_to_oss(SampleFormat format) { switch (format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_FLOAT: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::FLOAT: + case SampleFormat::DSD: return AFMT_QUERY; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: return AFMT_S8; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: return AFMT_S16_NE; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: #ifdef AFMT_S24_NE return AFMT_S24_NE; #else return AFMT_QUERY; #endif - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: #ifdef AFMT_S32_NE return AFMT_S32_NE; #else @@ -452,47 +452,47 @@ sample_format_to_oss(enum sample_format format) /** * Convert an OSS sample format to its MPD counterpart. Returns - * SAMPLE_FORMAT_UNDEFINED if there is no direct counterpart. + * SampleFormat::UNDEFINED if there is no direct counterpart. */ -static enum sample_format +static SampleFormat sample_format_from_oss(int format) { switch (format) { case AFMT_S8: - return SAMPLE_FORMAT_S8; + return SampleFormat::S8; case AFMT_S16_NE: - return SAMPLE_FORMAT_S16; + return SampleFormat::S16; #ifdef AFMT_S24_PACKED case AFMT_S24_PACKED: - return SAMPLE_FORMAT_S24_P32; + return SampleFormat::S24_P32; #endif #ifdef AFMT_S24_NE case AFMT_S24_NE: - return SAMPLE_FORMAT_S24_P32; + return SampleFormat::S24_P32; #endif #ifdef AFMT_S32_NE case AFMT_S32_NE: - return SAMPLE_FORMAT_S32; + return SampleFormat::S32; #endif default: - return SAMPLE_FORMAT_UNDEFINED; + return SampleFormat::UNDEFINED; } } /** * Probe one sample format. * - * @return the selected sample format or SAMPLE_FORMAT_UNDEFINED on + * @return the selected sample format or SampleFormat::UNDEFINED on * error */ static enum oss_setup_result -oss_probe_sample_format(int fd, enum sample_format sample_format, - enum sample_format *sample_format_r, +oss_probe_sample_format(int fd, SampleFormat sample_format, + SampleFormat *sample_format_r, int *oss_format_r, #ifdef AFMT_S24_PACKED PcmExport &pcm_export, @@ -509,7 +509,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format, "Failed to set sample format", error_r); #ifdef AFMT_S24_PACKED - if (result == UNSUPPORTED && sample_format == SAMPLE_FORMAT_S24_P32) { + if (result == UNSUPPORTED && sample_format == SampleFormat::S24_P32) { /* if the driver doesn't support padded 24 bit, try packed 24 bit */ oss_format = AFMT_S24_PACKED; @@ -523,7 +523,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format, return result; sample_format = sample_format_from_oss(oss_format); - if (sample_format == SAMPLE_FORMAT_UNDEFINED) + if (sample_format == SampleFormat::UNDEFINED) return UNSUPPORTED; *sample_format_r = sample_format; @@ -544,16 +544,16 @@ oss_probe_sample_format(int fd, enum sample_format sample_format, * specified format is not supported. */ static bool -oss_setup_sample_format(int fd, struct audio_format *audio_format, +oss_setup_sample_format(int fd, AudioFormat &audio_format, int *oss_format_r, #ifdef AFMT_S24_PACKED PcmExport &pcm_export, #endif GError **error_r) { - enum sample_format mpd_format; + SampleFormat mpd_format; enum oss_setup_result result = - oss_probe_sample_format(fd, sample_format(audio_format->format), + oss_probe_sample_format(fd, audio_format.format, &mpd_format, oss_format_r, #ifdef AFMT_S24_PACKED pcm_export, @@ -561,7 +561,7 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format, error_r); switch (result) { case SUCCESS: - audio_format->format = mpd_format; + audio_format.format = mpd_format; return true; case ERROR: @@ -577,17 +577,17 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format, /* the requested sample format is not available - probe for other formats supported by MPD */ - static const enum sample_format sample_formats[] = { - SAMPLE_FORMAT_S24_P32, - SAMPLE_FORMAT_S32, - SAMPLE_FORMAT_S16, - SAMPLE_FORMAT_S8, - SAMPLE_FORMAT_UNDEFINED /* sentinel */ + static const SampleFormat sample_formats[] = { + SampleFormat::S24_P32, + SampleFormat::S32, + SampleFormat::S16, + SampleFormat::S8, + SampleFormat::UNDEFINED /* sentinel */ }; - for (unsigned i = 0; sample_formats[i] != SAMPLE_FORMAT_UNDEFINED; ++i) { + for (unsigned i = 0; sample_formats[i] != SampleFormat::UNDEFINED; ++i) { mpd_format = sample_formats[i]; - if (mpd_format == audio_format->format) + if (mpd_format == audio_format.format) /* don't try that again */ continue; @@ -599,7 +599,7 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format, error_r); switch (result) { case SUCCESS: - audio_format->format = mpd_format; + audio_format.format = mpd_format; return true; case ERROR: @@ -619,7 +619,7 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format, * Sets up the OSS device which was opened before. */ static bool -oss_setup(OssOutput *od, struct audio_format *audio_format, +oss_setup(OssOutput *od, AudioFormat &audio_format, GError **error_r) { return oss_setup_channels(od->fd, audio_format, error_r) && @@ -687,7 +687,7 @@ oss_reopen(OssOutput *od, GError **error_r) } static bool -oss_output_open(struct audio_output *ao, struct audio_format *audio_format, +oss_output_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { OssOutput *od = (OssOutput *)ao; @@ -705,7 +705,7 @@ oss_output_open(struct audio_output *ao, struct audio_format *audio_format, return false; } - od->audio_format = *audio_format; + od->audio_format = audio_format; return true; } diff --git a/src/output/PipeOutputPlugin.cxx b/src/output/PipeOutputPlugin.cxx index 6e72eff12..f76dd04c1 100644 --- a/src/output/PipeOutputPlugin.cxx +++ b/src/output/PipeOutputPlugin.cxx @@ -95,7 +95,7 @@ pipe_output_finish(struct audio_output *ao) static bool pipe_output_open(struct audio_output *ao, - G_GNUC_UNUSED struct audio_format *audio_format, + G_GNUC_UNUSED AudioFormat &audio_format, G_GNUC_UNUSED GError **error) { PipeOutput *pd = (PipeOutput *)ao; diff --git a/src/output/PulseOutputPlugin.cxx b/src/output/PulseOutputPlugin.cxx index 5098e2adc..65526c8fb 100644 --- a/src/output/PulseOutputPlugin.cxx +++ b/src/output/PulseOutputPlugin.cxx @@ -578,7 +578,7 @@ pulse_output_setup_stream(PulseOutput *po, const pa_sample_spec *ss, } static bool -pulse_output_open(struct audio_output *ao, struct audio_format *audio_format, +pulse_output_open(struct audio_output *ao, AudioFormat &audio_format, GError **error_r) { PulseOutput *po = (PulseOutput *)ao; @@ -615,11 +615,11 @@ pulse_output_open(struct audio_output *ao, struct audio_format *audio_format, /* MPD doesn't support the other pulseaudio sample formats, so we just force MPD to send us everything as 16 bit */ - audio_format->format = SAMPLE_FORMAT_S16; + audio_format.format = SampleFormat::S16; ss.format = PA_SAMPLE_S16NE; - ss.rate = audio_format->sample_rate; - ss.channels = audio_format->channels; + ss.rate = audio_format.sample_rate; + ss.channels = audio_format.channels; /* create a stream .. */ diff --git a/src/output/RecorderOutputPlugin.cxx b/src/output/RecorderOutputPlugin.cxx index 6b31fcbc6..2e77463f0 100644 --- a/src/output/RecorderOutputPlugin.cxx +++ b/src/output/RecorderOutputPlugin.cxx @@ -192,7 +192,7 @@ RecorderOutput::EncoderToFile(GError **error_r) static bool recorder_output_open(struct audio_output *ao, - struct audio_format *audio_format, + AudioFormat &audio_format, GError **error_r) { RecorderOutput *recorder = (RecorderOutput *)ao; diff --git a/src/output/RoarOutputPlugin.cxx b/src/output/RoarOutputPlugin.cxx index 835c7cdd9..bf2bf1789 100644 --- a/src/output/RoarOutputPlugin.cxx +++ b/src/output/RoarOutputPlugin.cxx @@ -144,39 +144,41 @@ roar_finish(struct audio_output *ao) static void roar_use_audio_format(struct roar_audio_info *info, - struct audio_format *audio_format) + AudioFormat &audio_format) { - info->rate = audio_format->sample_rate; - info->channels = audio_format->channels; + info->rate = audio_format.sample_rate; + info->channels = audio_format.channels; info->codec = ROAR_CODEC_PCM_S; - switch (audio_format->format) { - case SAMPLE_FORMAT_UNDEFINED: + switch (audio_format.format) { + case SampleFormat::UNDEFINED: + case SampleFormat::FLOAT: + case SampleFormat::DSD: info->bits = 16; - audio_format->format = SAMPLE_FORMAT_S16; + audio_format.format = SampleFormat::S16; break; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: info->bits = 8; break; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: info->bits = 16; break; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: info->bits = 32; - audio_format->format = SAMPLE_FORMAT_S32; + audio_format.format = SampleFormat::S32; break; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: info->bits = 32; break; } } static bool -roar_open(struct audio_output *ao, struct audio_format *audio_format, GError **error) +roar_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { RoarOutput *self = (RoarOutput *)ao; const ScopeLock protect(self->mutex); diff --git a/src/output/ShoutOutputPlugin.cxx b/src/output/ShoutOutputPlugin.cxx index a0f75da1d..fbad24ec8 100644 --- a/src/output/ShoutOutputPlugin.cxx +++ b/src/output/ShoutOutputPlugin.cxx @@ -116,9 +116,8 @@ inline bool ShoutOutput::Configure(const config_param *param, GError **error_r) { - const struct audio_format *audio_format = - &base.config_audio_format; - if (!audio_format_fully_defined(audio_format)) { + const AudioFormat audio_format = base.config_audio_format; + if (!audio_format.IsFullyDefined()) { g_set_error(error_r, shout_output_quark(), 0, "Need full audio format specification"); return nullptr; @@ -269,10 +268,10 @@ ShoutOutput::Configure(const config_param *param, GError **error_r) char temp[11]; memset(temp, 0, sizeof(temp)); - snprintf(temp, sizeof(temp), "%u", audio_format->channels); + snprintf(temp, sizeof(temp), "%u", audio_format.channels); shout_set_audio_info(shout_conn, SHOUT_AI_CHANNELS, temp); - snprintf(temp, sizeof(temp), "%u", audio_format->sample_rate); + snprintf(temp, sizeof(temp), "%u", audio_format.sample_rate); shout_set_audio_info(shout_conn, SHOUT_AI_SAMPLERATE, temp); @@ -428,7 +427,7 @@ shout_connect(ShoutOutput *sd, GError **error) } static bool -my_shout_open_device(struct audio_output *ao, struct audio_format *audio_format, +my_shout_open_device(struct audio_output *ao, AudioFormat &audio_format, GError **error) { ShoutOutput *sd = (ShoutOutput *)ao; diff --git a/src/output/SolarisOutputPlugin.cxx b/src/output/SolarisOutputPlugin.cxx index c20de4a8b..81c793e21 100644 --- a/src/output/SolarisOutputPlugin.cxx +++ b/src/output/SolarisOutputPlugin.cxx @@ -113,7 +113,7 @@ solaris_output_finish(struct audio_output *ao) } static bool -solaris_output_open(struct audio_output *ao, struct audio_format *audio_format, +solaris_output_open(struct audio_output *ao, AudioFormat &audio_format, GError **error) { SolarisOutput *so = (SolarisOutput *)ao; @@ -122,7 +122,7 @@ solaris_output_open(struct audio_output *ao, struct audio_format *audio_format, /* support only 16 bit mono/stereo for now; nothing else has been tested */ - audio_format->format = SAMPLE_FORMAT_S16; + audio_format.format = SampleFormat::S16; /* open the device in non-blocking mode */ @@ -150,8 +150,8 @@ solaris_output_open(struct audio_output *ao, struct audio_format *audio_format, return false; } - info.play.sample_rate = audio_format->sample_rate; - info.play.channels = audio_format->channels; + info.play.sample_rate = audio_format.sample_rate; + info.play.channels = audio_format.channels; info.play.precision = 16; info.play.encoding = AUDIO_ENCODING_LINEAR; diff --git a/src/output/WinmmOutputPlugin.cxx b/src/output/WinmmOutputPlugin.cxx index 2418731a0..de6927bd2 100644 --- a/src/output/WinmmOutputPlugin.cxx +++ b/src/output/WinmmOutputPlugin.cxx @@ -142,7 +142,7 @@ winmm_output_finish(struct audio_output *ao) } static bool -winmm_output_open(struct audio_output *ao, struct audio_format *audio_format, +winmm_output_open(struct audio_output *ao, AudioFormat &audio_format, GError **error_r) { WinmmOutput *wo = (WinmmOutput *)ao; @@ -154,30 +154,32 @@ winmm_output_open(struct audio_output *ao, struct audio_format *audio_format, return false; } - switch (audio_format->format) { - case SAMPLE_FORMAT_S8: - case SAMPLE_FORMAT_S16: + switch (audio_format.format) { + case SampleFormat::S8: + case SampleFormat::S16: break; - case SAMPLE_FORMAT_S24_P32: - case SAMPLE_FORMAT_S32: - case SAMPLE_FORMAT_UNDEFINED: + case SampleFormat::S24_P32: + case SampleFormat::S32: + case SampleFormat::FLOAT: + case SampleFormat::DSD: + case SampleFormat::UNDEFINED: /* we havn't tested formats other than S16 */ - audio_format->format = SAMPLE_FORMAT_S16; + audio_format.format = SampleFormat::S16; break; } - if (audio_format->channels > 2) + if (audio_format.channels > 2) /* same here: more than stereo was not tested */ - audio_format->channels = 2; + audio_format.channels = 2; WAVEFORMATEX format; format.wFormatTag = WAVE_FORMAT_PCM; - format.nChannels = audio_format->channels; - format.nSamplesPerSec = audio_format->sample_rate; - format.nBlockAlign = audio_format_frame_size(audio_format); + format.nChannels = audio_format.channels; + format.nSamplesPerSec = audio_format.sample_rate; + format.nBlockAlign = audio_format.GetFrameSize(); format.nAvgBytesPerSec = format.nSamplesPerSec * format.nBlockAlign; - format.wBitsPerSample = audio_format_sample_size(audio_format) * 8; + format.wBitsPerSample = audio_format.GetSampleSize() * 8; format.cbSize = 0; MMRESULT result = waveOutOpen(&wo->handle, wo->device_id, &format, diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx index 69e1c5d04..7e71da283 100644 --- a/src/pcm/PcmConvert.cxx +++ b/src/pcm/PcmConvert.cxx @@ -22,7 +22,7 @@ #include "PcmChannels.hxx" #include "PcmFormat.hxx" #include "pcm_pack.h" -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -48,46 +48,46 @@ PcmConvert::Reset() } inline const int16_t * -PcmConvert::Convert16(const audio_format *src_format, +PcmConvert::Convert16(const AudioFormat src_format, const void *src_buffer, size_t src_size, - const audio_format *dest_format, size_t *dest_size_r, + const AudioFormat dest_format, size_t *dest_size_r, GError **error_r) { const int16_t *buf; size_t len; - assert(dest_format->format == SAMPLE_FORMAT_S16); + assert(dest_format.format == SampleFormat::S16); buf = pcm_convert_to_16(format_buffer, dither, - sample_format(src_format->format), + src_format.format, src_buffer, src_size, &len); if (buf == NULL) { g_set_error(error_r, pcm_convert_quark(), 0, "Conversion from %s to 16 bit is not implemented", - sample_format_to_string(sample_format(src_format->format))); + sample_format_to_string(src_format.format)); return NULL; } - if (src_format->channels != dest_format->channels) { + if (src_format.channels != dest_format.channels) { buf = pcm_convert_channels_16(channels_buffer, - dest_format->channels, - src_format->channels, + dest_format.channels, + src_format.channels, buf, len, &len); if (buf == NULL) { g_set_error(error_r, pcm_convert_quark(), 0, "Conversion from %u to %u channels " "is not implemented", - src_format->channels, - dest_format->channels); + src_format.channels, + dest_format.channels); return NULL; } } - if (src_format->sample_rate != dest_format->sample_rate) { - buf = resampler.Resample16(dest_format->channels, - src_format->sample_rate, buf, len, - dest_format->sample_rate, &len, + if (src_format.sample_rate != dest_format.sample_rate) { + buf = resampler.Resample16(dest_format.channels, + src_format.sample_rate, buf, len, + dest_format.sample_rate, &len, error_r); if (buf == NULL) return NULL; @@ -98,45 +98,45 @@ PcmConvert::Convert16(const audio_format *src_format, } inline const int32_t * -PcmConvert::Convert24(const audio_format *src_format, +PcmConvert::Convert24(const AudioFormat src_format, const void *src_buffer, size_t src_size, - const audio_format *dest_format, size_t *dest_size_r, + const AudioFormat dest_format, size_t *dest_size_r, GError **error_r) { const int32_t *buf; size_t len; - assert(dest_format->format == SAMPLE_FORMAT_S24_P32); + assert(dest_format.format == SampleFormat::S24_P32); buf = pcm_convert_to_24(format_buffer, - sample_format(src_format->format), + src_format.format, src_buffer, src_size, &len); if (buf == NULL) { g_set_error(error_r, pcm_convert_quark(), 0, "Conversion from %s to 24 bit is not implemented", - sample_format_to_string(sample_format(src_format->format))); + sample_format_to_string(src_format.format)); return NULL; } - if (src_format->channels != dest_format->channels) { + if (src_format.channels != dest_format.channels) { buf = pcm_convert_channels_24(channels_buffer, - dest_format->channels, - src_format->channels, + dest_format.channels, + src_format.channels, buf, len, &len); if (buf == NULL) { g_set_error(error_r, pcm_convert_quark(), 0, "Conversion from %u to %u channels " "is not implemented", - src_format->channels, - dest_format->channels); + src_format.channels, + dest_format.channels); return NULL; } } - if (src_format->sample_rate != dest_format->sample_rate) { - buf = resampler.Resample24(dest_format->channels, - src_format->sample_rate, buf, len, - dest_format->sample_rate, &len, + if (src_format.sample_rate != dest_format.sample_rate) { + buf = resampler.Resample24(dest_format.channels, + src_format.sample_rate, buf, len, + dest_format.sample_rate, &len, error_r); if (buf == NULL) return NULL; @@ -147,45 +147,45 @@ PcmConvert::Convert24(const audio_format *src_format, } inline const int32_t * -PcmConvert::Convert32(const audio_format *src_format, +PcmConvert::Convert32(const AudioFormat src_format, const void *src_buffer, size_t src_size, - const audio_format *dest_format, size_t *dest_size_r, + const AudioFormat dest_format, size_t *dest_size_r, GError **error_r) { const int32_t *buf; size_t len; - assert(dest_format->format == SAMPLE_FORMAT_S32); + assert(dest_format.format == SampleFormat::S32); buf = pcm_convert_to_32(format_buffer, - sample_format(src_format->format), + src_format.format, src_buffer, src_size, &len); if (buf == NULL) { g_set_error(error_r, pcm_convert_quark(), 0, "Conversion from %s to 32 bit is not implemented", - sample_format_to_string(sample_format(src_format->format))); + sample_format_to_string(src_format.format)); return NULL; } - if (src_format->channels != dest_format->channels) { + if (src_format.channels != dest_format.channels) { buf = pcm_convert_channels_32(channels_buffer, - dest_format->channels, - src_format->channels, + dest_format.channels, + src_format.channels, buf, len, &len); if (buf == NULL) { g_set_error(error_r, pcm_convert_quark(), 0, "Conversion from %u to %u channels " "is not implemented", - src_format->channels, - dest_format->channels); + src_format.channels, + dest_format.channels); return NULL; } } - if (src_format->sample_rate != dest_format->sample_rate) { - buf = resampler.Resample32(dest_format->channels, - src_format->sample_rate, buf, len, - dest_format->sample_rate, &len, + if (src_format.sample_rate != dest_format.sample_rate) { + buf = resampler.Resample32(dest_format.channels, + src_format.sample_rate, buf, len, + dest_format.sample_rate, &len, error_r); if (buf == NULL) return buf; @@ -196,41 +196,41 @@ PcmConvert::Convert32(const audio_format *src_format, } inline const float * -PcmConvert::ConvertFloat(const audio_format *src_format, +PcmConvert::ConvertFloat(const AudioFormat src_format, const void *src_buffer, size_t src_size, - const audio_format *dest_format, size_t *dest_size_r, + const AudioFormat dest_format, size_t *dest_size_r, GError **error_r) { const float *buffer = (const float *)src_buffer; size_t size = src_size; - assert(dest_format->format == SAMPLE_FORMAT_FLOAT); + assert(dest_format.format == SampleFormat::FLOAT); /* convert to float now */ buffer = pcm_convert_to_float(format_buffer, - sample_format(src_format->format), + src_format.format, buffer, size, &size); if (buffer == NULL) { g_set_error(error_r, pcm_convert_quark(), 0, "Conversion from %s to float is not implemented", - sample_format_to_string(sample_format(src_format->format))); + sample_format_to_string(src_format.format)); return NULL; } /* convert channels */ - if (src_format->channels != dest_format->channels) { + if (src_format.channels != dest_format.channels) { buffer = pcm_convert_channels_float(channels_buffer, - dest_format->channels, - src_format->channels, + dest_format.channels, + src_format.channels, buffer, size, &size); if (buffer == NULL) { g_set_error(error_r, pcm_convert_quark(), 0, "Conversion from %u to %u channels " "is not implemented", - src_format->channels, - dest_format->channels); + src_format.channels, + dest_format.channels); return NULL; } } @@ -238,11 +238,11 @@ PcmConvert::ConvertFloat(const audio_format *src_format, /* resample with float, because this is the best format for libsamplerate */ - if (src_format->sample_rate != dest_format->sample_rate) { - buffer = resampler.ResampleFloat(dest_format->channels, - src_format->sample_rate, + if (src_format.sample_rate != dest_format.sample_rate) { + buffer = resampler.ResampleFloat(dest_format.channels, + src_format.sample_rate, buffer, size, - dest_format->sample_rate, + dest_format.sample_rate, &size, error_r); if (buffer == NULL) return NULL; @@ -253,16 +253,16 @@ PcmConvert::ConvertFloat(const audio_format *src_format, } const void * -PcmConvert::Convert(const audio_format *src_format, +PcmConvert::Convert(AudioFormat src_format, const void *src, size_t src_size, - const audio_format *dest_format, + const AudioFormat dest_format, size_t *dest_size_r, GError **error_r) { - struct audio_format float_format; - if (src_format->format == SAMPLE_FORMAT_DSD) { + AudioFormat float_format; + if (src_format.format == SampleFormat::DSD) { size_t f_size; - const float *f = dsd.ToFloat(src_format->channels, + const float *f = dsd.ToFloat(src_format.channels, false, (const uint8_t *)src, src_size, &f_size); if (f == NULL) { @@ -271,31 +271,31 @@ PcmConvert::Convert(const audio_format *src_format, return NULL; } - float_format = *src_format; - float_format.format = SAMPLE_FORMAT_FLOAT; + float_format = src_format; + float_format.format = SampleFormat::FLOAT; - src_format = &float_format; + src_format = float_format; src = f; src_size = f_size; } - switch (sample_format(dest_format->format)) { - case SAMPLE_FORMAT_S16: + switch (dest_format.format) { + case SampleFormat::S16: return Convert16(src_format, src, src_size, dest_format, dest_size_r, error_r); - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: return Convert24(src_format, src, src_size, dest_format, dest_size_r, error_r); - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: return Convert32(src_format, src, src_size, dest_format, dest_size_r, error_r); - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: return ConvertFloat(src_format, src, src_size, dest_format, dest_size_r, error_r); @@ -303,7 +303,7 @@ PcmConvert::Convert(const audio_format *src_format, default: g_set_error(error_r, pcm_convert_quark(), 0, "PCM conversion to %s is not implemented", - sample_format_to_string(sample_format(dest_format->format))); + sample_format_to_string(dest_format.format)); return NULL; } } diff --git a/src/pcm/PcmConvert.hxx b/src/pcm/PcmConvert.hxx index f4503f5aa..42b59e407 100644 --- a/src/pcm/PcmConvert.hxx +++ b/src/pcm/PcmConvert.hxx @@ -27,7 +27,7 @@ #include -struct audio_format; +struct AudioFormat; /** * This object is statically allocated (within another struct), and @@ -71,34 +71,34 @@ public: * ignore errors * @return the destination buffer, or NULL on error */ - const void *Convert(const audio_format *src_format, + const void *Convert(AudioFormat src_format, const void *src, size_t src_size, - const audio_format *dest_format, + AudioFormat dest_format, size_t *dest_size_r, GError **error_r); private: - const int16_t *Convert16(const audio_format *src_format, + const int16_t *Convert16(AudioFormat src_format, const void *src_buffer, size_t src_size, - const audio_format *dest_format, + AudioFormat dest_format, size_t *dest_size_r, GError **error_r); - const int32_t *Convert24(const audio_format *src_format, + const int32_t *Convert24(AudioFormat src_format, const void *src_buffer, size_t src_size, - const audio_format *dest_format, + AudioFormat dest_format, size_t *dest_size_r, GError **error_r); - const int32_t *Convert32(const audio_format *src_format, + const int32_t *Convert32(AudioFormat src_format, const void *src_buffer, size_t src_size, - const audio_format *dest_format, + AudioFormat dest_format, size_t *dest_size_r, GError **error_r); - const float *ConvertFloat(const audio_format *src_format, + const float *ConvertFloat(AudioFormat src_format, const void *src_buffer, size_t src_size, - const audio_format *dest_format, + AudioFormat dest_format, size_t *dest_size_r, GError **error_r); }; diff --git a/src/pcm/PcmDsdUsb.cxx b/src/pcm/PcmDsdUsb.cxx index ef267e724..7d58dec4d 100644 --- a/src/pcm/PcmDsdUsb.cxx +++ b/src/pcm/PcmDsdUsb.cxx @@ -20,7 +20,7 @@ #include "config.h" #include "PcmDsdUsb.hxx" #include "PcmBuffer.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" G_GNUC_CONST static inline uint32_t diff --git a/src/pcm/PcmExport.cxx b/src/pcm/PcmExport.cxx index 8840dc865..762411f59 100644 --- a/src/pcm/PcmExport.cxx +++ b/src/pcm/PcmExport.cxx @@ -27,21 +27,21 @@ extern "C" { } void -PcmExport::Open(enum sample_format sample_format, unsigned _channels, +PcmExport::Open(SampleFormat sample_format, unsigned _channels, bool _dsd_usb, bool _shift8, bool _pack, bool _reverse_endian) { assert(audio_valid_sample_format(sample_format)); assert(!_dsd_usb || audio_valid_channel_count(_channels)); channels = _channels; - dsd_usb = _dsd_usb && sample_format == SAMPLE_FORMAT_DSD; + dsd_usb = _dsd_usb && sample_format == SampleFormat::DSD; if (dsd_usb) /* after the conversion to DSD-over-USB, the DSD samples are stuffed inside fake 24 bit samples */ - sample_format = SAMPLE_FORMAT_S24_P32; + sample_format = SampleFormat::S24_P32; - shift8 = _shift8 && sample_format == SAMPLE_FORMAT_S24_P32; - pack24 = _pack && sample_format == SAMPLE_FORMAT_S24_P32; + shift8 = _shift8 && sample_format == SampleFormat::S24_P32; + pack24 = _pack && sample_format == SampleFormat::S24_P32; assert(!shift8 || !pack24); @@ -58,7 +58,7 @@ PcmExport::Open(enum sample_format sample_format, unsigned _channels, } size_t -PcmExport::GetFrameSize(const struct audio_format &audio_format) const +PcmExport::GetFrameSize(const AudioFormat &audio_format) const { if (pack24) /* packed 24 bit samples (3 bytes per sample) */ @@ -71,7 +71,7 @@ PcmExport::GetFrameSize(const struct audio_format &audio_format) const bytes per sample) */ return channels * 4; - return audio_format_frame_size(&audio_format); + return audio_format.GetFrameSize(); } const void * diff --git a/src/pcm/PcmExport.hxx b/src/pcm/PcmExport.hxx index e420493f0..bd18c0534 100644 --- a/src/pcm/PcmExport.hxx +++ b/src/pcm/PcmExport.hxx @@ -22,9 +22,9 @@ #include "check.h" #include "PcmBuffer.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" -struct audio_format; +struct AudioFormat; /** * An object that handles export of PCM samples to some instance @@ -61,8 +61,8 @@ struct PcmExport { /** * Convert DSD to DSD-over-USB? Input format must be - * SAMPLE_FORMAT_DSD and output format must be - * SAMPLE_FORMAT_S24_P32. + * SampleFormat::DSD and output format must be + * SampleFormat::S24_P32. */ bool dsd_usb; @@ -94,14 +94,14 @@ struct PcmExport { * * @param channels the number of channels; ignored unless dsd_usb is set */ - void Open(enum sample_format sample_format, unsigned channels, + void Open(SampleFormat sample_format, unsigned channels, bool dsd_usb, bool shift8, bool pack, bool reverse_endian); /** * Calculate the size of one output frame. */ gcc_pure - size_t GetFrameSize(const struct audio_format &audio_format) const; + size_t GetFrameSize(const AudioFormat &audio_format) const; /** * Export a PCM buffer. diff --git a/src/pcm/PcmFormat.cxx b/src/pcm/PcmFormat.cxx index 2dea09d2c..6425c7cfd 100644 --- a/src/pcm/PcmFormat.cxx +++ b/src/pcm/PcmFormat.cxx @@ -142,36 +142,36 @@ pcm_allocate_float_to_16(PcmBuffer &buffer, const int16_t * pcm_convert_to_16(PcmBuffer &buffer, PcmDither &dither, - enum sample_format src_format, const void *src, + SampleFormat src_format, const void *src, size_t src_size, size_t *dest_size_r) { assert(src_size % sample_format_size(src_format) == 0); switch (src_format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: break; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: return pcm_allocate_8_to_16(buffer, (const int8_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: *dest_size_r = src_size; return (const int16_t *)src; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: return pcm_allocate_24p32_to_16(buffer, dither, (const int32_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: return pcm_allocate_32_to_16(buffer, dither, (const int32_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: return pcm_allocate_float_to_16(buffer, (const float *)src, src_size, dest_size_r); @@ -247,36 +247,36 @@ pcm_allocate_float_to_24(PcmBuffer &buffer, const int32_t * pcm_convert_to_24(PcmBuffer &buffer, - enum sample_format src_format, const void *src, + SampleFormat src_format, const void *src, size_t src_size, size_t *dest_size_r) { assert(src_size % sample_format_size(src_format) == 0); switch (src_format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: break; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: return pcm_allocate_8_to_24(buffer, (const int8_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: return pcm_allocate_16_to_24(buffer, (const int16_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: *dest_size_r = src_size; return (const int32_t *)src; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: return pcm_allocate_32_to_24(buffer, (const int32_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: return pcm_allocate_float_to_24(buffer, (const float *)src, src_size, dest_size_r); @@ -358,36 +358,36 @@ pcm_allocate_float_to_32(PcmBuffer &buffer, const int32_t * pcm_convert_to_32(PcmBuffer &buffer, - enum sample_format src_format, const void *src, + SampleFormat src_format, const void *src, size_t src_size, size_t *dest_size_r) { assert(src_size % sample_format_size(src_format) == 0); switch (src_format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: break; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: return pcm_allocate_8_to_32(buffer, (const int8_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: return pcm_allocate_16_to_32(buffer, (const int16_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: return pcm_allocate_24p32_to_32(buffer, (const int32_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: *dest_size_r = src_size; return (const int32_t *)src; - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: return pcm_allocate_float_to_32(buffer, (const float *)src, src_size, dest_size_r); @@ -463,35 +463,35 @@ pcm_allocate_32_to_float(PcmBuffer &buffer, const float * pcm_convert_to_float(PcmBuffer &buffer, - enum sample_format src_format, const void *src, + SampleFormat src_format, const void *src, size_t src_size, size_t *dest_size_r) { switch (src_format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: break; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: return pcm_allocate_8_to_float(buffer, (const int8_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: return pcm_allocate_16_to_float(buffer, (const int16_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: return pcm_allocate_24p32_to_float(buffer, (const int32_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: return pcm_allocate_32_to_float(buffer, (const int32_t *)src, src_size, dest_size_r); - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: *dest_size_r = src_size; return (const float *)src; } diff --git a/src/pcm/PcmFormat.hxx b/src/pcm/PcmFormat.hxx index bb5ad49ae..b18b4f932 100644 --- a/src/pcm/PcmFormat.hxx +++ b/src/pcm/PcmFormat.hxx @@ -20,7 +20,7 @@ #ifndef MPD_PCM_FORMAT_HXX #define MPD_PCM_FORMAT_HXX -#include "audio_format.h" +#include "AudioFormat.hxx" #include #include @@ -42,7 +42,7 @@ class PcmDither; */ const int16_t * pcm_convert_to_16(PcmBuffer &buffer, PcmDither &dither, - enum sample_format src_format, const void *src, + SampleFormat src_format, const void *src, size_t src_size, size_t *dest_size_r); /** @@ -57,7 +57,7 @@ pcm_convert_to_16(PcmBuffer &buffer, PcmDither &dither, */ const int32_t * pcm_convert_to_24(PcmBuffer &buffer, - enum sample_format src_format, const void *src, + SampleFormat src_format, const void *src, size_t src_size, size_t *dest_size_r); /** @@ -72,7 +72,7 @@ pcm_convert_to_24(PcmBuffer &buffer, */ const int32_t * pcm_convert_to_32(PcmBuffer &buffer, - enum sample_format src_format, const void *src, + SampleFormat src_format, const void *src, size_t src_size, size_t *dest_size_r); /** @@ -87,7 +87,7 @@ pcm_convert_to_32(PcmBuffer &buffer, */ const float * pcm_convert_to_float(PcmBuffer &buffer, - enum sample_format src_format, const void *src, + SampleFormat src_format, const void *src, size_t src_size, size_t *dest_size_r); #endif diff --git a/src/pcm/PcmMix.cxx b/src/pcm/PcmMix.cxx index b92258a83..f4a02fc47 100644 --- a/src/pcm/PcmMix.cxx +++ b/src/pcm/PcmMix.cxx @@ -21,7 +21,7 @@ #include "PcmMix.hxx" #include "PcmVolume.hxx" #include "PcmUtils.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -74,35 +74,35 @@ pcm_add_vol_float(float *buffer1, const float *buffer2, static bool pcm_add_vol(void *buffer1, const void *buffer2, size_t size, int vol1, int vol2, - enum sample_format format) + SampleFormat format) { switch (format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: /* not implemented */ return false; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: PcmAddVolumeVoid(buffer1, buffer2, size, vol1, vol2); return true; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: PcmAddVolumeVoid(buffer1, buffer2, size, vol1, vol2); return true; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: PcmAddVolumeVoid(buffer1, buffer2, size, vol1, vol2); return true; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: PcmAddVolumeVoid(buffer1, buffer2, size, vol1, vol2); return true; - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: pcm_add_vol_float((float *)buffer1, (const float *)buffer2, size / 4, pcm_volume_to_float(vol1), @@ -153,31 +153,31 @@ pcm_add_float(float *buffer1, const float *buffer2, unsigned num_samples) static bool pcm_add(void *buffer1, const void *buffer2, size_t size, - enum sample_format format) + SampleFormat format) { switch (format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: /* not implemented */ return false; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: PcmAddVoid(buffer1, buffer2, size); return true; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: PcmAddVoid(buffer1, buffer2, size); return true; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: PcmAddVoid(buffer1, buffer2, size); return true; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: PcmAddVoid(buffer1, buffer2, size); return true; - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: pcm_add_float((float *)buffer1, (const float *)buffer2, size / 4); return true; @@ -189,7 +189,7 @@ pcm_add(void *buffer1, const void *buffer2, size_t size, bool pcm_mix(void *buffer1, const void *buffer2, size_t size, - enum sample_format format, float portion1) + SampleFormat format, float portion1) { int vol1; float s; diff --git a/src/pcm/PcmMix.hxx b/src/pcm/PcmMix.hxx index bb7110d04..b50a163fd 100644 --- a/src/pcm/PcmMix.hxx +++ b/src/pcm/PcmMix.hxx @@ -20,7 +20,7 @@ #ifndef MPD_PCM_MIX_HXX #define MPD_PCM_MIX_HXX -#include "audio_format.h" +#include "AudioFormat.hxx" #include "gcc.h" #include @@ -44,6 +44,6 @@ gcc_warn_unused_result bool pcm_mix(void *buffer1, const void *buffer2, size_t size, - enum sample_format format, float portion1); + SampleFormat format, float portion1); #endif diff --git a/src/pcm/PcmVolume.cxx b/src/pcm/PcmVolume.cxx index c22017ddd..2a8027400 100644 --- a/src/pcm/PcmVolume.cxx +++ b/src/pcm/PcmVolume.cxx @@ -20,7 +20,7 @@ #include "config.h" #include "PcmVolume.hxx" #include "PcmUtils.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -144,7 +144,7 @@ pcm_volume_change_float(float *buffer, const float *end, float volume) bool pcm_volume(void *buffer, size_t length, - enum sample_format format, + SampleFormat format, int volume) { if (volume == PCM_VOLUME_1) @@ -157,32 +157,32 @@ pcm_volume(void *buffer, size_t length, const void *end = pcm_end_pointer(buffer, length); switch (format) { - case SAMPLE_FORMAT_UNDEFINED: - case SAMPLE_FORMAT_DSD: + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: /* not implemented */ return false; - case SAMPLE_FORMAT_S8: + case SampleFormat::S8: pcm_volume_change_8((int8_t *)buffer, (const int8_t *)end, volume); return true; - case SAMPLE_FORMAT_S16: + case SampleFormat::S16: pcm_volume_change_16((int16_t *)buffer, (const int16_t *)end, volume); return true; - case SAMPLE_FORMAT_S24_P32: + case SampleFormat::S24_P32: pcm_volume_change_24((int32_t *)buffer, (const int32_t *)end, volume); return true; - case SAMPLE_FORMAT_S32: + case SampleFormat::S32: pcm_volume_change_32((int32_t *)buffer, (const int32_t *)end, volume); return true; - case SAMPLE_FORMAT_FLOAT: + case SampleFormat::FLOAT: pcm_volume_change_float((float *)buffer, (const float *)end, pcm_volume_to_float(volume)); return true; diff --git a/src/pcm/PcmVolume.hxx b/src/pcm/PcmVolume.hxx index ed660b22a..8cd82acf7 100644 --- a/src/pcm/PcmVolume.hxx +++ b/src/pcm/PcmVolume.hxx @@ -21,7 +21,7 @@ #define MPD_PCM_VOLUME_HXX #include "PcmPrng.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include #include @@ -31,7 +31,7 @@ enum { PCM_VOLUME_1 = 1024, }; -struct audio_format; +struct AudioFormat; /** * Converts a float value (0.0 = silence, 1.0 = 100% volume) to an @@ -75,7 +75,7 @@ pcm_volume_dither(void) */ bool pcm_volume(void *buffer, size_t length, - enum sample_format format, + SampleFormat format, int volume); #endif diff --git a/test/dump_playlist.cxx b/test/dump_playlist.cxx index ea341ab95..bf8ddcdfe 100644 --- a/test/dump_playlist.cxx +++ b/test/dump_playlist.cxx @@ -51,7 +51,7 @@ my_log_func(const gchar *log_domain, G_GNUC_UNUSED GLogLevelFlags log_level, void decoder_initialized(G_GNUC_UNUSED struct decoder *decoder, - G_GNUC_UNUSED const struct audio_format *audio_format, + G_GNUC_UNUSED const AudioFormat audio_format, G_GNUC_UNUSED bool seekable, G_GNUC_UNUSED float total_time) { diff --git a/test/read_mixer.cxx b/test/read_mixer.cxx index 8fc89e93d..1d8372a98 100644 --- a/test/read_mixer.cxx +++ b/test/read_mixer.cxx @@ -101,7 +101,7 @@ filter_plugin_by_name(G_GNUC_UNUSED const char *name) bool pcm_volume(G_GNUC_UNUSED void *buffer, G_GNUC_UNUSED size_t length, - G_GNUC_UNUSED enum sample_format format, + G_GNUC_UNUSED SampleFormat format, G_GNUC_UNUSED int volume) { assert(false); diff --git a/test/read_tags.cxx b/test/read_tags.cxx index ee9464b2c..420cf9f0e 100644 --- a/test/read_tags.cxx +++ b/test/read_tags.cxx @@ -23,7 +23,7 @@ #include "DecoderAPI.hxx" #include "InputInit.hxx" #include "InputStream.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "TagHandler.hxx" #include "TagId3.hxx" #include "ApeTag.hxx" @@ -40,7 +40,7 @@ void decoder_initialized(G_GNUC_UNUSED struct decoder *decoder, - G_GNUC_UNUSED const struct audio_format *audio_format, + G_GNUC_UNUSED const AudioFormat audio_format, G_GNUC_UNUSED bool seekable, G_GNUC_UNUSED float total_time) { diff --git a/test/run_convert.cxx b/test/run_convert.cxx index 2b0214912..ce7df42c3 100644 --- a/test/run_convert.cxx +++ b/test/run_convert.cxx @@ -25,7 +25,7 @@ #include "config.h" #include "AudioParser.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "pcm/PcmConvert.hxx" #include "conf.h" #include "util/fifo_buffer.h" @@ -57,7 +57,7 @@ config_get_string(gcc_unused enum ConfigOption option, int main(int argc, char **argv) { GError *error = NULL; - struct audio_format in_audio_format, out_audio_format; + AudioFormat in_audio_format, out_audio_format; const void *output; ssize_t nbytes; size_t length; @@ -69,15 +69,15 @@ int main(int argc, char **argv) g_log_set_default_handler(my_log_func, NULL); - if (!audio_format_parse(&in_audio_format, argv[1], + if (!audio_format_parse(in_audio_format, argv[1], false, &error)) { g_printerr("Failed to parse audio format: %s\n", error->message); return 1; } - struct audio_format out_audio_format_mask; - if (!audio_format_parse(&out_audio_format_mask, argv[2], + AudioFormat out_audio_format_mask; + if (!audio_format_parse(out_audio_format_mask, argv[2], true, &error)) { g_printerr("Failed to parse audio format: %s\n", error->message); @@ -85,9 +85,9 @@ int main(int argc, char **argv) } out_audio_format = in_audio_format; - audio_format_mask_apply(&out_audio_format, &out_audio_format_mask); + out_audio_format.ApplyMask(out_audio_format_mask); - const size_t in_frame_size = audio_format_frame_size(&in_audio_format); + const size_t in_frame_size = in_audio_format.GetFrameSize(); PcmConvert state; @@ -112,8 +112,8 @@ int main(int argc, char **argv) fifo_buffer_consume(buffer, length); - output = state.Convert(&in_audio_format, src, length, - &out_audio_format, &length, &error); + output = state.Convert(in_audio_format, src, length, + out_audio_format, &length, &error); if (output == NULL) { g_printerr("Failed to convert: %s\n", error->message); return 2; diff --git a/test/run_decoder.cxx b/test/run_decoder.cxx index a05afb113..a5826b278 100644 --- a/test/run_decoder.cxx +++ b/test/run_decoder.cxx @@ -23,7 +23,7 @@ #include "DecoderAPI.hxx" #include "InputInit.hxx" #include "input_stream.h" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "stdbin.h" #include @@ -52,14 +52,14 @@ struct decoder { void decoder_initialized(struct decoder *decoder, - const struct audio_format *audio_format, + const AudioFormat audio_format, G_GNUC_UNUSED bool seekable, G_GNUC_UNUSED float total_time) { struct audio_format_string af_string; assert(!decoder->initialized); - assert(audio_format_valid(audio_format)); + assert(audio_format.IsValid()); g_printerr("audio_format=%s\n", audio_format_to_string(audio_format, &af_string)); diff --git a/test/run_encoder.cxx b/test/run_encoder.cxx index 1beb434df..4b1521b99 100644 --- a/test/run_encoder.cxx +++ b/test/run_encoder.cxx @@ -20,7 +20,7 @@ #include "config.h" #include "EncoderList.hxx" #include "EncoderPlugin.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "AudioParser.hxx" #include "conf.h" #include "stdbin.h" @@ -44,7 +44,6 @@ encoder_to_stdout(Encoder &encoder) int main(int argc, char **argv) { GError *error = NULL; - struct audio_format audio_format; bool ret; const char *encoder_name; static char buffer[32768]; @@ -61,8 +60,6 @@ int main(int argc, char **argv) else encoder_name = "vorbis"; - audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2); - /* create the encoder */ const auto plugin = encoder_plugin_get(encoder_name); @@ -84,8 +81,9 @@ int main(int argc, char **argv) /* open the encoder */ + AudioFormat audio_format(44100, SampleFormat::S16, 2); if (argc > 2) { - ret = audio_format_parse(&audio_format, argv[2], + ret = audio_format_parse(audio_format, argv[2], false, &error); if (!ret) { g_printerr("Failed to parse audio format: %s\n", @@ -95,7 +93,7 @@ int main(int argc, char **argv) } } - if (!encoder_open(encoder, &audio_format, &error)) { + if (!encoder_open(encoder, audio_format, &error)) { g_printerr("Failed to open encoder: %s\n", error->message); g_error_free(error); diff --git a/test/run_filter.cxx b/test/run_filter.cxx index 8df3fa732..db4a2ee89 100644 --- a/test/run_filter.cxx +++ b/test/run_filter.cxx @@ -21,7 +21,7 @@ #include "conf.h" #include "fs/Path.hxx" #include "AudioParser.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "FilterPlugin.hxx" #include "FilterInternal.hxx" #include "pcm/PcmVolume.hxx" @@ -91,11 +91,9 @@ load_filter(const char *name) int main(int argc, char **argv) { - struct audio_format audio_format; struct audio_format_string af_string; bool success; GError *error = NULL; - const struct audio_format *out_audio_format; char buffer[4096]; if (argc < 3 || argc > 4) { @@ -105,7 +103,7 @@ int main(int argc, char **argv) const Path config_path = Path::FromFS(argv[1]); - audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2); + AudioFormat audio_format(44100, SampleFormat::S16, 2); /* initialize GLib */ @@ -127,7 +125,7 @@ int main(int argc, char **argv) /* parse the audio format */ if (argc > 3) { - success = audio_format_parse(&audio_format, argv[3], + success = audio_format_parse(audio_format, argv[3], false, &error); if (!success) { g_printerr("Failed to parse audio format: %s\n", @@ -145,8 +143,9 @@ int main(int argc, char **argv) /* open the filter */ - out_audio_format = filter->Open(audio_format, &error); - if (out_audio_format == NULL) { + const AudioFormat out_audio_format = + filter->Open(audio_format, &error); + if (!out_audio_format.IsDefined()) { g_printerr("Failed to open filter: %s\n", error->message); g_error_free(error); delete filter; diff --git a/test/run_normalize.cxx b/test/run_normalize.cxx index 070dbaf5a..72e23054f 100644 --- a/test/run_normalize.cxx +++ b/test/run_normalize.cxx @@ -26,7 +26,7 @@ #include "config.h" #include "AudioCompress/compress.h" #include "AudioParser.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "stdbin.h" #include @@ -38,7 +38,6 @@ int main(int argc, char **argv) { GError *error = NULL; - struct audio_format audio_format; bool ret; struct Compressor *compressor; static char buffer[4096]; @@ -49,16 +48,16 @@ int main(int argc, char **argv) return 1; } + AudioFormat audio_format(48000, SampleFormat::S16, 2); if (argc > 1) { - ret = audio_format_parse(&audio_format, argv[1], + ret = audio_format_parse(audio_format, argv[1], false, &error); if (!ret) { g_printerr("Failed to parse audio format: %s\n", error->message); return 1; } - } else - audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, 2); + } compressor = Compressor_new(0); diff --git a/test/run_output.cxx b/test/run_output.cxx index 399d8e731..34879c657 100644 --- a/test/run_output.cxx +++ b/test/run_output.cxx @@ -52,9 +52,9 @@ PcmConvert::PcmConvert() {} PcmConvert::~PcmConvert() {} const void * -PcmConvert::Convert(gcc_unused const audio_format *src_format, +PcmConvert::Convert(gcc_unused const AudioFormat src_format, gcc_unused const void *src, gcc_unused size_t src_size, - gcc_unused const audio_format *dest_format, + gcc_unused const AudioFormat dest_format, gcc_unused size_t *dest_size_r, gcc_unused GError **error_r) { @@ -114,7 +114,7 @@ load_audio_output(const char *name) } static bool -run_output(struct audio_output *ao, struct audio_format *audio_format) +run_output(struct audio_output *ao, AudioFormat audio_format) { /* open the audio output */ @@ -138,7 +138,7 @@ run_output(struct audio_output *ao, struct audio_format *audio_format) g_printerr("audio_format=%s\n", audio_format_to_string(audio_format, &af_string)); - size_t frame_size = audio_format_frame_size(audio_format); + size_t frame_size = audio_format.GetFrameSize(); /* play */ @@ -183,7 +183,6 @@ run_output(struct audio_output *ao, struct audio_format *audio_format) int main(int argc, char **argv) { - struct audio_format audio_format; bool success; GError *error = NULL; @@ -194,7 +193,7 @@ int main(int argc, char **argv) const Path config_path = Path::FromFS(argv[1]); - audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2); + AudioFormat audio_format(44100, SampleFormat::S16, 2); #if !GLIB_CHECK_VERSION(2,32,0) g_thread_init(NULL); @@ -227,7 +226,7 @@ int main(int argc, char **argv) /* parse the audio format */ if (argc > 3) { - success = audio_format_parse(&audio_format, argv[3], + success = audio_format_parse(audio_format, argv[3], false, &error); if (!success) { g_printerr("Failed to parse audio format: %s\n", @@ -239,7 +238,7 @@ int main(int argc, char **argv) /* do it */ - success = run_output(ao, &audio_format); + success = run_output(ao, audio_format); /* cleanup and exit */ diff --git a/test/software_volume.cxx b/test/software_volume.cxx index fee4eeff3..c46804731 100644 --- a/test/software_volume.cxx +++ b/test/software_volume.cxx @@ -26,7 +26,7 @@ #include "config.h" #include "pcm/PcmVolume.hxx" #include "AudioParser.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "stdbin.h" #include @@ -37,7 +37,6 @@ int main(int argc, char **argv) { GError *error = NULL; - struct audio_format audio_format; bool ret; static char buffer[4096]; ssize_t nbytes; @@ -47,20 +46,20 @@ int main(int argc, char **argv) return 1; } + AudioFormat audio_format(48000, SampleFormat::S16, 2); if (argc > 1) { - ret = audio_format_parse(&audio_format, argv[1], + ret = audio_format_parse(audio_format, argv[1], false, &error); if (!ret) { g_printerr("Failed to parse audio format: %s\n", error->message); return 1; } - } else - audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, 2); + } while ((nbytes = read(0, buffer, sizeof(buffer))) > 0) { if (!pcm_volume(buffer, nbytes, - sample_format(audio_format.format), + audio_format.format, PCM_VOLUME_1 / 2)) { g_printerr("pcm_volume() has failed\n"); return 2; diff --git a/test/test_pcm_format.cxx b/test/test_pcm_format.cxx index d7a0a2698..65d744671 100644 --- a/test/test_pcm_format.cxx +++ b/test/test_pcm_format.cxx @@ -24,7 +24,7 @@ #include "pcm/PcmDither.hxx" #include "pcm/PcmUtils.hxx" #include "pcm/PcmBuffer.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include @@ -38,7 +38,7 @@ test_pcm_format_8_to_16() size_t d_size; PcmDither dither; - auto d = pcm_convert_to_16(buffer, dither, SAMPLE_FORMAT_S8, + auto d = pcm_convert_to_16(buffer, dither, SampleFormat::S8, src, sizeof(src), &d_size); auto d_end = pcm_end_pointer(d, d_size); g_assert_cmpint(d_end - d, ==, N); @@ -56,7 +56,7 @@ test_pcm_format_16_to_24() PcmBuffer buffer; size_t d_size; - auto d = pcm_convert_to_24(buffer, SAMPLE_FORMAT_S16, + auto d = pcm_convert_to_24(buffer, SampleFormat::S16, src, sizeof(src), &d_size); auto d_end = pcm_end_pointer(d, d_size); g_assert_cmpint(d_end - d, ==, N); @@ -74,7 +74,7 @@ test_pcm_format_16_to_32() PcmBuffer buffer; size_t d_size; - auto d = pcm_convert_to_32(buffer, SAMPLE_FORMAT_S16, + auto d = pcm_convert_to_32(buffer, SampleFormat::S16, src, sizeof(src), &d_size); auto d_end = pcm_end_pointer(d, d_size); g_assert_cmpint(d_end - d, ==, N); @@ -92,7 +92,7 @@ test_pcm_format_float() PcmBuffer buffer1, buffer2; size_t f_size; - auto f = pcm_convert_to_float(buffer1, SAMPLE_FORMAT_S16, + auto f = pcm_convert_to_float(buffer1, SampleFormat::S16, src, sizeof(src), &f_size); auto f_end = pcm_end_pointer(f, f_size); g_assert_cmpint(f_end - f, ==, N); @@ -106,7 +106,7 @@ test_pcm_format_float() size_t d_size; auto d = pcm_convert_to_16(buffer2, dither, - SAMPLE_FORMAT_FLOAT, + SampleFormat::FLOAT, f, f_size, &d_size); auto d_end = pcm_end_pointer(d, d_size); g_assert_cmpint(d_end - d, ==, N); diff --git a/test/test_pcm_mix.cxx b/test/test_pcm_mix.cxx index b4d8486bf..b0e89639c 100644 --- a/test/test_pcm_mix.cxx +++ b/test/test_pcm_mix.cxx @@ -24,7 +24,7 @@ #include -template> +template> void TestPcmMix(G g=G()) { @@ -62,23 +62,23 @@ TestPcmMix(G g=G()) void test_pcm_mix_8() { - TestPcmMix(); + TestPcmMix(); } void test_pcm_mix_16() { - TestPcmMix(); + TestPcmMix(); } void test_pcm_mix_24() { - TestPcmMix(GlibRandomInt24()); + TestPcmMix(GlibRandomInt24()); } void test_pcm_mix_32() { - TestPcmMix(); + TestPcmMix(); } diff --git a/test/test_pcm_volume.cxx b/test/test_pcm_volume.cxx index ec3ac23b9..d5aa3782e 100644 --- a/test/test_pcm_volume.cxx +++ b/test/test_pcm_volume.cxx @@ -37,17 +37,17 @@ test_pcm_volume_8() int8_t dest[N]; std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S8, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S8, 0), ==, true); g_assert_cmpint(memcmp(dest, zero, sizeof(zero)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S8, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S8, PCM_VOLUME_1), ==, true); g_assert_cmpint(memcmp(dest, src, sizeof(src)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S8, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S8, PCM_VOLUME_1 / 2), ==, true); for (unsigned i = 0; i < N; ++i) { @@ -66,17 +66,17 @@ test_pcm_volume_16() int16_t dest[N]; std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S16, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S16, 0), ==, true); g_assert_cmpint(memcmp(dest, zero, sizeof(zero)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S16, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S16, PCM_VOLUME_1), ==, true); g_assert_cmpint(memcmp(dest, src, sizeof(src)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S16, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S16, PCM_VOLUME_1 / 2), ==, true); for (unsigned i = 0; i < N; ++i) { @@ -95,17 +95,17 @@ test_pcm_volume_24() int32_t dest[N]; std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S24_P32, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S24_P32, 0), ==, true); g_assert_cmpint(memcmp(dest, zero, sizeof(zero)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S24_P32, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S24_P32, PCM_VOLUME_1), ==, true); g_assert_cmpint(memcmp(dest, src, sizeof(src)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S24_P32, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S24_P32, PCM_VOLUME_1 / 2), ==, true); for (unsigned i = 0; i < N; ++i) { @@ -124,17 +124,17 @@ test_pcm_volume_32() int32_t dest[N]; std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S32, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S32, 0), ==, true); g_assert_cmpint(memcmp(dest, zero, sizeof(zero)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S32, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S32, PCM_VOLUME_1), ==, true); g_assert_cmpint(memcmp(dest, src, sizeof(src)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_S32, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::S32, PCM_VOLUME_1 / 2), ==, true); for (unsigned i = 0; i < N; ++i) { @@ -153,17 +153,17 @@ test_pcm_volume_float() float dest[N]; std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_FLOAT, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::FLOAT, 0), ==, true); g_assert_cmpint(memcmp(dest, zero, sizeof(zero)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_FLOAT, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::FLOAT, PCM_VOLUME_1), ==, true); g_assert_cmpint(memcmp(dest, src, sizeof(src)), ==, 0); std::copy(src.begin(), src.end(), dest); - g_assert_cmpint(pcm_volume(dest, sizeof(dest), SAMPLE_FORMAT_FLOAT, + g_assert_cmpint(pcm_volume(dest, sizeof(dest), SampleFormat::FLOAT, PCM_VOLUME_1 / 2), ==, true); for (unsigned i = 0; i < N; ++i) diff --git a/test/test_vorbis_encoder.cxx b/test/test_vorbis_encoder.cxx index 650480319..a523f9d72 100644 --- a/test/test_vorbis_encoder.cxx +++ b/test/test_vorbis_encoder.cxx @@ -20,7 +20,7 @@ #include "config.h" #include "EncoderList.hxx" #include "EncoderPlugin.hxx" -#include "audio_format.h" +#include "AudioFormat.hxx" #include "conf.h" #include "stdbin.h" #include "Tag.hxx" @@ -61,10 +61,8 @@ main(G_GNUC_UNUSED int argc, G_GNUC_UNUSED char **argv) /* open the encoder */ - struct audio_format audio_format; - - audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2); - success = encoder_open(encoder, &audio_format, NULL); + AudioFormat audio_format(44100, SampleFormat::S16, 2); + success = encoder_open(encoder, audio_format, NULL); assert(success); encoder_to_stdout(*encoder);