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 <assert.h>
#include <stdio.h>
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 <stdint.h>
+#include <assert.h>
+
+#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 <assert.h>
@@ -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 <glib.h>
@@ -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 <cmath>
@@ -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 <stddef.h>
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 <assert.h>
@@ -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 <stdint.h>
@@ -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 <stddef.h>
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 <stddef.h>
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 <glib.h>
@@ -28,10 +28,10 @@
#include <limits.h>
#include <stddef.h>
-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 <stdint.h>
-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 <stdint.h>
-#include <stdbool.h>
-#include <assert.h>
-
-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 <libavutil/dict.h>
}
+/* 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 <FLAC/ordinals.h>
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 <lame/lame.h>
@@ -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 <opus.h>
@@ -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 <twolame.h>
@@ -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 <vorbis/vorbisenc.h>
@@ -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 <assert.h>
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 <glib.h>
@@ -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 <assert.h>
@@ -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<PcmConvert> 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 <assert.h>
@@ -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 <assert.h>
#include <string.h>
@@ -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 <glib.h>
@@ -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 <glib.h>
-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 <stdint.h>
#include <stddef.h>
@@ -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 <math.h>
@@ -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<int8_t, int32_t, 8>(buffer1, buffer2, size,
vol1, vol2);
return true;
- case SAMPLE_FORMAT_S16:
+ case SampleFormat::S16:
PcmAddVolumeVoid<int16_t, int32_t, 16>(buffer1, buffer2, size,
vol1, vol2);
return true;
- case SAMPLE_FORMAT_S24_P32:
+ case SampleFormat::S24_P32:
PcmAddVolumeVoid<int32_t, int64_t, 24>(buffer1, buffer2, size,
vol1, vol2);
return true;
- case SAMPLE_FORMAT_S32:
+ case SampleFormat::S32:
PcmAddVolumeVoid<int32_t, int64_t, 32>(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<int8_t, int32_t, 8>(buffer1, buffer2, size);
return true;
- case SAMPLE_FORMAT_S16:
+ case SampleFormat::S16:
PcmAddVoid<int16_t, int32_t, 16>(buffer1, buffer2, size);
return true;
- case SAMPLE_FORMAT_S24_P32:
+ case SampleFormat::S24_P32:
PcmAddVoid<int32_t, int64_t, 24>(buffer1, buffer2, size);
return true;
- case SAMPLE_FORMAT_S32:
+ case SampleFormat::S32:
PcmAddVoid<int32_t, int64_t, 32>(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 <stddef.h>
@@ -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 <glib.h>
@@ -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 <stdint.h>
#include <stddef.h>
@@ -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 <glib.h>
@@ -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 <glib.h>
@@ -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 <glib.h>
@@ -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 <glib.h>
@@ -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 <glib.h>
-template<typename T, sample_format format, typename G=GlibRandomInt<T>>
+template<typename T, SampleFormat format, typename G=GlibRandomInt<T>>
void
TestPcmMix(G g=G())
{
@@ -62,23 +62,23 @@ TestPcmMix(G g=G())
void
test_pcm_mix_8()
{
- TestPcmMix<int8_t, SAMPLE_FORMAT_S8>();
+ TestPcmMix<int8_t, SampleFormat::S8>();
}
void
test_pcm_mix_16()
{
- TestPcmMix<int16_t, SAMPLE_FORMAT_S16>();
+ TestPcmMix<int16_t, SampleFormat::S16>();
}
void
test_pcm_mix_24()
{
- TestPcmMix<int32_t, SAMPLE_FORMAT_S24_P32>(GlibRandomInt24());
+ TestPcmMix<int32_t, SampleFormat::S24_P32>(GlibRandomInt24());
}
void
test_pcm_mix_32()
{
- TestPcmMix<int32_t, SAMPLE_FORMAT_S32>();
+ TestPcmMix<int32_t, SampleFormat::S32>();
}
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);