audio_format: convert to C++

26 files changed, 143 insertions, 139 deletions

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 */