audio_format: convert to C++

7 files changed, 67 insertions, 65 deletions

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;