audio_format: changed "bits" to "enum sample_format"

This patch prepares support for floating point samples (and probably
other formats).  It changes the meaning of the "bits" attribute from a
bit count to a symbolic value.

19 files changed, 138 insertions, 43 deletions

diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index f12b8bff0..70b2c0202 100644
--- a/src/decoder/_flac_common.c
+++ b/src/decoder/_flac_common.c
@@ -60,6 +60,27 @@ flac_data_deinit(struct flac_data *data)
 		tag_free(data->tag);
 }
 
+static enum sample_format
+flac_sample_format(const FLAC__StreamMetadata_StreamInfo *si)
+{
+	switch (si->bits_per_sample) {
+	case 8:
+		return SAMPLE_FORMAT_S8;
+
+	case 16:
+		return SAMPLE_FORMAT_S16;
+
+	case 24:
+		return SAMPLE_FORMAT_S24_P32;
+
+	case 32:
+		return SAMPLE_FORMAT_S32;
+
+	default:
+		return SAMPLE_FORMAT_UNDEFINED;
+	}
+}
+
 bool
 flac_data_get_audio_format(struct flac_data *data,
 			   struct audio_format *audio_format)
@@ -71,9 +92,11 @@ flac_data_get_audio_format(struct flac_data *data,
 		return false;
 	}
 
+	data->sample_format = flac_sample_format(&data->stream_info);
+
 	if (!audio_format_init_checked(audio_format,
 				       data->stream_info.sample_rate,
-				       data->stream_info.bits_per_sample,
+				       data->sample_format,
 				       data->stream_info.channels, &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);
@@ -144,7 +167,7 @@ flac_common_write(struct flac_data *data, const FLAC__Frame * frame,
 	buffer = pcm_buffer_get(&data->buffer, buffer_size);
 
 	flac_convert(buffer, frame->header.channels,
-		     frame->header.bits_per_sample, buf,
+		     data->sample_format, buf,
 		     0, frame->header.blocksize);
 
 	if (data->next_frame >= data->first_frame)
diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h
index 1d211fcfb..2f328afa6 100644
--- a/src/decoder/_flac_common.h
+++ b/src/decoder/_flac_common.h
@@ -38,6 +38,8 @@
 struct flac_data {
 	struct pcm_buffer buffer;
 
+	enum sample_format sample_format;
+
 	/**
 	 * The size of one frame in the output buffer.
 	 */
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
index 5a2096d00..fcb431db7 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_plugin.c
@@ -101,13 +101,33 @@ setup_virtual_fops(struct input_stream *stream)
 	return vf;
 }
 
-static uint8_t
+static enum sample_format
+audiofile_bits_to_sample_format(int bits)
+{
+	switch (bits) {
+	case 8:
+		return SAMPLE_FORMAT_S8;
+
+	case 16:
+		return SAMPLE_FORMAT_S16;
+
+	case 24:
+		return SAMPLE_FORMAT_S24_P32;
+
+	case 32:
+		return SAMPLE_FORMAT_S32;
+	}
+
+	return SAMPLE_FORMAT_UNDEFINED;
+}
+
+static enum sample_format
 audiofile_setup_sample_format(AFfilehandle af_fp)
 {
 	int fs, bits;
 
 	afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	if (!audio_valid_sample_format(bits)) {
+	if (!audio_valid_sample_format(audiofile_bits_to_sample_format(bits))) {
 		g_debug("input file has %d bit samples, converting to 16",
 			bits);
 		bits = 16;
@@ -117,7 +137,7 @@ audiofile_setup_sample_format(AFfilehandle af_fp)
 	                         AF_SAMPFMT_TWOSCOMP, bits);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
 
-	return bits;
+	return audiofile_bits_to_sample_format(bits);
 }
 
 static void
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
index 55df15555..2a05e33e8 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_plugin.c
@@ -283,7 +283,7 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
 	decoder_buffer_consume(buffer, nbytes);
 
 	return audio_format_init_checked(audio_format, sample_rate,
-					 16, channels, error_r);
+					 SAMPLE_FORMAT_S16, channels, error_r);
 }
 
 /**
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
index 9b025153b..af9320322 100644
--- a/src/decoder/ffmpeg_plugin.c
+++ b/src/decoder/ffmpeg_plugin.c
@@ -277,6 +277,26 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 	return cmd;
 }
 
+static enum sample_format
+ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
+{
+#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
+	int bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
+
+	/* XXX implement & test other sample formats */
+
+	switch (bits) {
+	case 16:
+		return SAMPLE_FORMAT_S16;
+	}
+
+	return SAMPLE_FORMAT_UNDEFINED;
+#else
+	/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
+	return SAMPLE_FORMAT_S16;
+#endif
+}
+
 static bool
 ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 {
@@ -288,7 +308,6 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 	struct audio_format audio_format;
 	enum decoder_command cmd;
 	int total_time;
-	uint8_t bits;
 
 	total_time = 0;
 
@@ -296,14 +315,9 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 		codec_context->channels = 2;
 	}
 
-#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
-	bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
-#else
-	/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
-	bits = (uint8_t) 16;
-#endif
 	if (!audio_format_init_checked(&audio_format,
-				       codec_context->sample_rate, bits,
+				       codec_context->sample_rate,
+				       ffmpeg_sample_format(codec_context),
 				       codec_context->channels, &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);
diff --git a/src/decoder/flac_pcm.c b/src/decoder/flac_pcm.c
index 737d5b043..a8bf6f293 100644
--- a/src/decoder/flac_pcm.c
+++ b/src/decoder/flac_pcm.c
@@ -20,6 +20,8 @@
 #include "config.h"
 #include "flac_pcm.h"
 
+#include <assert.h>
+
 static void flac_convert_stereo16(int16_t *dest,
 				  const FLAC__int32 * const buf[],
 				  unsigned int position, unsigned int end)
@@ -74,12 +76,12 @@ flac_convert_8(int8_t *dest,
 
 void
 flac_convert(void *dest,
-	     unsigned int num_channels, unsigned sample_format,
+	     unsigned int num_channels, enum sample_format sample_format,
 	     const FLAC__int32 *const buf[],
 	     unsigned int position, unsigned int end)
 {
 	switch (sample_format) {
-	case 16:
+	case SAMPLE_FORMAT_S16:
 		if (num_channels == 2)
 			flac_convert_stereo16((int16_t*)dest, buf,
 					      position, end);
@@ -88,15 +90,19 @@ flac_convert(void *dest,
 					position, end);
 		break;
 
-	case 24:
-	case 32:
+	case SAMPLE_FORMAT_S24_P32:
+	case SAMPLE_FORMAT_S32:
 		flac_convert_32((int32_t*)dest, num_channels, buf,
 				position, end);
 		break;
 
-	case 8:
+	case SAMPLE_FORMAT_S8:
 		flac_convert_8((int8_t*)dest, num_channels, buf,
 			       position, end);
 		break;
+
+	case SAMPLE_FORMAT_UNDEFINED:
+		/* unreachable */
+		assert(false);
 	}
 }
diff --git a/src/decoder/flac_pcm.h b/src/decoder/flac_pcm.h
index dca9d6824..4d7a51c4d 100644
--- a/src/decoder/flac_pcm.h
+++ b/src/decoder/flac_pcm.h
@@ -20,11 +20,13 @@
 #ifndef MPD_FLAC_PCM_H
 #define MPD_FLAC_PCM_H
 
+#include "audio_format.h"
+
 #include <FLAC/ordinals.h>
 
 void
 flac_convert(void *dest,
-	     unsigned int num_channels, unsigned sample_format,
+	     unsigned int num_channels, enum sample_format sample_format,
 	     const FLAC__int32 *const buf[],
 	     unsigned int position, unsigned int end);
 
diff --git a/src/decoder/fluidsynth_plugin.c b/src/decoder/fluidsynth_plugin.c
index 3e8a4edc4..1b1e6a531 100644
--- a/src/decoder/fluidsynth_plugin.c
+++ b/src/decoder/fluidsynth_plugin.c
@@ -88,7 +88,7 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs)
 {
 	static const struct audio_format audio_format = {
 		.sample_rate = 48000,
-		.bits = 16,
+		.format = SAMPLE_FORMAT_S16,
 		.channels = 2,
 	};
 	char setting_sample_rate[] = "synth.sample-rate";
diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c
index da93fe45b..cba40aea0 100644
--- a/src/decoder/mad_plugin.c
+++ b/src/decoder/mad_plugin.c
@@ -1188,7 +1188,8 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 	}
 
 	if (!audio_format_init_checked(&audio_format,
-				       data.frame.header.samplerate, 24,
+				       data.frame.header.samplerate,
+				       SAMPLE_FORMAT_S24_P32,
 				       MAD_NCHANNELS(&data.frame.header),
 				       &error)) {
 		g_warning("%s", error->message);
diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c
index 204dd5ce0..1e64aeffd 100644
--- a/src/decoder/mikmod_plugin.c
+++ b/src/decoder/mikmod_plugin.c
@@ -163,7 +163,7 @@ 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, 16, 2);
+	audio_format_init(&audio_format, mikmod_sample_rate, SAMPLE_FORMAT_S16, 2);
 	assert(audio_format_valid(&audio_format));
 
 	decoder_initialized(decoder, &audio_format, false, 0);
diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_plugin.c
index 6c08c2199..02292992d 100644
--- a/src/decoder/modplug_plugin.c
+++ b/src/decoder/modplug_plugin.c
@@ -122,7 +122,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
 		return;
 	}
 
-	audio_format_init(&audio_format, 44100, 16, 2);
+	audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2);
 	assert(audio_format_valid(&audio_format));
 
 	decoder_initialized(decoder, &audio_format,
diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c
index 70ca4bdc3..8d3a4b9e9 100644
--- a/src/decoder/mp4ff_plugin.c
+++ b/src/decoder/mp4ff_plugin.c
@@ -132,7 +132,8 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
 		return NULL;
 	}
 
-	if (!audio_format_init_checked(audio_format, sample_rate, 16, channels,
+	if (!audio_format_init_checked(audio_format, sample_rate,
+				       SAMPLE_FORMAT_S16, channels,
 				       &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);
diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c
index d985f8459..2f1936e55 100644
--- a/src/decoder/mpcdec_plugin.c
+++ b/src/decoder/mpcdec_plugin.c
@@ -196,7 +196,8 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 	mpc_demux_get_info(demux, &info);
 #endif
 
-	if (!audio_format_init_checked(&audio_format, info.sample_freq, 24,
+	if (!audio_format_init_checked(&audio_format, info.sample_freq,
+				       SAMPLE_FORMAT_S24_P32,
 				       info.channels, &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);
diff --git a/src/decoder/mpg123_decoder_plugin.c b/src/decoder/mpg123_decoder_plugin.c
index 922e56484..62e6b00b0 100644
--- a/src/decoder/mpg123_decoder_plugin.c
+++ b/src/decoder/mpg123_decoder_plugin.c
@@ -87,7 +87,7 @@ mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
 		return false;
 	}
 
-	if (!audio_format_init_checked(audio_format, rate, 16,
+	if (!audio_format_init_checked(audio_format, rate, SAMPLE_FORMAT_S16,
 				       channels, &gerror)) {
 		g_warning("%s", gerror->message);
 		g_error_free(gerror);
diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx
index b6e557e08..63e46a285 100644
--- a/src/decoder/sidplay_plugin.cxx
+++ b/src/decoder/sidplay_plugin.cxx
@@ -277,7 +277,7 @@ 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, 16, 2);
+	audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, 2);
 	assert(audio_format_valid(&audio_format));
 
 	decoder_initialized(decoder, &audio_format, true, (float)song_len);
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
index 84835c449..fee0f8292 100644
--- a/src/decoder/sndfile_decoder_plugin.c
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -130,7 +130,8 @@ 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, 32,
+	if (!audio_format_init_checked(&audio_format, info.samplerate,
+				       SAMPLE_FORMAT_S32,
 				       info.channels, &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c
index 3a41869a0..cb61e5999 100755
--- a/src/decoder/vorbis_plugin.c
+++ b/src/decoder/vorbis_plugin.c
@@ -311,7 +311,8 @@ vorbis_stream_decode(struct decoder *decoder,
 		return;
 	}
 
-	if (!audio_format_init_checked(&audio_format, vi->rate, 16,
+	if (!audio_format_init_checked(&audio_format, vi->rate,
+				       SAMPLE_FORMAT_S16,
 				       vi->channels, &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
index 9b32a79f2..dffa078f9 100644
--- a/src/decoder/wavpack_plugin.c
+++ b/src/decoder/wavpack_plugin.c
@@ -123,6 +123,33 @@ 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
+wavpack_bits_to_sample_format(bool is_float, int bytes_per_sample)
+{
+	if (is_float)
+		return SAMPLE_FORMAT_S24_P32;
+
+	switch (bytes_per_sample) {
+	case 1:
+		return SAMPLE_FORMAT_S8;
+
+	case 2:
+		return SAMPLE_FORMAT_S16;
+
+	case 3:
+		return SAMPLE_FORMAT_S24_P32;
+
+	case 4:
+		return SAMPLE_FORMAT_S32;
+
+	default:
+		return SAMPLE_FORMAT_UNDEFINED;
+	}
+}
+
 /*
  * This does the main decoding thing.
  * Requires an already opened WavpackContext.
@@ -132,7 +159,8 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 	       struct replay_gain_info *replay_gain_info)
 {
 	GError *error = NULL;
-	unsigned bits;
+	bool is_float;
+	enum sample_format sample_format;
 	struct audio_format audio_format;
 	format_samples_t format_samples;
 	char chunk[CHUNK_SIZE];
@@ -141,19 +169,14 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 	int bytes_per_sample, output_sample_size;
 	int position;
 
-	bits = WavpackGetBitsPerSample(wpc);
-
-	/* round bitwidth to 8-bit units */
-	bits = (bits + 7) & (~7);
-	/* MPD handles max 32-bit samples */
-	if (bits > 32)
-		bits = 32;
-
-	if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
-		bits = 24;
+	is_float = (WavpackGetMode(wpc) & MODE_FLOAT) != 0;
+	sample_format =
+		wavpack_bits_to_sample_format(is_float,
+					      WavpackGetBytesPerSample(wpc));
 
 	if (!audio_format_init_checked(&audio_format,
-				       WavpackGetSampleRate(wpc), bits,
+				       WavpackGetSampleRate(wpc),
+				       sample_format,
 				       WavpackGetNumChannels(wpc), &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);
diff --git a/src/decoder/wildmidi_plugin.c b/src/decoder/wildmidi_plugin.c
index 718f24c2e..70b4d5ef9 100644
--- a/src/decoder/wildmidi_plugin.c
+++ b/src/decoder/wildmidi_plugin.c
@@ -59,7 +59,7 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs)
 {
 	static const struct audio_format audio_format = {
 		.sample_rate = WILDMIDI_SAMPLE_RATE,
-		.bits = 16,
+		.format = SAMPLE_FORMAT_S16,
 		.channels = 2,
 	};
 	midi *wm;