pcm/export: move Open() parameters to struct Params

src/output/plugins/AlsaOutputPlugin.cxx

@@ -716,22 +716,24 @@ AlsaOutput::SetupDop(const AudioFormat audio_format,
 inline bool
 AlsaOutput::SetupOrDop(AudioFormat &audio_format, Error &error)
 {
-	bool shift8 = false, packed, reverse_endian;
+	PcmExport::Params params;
+	params.alsa_channel_order = true;
 
-	const bool dop2 = dop &&
-		audio_format.format == SampleFormat::DSD;
-	const bool success = dop2
+	params.dop = dop && audio_format.format == SampleFormat::DSD;
+	const bool success = params.dop
 		? SetupDop(audio_format,
-			   &shift8, &packed, &reverse_endian,
+			   &params.shift8,
+			   &params.pack24, &params.reverse_endian,
 			   error)
-		: alsa_setup(this, audio_format, &packed, &reverse_endian,
+		: alsa_setup(this, audio_format,
+			     &params.pack24, &params.reverse_endian,
 			     error);
 	if (!success)
 		return false;
 
 	pcm_export->Open(audio_format.format,
 			 audio_format.channels,
-			 true, dop2, shift8, packed, reverse_endian);
+			 params);
 	return true;
 }
 

src/output/plugins/OssOutputPlugin.cxx

@@ -537,10 +537,13 @@ oss_probe_sample_format(int fd, SampleFormat sample_format,
 	*oss_format_r = oss_format;
 
 #ifdef AFMT_S24_PACKED
-	pcm_export.Open(sample_format, 0, true, false, false,
-			oss_format == AFMT_S24_PACKED,
-			oss_format == AFMT_S24_PACKED &&
-			!IsLittleEndian());
+	PcmExport::Params params;
+	params.alsa_channel_order = true;
+	params.pack24 = oss_format == AFMT_S24_PACKED;
+	params.reverse_endian = oss_format == AFMT_S24_PACKED &&
+		!IsLittleEndian();
+
+	pcm_export.Open(sample_format, 0, params);
 #endif
 
 	return SUCCESS;

src/pcm/PcmExport.cxx

@@ -32,35 +32,33 @@
 
 void
 PcmExport::Open(SampleFormat sample_format, unsigned _channels,
-		bool _alsa_channel_order,
-		bool _dop, bool _shift8, bool _pack, bool _reverse_endian)
+		Params params)
 {
 	assert(audio_valid_sample_format(sample_format));
 
 	channels = _channels;
-	alsa_channel_order = _alsa_channel_order
+	alsa_channel_order = params.alsa_channel_order
 		? sample_format
 		: SampleFormat::UNDEFINED;
 
 #ifdef ENABLE_DSD
-	assert(!_dop || audio_valid_channel_count(_channels));
-	dop = _dop && sample_format == SampleFormat::DSD;
+	assert(!params.dop || audio_valid_channel_count(_channels));
+	dop = params.dop && sample_format == SampleFormat::DSD;
 	if (dop)
 		/* after the conversion to DoP, the DSD
 		   samples are stuffed inside fake 24 bit samples */
 		sample_format = SampleFormat::S24_P32;
 #else
 	(void)_channels;
-	(void)_dop;
 #endif
 
-	shift8 = _shift8 && sample_format == SampleFormat::S24_P32;
-	pack24 = _pack && sample_format == SampleFormat::S24_P32;
+	shift8 = params.shift8 && sample_format == SampleFormat::S24_P32;
+	pack24 = params.pack24 && sample_format == SampleFormat::S24_P32;
 
 	assert(!shift8 || !pack24);
 
 	reverse_endian = 0;
-	if (_reverse_endian) {
+	if (params.reverse_endian) {
 		size_t sample_size = pack24
 			? 3
 			: sample_format_size(sample_format);

src/pcm/PcmExport.hxx

@@ -33,6 +33,14 @@ template<typename T> struct ConstBuffer;
  * representation which are not supported by the pcm_convert library.
  */
 struct PcmExport {
+	struct Params {
+		bool alsa_channel_order = false;
+		bool dop = false;
+		bool shift8 = false;
+		bool pack24 = false;
+		bool reverse_endian = false;
+	};
+
 	/**
 	 * This buffer is used to reorder channels.
 	 *
@@ -117,8 +125,7 @@ struct PcmExport {
 	 * @param channels the number of channels; ignored unless dop is set
 	 */
 	void Open(SampleFormat sample_format, unsigned channels,
-		  bool _alsa_channel_order,
-		  bool dop, bool shift8, bool pack, bool reverse_endian);
+		  Params params);
 
 	/**
 	 * Calculate the size of one output frame.