release v0.20.2

If the input AudioFormat changes but the out_audio_format doesn't
change (e.g. because there is a fixed "format" setting in this
"audio_output" section), the ConvertFilter needs to be reconfigured.
This didn't happen, resulting in awful static noise after changing
songs.

Makefile.am

@@ -575,6 +575,8 @@ PCM_LIBS = \
 
 if ENABLE_DSD
 libpcm_a_SOURCES += \
+	src/pcm/Dsd16.cxx src/pcm/Dsd16.hxx \
+	src/pcm/Dsd32.cxx src/pcm/Dsd32.hxx \
 	src/pcm/PcmDsd.cxx src/pcm/PcmDsd.hxx \
 	src/pcm/dsd2pcm/dsd2pcm.c src/pcm/dsd2pcm/dsd2pcm.h
 endif

NEWS

@@ -1,3 +1,20 @@
+ver 0.20.2 (2017/01/15)
+* input
+  - alsa: fix crash bug
+  - alsa: fix buffer overruns
+* decoder
+  - flac: add options "probesize" and "analyzeduration"
+* resampler
+  - libsamplerate: reset state after seeking
+* output
+  - fix static noise after changing to a different audio format
+  - alsa: fix the DSD_U32 sample rate
+  - alsa: fix the DSD_U32 byte order
+  - alsa: support DSD_U16
+  - recorder: fix error "Failed to create : No such file or directory"
+* playlist
+  - cue: fix skipping songs
+
 ver 0.20.1 (2017/01/09)
 * input
   - curl: fix crash bug

configure.ac

@@ -1,6 +1,6 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.20.1, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.20.2, musicpd-dev-team@lists.sourceforge.net)
 
 VERSION_MAJOR=0
 VERSION_MINOR=20

doc/user.xml

@@ -2206,6 +2206,48 @@ run</programlisting>
           Decodes various codecs using
           <application>FFmpeg</application>.
         </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>analyzeduration</varname>
+                  <parameter>VALUE</parameter>
+                </entry>
+                <entry>
+                  Sets the FFmpeg muxer option
+                  <varname>analyzeduration</varname>, which specifies
+                  how many microseconds are analyzed to probe the
+                  input.  The <ulink
+                  url="https://ffmpeg.org/ffmpeg-formats.html">FFmpeg
+                  formats documentation</ulink> has more information.
+                </entry>
+              </row>
+
+              <row>
+                <entry>
+                  <varname>probesize</varname>
+                  <parameter>VALUE</parameter>
+                </entry>
+                <entry>
+                  Sets the FFmpeg muxer option
+                  <varname>probesize</varname>, which specifies
+                  probing size in bytes, i.e. the size of the data to
+                  analyze to get stream information.  The <ulink
+                  url="https://ffmpeg.org/ffmpeg-formats.html">FFmpeg
+                  formats documentation</ulink> has more information.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
       </section>
 
       <section id="flac_decoder">

src/DetachedSong.hxx

@@ -146,7 +146,9 @@ public:
 	 */
 	gcc_pure
 	bool IsSame(const DetachedSong &other) const {
-		return uri == other.uri;
+		return uri == other.uri &&
+			start_time == other.start_time &&
+			end_time == other.end_time;
 	}
 
 	gcc_pure gcc_nonnull_all

src/decoder/Bridge.cxx

@@ -317,6 +317,9 @@ DecoderBridge::CommandFinished()
 
 		dc.pipe->Clear(*dc.buffer);
 
+		if (convert != nullptr)
+			convert->Reset();
+
 		timestamp = dc.seek_time.ToDoubleS();
 	}
 

src/decoder/plugins/FfmpegDecoderPlugin.cxx

@@ -56,6 +56,11 @@ extern "C" {
 #include <assert.h>
 #include <string.h>
 
+/**
+ * Muxer options to be passed to avformat_open_input().
+ */
+static AVDictionary *avformat_options = nullptr;
+
 static AVFormatContext *
 FfmpegOpenInput(AVIOContext *pb,
 		const char *filename,
@@ -67,7 +72,11 @@ FfmpegOpenInput(AVIOContext *pb,
 
 	context->pb = pb;
 
-	int err = avformat_open_input(&context, filename, fmt, nullptr);
+	AVDictionary *options = nullptr;
+	AtScopeExit(&options) { av_dict_free(&options); };
+	av_dict_copy(&options, avformat_options, 0);
+
+	int err = avformat_open_input(&context, filename, fmt, &options);
 	if (err < 0)
 		throw MakeFfmpegError(err, "avformat_open_input() failed");
 
@@ -75,12 +84,30 @@ FfmpegOpenInput(AVIOContext *pb,
 }
 
 static bool
-ffmpeg_init(gcc_unused const ConfigBlock &block)
+ffmpeg_init(const ConfigBlock &block)
 {
 	FfmpegInit();
+
+	static constexpr const char *option_names[] = {
+		"probesize",
+		"analyzeduration",
+	};
+
+	for (const char *name : option_names) {
+		const char *value = block.GetBlockValue(name);
+		if (value != nullptr)
+			av_dict_set(&avformat_options, name, value, 0);
+	}
+
 	return true;
 }
 
+static void
+ffmpeg_finish()
+{
+	av_dict_free(&avformat_options);
+}
+
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(57, 25, 0) /* FFmpeg 3.1 */
 
 gcc_pure
@@ -967,7 +994,7 @@ static const char *const ffmpeg_mime_types[] = {
 const struct DecoderPlugin ffmpeg_decoder_plugin = {
 	"ffmpeg",
 	ffmpeg_init,
-	nullptr,
+	ffmpeg_finish,
 	ffmpeg_decode,
 	nullptr,
 	nullptr,

src/filter/FilterInternal.hxx

@@ -52,6 +52,12 @@ public:
 		return out_audio_format;
 	}
 
+	/**
+	 * Reset the filter's state, e.g. drop/flush buffers.
+	 */
+	virtual void Reset() {
+	}
+
 	/**
 	 * Filters a block of PCM data.
 	 *
@@ -60,7 +66,7 @@ public:
 	 * @param src the input buffer
 	 * @return the destination buffer on success (will be
 	 * invalidated by deleting this object or the next FilterPCM()
-	 * call)
+	 * or Reset() call)
 	 */
 	virtual ConstBuffer<void> FilterPCM(ConstBuffer<void> src) = 0;
 };

src/filter/plugins/AutoConvertFilterPlugin.cxx

@@ -48,6 +48,13 @@ public:
 		:Filter(_filter->GetOutAudioFormat()),
 		 filter(std::move(_filter)), convert(std::move(_convert)) {}
 
+	void Reset() override {
+		filter->Reset();
+
+		if (convert)
+			convert->Reset();
+	}
+
 	ConstBuffer<void> FilterPCM(ConstBuffer<void> src) override;
 };
 

src/filter/plugins/ChainFilterPlugin.cxx

@@ -61,6 +61,7 @@ public:
 	}
 
 	/* virtual methods from class Filter */
+	void Reset() override;
 	ConstBuffer<void> FilterPCM(ConstBuffer<void> src) override;
 };
 
@@ -130,6 +131,13 @@ PreparedChainFilter::Open(AudioFormat &in_audio_format)
 	return chain.release();
 }
 
+void
+ChainFilter::Reset()
+{
+	for (auto &child : children)
+		child.filter->Reset();
+}
+
 ConstBuffer<void>
 ChainFilter::FilterPCM(ConstBuffer<void> src)
 {

src/filter/plugins/ConvertFilterPlugin.cxx

@@ -52,6 +52,10 @@ public:
 
 	void Set(const AudioFormat &_out_audio_format);
 
+	void Reset() override {
+		state.Reset();
+	}
+
 	ConstBuffer<void> FilterPCM(ConstBuffer<void> src) override;
 };
 

src/input/plugins/AlsaInputPlugin.cxx

@@ -28,6 +28,7 @@
 #include "AlsaInputPlugin.hxx"
 #include "../InputPlugin.hxx"
 #include "../AsyncInputStream.hxx"
+#include "event/Call.hxx"
 #include "util/Domain.hxx"
 #include "util/RuntimeError.hxx"
 #include "util/StringCompare.hxx"
@@ -56,31 +57,30 @@ static constexpr unsigned int default_rate = 44100; // cd quality
 static constexpr size_t ALSA_MAX_BUFFERED = default_rate * default_channels * 2;
 static constexpr size_t ALSA_RESUME_AT = ALSA_MAX_BUFFERED / 2;
 
-/**
- * This value should be the same as the read buffer size defined in
- * PcmDecoderPlugin.cxx:pcm_stream_decode().
- * We use it to calculate how many audio frames to buffer in the alsa driver
- * before reading from the device. snd_pcm_readi() blocks until that many
- * frames are ready.
- */
-static constexpr size_t read_buffer_size = 4096;
-
 class AlsaInputStream final
 	: public AsyncInputStream,
 	  MultiSocketMonitor, DeferredMonitor {
-	snd_pcm_t *capture_handle;
-	size_t frame_size;
+
+	/**
+	 * The configured name of the ALSA device.
+	 */
+	const std::string device;
+
+	snd_pcm_t *const capture_handle;
+	const size_t frame_size;
 
 	ReusableArray<pollfd> pfd_buffer;
 
 public:
 	AlsaInputStream(EventLoop &loop,
 			const char *_uri, Mutex &_mutex, Cond &_cond,
+			const char *_device,
 			snd_pcm_t *_handle, int _frame_size)
 		:AsyncInputStream(_uri, _mutex, _cond,
 				  ALSA_MAX_BUFFERED, ALSA_RESUME_AT),
 		 MultiSocketMonitor(loop),
 		 DeferredMonitor(loop),
+		 device(_device),
 		 capture_handle(_handle),
 		 frame_size(_frame_size)
 	{
@@ -99,6 +99,14 @@ public:
 	}
 
 	~AlsaInputStream() {
+		/* ClearSocketList must be called from within the
+		   IOThread; if we don't do it manually here, the
+		   ~MultiSocketMonitor() will do it in the current
+		   thread */
+		BlockingCall(MultiSocketMonitor::GetEventLoop(), [this](){
+				ClearSocketList();
+			});
+
 		snd_pcm_close(capture_handle);
 	}
 
@@ -162,7 +170,7 @@ AlsaInputStream::Create(const char *uri, Mutex &mutex, Cond &cond)
 	int frame_size = snd_pcm_format_width(format) / 8 * channels;
 	return new AlsaInputStream(io_thread_get(),
 				   uri, mutex, cond,
-				   handle, frame_size);
+				   device, handle, frame_size);
 }
 
 std::chrono::steady_clock::duration
@@ -205,6 +213,9 @@ AlsaInputStream::DispatchSockets()
 	snd_pcm_sframes_t n_frames;
 	while ((n_frames = snd_pcm_readi(capture_handle,
 					 w.data, w_frames)) < 0) {
+		if (n_frames == -EAGAIN)
+			return;
+
 		if (Recover(n_frames) < 0) {
 			postponed_exception = std::make_exception_ptr(std::runtime_error("PCM error - stream aborted"));
 			cond.broadcast();
@@ -221,20 +232,51 @@ AlsaInputStream::Recover(int err)
 {
 	switch(err) {
 	case -EPIPE:
-		LogDebug(alsa_input_domain, "Buffer Overrun");
-		// drop through
+		FormatDebug(alsa_input_domain,
+			    "Overrun on ALSA capture device \"%s\"",
+			    device.c_str());
+		break;
+
+	case -ESTRPIPE:
+		FormatDebug(alsa_input_domain,
+			    "ALSA capture device \"%s\" was suspended",
+			    device.c_str());
+		break;
+	}
+
+	switch (snd_pcm_state(capture_handle)) {
+	case SND_PCM_STATE_PAUSED:
+		err = snd_pcm_pause(capture_handle, /* disable */ 0);
+		break;
+
+	case SND_PCM_STATE_SUSPENDED:
+		err = snd_pcm_resume(capture_handle);
+		if (err == -EAGAIN)
+			return 0;
+		/* fall-through to snd_pcm_prepare: */
 #if GCC_CHECK_VERSION(7,0)
 		[[fallthrough]];
 #endif
-
-	case -ESTRPIPE:
-	case -EINTR:
-		err = snd_pcm_recover(capture_handle, err, 1);
+	case SND_PCM_STATE_OPEN:
+	case SND_PCM_STATE_SETUP:
+	case SND_PCM_STATE_XRUN:
+		err = snd_pcm_prepare(capture_handle);
+		if (err == 0)
+			err = snd_pcm_start(capture_handle);
+		break;
+
+	case SND_PCM_STATE_DISCONNECTED:
+		break;
+
+	case SND_PCM_STATE_PREPARED:
+	case SND_PCM_STATE_RUNNING:
+	case SND_PCM_STATE_DRAINING:
+		/* this is no error, so just keep running */
+		err = 0;
 		break;
-	default:
-		// something broken somewhere, give up
-		err = -1;
 	}
+
+
 	return err;
 }
 
@@ -273,23 +315,63 @@ ConfigureCapture(snd_pcm_t *capture_handle,
 		throw FormatRuntimeError("Cannot set sample rate (%s)",
 					 snd_strerror(err));
 
-	/* period needs to be big enough so that poll() doesn't fire too often,
-	 * but small enough that buffer overruns don't occur if Read() is not
-	 * invoked often enough.
-	 * the calculation here is empirical; however all measurements were
-	 * done using 44100:16:2. When we extend this plugin to support
-	 * other audio formats then this may need to be revisited */
-	snd_pcm_uframes_t period = read_buffer_size * 2;
-	int direction = -1;
-	if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params,
-							  &period, &direction)) < 0)
-		throw FormatRuntimeError("Cannot set period size (%s)",
-					 snd_strerror(err));
+	snd_pcm_uframes_t buffer_size_min, buffer_size_max;
+	snd_pcm_hw_params_get_buffer_size_min(hw_params, &buffer_size_min);
+	snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size_max);
+	unsigned buffer_time_min, buffer_time_max;
+	snd_pcm_hw_params_get_buffer_time_min(hw_params, &buffer_time_min, 0);
+	snd_pcm_hw_params_get_buffer_time_max(hw_params, &buffer_time_max, 0);
+	FormatDebug(alsa_input_domain, "buffer: size=%u..%u time=%u..%u",
+		    (unsigned)buffer_size_min, (unsigned)buffer_size_max,
+		    buffer_time_min, buffer_time_max);
+
+	snd_pcm_uframes_t period_size_min, period_size_max;
+	snd_pcm_hw_params_get_period_size_min(hw_params, &period_size_min, 0);
+	snd_pcm_hw_params_get_period_size_max(hw_params, &period_size_max, 0);
+	unsigned period_time_min, period_time_max;
+	snd_pcm_hw_params_get_period_time_min(hw_params, &period_time_min, 0);
+	snd_pcm_hw_params_get_period_time_max(hw_params, &period_time_max, 0);
+	FormatDebug(alsa_input_domain, "period: size=%u..%u time=%u..%u",
+		    (unsigned)period_size_min, (unsigned)period_size_max,
+		    period_time_min, period_time_max);
+
+	/* choose the maximum possible buffer_size ... */
+	snd_pcm_hw_params_set_buffer_size(capture_handle, hw_params,
+					  buffer_size_max);
+
+	/* ... and calculate the period_size to have four periods in
+	   one buffer; this way, we get woken up often enough to avoid
+	   buffer overruns, but not too often */
+	snd_pcm_uframes_t buffer_size;
+	if (snd_pcm_hw_params_get_buffer_size(hw_params, &buffer_size) == 0) {
+		snd_pcm_uframes_t period_size = buffer_size / 4;
+		int direction = -1;
+		if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params,
+								  &period_size, &direction)) < 0)
+			throw FormatRuntimeError("Cannot set period size (%s)",
+						 snd_strerror(err));
+	}
 
 	if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0)
 		throw FormatRuntimeError("Cannot set parameters (%s)",
 					 snd_strerror(err));
 
+	snd_pcm_uframes_t alsa_buffer_size;
+	err = snd_pcm_hw_params_get_buffer_size(hw_params, &alsa_buffer_size);
+	if (err < 0)
+		throw FormatRuntimeError("snd_pcm_hw_params_get_buffer_size() failed: %s",
+					 snd_strerror(-err));
+
+	snd_pcm_uframes_t alsa_period_size;
+	err = snd_pcm_hw_params_get_period_size(hw_params, &alsa_period_size,
+						nullptr);
+	if (err < 0)
+		throw FormatRuntimeError("snd_pcm_hw_params_get_period_size() failed: %s",
+					 snd_strerror(-err));
+
+	FormatDebug(alsa_input_domain, "buffer_size=%u period_size=%u",
+		    (unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
+
 	snd_pcm_sw_params_t *sw_params;
 
 	snd_pcm_sw_params_malloc(&sw_params);
@@ -299,11 +381,6 @@ ConfigureCapture(snd_pcm_t *capture_handle,
 		snd_pcm_sw_params_free(sw_params);
 	};
 
-	if ((err = snd_pcm_sw_params_set_start_threshold(capture_handle, sw_params,
-							 period)) < 0)
-		throw FormatRuntimeError("unable to set start threshold (%s)",
-					 snd_strerror(err));
-
 	if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0)
 		throw FormatRuntimeError("unable to install sw params (%s)",
 					 snd_strerror(err));
@@ -316,7 +393,8 @@ AlsaInputStream::OpenDevice(const char *device,
 	snd_pcm_t *capture_handle;
 	int err;
 	if ((err = snd_pcm_open(&capture_handle, device,
-				SND_PCM_STREAM_CAPTURE, 0)) < 0)
+				SND_PCM_STREAM_CAPTURE,
+				SND_PCM_NONBLOCK)) < 0)
 		throw FormatRuntimeError("Failed to open device: %s (%s)",
 					 device, snd_strerror(err));
 

src/lib/nfs/Callback.hxx

@@ -24,9 +24,21 @@
 
 #include <exception>
 
+/**
+ * Callbacks for an asynchronous libnfs operation.
+ *
+ * Note that no callback is invoked for cancelled operations.
+ */
 class NfsCallback {
 public:
+	/**
+	 * The operation completed successfully.
+	 */
 	virtual void OnNfsCallback(unsigned status, void *data) = 0;
+
+	/**
+	 * An error has occurred.
+	 */
 	virtual void OnNfsError(std::exception_ptr &&e) = 0;
 };
 

src/lib/nfs/FileReader.hxx

@@ -35,6 +35,14 @@
 struct nfsfh;
 class NfsConnection;
 
+/**
+ * A helper class which helps with reading from a file.  It obtains a
+ * connection lease (#NfsLease), opens the given file, "stats" the
+ * file, and finally allos you to read its contents.
+ *
+ * To get started, derive your class from it and implement the pure
+ * virtual methods, construct an instance, and call Open().
+ */
 class NfsFileReader : NfsLease, NfsCallback, DeferredMonitor {
 	enum class State {
 		INITIAL,
@@ -63,14 +71,30 @@ public:
 	void DeferClose();
 
 	/**
+	 * Open the file.  This method is thread-safe.
+	 *
 	 * Throws std::runtime_error on error.
 	 */
 	void Open(const char *uri);
 
 	/**
+	 * Attempt to read from the file.  This may only be done after
+	 * OnNfsFileOpen() has been called.  Only one read operation
+	 * may be performed at a time.
+	 *
+	 * This method is not thread-safe and must be called from
+	 * within the I/O thread.
+	 *
 	 * Throws std::runtime_error on error.
 	 */
 	void Read(uint64_t offset, size_t size);
+
+	/**
+	 * Cancel the most recent Read() call.
+	 *
+	 * This method is not thread-safe and must be called from
+	 * within the I/O thread.
+	 */
 	void CancelRead();
 
 	bool IsIdle() const {
@@ -78,8 +102,27 @@ public:
 	}
 
 protected:
+	/**
+	 * The file has been opened successfully.  It is a regular
+	 * file, and its size is known.  It is ready to be read from
+	 * using Read().
+	 *
+	 * This method will be called from within the I/O thread.
+	 */
 	virtual void OnNfsFileOpen(uint64_t size) = 0;
+
+	/**
+	 * A Read() has completed successfully.
+	 *
+	 * This method will be called from within the I/O thread.
+	 */
 	virtual void OnNfsFileRead(const void *data, size_t size) = 0;
+
+	/**
+	 * An error has occurred, which can be either while waiting
+	 * for OnNfsFileOpen(), or while waiting for OnNfsFileRead(),
+	 * or if disconnected while idle.
+	 */
 	virtual void OnNfsFileError(std::exception_ptr &&e) = 0;
 
 private:

src/output/OutputThread.cxx

@@ -111,8 +111,7 @@ AudioOutput::Open()
 		f = source.Open(request.audio_format, *request.pipe,
 				prepared_replay_gain_filter,
 				prepared_other_replay_gain_filter,
-				prepared_filter)
-			.WithMask(config_audio_format);
+				prepared_filter);
 
 		if (mixer != nullptr && mixer->IsPlugin(software_mixer_plugin))
 			software_mixer_set_filter(*mixer, volume_filter.Get());
@@ -121,14 +120,16 @@ AudioOutput::Open()
 							  name, plugin.name));
 	}
 
-	if (open && f != filter_audio_format) {
+	const auto cf = f.WithMask(config_audio_format);
+
+	if (open && cf != filter_audio_format) {
 		/* if the filter's output format changes, the output
 		   must be reopened as well */
 		CloseOutput(true);
 		open = false;
 	}
 
-	filter_audio_format = f;
+	filter_audio_format = cf;
 
 	if (!open) {
 		try {
@@ -139,6 +140,27 @@ AudioOutput::Open()
 		}
 
 		open = true;
+	} else if (f != out_audio_format) {
+		/* reconfigure the final ConvertFilter for its new
+		   input AudioFormat */
+
+		try {
+			convert_filter_set(convert_filter.Get(),
+					   out_audio_format);
+		} catch (const std::runtime_error &e) {
+			Close(false);
+			std::throw_with_nested(FormatRuntimeError("Failed to convert for \"%s\" [%s]",
+								  name, plugin.name));
+		}
+	}
+
+	if (f != source.GetInputAudioFormat() || f != out_audio_format) {
+		struct audio_format_string afs1, afs2, afs3;
+		FormatDebug(output_domain, "converting in=%s -> f=%s -> out=%s",
+			    audio_format_to_string(source.GetInputAudioFormat(),
+						   &afs1),
+			    audio_format_to_string(f, &afs2),
+			    audio_format_to_string(out_audio_format, &afs3));
 	}
 }
 
@@ -154,6 +176,12 @@ AudioOutput::OpenOutputAndConvert(AudioFormat desired_audio_format)
 							  name, plugin.name));
 	}
 
+	struct audio_format_string af_string;
+	FormatDebug(output_domain,
+		    "opened plugin=%s name=\"%s\" audio_format=%s",
+		    plugin.name, name,
+		    audio_format_to_string(out_audio_format, &af_string));
+
 	try {
 		convert_filter_set(convert_filter.Get(), out_audio_format);
 	} catch (const std::runtime_error &e) {
@@ -177,17 +205,6 @@ AudioOutput::OpenOutputAndConvert(AudioFormat desired_audio_format)
 		std::throw_with_nested(FormatRuntimeError("Failed to convert for \"%s\" [%s]",
 							  name, plugin.name));
 	}
-
-	struct audio_format_string af_string;
-	FormatDebug(output_domain,
-		    "opened plugin=%s name=\"%s\" audio_format=%s",
-		    plugin.name, name,
-		    audio_format_to_string(out_audio_format, &af_string));
-
-	if (source.GetInputAudioFormat() != out_audio_format)
-		FormatDebug(output_domain, "converting from %s",
-			    audio_format_to_string(source.GetInputAudioFormat(),
-						   &af_string));
 }
 
 void

src/output/Source.cxx

@@ -70,6 +70,22 @@ AudioOutputSource::Close()
 	CloseFilter();
 }
 
+void
+AudioOutputSource::Cancel()
+{
+	current_chunk = nullptr;
+	pipe.Cancel();
+
+	if (replay_gain_filter_instance != nullptr)
+		replay_gain_filter_instance->Reset();
+
+	if (other_replay_gain_filter_instance != nullptr)
+		other_replay_gain_filter_instance->Reset();
+
+	if (filter_instance != nullptr)
+		filter_instance->Reset();
+}
+
 void
 AudioOutputSource::OpenFilter(AudioFormat audio_format,
 			      PreparedFilter *prepared_replay_gain_filter,

src/output/Source.hxx

@@ -139,11 +139,7 @@ public:
 			 PreparedFilter *prepared_filter);
 
 	void Close();
-
-	void Cancel() {
-		current_chunk = nullptr;
-		pipe.Cancel();
-	}
+	void Cancel();
 
 	/**
 	 * Ensure that ReadTag() or PeekData() return any input.

src/output/plugins/AlsaOutputPlugin.cxx

@@ -141,6 +141,7 @@ struct AlsaOutput {
 	void Open(AudioFormat &audio_format);
 	void Close();
 
+	size_t PlayRaw(ConstBuffer<void> data);
 	size_t Play(const void *chunk, size_t size);
 	void Drain();
 	void Cancel();
@@ -384,7 +385,7 @@ AlsaTryFormatOrByteSwap(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 
 /**
  * Attempts to configure the specified sample format.  On DSD_U8
- * failure, attempt to switch to DSD_U32.
+ * failure, attempt to switch to DSD_U32 or DSD_U16.
  */
 static int
 AlsaTryFormatDsd(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
@@ -393,8 +394,10 @@ AlsaTryFormatDsd(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 	int err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
 
 #if defined(ENABLE_DSD) && defined(HAVE_ALSA_DSD_U32)
-	if (err == 0)
+	if (err == 0) {
+		params.dsd_u16 = false;
 		params.dsd_u32 = false;
+	}
 
 	if (err == -EINVAL && fmt == SND_PCM_FORMAT_DSD_U8) {
 		/* attempt to switch to DSD_U32 */
@@ -404,6 +407,20 @@ AlsaTryFormatDsd(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 		err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
 		if (err == 0)
 			params.dsd_u32 = true;
+		else
+			fmt = SND_PCM_FORMAT_DSD_U8;
+	}
+
+	if (err == -EINVAL && fmt == SND_PCM_FORMAT_DSD_U8) {
+		/* attempt to switch to DSD_U16 */
+		fmt = IsLittleEndian()
+			? SND_PCM_FORMAT_DSD_U16_LE
+			: SND_PCM_FORMAT_DSD_U16_BE;
+		err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
+		if (err == 0)
+			params.dsd_u16 = true;
+		else
+			fmt = SND_PCM_FORMAT_DSD_U8;
 	}
 #endif
 
@@ -469,7 +486,6 @@ static void
 AlsaSetup(AlsaOutput *ad, AudioFormat &audio_format,
 	  PcmExport::Params &params)
 {
-	unsigned int sample_rate = audio_format.sample_rate;
 	unsigned int channels = audio_format.channels;
 	int err;
 	unsigned retry = MPD_ALSA_RETRY_NR;
@@ -513,18 +529,23 @@ configure_hw:
 
 	audio_format.channels = (int8_t)channels;
 
+	const unsigned requested_sample_rate =
+		params.CalcOutputSampleRate(audio_format.sample_rate);
+	unsigned output_sample_rate = requested_sample_rate;
+
 	err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
-					      &sample_rate, nullptr);
+					      &output_sample_rate, nullptr);
 	if (err < 0)
 		throw FormatRuntimeError("Failed to configure sample rate %u Hz: %s",
-					 audio_format.sample_rate,
+					 requested_sample_rate,
 					 snd_strerror(-err));
 
-	if (sample_rate == 0)
+	if (output_sample_rate == 0)
 		throw FormatRuntimeError("Failed to configure sample rate %u Hz",
 					 audio_format.sample_rate);
 
-	audio_format.sample_rate = sample_rate;
+	if (output_sample_rate != requested_sample_rate)
+		audio_format.sample_rate = params.CalcInputSampleRate(output_sample_rate);
 
 	snd_pcm_uframes_t buffer_size_min, buffer_size_max;
 	snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min);
@@ -662,7 +683,6 @@ AlsaOutput::SetupDop(const AudioFormat audio_format,
 
 	AudioFormat dop_format = audio_format;
 	dop_format.format = SampleFormat::S24_P32;
-	dop_format.sample_rate /= 2;
 
 	const AudioFormat check = dop_format;
 
@@ -694,11 +714,12 @@ AlsaOutput::SetupOrDop(AudioFormat &audio_format, PcmExport::Params &params)
 	std::exception_ptr dop_error;
 	if (dop && audio_format.format == SampleFormat::DSD) {
 		try {
-			SetupDop(audio_format, params);
 			params.dop = true;
+			SetupDop(audio_format, params);
 			return;
 		} catch (...) {
 			dop_error = std::current_exception();
+			params.dop = false;
 		}
 	}
 
@@ -742,6 +763,11 @@ AlsaOutput::Open(AudioFormat &audio_format)
 							  GetDevice()));
 	}
 
+#ifdef ENABLE_DSD
+	if (params.dop)
+		FormatDebug(alsa_output_domain, "DoP (DSD over PCM) enabled");
+#endif
+
 	pcm_export->Open(audio_format.format,
 			 audio_format.channels,
 			 params);
@@ -777,6 +803,7 @@ AlsaOutput::Recover(int err)
 #if GCC_CHECK_VERSION(7,0)
 		[[fallthrough]];
 #endif
+	case SND_PCM_STATE_OPEN:
 	case SND_PCM_STATE_SETUP:
 	case SND_PCM_STATE_XRUN:
 		period_position = 0;
@@ -785,12 +812,11 @@ AlsaOutput::Recover(int err)
 	case SND_PCM_STATE_DISCONNECTED:
 		break;
 	/* this is no error, so just keep running */
+	case SND_PCM_STATE_PREPARED:
 	case SND_PCM_STATE_RUNNING:
+	case SND_PCM_STATE_DRAINING:
 		err = 0;
 		break;
-	default:
-		/* unknown state, do nothing */
-		break;
 	}
 
 	return err;
@@ -822,6 +848,8 @@ AlsaOutput::Cancel()
 	must_prepare = true;
 
 	snd_pcm_drop(pcm);
+
+	pcm_export->Reset();
 }
 
 inline void
@@ -831,6 +859,36 @@ AlsaOutput::Close()
 	delete[] silence;
 }
 
+inline size_t
+AlsaOutput::PlayRaw(ConstBuffer<void> data)
+{
+	if (data.IsEmpty())
+		return 0;
+
+	assert(data.size % out_frame_size == 0);
+
+	const size_t n_frames = data.size / out_frame_size;
+	assert(n_frames > 0);
+
+	while (true) {
+		const auto frames_written = snd_pcm_writei(pcm, data.data,
+							   n_frames);
+		if (frames_written > 0) {
+			period_position = (period_position + frames_written)
+				% period_frames;
+
+			return frames_written * out_frame_size;
+		}
+
+		if (frames_written < 0 && frames_written != -EAGAIN &&
+		    frames_written != -EINTR &&
+		    Recover(frames_written) < 0)
+			throw FormatRuntimeError("snd_pcm_writei() failed: %s",
+						 snd_strerror(-frames_written));
+	}
+
+}
+
 inline size_t
 AlsaOutput::Play(const void *chunk, size_t size)
 {
@@ -856,29 +914,8 @@ AlsaOutput::Play(const void *chunk, size_t size)
 		   been played */
 		return size;
 
-	chunk = e.data;
-	size = e.size;
-
-	assert(size % out_frame_size == 0);
-
-	size /= out_frame_size;
-	assert(size > 0);
-
-	while (true) {
-		snd_pcm_sframes_t ret = snd_pcm_writei(pcm, chunk, size);
-		if (ret > 0) {
-			period_position = (period_position + ret)
-				% period_frames;
-
-			size_t bytes_written = ret * out_frame_size;
-			return pcm_export->CalcSourceSize(bytes_written);
-		}
-
-		if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
-		    Recover(ret) < 0)
-			throw FormatRuntimeError("snd_pcm_writei() failed: %s",
-						 snd_strerror(-ret));
-	}
+	const size_t bytes_written = PlayRaw(e);
+	return pcm_export->CalcSourceSize(bytes_written);
 }
 
 typedef AudioOutputWrapper<AlsaOutput> Wrapper;

src/output/plugins/OssOutputPlugin.cxx

@@ -659,6 +659,10 @@ OssOutput::Cancel()
 		ioctl(fd, SNDCTL_DSP_RESET, 0);
 		DoClose();
 	}
+
+#ifdef AFMT_S24_PACKED
+	pcm_export->Reset();
+#endif
 }
 
 inline size_t

src/output/plugins/RecorderOutputPlugin.cxx

@@ -274,7 +274,7 @@ RecorderOutput::ReopenFormat(AllocatedPath &&new_path)
 	assert(path.IsNull());
 	assert(file == nullptr);
 
-	FileOutputStream *new_file = new FileOutputStream(path);
+	FileOutputStream *new_file = new FileOutputStream(new_path);
 
 	AudioFormat new_audio_format = effective_audio_format;
 

src/pcm/Dsd16.cxx

@@ -0,0 +1,62 @@
+/*
+ * Copyright 2003-2017 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.
+ */
+
+#include "config.h"
+#include "Dsd16.hxx"
+#include "PcmBuffer.hxx"
+#include "util/ConstBuffer.hxx"
+
+/**
+ * Construct a 16 bit integer from two bytes.
+ */
+static constexpr inline uint16_t
+Construct16(uint8_t a, uint8_t b)
+{
+	/* "a" is the oldest byte, which must be in the most
+	   significant byte */
+
+	return uint16_t(b) | (uint16_t(a) << 8);
+}
+
+static constexpr inline uint16_t
+Dsd8To16Sample(const uint8_t *src, unsigned channels)
+{
+	return Construct16(src[0], src[channels]);
+}
+
+ConstBuffer<uint16_t>
+Dsd8To16(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> _src)
+{
+	const size_t in_frames = _src.size / channels;
+	const size_t out_frames = in_frames / 2;
+	const size_t out_samples = out_frames * channels;
+
+	const uint8_t *src = _src.data;
+	uint16_t *const dest0 = buffer.GetT<uint16_t>(out_samples);
+	uint16_t *dest = dest0;
+
+	for (size_t i = 0; i < out_frames; ++i) {
+		for (size_t c = 0; c < channels; ++c)
+			*dest++ = Dsd8To16Sample(src++, channels);
+
+		src += channels;
+	}
+
+	return {dest0, out_samples};
+}

src/pcm/Dsd16.hxx

@@ -0,0 +1,36 @@
+/*
+ * Copyright 2003-2017 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_PCM_DSD_16_HXX
+#define MPD_PCM_DSD_16_HXX
+
+#include "check.h"
+
+#include <stdint.h>
+
+template<typename T> struct ConstBuffer;
+class PcmBuffer;
+
+/**
+ * Convert DSD_U8 to DSD_U16 (native endian, oldest bits in MSB).
+ */
+ConstBuffer<uint16_t>
+Dsd8To16(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> src);
+
+#endif

src/pcm/Dsd32.cxx

@@ -0,0 +1,64 @@
+/*
+ * Copyright 2003-2017 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.
+ */
+
+#include "config.h"
+#include "Dsd32.hxx"
+#include "PcmBuffer.hxx"
+#include "util/ConstBuffer.hxx"
+
+/**
+ * Construct a 32 bit integer from four bytes.
+ */
+static constexpr inline uint32_t
+Construct32(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
+{
+	/* "a" is the oldest byte, which must be in the most
+	   significant byte */
+
+	return uint32_t(d) | (uint32_t(c) << 8) |
+		(uint32_t(b) << 16) | (uint32_t(a) << 24);
+}
+
+static constexpr inline uint32_t
+Dsd8To32Sample(const uint8_t *src, unsigned channels)
+{
+	return Construct32(src[0], src[channels],
+			   src[2 * channels], src[3 * channels]);
+}
+
+ConstBuffer<uint32_t>
+Dsd8To32(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> _src)
+{
+	const size_t in_frames = _src.size / channels;
+	const size_t out_frames = in_frames / 4;
+	const size_t out_samples = out_frames * channels;
+
+	const uint8_t *src = _src.data;
+	uint32_t *const dest0 = buffer.GetT<uint32_t>(out_samples);
+	uint32_t *dest = dest0;
+
+	for (size_t i = 0; i < out_frames; ++i) {
+		for (size_t c = 0; c < channels; ++c)
+			*dest++ = Dsd8To32Sample(src++, channels);
+
+		src += 3 * channels;
+	}
+
+	return {dest0, out_samples};
+}

src/pcm/Dsd32.hxx

@@ -0,0 +1,36 @@
+/*
+ * Copyright 2003-2017 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_PCM_DSD_32_HXX
+#define MPD_PCM_DSD_32_HXX
+
+#include "check.h"
+
+#include <stdint.h>
+
+template<typename T> struct ConstBuffer;
+class PcmBuffer;
+
+/**
+ * Convert DSD_U8 to DSD_U32 (native endian, oldest bits in MSB).
+ */
+ConstBuffer<uint32_t>
+Dsd8To32(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> src);
+
+#endif

src/pcm/GlueResampler.cxx

@@ -65,6 +65,12 @@ GluePcmResampler::Close()
 	resampler->Close();
 }
 
+void
+GluePcmResampler::Reset()
+{
+	resampler->Reset();
+}
+
 ConstBuffer<void>
 GluePcmResampler::Resample(ConstBuffer<void> src)
 {

src/pcm/GlueResampler.hxx

@@ -55,6 +55,11 @@ public:
 		return output_sample_format;
 	}
 
+	/**
+	 * @see PcmResampler::Reset()
+	 */
+	void Reset();
+
 	ConstBuffer<void> Resample(ConstBuffer<void> src);
 };
 

src/pcm/LibsamplerateResampler.cxx

@@ -114,6 +114,12 @@ LibsampleratePcmResampler::Close()
 	state = src_delete(state);
 }
 
+void
+LibsampleratePcmResampler::Reset()
+{
+	src_reset(state);
+}
+
 inline ConstBuffer<float>
 LibsampleratePcmResampler::Resample2(ConstBuffer<float> src)
 {

src/pcm/LibsamplerateResampler.hxx

@@ -44,6 +44,7 @@ class LibsampleratePcmResampler final : public PcmResampler {
 public:
 	AudioFormat Open(AudioFormat &af, unsigned new_sample_rate) override;
 	void Close() override;
+	void Reset() override;
 	ConstBuffer<void> Resample(ConstBuffer<void> src) override;
 
 private:

src/pcm/PcmBuffer.hxx

@@ -40,7 +40,7 @@ public:
 
 	/**
 	 * Get the buffer, and guarantee a minimum size.  This buffer becomes
-	 * invalid with the next pcm_buffer_get() call.
+	 * invalid with the next Get() call.
 	 *
 	 * This function will never return nullptr, even if size is
 	 * zero, because the PCM library uses the nullptr return value

src/pcm/PcmConvert.cxx

@@ -117,6 +117,17 @@ PcmConvert::Close()
 #endif
 }
 
+void
+PcmConvert::Reset()
+{
+	if (enable_resampler)
+		resampler.Reset();
+
+#ifdef ENABLE_DSD
+	dsd.Reset();
+#endif
+}
+
 ConstBuffer<void>
 PcmConvert::Convert(ConstBuffer<void> buffer)
 {

src/pcm/PcmConvert.hxx

@@ -67,6 +67,11 @@ public:
 	 */
 	void Close();
 
+	/**
+	 * Reset the filter's state, e.g. drop/flush buffers.
+	 */
+	void Reset();
+
 	/**
 	 * Converts PCM data between two audio formats.
 	 *

src/pcm/PcmDop.cxx

@@ -44,8 +44,6 @@ pcm_dsd_to_dop(PcmBuffer &buffer, unsigned channels,
 	       ConstBuffer<uint8_t> _src)
 {
 	assert(audio_valid_channel_count(channels));
-	assert(!_src.IsNull());
-	assert(_src.size > 0);
 	assert(_src.size % channels == 0);
 
 	const unsigned num_src_samples = _src.size;

src/pcm/PcmDsd.cxx

@@ -71,41 +71,3 @@ PcmDsd::ToFloat(unsigned channels, ConstBuffer<uint8_t> src)
 
 	return { dest, num_samples };
 }
-
-/**
- * Construct a 32 bit integer from four bytes.
- */
-static constexpr inline uint32_t
-Construct32(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
-{
-	return uint32_t(a) | (uint32_t(b) << 8) |
-		(uint32_t(c) << 16) | (uint32_t(d) << 24);
-}
-
-static constexpr inline uint32_t
-Dsd8To32Sample(const uint8_t *src, unsigned channels)
-{
-	return Construct32(src[0], src[channels],
-			   src[2 * channels], src[3 * channels]);
-}
-
-ConstBuffer<uint32_t>
-Dsd8To32(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> _src)
-{
-	const size_t in_frames = _src.size / channels;
-	const size_t out_frames = in_frames / 4;
-	const size_t out_samples = out_frames * channels;
-
-	const uint8_t *src = _src.data;
-	uint32_t *const dest0 = buffer.GetT<uint32_t>(out_samples);
-	uint32_t *dest = dest0;
-
-	for (size_t i = 0; i < out_frames; ++i) {
-		for (size_t c = 0; c < channels; ++c)
-			*dest++ = Dsd8To32Sample(src++, channels);
-
-		src += 3 * channels;
-	}
-
-	return {dest0, out_samples};
-}

src/pcm/PcmDsd.hxx

@@ -48,10 +48,4 @@ public:
 				   ConstBuffer<uint8_t> src);
 };
 
-/**
- * Convert DSD_U8 to DSD_U32 (native endian).
- */
-ConstBuffer<uint32_t>
-Dsd8To32(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> src);
-
 #endif

src/pcm/PcmExport.cxx

@@ -25,6 +25,8 @@
 #include "util/ConstBuffer.hxx"
 
 #ifdef ENABLE_DSD
+#include "Dsd16.hxx"
+#include "Dsd32.hxx"
 #include "PcmDsd.hxx"
 #include "PcmDop.hxx"
 #endif
@@ -41,9 +43,15 @@ PcmExport::Open(SampleFormat sample_format, unsigned _channels,
 		: SampleFormat::UNDEFINED;
 
 #ifdef ENABLE_DSD
-	assert(!params.dsd_u32 || !params.dop);
+	assert((params.dsd_u16 + params.dsd_u32 + params.dop) <= 1);
 	assert(!params.dop || audio_valid_channel_count(_channels));
 
+	dsd_u16 = params.dsd_u16 && sample_format == SampleFormat::DSD;
+	if (dsd_u16)
+		/* after the conversion to DSD_U16, the DSD samples
+		   are stuffed inside fake 16 bit samples */
+		sample_format = SampleFormat::S16;
+
 	dsd_u32 = params.dsd_u32 && sample_format == SampleFormat::DSD;
 	if (dsd_u32)
 		/* after the conversion to DSD_U32, the DSD samples
@@ -55,8 +63,6 @@ PcmExport::Open(SampleFormat sample_format, unsigned _channels,
 		/* after the conversion to DoP, the DSD
 		   samples are stuffed inside fake 24 bit samples */
 		sample_format = SampleFormat::S24_P32;
-#else
-	(void)_channels;
 #endif
 
 	shift8 = params.shift8 && sample_format == SampleFormat::S24_P32;
@@ -84,6 +90,9 @@ PcmExport::GetFrameSize(const AudioFormat &audio_format) const
 		return audio_format.channels * 3;
 
 #ifdef ENABLE_DSD
+	if (dsd_u16)
+		return channels * 2;
+
 	if (dsd_u32)
 		return channels * 4;
 
@@ -98,6 +107,46 @@ PcmExport::GetFrameSize(const AudioFormat &audio_format) const
 	return audio_format.GetFrameSize();
 }
 
+unsigned
+PcmExport::Params::CalcOutputSampleRate(unsigned sample_rate) const
+{
+#ifdef ENABLE_DSD
+	if (dsd_u16)
+		/* DSD_U16 combines two 8-bit "samples" in one 16-bit
+		   "sample" */
+		sample_rate /= 2;
+
+	if (dsd_u32)
+		/* DSD_U32 combines four 8-bit "samples" in one 32-bit
+		   "sample" */
+		sample_rate /= 4;
+
+	if (dop)
+		/* DoP packs two 8-bit "samples" in one 24-bit
+		   "sample" */
+		sample_rate /= 2;
+#endif
+
+	return sample_rate;
+}
+
+unsigned
+PcmExport::Params::CalcInputSampleRate(unsigned sample_rate) const
+{
+#ifdef ENABLE_DSD
+	if (dsd_u16)
+		sample_rate *= 2;
+
+	if (dsd_u32)
+		sample_rate *= 4;
+
+	if (dop)
+		sample_rate *= 2;
+#endif
+
+	return sample_rate;
+}
+
 ConstBuffer<void>
 PcmExport::Export(ConstBuffer<void> data)
 {
@@ -106,6 +155,11 @@ PcmExport::Export(ConstBuffer<void> data)
 					  alsa_channel_order, channels);
 
 #ifdef ENABLE_DSD
+	if (dsd_u16)
+		data = Dsd8To16(dop_buffer, channels,
+				ConstBuffer<uint8_t>::FromVoid(data))
+			.ToVoid();
+
 	if (dsd_u32)
 		data = Dsd8To32(dop_buffer, channels,
 				ConstBuffer<uint8_t>::FromVoid(data))

src/pcm/PcmExport.hxx

@@ -31,18 +31,7 @@ template<typename T> struct ConstBuffer;
  * outside of MPD.  It has a few more options to tweak the binary
  * representation which are not supported by the pcm_convert library.
  */
-struct PcmExport {
-	struct Params {
-		bool alsa_channel_order = false;
-#ifdef ENABLE_DSD
-		bool dsd_u32 = false;
-		bool dop = false;
-#endif
-		bool shift8 = false;
-		bool pack24 = false;
-		bool reverse_endian = false;
-	};
-
+class PcmExport {
 	/**
 	 * This buffer is used to reorder channels.
 	 *
@@ -90,6 +79,11 @@ struct PcmExport {
 	SampleFormat alsa_channel_order;
 
 #ifdef ENABLE_DSD
+	/**
+	 * Convert DSD (U8) to DSD_U16?
+	 */
+	bool dsd_u16;
+
 	/**
 	 * Convert DSD (U8) to DSD_U32?
 	 */
@@ -121,6 +115,34 @@ struct PcmExport {
 	 */
 	uint8_t reverse_endian;
 
+public:
+	struct Params {
+		bool alsa_channel_order = false;
+#ifdef ENABLE_DSD
+		bool dsd_u16 = false;
+		bool dsd_u32 = false;
+		bool dop = false;
+#endif
+		bool shift8 = false;
+		bool pack24 = false;
+		bool reverse_endian = false;
+
+		/**
+		 * Calculate the output sample rate, given a specific input
+		 * sample rate.  Usually, both are the same; however, with
+		 * DSD_U32, four input bytes (= 4 * 8 bits) are combined to
+		 * one output word (32 bits), dividing the sample rate by 4.
+		 */
+		gcc_pure
+		unsigned CalcOutputSampleRate(unsigned input_sample_rate) const;
+
+		/**
+		 * The inverse of CalcOutputSampleRate().
+		 */
+		gcc_pure
+		unsigned CalcInputSampleRate(unsigned output_sample_rate) const;
+	};
+
 	/**
 	 * Open the object.
 	 *
@@ -134,6 +156,12 @@ struct PcmExport {
 	void Open(SampleFormat sample_format, unsigned channels,
 		  Params params);
 
+	/**
+	 * Reset the filter's state, e.g. drop/flush buffers.
+	 */
+	void Reset() {
+	}
+
 	/**
 	 * Calculate the size of one output frame.
 	 */

src/pcm/Resampler.hxx

@@ -54,6 +54,12 @@ public:
 	 */
 	virtual void Close() = 0;
 
+	/**
+	 * Reset the filter's state, e.g. drop/flush buffers.
+	 */
+	virtual void Reset() {
+	}
+
 	/**
 	 * Resamples a block of PCM data.
 	 *

src/util/ReusableArray.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2013 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2013-2017 Max Kellermann <max@duempel.org>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,10 +30,12 @@
 #ifndef REUSABLE_ARRAY_HXX
 #define REUSABLE_ARRAY_HXX
 
-#include <stddef.h>
-
 #include "Compiler.h"
 
+#include <utility>
+
+#include <stddef.h>
+
 /**
  * Manager for a temporary array which grows as needed.  This attempts
  * to reduce the number of consecutive heap allocations and
@@ -44,19 +46,30 @@
  */
 template<typename T, size_t M=1>
 class ReusableArray {
-	T *buffer;
-	size_t capacity;
+	T *buffer = nullptr;
+	size_t capacity = 0;
 
 public:
-	ReusableArray():buffer(nullptr), capacity(0) {}
+	ReusableArray() = default;
 
-	ReusableArray(const ReusableArray &other) = delete;
-	ReusableArray &operator=(const ReusableArray &other) = delete;
+	ReusableArray(ReusableArray &&src)
+		:buffer(std::exchange(src.buffer, nullptr)),
+		 capacity(std::exchange(src.capacity, 0)) {}
+
+	ReusableArray &operator=(const ReusableArray &&src) {
+		std::swap(buffer, src.buffer);
+		std::swap(capacity, src.capacity);
+		return *this;
+	}
 
 	~ReusableArray() {
 		delete[] buffer;
 	}
 
+	size_t GetCapacity() const {
+		return capacity;
+	}
+
 	/**
 	 * Free resources allocated by this object.  This invalidates
 	 * the buffer returned by Get().

test/test_pcm_all.hxx

@@ -20,6 +20,8 @@
 #ifndef MPD_TEST_PCM_ALL_HXX
 #define MPD_TEST_PCM_ALL_HXX
 
+#include "check.h"
+
 #include <cppunit/TestFixture.h>
 #include <cppunit/extensions/HelperMacros.h>
 
@@ -126,6 +128,7 @@ class PcmExportTest : public CppUnit::TestFixture {
 	CPPUNIT_TEST(TestPack24);
 	CPPUNIT_TEST(TestReverseEndian);
 #ifdef ENABLE_DSD
+	CPPUNIT_TEST(TestDsdU16);
 	CPPUNIT_TEST(TestDsdU32);
 	CPPUNIT_TEST(TestDop);
 #endif
@@ -137,6 +140,7 @@ public:
 	void TestPack24();
 	void TestReverseEndian();
 #ifdef ENABLE_DSD
+	void TestDsdU16();
 	void TestDsdU32();
 	void TestDop();
 #endif

test/test_pcm_dither.cxx

@@ -17,6 +17,7 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
+#include "config.h"
 #include "test_pcm_all.hxx"
 #include "test_pcm_util.hxx"
 #include "pcm/PcmDither.cxx"

test/test_pcm_export.cxx

@@ -35,6 +35,9 @@ PcmExportTest::TestShift8()
 	PcmExport::Params params;
 	params.shift8 = true;
 
+	CPPUNIT_ASSERT_EQUAL(params.CalcOutputSampleRate(42u), 42u);
+	CPPUNIT_ASSERT_EQUAL(params.CalcInputSampleRate(42u), 42u);
+
 	PcmExport e;
 	e.Open(SampleFormat::S24_P32, 2, params);
 
@@ -71,6 +74,9 @@ PcmExportTest::TestPack24()
 	PcmExport::Params params;
 	params.pack24 = true;
 
+	CPPUNIT_ASSERT_EQUAL(params.CalcOutputSampleRate(42u), 42u);
+	CPPUNIT_ASSERT_EQUAL(params.CalcInputSampleRate(42u), 42u);
+
 	PcmExport e;
 	e.Open(SampleFormat::S24_P32, 2, params);
 
@@ -97,6 +103,9 @@ PcmExportTest::TestReverseEndian()
 	PcmExport::Params params;
 	params.reverse_endian = true;
 
+	CPPUNIT_ASSERT_EQUAL(params.CalcOutputSampleRate(42u), 42u);
+	CPPUNIT_ASSERT_EQUAL(params.CalcInputSampleRate(42u), 42u);
+
 	PcmExport e;
 	e.Open(SampleFormat::S8, 2, params);
 
@@ -117,6 +126,37 @@ PcmExportTest::TestReverseEndian()
 
 #ifdef ENABLE_DSD
 
+void
+PcmExportTest::TestDsdU16()
+{
+	static constexpr uint8_t src[] = {
+		0x01, 0x23, 0x45, 0x67,
+		0x89, 0xab, 0xcd, 0xef,
+		0x11, 0x22, 0x33, 0x44,
+		0x55, 0x66, 0x77, 0x88,
+	};
+
+	static constexpr uint16_t expected[] = {
+		0x0145, 0x2367,
+		0x89cd, 0xabef,
+		0x1133, 0x2244,
+		0x5577, 0x6688,
+	};
+
+	PcmExport::Params params;
+	params.dsd_u16 = true;
+
+	CPPUNIT_ASSERT_EQUAL(params.CalcOutputSampleRate(705600u), 352800u);
+	CPPUNIT_ASSERT_EQUAL(params.CalcInputSampleRate(352800u), 705600u);
+
+	PcmExport e;
+	e.Open(SampleFormat::DSD, 2, params);
+
+	auto dest = e.Export({src, sizeof(src)});
+	CPPUNIT_ASSERT_EQUAL(sizeof(expected), dest.size);
+	CPPUNIT_ASSERT(memcmp(dest.data, expected, dest.size) == 0);
+}
+
 void
 PcmExportTest::TestDsdU32()
 {
@@ -128,15 +168,18 @@ PcmExportTest::TestDsdU32()
 	};
 
 	static constexpr uint32_t expected[] = {
-		0xcd894501,
-		0xefab6723,
-		0x77553311,
-		0x88664422,
+		0x014589cd,
+		0x2367abef,
+		0x11335577,
+		0x22446688,
 	};
 
 	PcmExport::Params params;
 	params.dsd_u32 = true;
 
+	CPPUNIT_ASSERT_EQUAL(params.CalcOutputSampleRate(705600u), 176400u);
+	CPPUNIT_ASSERT_EQUAL(params.CalcInputSampleRate(176400u), 705600u);
+
 	PcmExport e;
 	e.Open(SampleFormat::DSD, 2, params);
 
@@ -163,6 +206,9 @@ PcmExportTest::TestDop()
 	PcmExport::Params params;
 	params.dop = true;
 
+	CPPUNIT_ASSERT_EQUAL(params.CalcOutputSampleRate(705600u), 352800u);
+	CPPUNIT_ASSERT_EQUAL(params.CalcInputSampleRate(352800u), 705600u);
+
 	PcmExport e;
 	e.Open(SampleFormat::DSD, 2, params);
 
@@ -192,6 +238,9 @@ TestAlsaChannelOrder51()
 	PcmExport::Params params;
 	params.alsa_channel_order = true;
 
+	CPPUNIT_ASSERT_EQUAL(params.CalcOutputSampleRate(42u), 42u);
+	CPPUNIT_ASSERT_EQUAL(params.CalcInputSampleRate(42u), 42u);
+
 	PcmExport e;
 	e.Open(F, 6, params);
 
@@ -219,6 +268,9 @@ TestAlsaChannelOrder71()
 	PcmExport::Params params;
 	params.alsa_channel_order = true;
 
+	CPPUNIT_ASSERT_EQUAL(params.CalcOutputSampleRate(42u), 42u);
+	CPPUNIT_ASSERT_EQUAL(params.CalcInputSampleRate(42u), 42u);
+
 	PcmExport e;
 	e.Open(F, 8, params);
 

test/test_pcm_main.cxx

@@ -17,6 +17,7 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
+#include "config.h"
 #include "test_pcm_all.hxx"
 #include "Compiler.h"
 

test/test_pcm_pack.cxx

@@ -17,6 +17,7 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
+#include "config.h"
 #include "test_pcm_all.hxx"
 #include "test_pcm_util.hxx"
 #include "pcm/PcmPack.hxx"