* Moved _SoundPlayNode into the BPrivate namespace and dismissed the '_' prefix.

* Further cleanup.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@34482 a95241bf-73f2-0310-859d-f6bbb57e9c96

headers/os/media/SoundPlayer.h

@@ -16,7 +16,9 @@
 class BContinuousParameter;
 class BParameterWeb;
 class BSound;
-class _SoundPlayNode;
+namespace BPrivate {
+	class SoundPlayNode;
+}
 
 
 class sound_error : public std::exception {
@@ -155,10 +157,10 @@ private:
 									const media_raw_audio_format& format);
 
 private:
-	friend class _SoundPlayNode;
+	friend class BPrivate::SoundPlayNode;
 
-	struct _playing_sound {
+	struct playing_sound {
-		_playing_sound*	next;
+		playing_sound*	next;
 		off_t			current_offset;
 		BSound*			sound;
 		play_id			id;
@@ -168,8 +170,8 @@ private:
 		float			volume;
 	};
 
-	struct _waiting_sound {
+	struct waiting_sound {
-		_waiting_sound*	next;
+		waiting_sound*	next;
 		bigtime_t		start_time;
 		BSound*			sound;
 		play_id			id;
@@ -177,10 +179,10 @@ private:
 		float			volume;
 	};
 
-			_SoundPlayNode*		fPlayerNode;
+			BPrivate::SoundPlayNode* fPlayerNode;
 
-			_playing_sound*		fPlayingSounds;
+			playing_sound*		fPlayingSounds;
-			_waiting_sound*		fWaitingSounds;
+			waiting_sound*		fWaitingSounds;
 
 			BufferPlayerFunc	fPlayBufferFunc;
 			EventNotifierFunc	fNotifierFunc;

src/kits/media/SoundPlayNode.cpp

@@ -8,9 +8,7 @@
  */
 
 
-/*!	This is the BBufferProducer, used internally by BSoundPlayer
+/*!	This is the BBufferProducer used internally by BSoundPlayer.
-	This belongs into a private namespace, but isn't for
-	compatibility reasons.
 */
 
 
@@ -28,25 +26,28 @@
 #define SEND_NEW_BUFFER_EVENT (BTimedEventQueue::B_USER_EVENT + 1)
 
 
-_SoundPlayNode::_SoundPlayNode(const char* name, BSoundPlayer* player)
+namespace BPrivate {
+
+
+SoundPlayNode::SoundPlayNode(const char* name, BSoundPlayer* player)
 	:
 	BMediaNode(name),
 	BBufferProducer(B_MEDIA_RAW_AUDIO),
 	BMediaEventLooper(),
-	mPlayer(player),
+	fPlayer(player),
-	mInitCheckStatus(B_OK),
+	fInitStatus(B_OK),
-	mOutputEnabled(true),
+	fOutputEnabled(true),
-	mBufferGroup(NULL),
+	fBufferGroup(NULL),
-	mFramesSent(0),
+	fFramesSent(0),
-	mTooEarlyCount(0)
+	fTooEarlyCount(0)
 {
 	CALLED();
-	mOutput.format.type = B_MEDIA_RAW_AUDIO;
+	fOutput.format.type = B_MEDIA_RAW_AUDIO;
-	mOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
+	fOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
 }
 
 
-_SoundPlayNode::~_SoundPlayNode()
+SoundPlayNode::~SoundPlayNode()
 {
 	CALLED();
 	Quit();
@@ -54,25 +55,25 @@ _SoundPlayNode::~_SoundPlayNode()
 
 
 bool
-_SoundPlayNode::IsPlaying()
+SoundPlayNode::IsPlaying()
 {
 	return RunState() == B_STARTED;
 }
 
 
 bigtime_t
-_SoundPlayNode::CurrentTime()
+SoundPlayNode::CurrentTime()
 {
-	int frameRate = (int)mOutput.format.u.raw_audio.frame_rate;
+	int frameRate = (int)fOutput.format.u.raw_audio.frame_rate;
 	return frameRate == 0 ? 0
-		: bigtime_t((1000000LL * mFramesSent) / frameRate);
+		: bigtime_t((1000000LL * fFramesSent) / frameRate);
 }
 
 
 media_multi_audio_format
-_SoundPlayNode::Format() const
+SoundPlayNode::Format() const
 {
-	return mOutput.format.u.raw_audio;
+	return fOutput.format.u.raw_audio;
 }
 
 
@@ -80,7 +81,7 @@ _SoundPlayNode::Format() const
 
 
 BMediaAddOn*
-_SoundPlayNode::AddOn(int32* _internalID) const
+SoundPlayNode::AddOn(int32* _internalID) const
 {
 	CALLED();
 	// This only gets called if we are in an add-on.
@@ -89,7 +90,7 @@ _SoundPlayNode::AddOn(int32* _internalID) const
 
 
 void
-_SoundPlayNode::Preroll()
+SoundPlayNode::Preroll()
 {
 	CALLED();
 	// TODO: Performance opportunity
@@ -98,7 +99,7 @@ _SoundPlayNode::Preroll()
 
 
 status_t
-_SoundPlayNode::HandleMessage(int32 message, const void* data, size_t size)
+SoundPlayNode::HandleMessage(int32 message, const void* data, size_t size)
 {
 	CALLED();
 	return B_ERROR;
@@ -106,31 +107,31 @@ _SoundPlayNode::HandleMessage(int32 message, const void* data, size_t size)
 
 
 void
-_SoundPlayNode::NodeRegistered()
+SoundPlayNode::NodeRegistered()
 {
 	CALLED();
 
-	if (mInitCheckStatus != B_OK) {
+	if (fInitStatus != B_OK) {
 		ReportError(B_NODE_IN_DISTRESS);
 		return;
 	}
 
 	SetPriority(B_URGENT_PRIORITY);
 
-	mOutput.format.type = B_MEDIA_RAW_AUDIO;
+	fOutput.format.type = B_MEDIA_RAW_AUDIO;
-	mOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
+	fOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
-	mOutput.destination = media_destination::null;
+	fOutput.destination = media_destination::null;
-	mOutput.source.port = ControlPort();
+	fOutput.source.port = ControlPort();
-	mOutput.source.id = 0;
+	fOutput.source.id = 0;
-	mOutput.node = Node();
+	fOutput.node = Node();
-	strcpy(mOutput.name, Name());
+	strcpy(fOutput.name, Name());
 
 	Run();
 }
 
 
 status_t
-_SoundPlayNode::RequestCompleted(const media_request_info& info)
+SoundPlayNode::RequestCompleted(const media_request_info& info)
 {
 	CALLED();
 	return B_OK;
@@ -138,7 +139,7 @@ _SoundPlayNode::RequestCompleted(const media_request_info& info)
 
 
 void
-_SoundPlayNode::SetTimeSource(BTimeSource* timeSource)
+SoundPlayNode::SetTimeSource(BTimeSource* timeSource)
 {
 	CALLED();
 	BMediaNode::SetTimeSource(timeSource);
@@ -146,9 +147,9 @@ _SoundPlayNode::SetTimeSource(BTimeSource* timeSource)
 
 
 void
-_SoundPlayNode::SetRunMode(run_mode mode)
+SoundPlayNode::SetRunMode(run_mode mode)
 {
-	TRACE("_SoundPlayNode::SetRunMode mode:%i\n", mode);
+	TRACE("SoundPlayNode::SetRunMode mode:%i\n", mode);
 	BMediaNode::SetRunMode(mode);
 }
 
@@ -157,7 +158,7 @@ _SoundPlayNode::SetRunMode(run_mode mode)
 
 
 status_t
-_SoundPlayNode::FormatSuggestionRequested(media_type type, int32 /*quality*/,
+SoundPlayNode::FormatSuggestionRequested(media_type type, int32 /*quality*/,
 	media_format* format)
 {
 	// FormatSuggestionRequested() is not necessarily part of the format
@@ -179,7 +180,7 @@ _SoundPlayNode::FormatSuggestionRequested(media_type type, int32 /*quality*/,
 
 
 status_t
-_SoundPlayNode::FormatProposal(const media_source& output, media_format* format)
+SoundPlayNode::FormatProposal(const media_source& output, media_format* format)
 {
 	// FormatProposal() is the first stage in the BMediaRoster::Connect()
 	// process. We hand out a suggested format, with wildcards for any
@@ -187,8 +188,8 @@ _SoundPlayNode::FormatProposal(const media_source& output, media_format* format)
 	CALLED();
 
 	// is this a proposal for our one output?
-	if (output != mOutput.source) {
+	if (output != fOutput.source) {
-		TRACE("_SoundPlayNode::FormatProposal returning B_MEDIA_BAD_SOURCE\n");
+		TRACE("SoundPlayNode::FormatProposal returning B_MEDIA_BAD_SOURCE\n");
 		return B_MEDIA_BAD_SOURCE;
 	}
 
@@ -198,14 +199,14 @@ _SoundPlayNode::FormatProposal(const media_source& output, media_format* format)
 
 	// if not raw audio, we can't support it
 	if (format->type != B_MEDIA_RAW_AUDIO) {
-		TRACE("_SoundPlayNode::FormatProposal returning B_MEDIA_BAD_FORMAT\n");
+		TRACE("SoundPlayNode::FormatProposal returning B_MEDIA_BAD_FORMAT\n");
 		return B_MEDIA_BAD_FORMAT;
 	}
 
 #if DEBUG >0
 	char buf[100];
 	string_for_format(*format, buf, sizeof(buf));
-	TRACE("_SoundPlayNode::FormatProposal: format %s\n", buf);
+	TRACE("SoundPlayNode::FormatProposal: format %s\n", buf);
 #endif
 
 	return B_OK;
@@ -213,7 +214,7 @@ _SoundPlayNode::FormatProposal(const media_source& output, media_format* format)
 
 
 status_t
-_SoundPlayNode::FormatChangeRequested(const media_source& source,
+SoundPlayNode::FormatChangeRequested(const media_source& source,
 	const media_destination& destination, media_format* _format,
 	int32* /* deprecated */)
 {
@@ -225,12 +226,12 @@ _SoundPlayNode::FormatChangeRequested(const media_source& source,
 
 
 status_t
-_SoundPlayNode::GetNextOutput(int32* cookie, media_output* _output)
+SoundPlayNode::GetNextOutput(int32* cookie, media_output* _output)
 {
 	CALLED();
 
 	if (*cookie == 0) {
-		*_output = mOutput;
+		*_output = fOutput;
 		*cookie += 1;
 		return B_OK;
 	} else {
@@ -240,7 +241,7 @@ _SoundPlayNode::GetNextOutput(int32* cookie, media_output* _output)
 
 
 status_t
-_SoundPlayNode::DisposeOutputCookie(int32 cookie)
+SoundPlayNode::DisposeOutputCookie(int32 cookie)
 {
 	CALLED();
 	// do nothing because we don't use the cookie for anything special
@@ -249,19 +250,19 @@ _SoundPlayNode::DisposeOutputCookie(int32 cookie)
 
 
 status_t
-_SoundPlayNode::SetBufferGroup(const media_source& forSource,
+SoundPlayNode::SetBufferGroup(const media_source& forSource,
 	BBufferGroup* newGroup)
 {
 	CALLED();
 
 	// is this our output?
-	if (forSource != mOutput.source) {
+	if (forSource != fOutput.source) {
-		TRACE("_SoundPlayNode::SetBufferGroup returning B_MEDIA_BAD_SOURCE\n");
+		TRACE("SoundPlayNode::SetBufferGroup returning B_MEDIA_BAD_SOURCE\n");
 		return B_MEDIA_BAD_SOURCE;
 	}
 
 	// Are we being passed the buffer group we're already using?
-	if (newGroup == mBufferGroup)
+	if (newGroup == fBufferGroup)
 		return B_OK;
 
 	// Ahh, someone wants us to use a different buffer group. At this point we
@@ -270,19 +271,19 @@ _SoundPlayNode::SetBufferGroup(const media_source& forSource,
 	// use *that*. Note that if we're caching a BBuffer that we requested
 	// earlier, we have to Recycle() that buffer *before* deleting the buffer
 	// group, otherwise we'll deadlock waiting for that buffer to be recycled!
-	delete mBufferGroup;
+	delete fBufferGroup;
 		// waits for all buffers to recycle
 	if (newGroup != NULL) {
 		// we were given a valid group; just use that one from now on
-		mBufferGroup = newGroup;
+		fBufferGroup = newGroup;
 	} else {
 		// we were passed a NULL group pointer; that means we construct
 		// our own buffer group to use from now on
-		size_t size = mOutput.format.u.raw_audio.buffer_size;
+		size_t size = fOutput.format.u.raw_audio.buffer_size;
-		int32 count = int32(mLatency / BufferDuration() + 1 + 1);
+		int32 count = int32(fLatency / BufferDuration() + 1 + 1);
 		if (count < 3)
 			count = 3;
-		mBufferGroup = new BBufferGroup(size, count);
+		fBufferGroup = new BBufferGroup(size, count);
 	}
 
 	return B_OK;
@@ -290,7 +291,7 @@ _SoundPlayNode::SetBufferGroup(const media_source& forSource,
 
 
 status_t
-_SoundPlayNode::GetLatency(bigtime_t* _latency)
+SoundPlayNode::GetLatency(bigtime_t* _latency)
 {
 	CALLED();
 
@@ -301,7 +302,7 @@ _SoundPlayNode::GetLatency(bigtime_t* _latency)
 
 
 status_t
-_SoundPlayNode::PrepareToConnect(const media_source& what,
+SoundPlayNode::PrepareToConnect(const media_source& what,
 	const media_destination& where, media_format* format,
 	media_source* _source, char* _name)
 {
@@ -314,14 +315,14 @@ _SoundPlayNode::PrepareToConnect(const media_source& what,
 	CALLED();
 
 	// is this our output?
-	if (what != mOutput.source)	{
+	if (what != fOutput.source)	{
-		TRACE("_SoundPlayNode::PrepareToConnect returning "
+		TRACE("SoundPlayNode::PrepareToConnect returning "
 			"B_MEDIA_BAD_SOURCE\n");
 		return B_MEDIA_BAD_SOURCE;
 	}
 
 	// are we already connected?
-	if (mOutput.destination != media_destination::null)
+	if (fOutput.destination != media_destination::null)
 		return B_MEDIA_ALREADY_CONNECTED;
 
 	// the format may not yet be fully specialized (the consumer might have
@@ -331,13 +332,13 @@ _SoundPlayNode::PrepareToConnect(const media_source& what,
 #if DEBUG > 0
 	char buf[100];
 	string_for_format(*format, buf, sizeof(buf));
-	TRACE("_SoundPlayNode::PrepareToConnect: input format %s\n", buf);
+	TRACE("SoundPlayNode::PrepareToConnect: input format %s\n", buf);
 #endif
 
 	// if not raw audio, we can't support it
 	if (format->type != B_MEDIA_UNKNOWN_TYPE
 		&& format->type != B_MEDIA_RAW_AUDIO) {
-		TRACE("_SoundPlayNode::PrepareToConnect: non raw format, returning "
+		TRACE("SoundPlayNode::PrepareToConnect: non raw format, returning "
 			"B_MEDIA_BAD_FORMAT\n");
 		return B_MEDIA_BAD_FORMAT;
 	}
@@ -354,7 +355,7 @@ _SoundPlayNode::PrepareToConnect(const media_source& what,
 		&& *(uint32 *)&format->user_data[44] == FORMAT_USER_DATA_MAGIC_2) {
 		channel_count = *(uint32 *)&format->user_data[4];
 		frame_rate = *(float *)&format->user_data[20];
-		TRACE("_SoundPlayNode::PrepareToConnect: found mixer info: "
+		TRACE("SoundPlayNode::PrepareToConnect: found mixer info: "
 			"channel_count %ld, frame_rate %.1f\n", channel_count, frame_rate);
 	}
 
@@ -377,28 +378,28 @@ _SoundPlayNode::PrepareToConnect(const media_source& what,
 
 #if DEBUG > 0
 	string_for_format(*format, buf, sizeof(buf));
-	TRACE("_SoundPlayNode::PrepareToConnect: output format %s\n", buf);
+	TRACE("SoundPlayNode::PrepareToConnect: output format %s\n", buf);
 #endif
 
 	// Now reserve the connection, and return information about it
-	mOutput.destination = where;
+	fOutput.destination = where;
-	mOutput.format = *format;
+	fOutput.format = *format;
-	*_source = mOutput.source;
+	*_source = fOutput.source;
 	strcpy(_name, Name());
 	return B_OK;
 }
 
 
 void
-_SoundPlayNode::Connect(status_t error, const media_source& source,
+SoundPlayNode::Connect(status_t error, const media_source& source,
 	const media_destination& destination, const media_format& format,
 	char* name)
 {
 	CALLED();
 
 	// is this our output?
-	if (source != mOutput.source) {
+	if (source != fOutput.source) {
-		TRACE("_SoundPlayNode::Connect returning\n");
+		TRACE("SoundPlayNode::Connect returning\n");
 		return;
 	}
 
@@ -406,85 +407,85 @@ _SoundPlayNode::Connect(status_t error, const media_source& source,
 	// a non-zero error code.  When that happens we simply unreserve the
 	// connection and do nothing else.
 	if (error) {
-		mOutput.destination = media_destination::null;
+		fOutput.destination = media_destination::null;
-		mOutput.format.type = B_MEDIA_RAW_AUDIO;
+		fOutput.format.type = B_MEDIA_RAW_AUDIO;
-		mOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
+		fOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
 		return;
 	}
 
 	// Okay, the connection has been confirmed.  Record the destination and
 	// format that we agreed on, and report our connection name again.
-	mOutput.destination = destination;
+	fOutput.destination = destination;
-	mOutput.format = format;
+	fOutput.format = format;
 	strcpy(name, Name());
 
 	// Now that we're connected, we can determine our downstream latency.
 	// Do so, then make sure we get our events early enough.
 	media_node_id id;
-	FindLatencyFor(mOutput.destination, &mLatency, &id);
+	FindLatencyFor(fOutput.destination, &fLatency, &id);
-	TRACE("_SoundPlayNode::Connect: downstream latency = %Ld\n", mLatency);
+	TRACE("SoundPlayNode::Connect: downstream latency = %Ld\n", fLatency);
 
 	// reset our buffer duration, etc. to avoid later calculations
-	bigtime_t duration = ((mOutput.format.u.raw_audio.buffer_size * 1000000LL)
+	bigtime_t duration = ((fOutput.format.u.raw_audio.buffer_size * 1000000LL)
-		/ ((mOutput.format.u.raw_audio.format
+		/ ((fOutput.format.u.raw_audio.format
 				& media_raw_audio_format::B_AUDIO_SIZE_MASK)
-			* mOutput.format.u.raw_audio.channel_count))
+			* fOutput.format.u.raw_audio.channel_count))
-		/ (int32)mOutput.format.u.raw_audio.frame_rate;
+		/ (int32)fOutput.format.u.raw_audio.frame_rate;
 	SetBufferDuration(duration);
-	TRACE("_SoundPlayNode::Connect: buffer duration is %Ld\n", duration);
+	TRACE("SoundPlayNode::Connect: buffer duration is %Ld\n", duration);
 
-	mInternalLatency = (3 * BufferDuration()) / 4;
+	fInternalLatency = (3 * BufferDuration()) / 4;
-	TRACE("_SoundPlayNode::Connect: using %Ld as internal latency\n",
+	TRACE("SoundPlayNode::Connect: using %Ld as internal latency\n",
-		mInternalLatency);
+		fInternalLatency);
-	SetEventLatency(mLatency + mInternalLatency);
+	SetEventLatency(fLatency + fInternalLatency);
 
 	// Set up the buffer group for our connection, as long as nobody handed us
 	// a buffer group (via SetBufferGroup()) prior to this.
 	// That can happen, for example, if the consumer calls SetOutputBuffersFor()
 	// on us from within its Connected() method.
-	if (!mBufferGroup)
+	if (!fBufferGroup)
 		AllocateBuffers();
 }
 
 
 void
-_SoundPlayNode::Disconnect(const media_source& what,
+SoundPlayNode::Disconnect(const media_source& what,
 	const media_destination& where)
 {
 	CALLED();
 
 	// is this our output?
-	if (what != mOutput.source) {
+	if (what != fOutput.source) {
-		TRACE("_SoundPlayNode::Disconnect returning\n");
+		TRACE("SoundPlayNode::Disconnect returning\n");
 		return;
 	}
 
 	// Make sure that our connection is the one being disconnected
-	if (where == mOutput.destination && what == mOutput.source) {
+	if (where == fOutput.destination && what == fOutput.source) {
-		mOutput.destination = media_destination::null;
+		fOutput.destination = media_destination::null;
-		mOutput.format.type = B_MEDIA_RAW_AUDIO;
+		fOutput.format.type = B_MEDIA_RAW_AUDIO;
-		mOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
+		fOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
-		delete mBufferGroup;
+		delete fBufferGroup;
-		mBufferGroup = NULL;
+		fBufferGroup = NULL;
 	} else {
 		fprintf(stderr, "\tDisconnect() called with wrong source/destination (%ld/%ld), ours is (%ld/%ld)\n",
-			what.id, where.id, mOutput.source.id, mOutput.destination.id);
+			what.id, where.id, fOutput.source.id, fOutput.destination.id);
 	}
 }
 
 
 void
-_SoundPlayNode::LateNoticeReceived(const media_source& what, bigtime_t howMuch,
+SoundPlayNode::LateNoticeReceived(const media_source& what, bigtime_t howMuch,
 	bigtime_t performanceTime)
 {
 	CALLED();
 
-	TRACE("_SoundPlayNode::LateNoticeReceived, %Ld too late at %Ld\n", howMuch,
+	TRACE("SoundPlayNode::LateNoticeReceived, %Ld too late at %Ld\n", howMuch,
 		performanceTime);
 
 	// is this our output?
-	if (what != mOutput.source) {
+	if (what != fOutput.source) {
-		TRACE("_SoundPlayNode::LateNoticeReceived returning\n");
+		TRACE("SoundPlayNode::LateNoticeReceived returning\n");
 		return;
 	}
 
@@ -495,31 +496,31 @@ _SoundPlayNode::LateNoticeReceived(const media_source& what, bigtime_t howMuch,
 		// that at the moment, so we try to start producing buffers earlier to
 		// compensate.
 
-		mInternalLatency += howMuch;
+		fInternalLatency += howMuch;
 
-		if (mInternalLatency > 30000)	// avoid getting a too high latency
+		if (fInternalLatency > 30000)	// avoid getting a too high latency
-			mInternalLatency = 30000;
+			fInternalLatency = 30000;
 
-		SetEventLatency(mLatency + mInternalLatency);
+		SetEventLatency(fLatency + fInternalLatency);
-		TRACE("_SoundPlayNode::LateNoticeReceived: increasing latency to %Ld\n", mLatency + mInternalLatency);
+		TRACE("SoundPlayNode::LateNoticeReceived: increasing latency to %Ld\n", fLatency + fInternalLatency);
 	} else {
 		// The other run modes dictate various strategies for sacrificing data quality
 		// in the interests of timely data delivery.  The way *we* do this is to skip
 		// a buffer, which catches us up in time by one buffer duration.
 
-		size_t nFrames = mOutput.format.u.raw_audio.buffer_size
+		size_t nFrames = fOutput.format.u.raw_audio.buffer_size
-			/ ((mOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK)
+			/ ((fOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK)
-			* mOutput.format.u.raw_audio.channel_count);
+			* fOutput.format.u.raw_audio.channel_count);
 
-		mFramesSent += nFrames;
+		fFramesSent += nFrames;
 
-		TRACE("_SoundPlayNode::LateNoticeReceived: skipping a buffer to try to catch up\n");
+		TRACE("SoundPlayNode::LateNoticeReceived: skipping a buffer to try to catch up\n");
 	}
 }
 
 
 void
-_SoundPlayNode::EnableOutput(const media_source& what, bool enabled,
+SoundPlayNode::EnableOutput(const media_source& what, bool enabled,
 	int32* /* deprecated */)
 {
 	CALLED();
@@ -531,17 +532,17 @@ _SoundPlayNode::EnableOutput(const media_source& what, bool enabled,
 	// matches, then set the enable state accordingly.
 
 	// is this our output?
-	if (what != mOutput.source) {
+	if (what != fOutput.source) {
-		fprintf(stderr, "_SoundPlayNode::EnableOutput returning\n");
+		fprintf(stderr, "SoundPlayNode::EnableOutput returning\n");
 		return;
 	}
 
-	mOutputEnabled = enabled;
+	fOutputEnabled = enabled;
 }
 
 
 void
-_SoundPlayNode::AdditionalBufferRequested(const media_source& source,
+SoundPlayNode::AdditionalBufferRequested(const media_source& source,
 	media_buffer_id previousBuffer, bigtime_t previousTime,
 	const media_seek_tag* previousTag)
 {
@@ -552,22 +553,22 @@ _SoundPlayNode::AdditionalBufferRequested(const media_source& source,
 
 
 void
-_SoundPlayNode::LatencyChanged(const media_source& source,
+SoundPlayNode::LatencyChanged(const media_source& source,
 	const media_destination& destination, bigtime_t newLatency, uint32 flags)
 {
 	CALLED();
 
-	TRACE("_SoundPlayNode::LatencyChanged: new_latency %Ld\n", newLatency);
+	TRACE("SoundPlayNode::LatencyChanged: new_latency %Ld\n", newLatency);
 
 	// something downstream changed latency, so we need to start producing
 	// buffers earlier (or later) than we were previously.  Make sure that the
 	// connection that changed is ours, and adjust to the new downstream
 	// latency if so.
-	if (source == mOutput.source && destination == mOutput.destination) {
+	if (source == fOutput.source && destination == fOutput.destination) {
-		mLatency = newLatency;
+		fLatency = newLatency;
-		SetEventLatency(mLatency + mInternalLatency);
+		SetEventLatency(fLatency + fInternalLatency);
 	} else {
-		TRACE("_SoundPlayNode::LatencyChanged: ignored\n");
+		TRACE("SoundPlayNode::LatencyChanged: ignored\n");
 	}
 }
 
@@ -576,7 +577,7 @@ _SoundPlayNode::LatencyChanged(const media_source& source,
 
 
 void
-_SoundPlayNode::HandleEvent(const media_timed_event* event, bigtime_t lateness,
+SoundPlayNode::HandleEvent(const media_timed_event* event, bigtime_t lateness,
 	bool realTimeEvent)
 {
 	CALLED();
@@ -619,7 +620,7 @@ _SoundPlayNode::HandleEvent(const media_timed_event* event, bigtime_t lateness,
 // how should we handle late buffers?  drop them?
 // notify the producer?
 status_t
-_SoundPlayNode::SendNewBuffer(const media_timed_event* event,
+SoundPlayNode::SendNewBuffer(const media_timed_event* event,
 	bigtime_t lateness, bool realTimeEvent)
 {
 	CALLED();
@@ -627,7 +628,7 @@ _SoundPlayNode::SendNewBuffer(const media_timed_event* event,
 
 	// make sure we're both started *and* connected before delivering a buffer
 	if (RunState() != BMediaEventLooper::B_STARTED
-		|| mOutput.destination == media_destination::null)
+		|| fOutput.destination == media_destination::null)
 		return B_OK;
 
 	// The event->event_time is the time at which the buffer we are preparing
@@ -637,12 +638,12 @@ _SoundPlayNode::SendNewBuffer(const media_timed_event* event,
 	// lateness is independent of EventLatency()!
 
 	if (lateness > (BufferDuration() / 3) ) {
-		printf("_SoundPlayNode::SendNewBuffer, event scheduled much too late, "
+		printf("SoundPlayNode::SendNewBuffer, event scheduled much too late, "
 			"lateness is %Ld\n", lateness);
 	}
 
 	// skip buffer creation if output not enabled
-	if (mOutputEnabled) {
+	if (fOutputEnabled) {
 
 		// Get the next buffer of data
 		BBuffer* buffer = FillNextBuffer(event->event_time);
@@ -651,26 +652,26 @@ _SoundPlayNode::SendNewBuffer(const media_timed_event* event,
 
 			// If we are ready way too early, decrase internal latency
 /*
-			bigtime_t how_early = event->event_time - TimeSource()->Now() - mLatency - mInternalLatency;
+			bigtime_t how_early = event->event_time - TimeSource()->Now() - fLatency - fInternalLatency;
 			if (how_early > 5000) {
 
-				printf("_SoundPlayNode::SendNewBuffer, event scheduled too early, how_early is %Ld\n", how_early);
+				printf("SoundPlayNode::SendNewBuffer, event scheduled too early, how_early is %Ld\n", how_early);
 
-				if (mTooEarlyCount++ == 5) {
+				if (fTooEarlyCount++ == 5) {
-					mInternalLatency -= how_early;
+					fInternalLatency -= how_early;
-					if (mInternalLatency < 500)
+					if (fInternalLatency < 500)
-						mInternalLatency = 500;
+						fInternalLatency = 500;
-					printf("_SoundPlayNode::SendNewBuffer setting internal latency to %Ld\n", mInternalLatency);
+					printf("SoundPlayNode::SendNewBuffer setting internal latency to %Ld\n", fInternalLatency);
-					SetEventLatency(mLatency + mInternalLatency);
+					SetEventLatency(fLatency + fInternalLatency);
-					mTooEarlyCount = 0;
+					fTooEarlyCount = 0;
 				}
 			}
 */
 			// send the buffer downstream if and only if output is enabled
-			if (B_OK != SendBuffer(buffer, mOutput.destination)) {
+			if (B_OK != SendBuffer(buffer, fOutput.destination)) {
 				// we need to recycle the buffer
 				// if the call to SendBuffer() fails
-				printf("_SoundPlayNode::SendNewBuffer: Buffer sending "
+				printf("SoundPlayNode::SendNewBuffer: Buffer sending "
 					"failed\n");
 				buffer->Recycle();
 			}
@@ -678,17 +679,17 @@ _SoundPlayNode::SendNewBuffer(const media_timed_event* event,
 	}
 
 	// track how much media we've delivered so far
-	size_t nFrames = mOutput.format.u.raw_audio.buffer_size
+	size_t nFrames = fOutput.format.u.raw_audio.buffer_size
-		/ ((mOutput.format.u.raw_audio.format
+		/ ((fOutput.format.u.raw_audio.format
 			& media_raw_audio_format::B_AUDIO_SIZE_MASK)
-		* mOutput.format.u.raw_audio.channel_count);
+		* fOutput.format.u.raw_audio.channel_count);
-	mFramesSent += nFrames;
+	fFramesSent += nFrames;
 
 	// The buffer is on its way; now schedule the next one to go
 	// nextEvent is the time at which the buffer should arrive at it's
 	// destination
-	bigtime_t nextEvent = mStartTime + bigtime_t((1000000LL * mFramesSent)
+	bigtime_t nextEvent = fStartTime + bigtime_t((1000000LL * fFramesSent)
-		/ (int32)mOutput.format.u.raw_audio.frame_rate);
+		/ (int32)fOutput.format.u.raw_audio.frame_rate);
 	media_timed_event nextBufferEvent(nextEvent, SEND_NEW_BUFFER_EVENT);
 	EventQueue()->AddEvent(nextBufferEvent);
 
@@ -697,10 +698,10 @@ _SoundPlayNode::SendNewBuffer(const media_timed_event* event,
 
 
 status_t
-_SoundPlayNode::HandleDataStatus(const media_timed_event* event,
+SoundPlayNode::HandleDataStatus(const media_timed_event* event,
 	bigtime_t lateness, bool realTimeEvent)
 {
-	TRACE("_SoundPlayNode::HandleDataStatus status: %li, lateness: %Li\n",
+	TRACE("SoundPlayNode::HandleDataStatus status: %li, lateness: %Li\n",
 		event->data, lateness);
 
 	switch (event->data) {
@@ -718,7 +719,7 @@ _SoundPlayNode::HandleDataStatus(const media_timed_event* event,
 
 
 status_t
-_SoundPlayNode::HandleStart(const media_timed_event* event, bigtime_t lateness,
+SoundPlayNode::HandleStart(const media_timed_event* event, bigtime_t lateness,
 	bool realTimeEvent)
 {
 	CALLED();
@@ -727,8 +728,8 @@ _SoundPlayNode::HandleStart(const media_timed_event* event, bigtime_t lateness,
 		// We want to start sending buffers now, so we set up the buffer-sending
 		// bookkeeping and fire off the first "produce a buffer" event.
 
-		mFramesSent = 0;
+		fFramesSent = 0;
-		mStartTime = event->event_time;
+		fStartTime = event->event_time;
 		media_timed_event firstBufferEvent(event->event_time,
 			SEND_NEW_BUFFER_EVENT);
 
@@ -744,18 +745,18 @@ _SoundPlayNode::HandleStart(const media_timed_event* event, bigtime_t lateness,
 
 
 status_t
-_SoundPlayNode::HandleSeek(const media_timed_event* event, bigtime_t lateness,
+SoundPlayNode::HandleSeek(const media_timed_event* event, bigtime_t lateness,
 	bool realTimeEvent)
 {
 	CALLED();
-	TRACE("_SoundPlayNode::HandleSeek(t=%lld, d=%li, bd=%lld)\n",
+	TRACE("SoundPlayNode::HandleSeek(t=%lld, d=%li, bd=%lld)\n",
 		event->event_time, event->data, event->bigdata);
 	return B_OK;
 }
 
 
 status_t
-_SoundPlayNode::HandleWarp(const media_timed_event* event, bigtime_t lateness,
+SoundPlayNode::HandleWarp(const media_timed_event* event, bigtime_t lateness,
 	bool realTimeEvent)
 {
 	CALLED();
@@ -764,7 +765,7 @@ _SoundPlayNode::HandleWarp(const media_timed_event* event, bigtime_t lateness,
 
 
 status_t
-_SoundPlayNode::HandleStop(const media_timed_event* event, bigtime_t lateness,
+SoundPlayNode::HandleStop(const media_timed_event* event, bigtime_t lateness,
 	bool realTimeEvent)
 {
 	CALLED();
@@ -777,7 +778,7 @@ _SoundPlayNode::HandleStop(const media_timed_event* event, bigtime_t lateness,
 
 
 status_t
-_SoundPlayNode::HandleParameter(const media_timed_event* event,
+SoundPlayNode::HandleParameter(const media_timed_event* event,
 	bigtime_t lateness, bool realTimeEvent)
 {
 	CALLED();
@@ -786,60 +787,63 @@ _SoundPlayNode::HandleParameter(const media_timed_event* event,
 
 
 void
-_SoundPlayNode::AllocateBuffers()
+SoundPlayNode::AllocateBuffers()
 {
 	CALLED();
 
 	// allocate enough buffers to span our downstream latency, plus one
-	size_t size = mOutput.format.u.raw_audio.buffer_size;
+	size_t size = fOutput.format.u.raw_audio.buffer_size;
-	int32 count = int32(mLatency / BufferDuration() + 1 + 1);
+	int32 count = int32(fLatency / BufferDuration() + 1 + 1);
 
-	TRACE("_SoundPlayNode::AllocateBuffers: latency = %Ld, buffer duration "
+	TRACE("SoundPlayNode::AllocateBuffers: latency = %Ld, buffer duration "
-		"= %Ld, count %ld\n", mLatency, BufferDuration(), count);
+		"= %Ld, count %ld\n", fLatency, BufferDuration(), count);
 
 	if (count < 3)
 		count = 3;
 
-	TRACE("_SoundPlayNode::AllocateBuffers: creating group of %ld buffers, "
+	TRACE("SoundPlayNode::AllocateBuffers: creating group of %ld buffers, "
 		"size = %lu\n", count, size);
 
-	mBufferGroup = new BBufferGroup(size, count);
+	fBufferGroup = new BBufferGroup(size, count);
-	if (mBufferGroup->InitCheck() != B_OK) {
+	if (fBufferGroup->InitCheck() != B_OK) {
-		ERROR("_SoundPlayNode::AllocateBuffers: BufferGroup::InitCheck() "
+		ERROR("SoundPlayNode::AllocateBuffers: BufferGroup::InitCheck() "
 			"failed\n");
 	}
 }
 
 
 BBuffer*
-_SoundPlayNode::FillNextBuffer(bigtime_t event_time)
+SoundPlayNode::FillNextBuffer(bigtime_t eventTime)
 {
 	CALLED();
 
 	// get a buffer from our buffer group
-	BBuffer* buf = mBufferGroup->RequestBuffer(
+	BBuffer* buffer = fBufferGroup->RequestBuffer(
-		mOutput.format.u.raw_audio.buffer_size, BufferDuration() / 2);
+		fOutput.format.u.raw_audio.buffer_size, BufferDuration() / 2);
 
 	// If we fail to get a buffer (for example, if the request times out), we
 	// skip this buffer and go on to the next, to avoid locking up the control
 	// thread
-	if (!buf) {
+	if (buffer == NULL) {
-		ERROR("_SoundPlayNode::FillNextBuffer: RequestBuffer failed\n");
+		ERROR("SoundPlayNode::FillNextBuffer: RequestBuffer failed\n");
 		return NULL;
 	}
 
-	if (mPlayer->HasData()) {
+	if (fPlayer->HasData()) {
-		mPlayer->_PlayBuffer(buf->Data(),
+		fPlayer->_PlayBuffer(buffer->Data(),
-			mOutput.format.u.raw_audio.buffer_size, mOutput.format.u.raw_audio);
+			fOutput.format.u.raw_audio.buffer_size, fOutput.format.u.raw_audio);
-	} else {
+	} else
-		memset(buf->Data(), 0, mOutput.format.u.raw_audio.buffer_size);
+		memset(buffer->Data(), 0, fOutput.format.u.raw_audio.buffer_size);
-	}
 
 	// fill in the buffer header
-	media_header* hdr = buf->Header();
+	media_header* header = buffer->Header();
-	hdr->type = B_MEDIA_RAW_AUDIO;
+	header->type = B_MEDIA_RAW_AUDIO;
-	hdr->size_used = mOutput.format.u.raw_audio.buffer_size;
+	header->size_used = fOutput.format.u.raw_audio.buffer_size;
-	hdr->time_source = TimeSource()->ID();
+	header->time_source = TimeSource()->ID();
-	hdr->start_time = event_time;
+	header->start_time = eventTime;
-	return buf;
+
+	return buffer;
 }
+
+
+}	// namespace BPrivate

src/kits/media/SoundPlayNode.h

@@ -1,177 +1,152 @@
-#ifndef _SOUND_PLAY_NODE_
+/*
-#define _SOUND_PLAY_NODE_
+ * Copyright 2002-2009, Haiku.
+ * Distributed under the terms of the MIT License.
+ *
+ * Authors:
+ *		Marcus Overhagen
+ *		Jérôme Duval
+ */
+#ifndef _SOUND_PLAY_NODE_H
+#define _SOUND_PLAY_NODE_H
+
 
-#include <BufferProducer.h>
-#include <MediaEventLooper.h>
 #include <Buffer.h>
 #include <BufferGroup.h>
-#include "SoundPlayer.h"
+#include <BufferProducer.h>
+#include <MediaEventLooper.h>
+#include <SoundPlayer.h>
 
-/***********************************************************************
- * AUTHOR: Marcus Overhagen, Jérôme Duval
- *   FILE: SoundPlayNode.h
- *  DESCR: This is the BBufferProducer, used internally by BSoundPlayer
- *         This belongs into a private namespace, but isn't for 
- *         compatibility reasons.
- ***********************************************************************/
 
-class _SoundPlayNode 
+namespace BPrivate {
-	: public BBufferProducer, public BMediaEventLooper
+
-{
+
+class SoundPlayNode : public BBufferProducer, public BMediaEventLooper {
 public:
-	_SoundPlayNode(const char *name, BSoundPlayer *player);
+								SoundPlayNode(const char* name,
-	~_SoundPlayNode();
+									BSoundPlayer* player);
+	virtual						~SoundPlayNode();
 
-	bool IsPlaying();
+			bool				IsPlaying();
-	bigtime_t CurrentTime();
+			bigtime_t			CurrentTime();
 
-/*************************/
+	// BMediaNode methods
-/* begin from BMediaNode */
+
-public:
+	virtual	BMediaAddOn*		AddOn(int32* _internalID) const;
-virtual	BMediaAddOn* AddOn(
-				int32 * internal_id) const;	/* Who instantiated you -- or NULL for app class */
 
 protected:
-		/* These don't return errors; instead, they use the global error condition reporter. */
+	virtual	void				Preroll();
-		/* A node is required to have a queue of at least one pending command (plus TimeWarp) */
-		/* and is recommended to allow for at least one pending command of each type. */
-		/* Allowing an arbitrary number of outstanding commands might be nice, but apps */
-		/* cannot depend on that happening. */
-virtual	void Preroll(void);
 
 public:
-virtual	status_t HandleMessage(
+	virtual	status_t			HandleMessage(int32 message, const void* data,
-				int32 message,
+									size_t size);
-				const void * data,
-				size_t size);
 
 protected:
-virtual		void NodeRegistered(void);	/* reserved 2 */
+	virtual	void				NodeRegistered();
-virtual		status_t RequestCompleted(const media_request_info &info);
+	virtual	status_t			RequestCompleted(
-virtual		void SetTimeSource(BTimeSource *timeSource);
+									const media_request_info& info);
-virtual 	void SetRunMode(run_mode mode);
+	virtual	void				SetTimeSource(BTimeSource* timeSource);
+	virtual void				SetRunMode(run_mode mode);
 
-/* end from BMediaNode */
+	// BBufferProducer methods
-/***********************/
 
-/******************************/
+	virtual status_t 			FormatSuggestionRequested(media_type type,
-/* begin from BBufferProducer */
+									int32 quality, media_format* format);
 
-		virtual status_t 	FormatSuggestionRequested(	media_type type,
+	virtual status_t		 	FormatProposal(const media_source& output,
-														int32 quality,
+									media_format* format);
-														media_format* format);
 
-		virtual status_t 	FormatProposal(				const media_source& output,
+	virtual status_t 			FormatChangeRequested(
-														media_format* format);
+									const media_source& source,
+									const media_destination& destination,
+									media_format* format, int32* _deprecated_);
+	virtual status_t 			GetNextOutput(int32* cookie,
+									media_output* _output);
+	virtual status_t		 	DisposeOutputCookie(int32 cookie);
 
-		virtual status_t 	FormatChangeRequested(		const media_source& source,
+	virtual	status_t 			SetBufferGroup(const media_source& forSource,
-														const media_destination& destination,
+									BBufferGroup* group);
-														media_format* io_format,
+	virtual	status_t 			GetLatency(bigtime_t* _latency);
-														int32* _deprecated_);
-		virtual status_t 	GetNextOutput(				int32* cookie,
-														media_output* out_output);
-		virtual status_t 	DisposeOutputCookie(		int32 cookie);
 
-		virtual	status_t 	SetBufferGroup(				const media_source& for_source,
+	virtual status_t 			PrepareToConnect(const media_source& what,
-														BBufferGroup* group);
+									const media_destination& where,
-		virtual	status_t 	GetLatency(					bigtime_t * out_latency);
+									media_format* format, media_source* _source,
+									char* _name);
 
-		virtual status_t 	PrepareToConnect(			const media_source& what,
+	virtual void 				Connect(status_t error,
-														const media_destination& where,
+									const media_source& source,
-														media_format* format,
+									const media_destination& destination,
-														media_source* out_source,
+									const media_format& format,
-														char* out_name);
+									char* name);
 
-		virtual void 		Connect(					status_t error, 
+	virtual void 				Disconnect(const media_source& what,
-														const media_source& source,
+									const media_destination& where);
-														const media_destination& destination,
-														const media_format& format,
-														char* io_name);
 
-		virtual void 		Disconnect(					const media_source& what,
+	virtual void 				LateNoticeReceived(const media_source& what,
-														const media_destination& where);
+									bigtime_t howMuch,
+									bigtime_t performanceTime);
 
-		virtual void 		LateNoticeReceived(			const media_source& what,
+	virtual void 				EnableOutput(const media_source& what,
-														bigtime_t how_much,
+									bool enabled, int32* _deprecated_);
-														bigtime_t performance_time);
+	virtual void 				AdditionalBufferRequested(
+									const media_source& source,
+									media_buffer_id previousBuffer,
+									bigtime_t previousTime,
+									const media_seek_tag* previousTag);
+	virtual void 				LatencyChanged(const media_source& source,
+									const media_destination& destination,
+									bigtime_t newLatency, uint32 flags);
 
-		virtual void 		EnableOutput(				const media_source & what,
+	// BMediaEventLooper methods
-														bool enabled,
-														int32* _deprecated_);
-		virtual void 		AdditionalBufferRequested(	const media_source& source,
-														media_buffer_id prev_buffer,
-														bigtime_t prev_time,
-														const media_seek_tag* prev_tag);
-		virtual void 		LatencyChanged(				const media_source& source,
-														const media_destination& destination,
-														bigtime_t new_latency,
-														uint32 flags);
-/* end from BBufferProducer */
-/****************************/
-
-/********************************/
-/* start from BMediaEventLooper */
-
-	protected:
-		/* you must override to handle your events! */
-		/* you should not call HandleEvent directly */
-		virtual void		HandleEvent(	const media_timed_event *event,
-											bigtime_t lateness,
-											bool realTimeEvent = false);
-
-/* end from BMediaEventLooper */
-/******************************/
 
+protected:
+	virtual void				HandleEvent(const media_timed_event* event,
+									bigtime_t lateness,
+									bool realTimeEvent = false);
 public:
-	media_multi_audio_format Format() const;
+			media_multi_audio_format Format() const;
 
 protected:
+	virtual status_t			HandleStart(const media_timed_event* event,
+									bigtime_t lateness,
+									bool realTimeEvent = false);
+	virtual status_t			HandleSeek(const media_timed_event* event,
+									bigtime_t lateness,
+									bool realTimeEvent = false);
+	virtual status_t			HandleWarp(const media_timed_event* event,
+									bigtime_t lateness,
+									bool realTimeEvent = false);
+	virtual status_t			HandleStop(const media_timed_event* event,
+									bigtime_t lateness,
+									bool realTimeEvent = false);
+	virtual status_t			SendNewBuffer(const media_timed_event* event,
+									bigtime_t lateness,
+									bool realTimeEvent = false);
+	virtual status_t			HandleDataStatus(const media_timed_event* event,
+									bigtime_t lateness,
+									bool realTimeEvent = false);
+	virtual status_t			HandleParameter(const media_timed_event* event,
+									bigtime_t lateness,
+									bool realTimeEvent = false);
 
-virtual status_t HandleStart(
+			void 				AllocateBuffers();
-						const media_timed_event *event,
+			BBuffer*			FillNextBuffer(bigtime_t eventTime);
-						bigtime_t lateness,
-						bool realTimeEvent = false);
-virtual status_t HandleSeek(
-						const media_timed_event *event,
-						bigtime_t lateness,
-						bool realTimeEvent = false);
-virtual status_t HandleWarp(
-						const media_timed_event *event,
-						bigtime_t lateness,
-						bool realTimeEvent = false);
-virtual status_t HandleStop(
-						const media_timed_event *event,
-						bigtime_t lateness,
-						bool realTimeEvent = false);
-virtual status_t SendNewBuffer(
-						const media_timed_event *event,
-						bigtime_t lateness,
-						bool realTimeEvent = false);
-virtual status_t HandleDataStatus(
-						const media_timed_event *event,
-						bigtime_t lateness,
-						bool realTimeEvent = false);
-virtual status_t HandleParameter(
-						const media_timed_event *event,
-						bigtime_t lateness,
-						bool realTimeEvent = false);
-	
-	void 				AllocateBuffers();
-	BBuffer*			FillNextBuffer(bigtime_t event_time);
 
 private:
-	BSoundPlayer 		*mPlayer;
+			BSoundPlayer*		fPlayer;
 
-	status_t 			mInitCheckStatus;
+			status_t 			fInitStatus;
-	bool 				mOutputEnabled;
+			bool 				fOutputEnabled;
-	media_output		mOutput;
+			media_output		fOutput;
-	BBufferGroup		*mBufferGroup;
+			BBufferGroup*		fBufferGroup;
-	media_format		mFormat;
+			bigtime_t 			fLatency;
-	bigtime_t 			mLatency;
+			bigtime_t 			fInternalLatency;
-	bigtime_t 			mInternalLatency;
+			bigtime_t 			fStartTime;
-	bigtime_t 			mStartTime;
+			uint64 				fFramesSent;
-	uint64 				mFramesSent;
+			int32				fTooEarlyCount;
-	int32				mTooEarlyCount;
 };
 
-#endif
+
+}	// namespace BPrivate
+
+
+#endif	// _SOUND_PLAY_NODE_H

src/kits/media/SoundPlayer.cpp

@@ -23,7 +23,6 @@
 
 #include "debug.h"
 
-#define atomic_read(a)	atomic_or(a, 0)
 
 // Flags used internally in BSoundPlayer
 enum {
@@ -293,10 +292,10 @@ BSoundPlayer::Latency()
 
 
 void
-BSoundPlayer::SetHasData(bool has_data)
+BSoundPlayer::SetHasData(bool hasData)
 {
 	CALLED();
-	if (has_data)
+	if (hasData)
 		atomic_or(&fFlags, F_HAS_DATA);
 	else
 		atomic_and(&fFlags, ~F_HAS_DATA);
@@ -307,7 +306,7 @@ bool
 BSoundPlayer::HasData()
 {
 	CALLED();
-	return (atomic_read(&fFlags) & F_HAS_DATA) != 0;
+	return (atomic_get(&fFlags) & F_HAS_DATA) != 0;
 }
 
 
@@ -445,7 +444,7 @@ BSoundPlayer::StartPlaying(BSound* sound, bigtime_t atTime, float withVolume)
 	CALLED();
 
 	// TODO: support the at_time and with_volume parameters
-	_playing_sound *item = (_playing_sound *)malloc(sizeof(_playing_sound));
+	playing_sound* item = (playing_sound*)malloc(sizeof(playing_sound));
 	if (item == NULL)
 		return B_NO_MEMORY;
 
@@ -478,7 +477,7 @@ BSoundPlayer::SetSoundVolume(play_id id, float newVolume)
 	if (!fLocker.Lock())
 		return B_ERROR;
 
-	_playing_sound *item = fPlayingSounds;
+	playing_sound *item = fPlayingSounds;
 	while (item) {
 		if (item->id == id) {
 			item->volume = newVolume;
@@ -501,7 +500,7 @@ BSoundPlayer::IsPlaying(play_id id)
 	if (!fLocker.Lock())
 		return B_ERROR;
 
-	_playing_sound *item = fPlayingSounds;
+	playing_sound *item = fPlayingSounds;
 	while (item) {
 		if (item->id == id) {
 			fLocker.Unlock();
@@ -523,8 +522,8 @@ BSoundPlayer::StopPlaying(play_id id)
 	if (!fLocker.Lock())
 		return B_ERROR;
 
-	_playing_sound** link = &fPlayingSounds;
+	playing_sound** link = &fPlayingSounds;
-	_playing_sound* item = fPlayingSounds;
+	playing_sound* item = fPlayingSounds;
 
 	while (item != NULL) {
 		if (item->id == id) {
@@ -557,7 +556,7 @@ BSoundPlayer::WaitForSound(play_id id)
 	if (!fLocker.Lock())
 		return B_ERROR;
 
-	_playing_sound* item = fPlayingSounds;
+	playing_sound* item = fPlayingSounds;
 	while (item != NULL) {
 		if (item->id == id) {
 			sem_id waitSem = item->wait_sem;
@@ -684,7 +683,7 @@ BSoundPlayer::_SoundPlayBufferFunc(void *cookie, void *buffer, size_t size,
 		return;
 	}
 
-	_playing_sound *sound = player->fPlayingSounds;
+	playing_sound *sound = player->fPlayingSounds;
 	if (sound == NULL) {
 		player->SetHasData(false);
 		player->fLocker.Unlock();
@@ -774,7 +773,7 @@ BSoundPlayer::_Init(const media_node* node,
 	media_format tryFormat;
 
 	// Create the player node and register it
-	fPlayerNode = new _SoundPlayNode(name, this);
+	fPlayerNode = new BPrivate::SoundPlayNode(name, this);
 	fInitStatus = roster->RegisterNode(fPlayerNode);
 	if (fInitStatus != B_OK) {
 		TRACE("BSoundPlayer::_Init: Couldn't RegisterNode: %s\n",