* The AudioProducer now correctly handles late producer notices by ignoring

extra notices for buffers already scheduled.
* Also, the AudioSupplier/AudioReader classes now know their initial latency,
  and the AudioProducer is now using that one to advertize its own initial
  latency - this fixes late buffers on start, causing the latency to grow
  too large.
* Cleanup here and there.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@36199 a95241bf-73f2-0310-859d-f6bbb57e9c96

src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.cpp

@@ -77,6 +77,13 @@ AudioAdapter::~AudioAdapter()
 }
 
 
+bigtime_t
+AudioAdapter::InitialLatency() const
+{
+	return fSource->InitialLatency();
+}
+
+
 status_t
 AudioAdapter::Read(void* buffer, int64 pos, int64 frames)
 {

src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.h

@@ -1,7 +1,10 @@
 /*
- * Copyright © 2000-2006 Ingo Weinhold <[email protected]>
+ * Copyright 2000-2006 Ingo Weinhold <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
+#ifndef AUDIO_ADAPTER_H
+#define AUDIO_ADAPTER_H
+
 
 /*! This AudioReader slaves an AudioConverter and an AudioResampler
 	to convert the source data to a given format.
@@ -10,21 +13,22 @@
 	If input and output format are the same, the overhead is quit small.
 */
 
-#ifndef AUDIO_ADAPTER_H
-#define AUDIO_ADAPTER_H
 
 #include "AudioReader.h"
 
+
 class AudioChannelConverter;
 class AudioFormatConverter;
 class AudioResampler;
 
+
 class AudioAdapter : public AudioReader {
 public:
 								AudioAdapter(AudioReader* source,
 									const media_format& format);
 	virtual						~AudioAdapter();
 
+	virtual bigtime_t			InitialLatency() const;
 	virtual	status_t			Read(void* buffer, int64 pos, int64 frames);
 
 	virtual	status_t			InitCheck() const;

src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.cpp

@@ -1,7 +1,9 @@
 /*
- * Copyright © 2008 Stephan Aßmus <[email protected]>
+ * Copyright 2008 Stephan Aßmus <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
+
+
 #include "AudioChannelConverter.h"
 
 #include <new>
@@ -10,29 +12,15 @@
 
 using std::nothrow;
 
+
 //#define TRACE_AUDIO_CONVERTER
 #ifdef TRACE_AUDIO_CONVERTER
-# define TRACE(x...)	printf(x)
+#	define TRACE(x...)	printf(x)
 #else
-# define TRACE(x...)
+#	define TRACE(x...)
 #endif
 
 
-AudioChannelConverter::AudioChannelConverter(AudioReader* source,
-		const media_format& format)
-	: AudioReader(format),
-	  fSource(source)
-{
-	// TODO: check the format and make sure everything matches
-	// except for channel count
-}
-
-
-AudioChannelConverter::~AudioChannelConverter()
-{
-}
-
-
 template<typename Type, typename BigType>
 static void
 convert(Type* inBuffer, Type* outBuffer, int32 inChannels, int32 outChannels,
@@ -79,6 +67,31 @@ convert(Type* inBuffer, Type* outBuffer, int32 inChannels, int32 outChannels,
 }
 
 
+// #pragma mark -
+
+
+AudioChannelConverter::AudioChannelConverter(AudioReader* source,
+		const media_format& format)
+	: AudioReader(format),
+	  fSource(source)
+{
+	// TODO: check the format and make sure everything matches
+	// except for channel count
+}
+
+
+AudioChannelConverter::~AudioChannelConverter()
+{
+}
+
+
+bigtime_t
+AudioChannelConverter::InitialLatency() const
+{
+	fSource->InitialLatency();
+}
+
+
 status_t
 AudioChannelConverter::Read(void* outBuffer, int64 pos, int64 frames)
 {

src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.h

@@ -1,24 +1,27 @@
 /*
- * Copyright © 2008 Stephan Aßmus <[email protected]>
+ * Copyright 2008 Stephan Aßmus <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
+#ifndef AUDIO_CHANNEL_CONVERTER_H
+#define AUDIO_CHANNEL_CONVERTER_H
+
 
 /*! This AudioReader just converts the source channel count
 	into another one, e.g. 1 -> 2. Frame rate and sample format
 	remain unchanged.
 */
 
-#ifndef AUDIO_CHANNEL_CONVERTER_H
-#define AUDIO_CHANNEL_CONVERTER_H
 
 #include "AudioReader.h"
 
+
 class AudioChannelConverter : public AudioReader {
 public:
 								AudioChannelConverter(AudioReader* source,
 									const media_format& format);
 	virtual						~AudioChannelConverter();
 
+	virtual bigtime_t			InitialLatency() const;
 	virtual	status_t			Read(void* buffer, int64 pos, int64 frames);
 
 	virtual	status_t			InitCheck() const;

src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.cpp

@@ -1,9 +1,10 @@
 /*
- * Copyright © 2000-2006 Ingo Weinhold <[email protected]>
+ * Copyright 2000-2006 Ingo Weinhold <[email protected]>
- * Copyright © 2008 Stephan Aßmus <[email protected]>
+ * Copyright 2008 Stephan Aßmus <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
 
+
 #include "AudioFormatConverter.h"
 
 #include <ByteOrder.h>
@@ -12,44 +13,12 @@
 
 //#define TRACE_AUDIO_CONVERTER
 #ifdef TRACE_AUDIO_CONVERTER
-# define TRACE(x...)	printf(x)
+#	define TRACE(x...)	printf(x)
 #else
-# define TRACE(x...)
+#	define TRACE(x...)
 #endif
 
 
-AudioFormatConverter::AudioFormatConverter(AudioReader* source, uint32 format,
-		uint32 byte_order)
-	: AudioReader(),
-	  fSource(NULL)
-{
-	uint32 hostByteOrder
-		= (B_HOST_IS_BENDIAN) ? B_MEDIA_BIG_ENDIAN : B_MEDIA_LITTLE_ENDIAN;
-	if (source && source->Format().type == B_MEDIA_RAW_AUDIO
-		&& source->Format().u.raw_audio.byte_order == hostByteOrder) {
-		fFormat = source->Format();
-		fFormat.u.raw_audio.format = format;
-		fFormat.u.raw_audio.byte_order = byte_order;
-		int32 inSampleSize = source->Format().u.raw_audio.format
-			& media_raw_audio_format::B_AUDIO_SIZE_MASK;
-		int32 outSampleSize = fFormat.u.raw_audio.format
-			& media_raw_audio_format::B_AUDIO_SIZE_MASK;
-		if (inSampleSize != outSampleSize) {
-			fFormat.u.raw_audio.buffer_size
-				= source->Format().u.raw_audio.buffer_size * outSampleSize
-				  / inSampleSize;
-		}
-	} else
-		source = NULL;
-	fSource = source;
-}
-
-
-AudioFormatConverter::~AudioFormatConverter()
-{
-}
-
-
 struct ReadFloat {
 	inline int operator()(const void* buffer) const {
 		// 0 == mid, -1.0 == bottom, 1.0 == top
@@ -150,8 +119,7 @@ convert(const ReadT& read, const WriteT& write,
 }
 
 
-static
+static void
-void
 swap_sample_byte_order(void* buffer, uint32 format, size_t length)
 {
 	type_code type = B_ANY_TYPE;
@@ -175,6 +143,48 @@ swap_sample_byte_order(void* buffer, uint32 format, size_t length)
 }
 
 
+// #pragma mark -
+
+
+AudioFormatConverter::AudioFormatConverter(AudioReader* source, uint32 format,
+		uint32 byte_order)
+	:
+	AudioReader(),
+	fSource(NULL)
+{
+	uint32 hostByteOrder
+		= (B_HOST_IS_BENDIAN) ? B_MEDIA_BIG_ENDIAN : B_MEDIA_LITTLE_ENDIAN;
+	if (source && source->Format().type == B_MEDIA_RAW_AUDIO
+		&& source->Format().u.raw_audio.byte_order == hostByteOrder) {
+		fFormat = source->Format();
+		fFormat.u.raw_audio.format = format;
+		fFormat.u.raw_audio.byte_order = byte_order;
+		int32 inSampleSize = source->Format().u.raw_audio.format
+			& media_raw_audio_format::B_AUDIO_SIZE_MASK;
+		int32 outSampleSize = fFormat.u.raw_audio.format
+			& media_raw_audio_format::B_AUDIO_SIZE_MASK;
+		if (inSampleSize != outSampleSize) {
+			fFormat.u.raw_audio.buffer_size
+				= source->Format().u.raw_audio.buffer_size * outSampleSize
+				  / inSampleSize;
+		}
+	} else
+		source = NULL;
+	fSource = source;
+}
+
+
+AudioFormatConverter::~AudioFormatConverter()
+{
+}
+
+
+bigtime_t
+AudioFormatConverter::InitialLatency() const
+{
+	fSource->InitialLatency();
+}
+
 status_t
 AudioFormatConverter::Read(void* buffer, int64 pos, int64 frames)
 {

src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.h

@@ -1,25 +1,28 @@
 /*
- * Copyright © 2000-2006 Ingo Weinhold <[email protected]>
+ * Copyright 2000-2006 Ingo Weinhold <[email protected]>
- * Copyright © 2008 Stephan Aßmus <[email protected]>
+ * Copyright 2008 Stephan Aßmus <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
+#ifndef AUDIO_FORMAT_CONVERTER_H
+#define AUDIO_FORMAT_CONVERTER_H
+
 
 /*! This AudioReader just converts the source sample format (and byte order)
 	into another one, e.g. LE short -> BE float. Frame rate and channel
 	count remain unchanged.
 */
 
-#ifndef AUDIO_FORMAT_CONVERTER_H
-#define AUDIO_FORMAT_CONVERTER_H
 
 #include "AudioReader.h"
 
+
 class AudioFormatConverter : public AudioReader {
 public:
 								AudioFormatConverter(AudioReader* source,
 									uint32 format, uint32 byte_order);
 	virtual						~AudioFormatConverter();
 
+	virtual bigtime_t			InitialLatency() const;
 	virtual	status_t			Read(void* buffer, int64 pos, int64 frames);
 
 	virtual	status_t			InitCheck() const;

src/apps/mediaplayer/media_node_framework/audio/AudioProducer.cpp

@@ -37,13 +37,13 @@
 // debugging
 //#define TRACE_AUDIO_PRODUCER
 #ifdef TRACE_AUDIO_PRODUCER
-# define TRACE(x...)		printf(x)
+#	define TRACE(x...)		printf(x)
-# define TRACE_BUFFER(x...)
+#	define TRACE_BUFFER(x...)
-# define ERROR(x...)		fprintf(stderr, x)
+#	define ERROR(x...)		fprintf(stderr, x)
 #else
-# define TRACE(x...)
+#	define TRACE(x...)
-# define TRACE_BUFFER(x...)
+#	define TRACE_BUFFER(x...)
-# define ERROR(x...)		fprintf(stderr, x)
+#	define ERROR(x...)		fprintf(stderr, x)
 #endif
 
 
@@ -85,24 +85,63 @@ init_media_file(media_format format, BMediaTrack** _track)
 #endif // DEBUG_TO_FILE
 
 
+static bigtime_t
+estimate_internal_latency(const media_format& format)
+{
+	bigtime_t startTime = system_time();
+	// calculate the number of samples per buffer
+	int32 sampleSize = format.u.raw_audio.format
+		& media_raw_audio_format::B_AUDIO_SIZE_MASK;
+	int32 sampleCount = format.u.raw_audio.buffer_size / sampleSize;
+	// alloc float buffers of this size
+	const int bufferCount = 10;	// number of input buffers
+	float* buffers[bufferCount + 1];
+	for (int32 i = 0; i < bufferCount + 1; i++)
+		buffers[i] = new float[sampleCount];
+	float* outBuffer = buffers[bufferCount];
+	// fill all buffers save the last one with arbitrary data and merge them
+	// into the last one
+	for (int32 i = 0; i < bufferCount; i++) {
+		for (int32 k = 0; k < sampleCount; k++) {
+			buffers[i][k] = ((float)i * (float)k)
+				/ float(bufferCount * sampleCount);
+		}
+	}
+	for (int32 k = 0; k < sampleCount; k++) {
+		outBuffer[k] = 0;
+		for (int32 i = 0; i < bufferCount; i++)
+			outBuffer[k] += buffers[i][k];
+		outBuffer[k] /= bufferCount;
+	}
+	// cleanup
+	for (int32 i = 0; i < bufferCount + 1; i++)
+		delete[] buffers[i];
+	return system_time() - startTime;
+}
+
+
+// #pragma mark -
+
 
-// constructor
 AudioProducer::AudioProducer(const char* name, AudioSupplier* supplier,
 		bool lowLatency)
-	: BMediaNode(name),
+	:
-	  BBufferProducer(B_MEDIA_RAW_AUDIO),
+	BMediaNode(name),
-	  BMediaEventLooper(),
+	BBufferProducer(B_MEDIA_RAW_AUDIO),
+	BMediaEventLooper(),
 
-	  fBufferGroup(NULL),
+	fBufferGroup(NULL),
-	  fLatency(0),
+	fLatency(0),
-	  fInternalLatency(0),
+	fInternalLatency(0),
-	  fLowLatency(lowLatency),
+	fLastLateNotice(0),
-	  fOutputEnabled(true),
+	fNextScheduledBuffer(0),
-	  fFramesSent(0),
+	fLowLatency(lowLatency),
-	  fStartTime(0),
+	fOutputEnabled(true),
-	  fSupplier(supplier),
+	fFramesSent(0),
+	fStartTime(0),
+	fSupplier(supplier),
 
-	  fPeakListener(NULL)
+	fPeakListener(NULL)
 {
 	TRACE("%p->AudioProducer::AudioProducer(%s, %p, %d)\n", this, name,
 		supplier, lowLatency);
@@ -135,9 +174,11 @@ AudioProducer::AudioProducer(const char* name, AudioSupplier* supplier,
 	// we're not connected yet
 	fOutput.destination = media_destination::null;
 	fOutput.format = fPreferredFormat;
+
 	// init the audio supplier
-	if (fSupplier) {
+	if (fSupplier != NULL) {
 		fSupplier->SetAudioProducer(this);
+		SetInitialLatency(fSupplier->InitialLatency());
 	}
 }
 
@@ -359,41 +400,6 @@ AudioProducer::PrepareToConnect(const media_source& what,
 }
 
 
-static bigtime_t
-estimate_internal_latency(const media_format& format)
-{
-	bigtime_t startTime = system_time();
-	// calculate the number of samples per buffer
-	int32 sampleSize = format.u.raw_audio.format
-		& media_raw_audio_format::B_AUDIO_SIZE_MASK;
-	int32 sampleCount = format.u.raw_audio.buffer_size / sampleSize;
-	// alloc float buffers of this size
-	const int bufferCount = 10;	// number of input buffers
-	float* buffers[bufferCount + 1];
-	for (int32 i = 0; i < bufferCount + 1; i++)
-		buffers[i] = new float[sampleCount];
-	float* outBuffer = buffers[bufferCount];
-	// fill all buffers save the last one with arbitrary data and merge them
-	// into the last one
-	for (int32 i = 0; i < bufferCount; i++) {
-		for (int32 k = 0; k < sampleCount; k++) {
-			buffers[i][k] = ((float)i * (float)k)
-				/ float(bufferCount * sampleCount);
-		}
-	}
-	for (int32 k = 0; k < sampleCount; k++) {
-		outBuffer[k] = 0;
-		for (int32 i = 0; i < bufferCount; i++)
-			outBuffer[k] += buffers[i][k];
-		outBuffer[k] /= bufferCount;
-	}
-	// cleanup
-	for (int32 i = 0; i < bufferCount + 1; i++)
-		delete[] buffers[i];
-	return system_time() - startTime;
-}
-
-
 void
 AudioProducer::Connect(status_t error, const media_source& source,
 	 const media_destination& destination, const media_format& format,
@@ -490,6 +496,13 @@ AudioProducer::LateNoticeReceived(const media_source& what, bigtime_t howMuch,
 	// If we're late, we need to catch up. Respond in a manner appropriate
 	// to our current run mode.
 	if (what == fOutput.source) {
+		// Ignore the notices for buffers we already send out (or scheduled
+		// their event) before we processed the last notice
+		if (fLastLateNotice > performanceTime)
+			return;
+
+		fLastLateNotice = fNextScheduledBuffer;
+
 		if (RunMode() == B_RECORDING) {
 			// ...
 		} else if (RunMode() == B_INCREASE_LATENCY) {
@@ -595,9 +608,10 @@ AudioProducer::HandleEvent(const media_timed_event* event, bigtime_t lateness,
 			TRACE("AudioProducer::HandleEvent(B_START)\n");
 			if (RunState() != B_STARTED) {
 				fFramesSent = 0;
-				fStartTime = event->event_time;
+				fStartTime = event->event_time + fSupplier->InitialLatency();
 printf("B_START: start time: %lld\n", fStartTime);
-				media_timed_event firstBufferEvent(fStartTime,
+				media_timed_event firstBufferEvent(
+					fStartTime - fSupplier->InitialLatency(),
 					BTimedEventQueue::B_HANDLE_BUFFER);
 				EventQueue()->AddEvent(firstBufferEvent);
 			}
@@ -607,7 +621,7 @@ printf("B_START: start time: %lld\n", fStartTime);
 		case BTimedEventQueue::B_STOP:
 			TRACE("AudioProducer::HandleEvent(B_STOP)\n");
 			EventQueue()->FlushEvents(0, BTimedEventQueue::B_ALWAYS, true,
-									  BTimedEventQueue::B_HANDLE_BUFFER);
+				BTimedEventQueue::B_HANDLE_BUFFER);
 			TRACE("AudioProducer::HandleEvent(B_STOP) done\n");
 			break;
 
@@ -633,10 +647,10 @@ printf("B_START: start time: %lld\n", fStartTime);
 					/ (sampleSize * fOutput.format.u.raw_audio.channel_count);
 				fFramesSent += nFrames;
 
-				bigtime_t nextEvent = fStartTime
+				fNextScheduledBuffer = fStartTime
 					+ bigtime_t(double(fFramesSent) * 1000000.0
-					  / double(fOutput.format.u.raw_audio.frame_rate));
+						/ double(fOutput.format.u.raw_audio.frame_rate));
-				media_timed_event nextBufferEvent(nextEvent,
+				media_timed_event nextBufferEvent(fNextScheduledBuffer,
 					BTimedEventQueue::B_HANDLE_BUFFER);
 				EventQueue()->AddEvent(nextBufferEvent);
 			} else {
@@ -663,7 +677,8 @@ AudioProducer::ChangeFormat(media_format* format)
 {
 	TRACE("AudioProducer::ChangeFormat()\n");
 
-	format->u.raw_audio.buffer_size = media_raw_audio_format::wildcard.buffer_size;
+	format->u.raw_audio.buffer_size
+		= media_raw_audio_format::wildcard.buffer_size;
 
 	status_t ret = _SpecializeFormat(format);
 	if (ret != B_OK) {
@@ -677,7 +692,8 @@ AudioProducer::ChangeFormat(media_format* format)
 		return ret;
 	}
 
-	ret = BBufferProducer::ChangeFormat(fOutput.source, fOutput.destination, format);
+	ret = BBufferProducer::ChangeFormat(fOutput.source, fOutput.destination,
+		format);
 	if (ret != B_OK) {
 		TRACE("  ChangeFormat(): %s\n", strerror(ret));
 		return ret;
@@ -707,20 +723,20 @@ AudioProducer::_SpecializeFormat(media_format* format)
 	}
 
 	if (format->u.raw_audio.channel_count
-		== media_raw_audio_format::wildcard.channel_count) {
+			== media_raw_audio_format::wildcard.channel_count) {
 		format->u.raw_audio.channel_count = 2;
 		TRACE("  -> adjusting channel count, it was wildcard\n");
 	}
 
 	if (format->u.raw_audio.frame_rate
-		== media_raw_audio_format::wildcard.frame_rate) {
+			== media_raw_audio_format::wildcard.frame_rate) {
 		format->u.raw_audio.frame_rate = 44100.0;
 		TRACE("  -> adjusting frame rate, it was wildcard\n");
 	}
 
 	// check the buffer size, which may still be wildcarded
 	if (format->u.raw_audio.buffer_size
-		== media_raw_audio_format::wildcard.buffer_size) {
+			== media_raw_audio_format::wildcard.buffer_size) {
 		// pick something comfortable to suggest
 		TRACE("  -> adjusting buffer size, it was wildcard\n");
 
@@ -734,7 +750,6 @@ AudioProducer::_SpecializeFormat(media_format* format)
 
 		if (!fLowLatency)
 			format->u.raw_audio.buffer_size *= 3;
-
 	}
 
 	return B_OK;
@@ -794,9 +809,9 @@ AudioProducer::_FillNextBuffer(bigtime_t eventTime)
 		// number of sample in the buffer
 
 	// fill in the buffer header
-	media_header* hdr = buffer->Header();
+	media_header* header = buffer->Header();
-	hdr->type = B_MEDIA_RAW_AUDIO;
+	header->type = B_MEDIA_RAW_AUDIO;
-	hdr->time_source = TimeSource()->ID();
+	header->time_source = TimeSource()->ID();
 	buffer->SetSizeUsed(fOutput.format.u.raw_audio.buffer_size);
 
 	bigtime_t performanceTime = bigtime_t(double(fFramesSent)
@@ -817,9 +832,9 @@ AudioProducer::_FillNextBuffer(bigtime_t eventTime)
 
 	// stamp buffer
 	if (RunMode() == B_RECORDING) {
-		hdr->start_time = eventTime;
+		header->start_time = eventTime;
 	} else {
-		hdr->start_time = fStartTime + performanceTime;
+		header->start_time = fStartTime + performanceTime;
 	}
 
 #if DEBUG_TO_FILE

src/apps/mediaplayer/media_node_framework/audio/AudioProducer.h

@@ -8,10 +8,11 @@
 #ifndef AUDIO_PRODUCER_H
 #define AUDIO_PRODUCER_H
 
+
 #include <BufferProducer.h>
-//#include <Controllable.h>
 #include <MediaEventLooper.h>
 
+
 class AudioSupplier;
 class BHandler;
 
@@ -19,6 +20,7 @@ enum {
 	MSG_PEAK_NOTIFICATION		= 'pknt'
 };
 
+
 class AudioProducer : public BBufferProducer, public BMediaEventLooper {
 public:
 								AudioProducer(const char* name,
@@ -116,6 +118,8 @@ private:
 			bool				fUsingOurBuffers;
 			bigtime_t			fLatency;
 			bigtime_t			fInternalLatency;
+			bigtime_t			fLastLateNotice;
+			bigtime_t			fNextScheduledBuffer;
 			bool				fLowLatency;
 			media_output		fOutput;
 			bool				fOutputEnabled;

src/apps/mediaplayer/media_node_framework/audio/AudioReader.h

@@ -18,6 +18,7 @@ public:
 			void				SetFormat(const media_format& format);
 			const media_format&	Format() const;
 
+	virtual bigtime_t			InitialLatency() const = 0;
 	virtual	status_t			Read(void* buffer, int64 pos, int64 frames) = 0;
 
 			void				SetOutOffset(int64 offset);

src/apps/mediaplayer/media_node_framework/audio/AudioResampler.cpp

@@ -1,7 +1,9 @@
 /*
- * Copyright © 2000-2006 Ingo Weinhold <[email protected]>
+ * Copyright 2000-2006 Ingo Weinhold <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
+
+
 #include "AudioResampler.h"
 
 #include <stdio.h>
@@ -11,39 +13,12 @@
 
 //#define TRACE_AUDIO_RESAMPLER
 #ifdef TRACE_AUDIO_RESAMPLER
-# define TRACE(x...)	printf(x)
+#	define TRACE(x...)	printf(x)
 #else
-# define TRACE(x...)
+#	define TRACE(x...)
 #endif
 
 
-AudioResampler::AudioResampler()
-	: AudioReader(),
-	  fSource(NULL),
-	  fTimeScale(1.0),
-	  fInOffset(0)
-{
-}
-
-
-AudioResampler::AudioResampler(AudioReader* source, float frameRate,
-		float timeScale)
-	: AudioReader(),
-	  fSource(NULL),
-	  fTimeScale(timeScale),
-	  fInOffset(0)
-{
-	SetSource(source);
-	if (fSource)
-		fFormat.u.raw_audio.frame_rate = frameRate;
-}
-
-
-AudioResampler::~AudioResampler()
-{
-}
-
-
 //! Calculates the greatest common divider of /a/ and /b/.
 template<typename T> inline
 T
@@ -92,6 +67,45 @@ resample_linear(void* _inBuffer, void* _outBuffer, uint32 channelCount,
 }
 
 
+// #pragma mark -
+
+
+AudioResampler::AudioResampler()
+	:
+	AudioReader(),
+	fSource(NULL),
+	fTimeScale(1.0),
+	fInOffset(0)
+{
+}
+
+
+AudioResampler::AudioResampler(AudioReader* source, float frameRate,
+		float timeScale)
+	:
+	AudioReader(),
+	fSource(NULL),
+	fTimeScale(timeScale),
+	fInOffset(0)
+{
+	SetSource(source);
+	if (fSource)
+		fFormat.u.raw_audio.frame_rate = frameRate;
+}
+
+
+AudioResampler::~AudioResampler()
+{
+}
+
+
+bigtime_t
+AudioResampler::InitialLatency() const
+{
+	return fSource->InitialLatency();
+}
+
+
 status_t
 AudioResampler::Read(void* buffer, int64 pos, int64 frames)
 {
@@ -185,7 +199,7 @@ AudioResampler::SetSource(AudioReader* source)
 		TRACE("AudioResampler::SetSource() - NULL source\n");
 		return;
 	}
-		
+
 	if (source->Format().type != B_MEDIA_RAW_AUDIO) {
 		TRACE("AudioResampler::SetSource() - not B_MEDIA_RAW_AUDIO\n");
 		return;
@@ -197,7 +211,7 @@ AudioResampler::SetSource(AudioReader* source)
 		TRACE("AudioResampler::SetSource() - not host byte order\n");
 		return;
 	}
-	
+
 	float frameRate = FrameRate();
 		// don't overwrite previous audio frame rate
 	fSource = source;

src/apps/mediaplayer/media_node_framework/audio/AudioResampler.h

@@ -1,7 +1,10 @@
 /*
- * Copyright © 2000-2006 Ingo Weinhold <[email protected]>
+ * Copyright 2000-2006 Ingo Weinhold <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
+#ifndef AUDIO_RESAMPLER_H
+#define AUDIO_RESAMPLER_H
+
 
 /*! This AudioReader does both resampling an audio source to a different
 	sample rate and rescaling the time, e.g. it is possible to convert the
@@ -9,11 +12,10 @@
 	(time scale = -2).
 */
 
-#ifndef AUDIO_RESAMPLER_H
-#define AUDIO_RESAMPLER_H
 
 #include "AudioReader.h"
 
+
 class AudioResampler : public AudioReader {
 public:
 								AudioResampler();
@@ -21,6 +23,7 @@ public:
 									float frameRate, float timeScale = 1.0);
 	virtual						~AudioResampler();
 
+	virtual bigtime_t			InitialLatency() const;
 	virtual	status_t			Read(void* buffer, int64 pos, int64 frames);
 
 	virtual	status_t			InitCheck() const;

src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.cpp

@@ -1,6 +1,8 @@
-/*	Copyright (c) 2000-2008, Ingo Weinhold <[email protected]>,
+/*	Copyright 2000-2008, Ingo Weinhold <[email protected]>,
  *	All Rights Reserved. Distributed under the terms of the MIT license.
  */
+
+
 #include "AudioSupplier.h"
 
 #include "AudioProducer.h"
@@ -26,6 +28,6 @@ AudioSupplier::SetAudioProducer(AudioProducer* producer)
 status_t
 AudioSupplier::InitCheck() const
 {
-	return (fAudioProducer ? B_OK : B_NO_INIT);
+	return fAudioProducer ? B_OK : B_NO_INIT;
 }
 

src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.h

@@ -1,11 +1,12 @@
-/*	Copyright (c) 2000-2008, Ingo Weinhold <[email protected]>,
+/*	Copyright 2000-2008, Ingo Weinhold <[email protected]>,
  *	All Rights Reserved. Distributed under the terms of the MIT license.
  */
+#ifndef AUDIO_SUPPLIER_H
+#define AUDIO_SUPPLIER_H
+
 
 /*!	This class is an interface used by the AudioProducer to retreive the
 	audio data to be played. */
-#ifndef AUDIO_SUPPLIER_H
-#define AUDIO_SUPPLIER_H
 
 
 #include <MediaDefs.h>
@@ -14,21 +15,23 @@
 class AudioProducer;
 
 class AudioSupplier {
- public:
+public:
 								AudioSupplier();
 	virtual						~AudioSupplier();
 
 	virtual	void				SetAudioProducer(AudioProducer* producer);
 
+	virtual	status_t			InitCheck() const;
+
+	virtual bigtime_t			InitialLatency() const = 0;
+
 	virtual	status_t			GetFrames(void* buffer, int64 frameCount,
 									bigtime_t startTime,
 									bigtime_t endTime) = 0;
 
 	virtual	void				SetFormat(const media_format& format) = 0;
 
-	virtual	status_t			InitCheck() const;
+protected:
-
- protected:
 	AudioProducer*				fAudioProducer;
 };
 

src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.cpp

@@ -1,8 +1,9 @@
 /*
- * Copyright © 2008 Stephan Aßmus <[email protected]>
+ * Copyright 2008 Stephan Aßmus <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
 
+
 #include "AudioVolumeConverter.h"
 
 #include <stdio.h>
@@ -13,31 +14,12 @@
 
 //#define TRACE_AUDIO_CONVERTER
 #ifdef TRACE_AUDIO_CONVERTER
-# define TRACE(x...)	printf(x)
+#	define TRACE(x...)	printf(x)
 #else
-# define TRACE(x...)
+#	define TRACE(x...)
 #endif
 
 
-AudioVolumeConverter::AudioVolumeConverter(AudioReader* source, float volume)
-	: AudioReader(),
-	  fSource(NULL),
-	  fVolume(volume),
-	  fPreviousVolume(volume)
-{
-	if (source && source->Format().type == B_MEDIA_RAW_AUDIO)
-		fFormat = source->Format();
-	else
-		source = NULL;
-	fSource = source;
-}
-
-
-AudioVolumeConverter::~AudioVolumeConverter()
-{
-}
-
-
 template<typename SampleType>
 static void
 convert(SampleType* buffer, const int32 samples, const float volume,
@@ -66,6 +48,36 @@ convert(SampleType* buffer, const int32 frames, const int32 channels,
 }
 
 
+// #pragma mark -
+
+
+AudioVolumeConverter::AudioVolumeConverter(AudioReader* source, float volume)
+	:
+	AudioReader(),
+	fSource(NULL),
+	fVolume(volume),
+	fPreviousVolume(volume)
+{
+	if (source && source->Format().type == B_MEDIA_RAW_AUDIO)
+		fFormat = source->Format();
+	else
+		source = NULL;
+	fSource = source;
+}
+
+
+AudioVolumeConverter::~AudioVolumeConverter()
+{
+}
+
+
+bigtime_t
+AudioVolumeConverter::InitialLatency() const
+{
+	return fSource->InitialLatency();
+}
+
+
 status_t
 AudioVolumeConverter::Read(void* buffer, int64 pos, int64 frames)
 {

src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.h

@@ -1,23 +1,26 @@
 /*
- * Copyright © 2008 Stephan Aßmus <[email protected]>
+ * Copyright 2008 Stephan Aßmus <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT licensce.
  */
+#ifndef AUDIO_VOLUME_CONVERTER_H
+#define AUDIO_VOLUME_CONVERTER_H
+
 
 /*! This AudioReader just filters the volume. It depends on floating point
  *	audio format.
 */
 
-#ifndef AUDIO_VOLUME_CONVERTER_H
-#define AUDIO_VOLUME_CONVERTER_H
 
 #include "AudioReader.h"
 
+
 class AudioVolumeConverter : public AudioReader {
 public:
 								AudioVolumeConverter(AudioReader* source,
 									float volume = 1.0);
 	virtual						~AudioVolumeConverter();
 
+	virtual bigtime_t			InitialLatency() const;
 	virtual	status_t			Read(void* buffer, int64 pos, int64 frames);
 
 	virtual	status_t			InitCheck() const;

src/apps/mediaplayer/supplier/MediaTrackAudioSupplier.cpp

@@ -115,7 +115,14 @@ MediaTrackAudioSupplier::Duration() const
 // #pragma mark - AudioReader
 
 
-// Read
+bigtime_t
+MediaTrackAudioSupplier::InitialLatency() const
+{
+	// TODO: this is just a wild guess, and not really founded on anything.
+	return 100000;
+}
+
+
 status_t
 MediaTrackAudioSupplier::Read(void* buffer, int64 pos, int64 frames)
 {
@@ -518,15 +525,15 @@ MediaTrackAudioSupplier::_ReadCachedFrames(void*& dest, int64& pos,
 	}
 }
 
-// _ReadUncachedFrames
+
-//
+/*!	Reads /frames/ frames from /position/ into /buffer/. The frames are not
-// Reads /frames/ frames from /position/ into /buffer/. The frames are not
+	read from the cache, but read frames are cached, if possible.
-// read from the cache, but read frames are cached, if possible.
+	New cache buffers are stamped with the supplied time.
-// New cache buffers are stamped with the supplied time.
+	If an error occurs, the untouched part of the buffer is set to 0.
-// If an error occurs, the untouched part of the buffer is set to 0.
+*/
 status_t
 MediaTrackAudioSupplier::_ReadUncachedFrames(void* buffer, int64 position,
-									  int64 frames, bigtime_t time)
+	int64 frames, bigtime_t time)
 {
 	TRACE("_ReadUncachedFrames()\n");
 	status_t error = B_OK;

src/apps/mediaplayer/supplier/MediaTrackAudioSupplier.h

@@ -1,21 +1,24 @@
 /*
- * Copyright © 2000-2004 Ingo Weinhold <[email protected]>
+ * Copyright 2000-2004 Ingo Weinhold <[email protected]>
- * Copyright © 2006-2008 Stephan Aßmus <[email protected]>
+ * Copyright 2006-2008 Stephan Aßmus <[email protected]>
  * All rights reserved. Distributed under the terms of the MIT License.
  */
 #ifndef MEDIA_TRACK_AUDIO_SUPPLIER_H
 #define MEDIA_TRACK_AUDIO_SUPPLIER_H
 
+
 #include <List.h>
 
 #include "AudioTrackSupplier.h"
 
+
 class BMediaTrack;
 struct media_codec_info;
 struct media_format;
 
+
 class MediaTrackAudioSupplier : public AudioTrackSupplier {
- public:
+public:
 								MediaTrackAudioSupplier(BMediaTrack* track,
 									int32 trackIndex);
 	virtual						~MediaTrackAudioSupplier();
@@ -27,6 +30,7 @@ class MediaTrackAudioSupplier : public AudioTrackSupplier {
 
 	// AudioReader interface
 	// (needed to reuse the class as AudioResampler input)
+	virtual bigtime_t			InitialLatency() const;
 	virtual	status_t			Read(void* buffer, int64 pos, int64 frames);
 
 	virtual	status_t			InitCheck() const;
@@ -34,7 +38,7 @@ class MediaTrackAudioSupplier : public AudioTrackSupplier {
 	virtual	int32				TrackIndex() const
 									{ return fTrackIndex; }
 
- private:
+private:
 			struct Buffer;
 			void				_InitFromTrack();
 

src/apps/mediaplayer/supplier/ProxyAudioSupplier.cpp

@@ -1,7 +1,9 @@
-/*	
+/*
- * Copyright © 2008 Stephan Aßmus <[email protected]>
+ * Copyright 2008 Stephan Aßmus <[email protected]>
  * All Rights Reserved. Distributed under the terms of the MIT license.
  */
+
+
 #include "ProxyAudioSupplier.h"
 
 #include <algorithm>
@@ -33,8 +35,9 @@ using std::swap;
 
 struct PlayingInterval {
 	PlayingInterval(bigtime_t startTime, bigtime_t endTime)
-		: start_time(startTime)
+		:
-		, end_time(endTime)
+		start_time(startTime),
+		end_time(endTime)
 	{
 	}
 
@@ -47,16 +50,17 @@ struct PlayingInterval {
 
 
 ProxyAudioSupplier::ProxyAudioSupplier(PlaybackManager* playbackManager)
-	: fSupplierLock("audio supplier lock")
+	:
+	fSupplierLock("audio supplier lock"),
 
-	, fPlaybackManager(playbackManager)
+	fPlaybackManager(playbackManager),
-	, fVideoFrameRate(25.0)
+	fVideoFrameRate(25.0),
-	, fVolume(1.0)
+	fVolume(1.0),
 
-	, fSupplier(NULL)
+	fSupplier(NULL),
-	, fAdapter(NULL)
+	fAdapter(NULL),
-	, fVolumeConverter(NULL)
+	fVolumeConverter(NULL),
-	, fAudioResampler()
+	fAudioResampler()
 {
 	TRACE("ProxyAudioSupplier()\n");
 }
@@ -70,6 +74,18 @@ ProxyAudioSupplier::~ProxyAudioSupplier()
 }
 
 
+bigtime_t
+ProxyAudioSupplier::InitialLatency() const
+{
+	BAutolock _(fSupplierLock);
+
+	if (fSupplier == NULL)
+		return 0;
+
+	return fSupplier->InitialLatency();
+}
+
+
 status_t
 ProxyAudioSupplier::GetFrames(void* buffer, int64 frameCount,
 	bigtime_t startTime, bigtime_t endTime)
@@ -100,7 +116,7 @@ ProxyAudioSupplier::GetFrames(void* buffer, int64 frameCount,
 				ERROR("GetFrames() - zero duration audio interval! start "
 					"time: %lld\n", intervalStartTime);
 				break;
-			}				
+			}
 			if (!playingIntervals.AddItem(interval)) {
 				delete interval;
 				error = B_NO_MEMORY;
@@ -182,7 +198,7 @@ ProxyAudioSupplier::SetFormat(const media_format& format)
 {
 //printf("ProxyAudioSupplier::SetFormat()\n");
 	#ifdef TRACE_PROXY_AUDIO_SUPPLIER
-		char string[256];		
+		char string[256];
 		string_for_format(format, string, 256);
 		TRACE("SetFormat(%s)\n", string);
 	#endif
@@ -258,6 +274,7 @@ ProxyAudioSupplier::Volume()
 
 // #pragma mark - audio/video/frame/time conversion
 
+
 int64
 ProxyAudioSupplier::_AudioFrameForVideoFrame(int64 frame) const
 {

src/apps/mediaplayer/supplier/ProxyAudioSupplier.h

@@ -1,10 +1,11 @@
 /*
- * Copyright © 2008 Stephan Aßmus <[email protected]>
+ * Copyright 2008 Stephan Aßmus <[email protected]>
  * All Rights Reserved. Distributed under the terms of the MIT license.
  */
 #ifndef PROXY_AUDIO_SUPPLIER_H
 #define PROXY_AUDIO_SUPPLIER_H
 
+
 #include <Locker.h>
 
 #include "AudioResampler.h"
@@ -26,6 +27,7 @@ public:
 	virtual	status_t			GetFrames(void* buffer, int64 frameCount,
 									bigtime_t startTime, bigtime_t endTime);
 
+	virtual bigtime_t			InitialLatency() const;
 	virtual	void				SetFormat(const media_format& format);
 	virtual	const media_format&	Format() const;
 
@@ -49,16 +51,16 @@ private:
 			void*				_SkipFrames(void* buffer, int64 frames) const;
 
 private:
-	BLocker						fSupplierLock;
+	mutable BLocker				fSupplierLock;
 
-	PlaybackManager*			fPlaybackManager;
+			PlaybackManager*	fPlaybackManager;
-	float						fVideoFrameRate;
+			float				fVideoFrameRate;
-	float						fVolume;
+			float				fVolume;
 
-	AudioTrackSupplier*			fSupplier;
+			AudioTrackSupplier*	fSupplier;
-	AudioReader*				fAdapter;
+			AudioReader*		fAdapter;
-	AudioVolumeConverter*		fVolumeConverter;
+			AudioVolumeConverter* fVolumeConverter;
-	AudioResampler				fAudioResampler;
+			AudioResampler		fAudioResampler;
 };