[haiku-commits] r36199 - in haiku/trunk/src/apps/mediaplayer: media_node_framework/audio supplier
- From: axeld@xxxxxxxxxxxxxxxx
- To: haiku-commits@xxxxxxxxxxxxx
- Date: Mon, 12 Apr 2010 22:36:20 +0200 (CEST)
Author: axeld
Date: 2010-04-12 22:36:19 +0200 (Mon, 12 Apr 2010)
New Revision: 36199
Changeset: http://dev.haiku-os.org/changeset/36199/haiku
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.cpp
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.h
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.cpp
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.h
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.cpp
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.h
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioProducer.cpp
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioProducer.h
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioReader.h
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioResampler.cpp
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioResampler.h
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.cpp
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.h
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.cpp
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.h
haiku/trunk/src/apps/mediaplayer/supplier/MediaTrackAudioSupplier.cpp
haiku/trunk/src/apps/mediaplayer/supplier/MediaTrackAudioSupplier.h
haiku/trunk/src/apps/mediaplayer/supplier/ProxyAudioSupplier.cpp
haiku/trunk/src/apps/mediaplayer/supplier/ProxyAudioSupplier.h
Log:
* 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.
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.cpp
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.cpp
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.cpp
2010-04-12 20:36:19 UTC (rev 36199)
@@ -77,6 +77,13 @@
}
+bigtime_t
+AudioAdapter::InitialLatency() const
+{
+ return fSource->InitialLatency();
+}
+
+
status_t
AudioAdapter::Read(void* buffer, int64 pos, int64 frames)
{
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.h
===================================================================
--- haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.h
2010-04-12 20:34:04 UTC (rev 36198)
+++ haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioAdapter.h
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,8 +1,11 @@
/*
- * Copyright © 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
+ * Copyright 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
* 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.
At this time the number of channels cannot be changed and the output
@@ -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;
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.cpp
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.cpp
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.cpp
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,7 +1,9 @@
/*
- * Copyright © 2008 Stephan Aßmus <superstippi@xxxxxx>
+ * Copyright 2008 Stephan Aßmus <superstippi@xxxxxx>
* 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 @@
}
+// #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)
{
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.h
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.h
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioChannelConverter.h
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,24 +1,27 @@
/*
- * Copyright © 2008 Stephan Aßmus <superstippi@xxxxxx>
+ * Copyright 2008 Stephan Aßmus <superstippi@xxxxxx>
* 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;
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.cpp
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.cpp
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.cpp
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,9 +1,10 @@
/*
- * Copyright © 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
- * Copyright © 2008 Stephan Aßmus <superstippi@xxxxxx>
+ * Copyright 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
+ * Copyright 2008 Stephan Aßmus <superstippi@xxxxxx>
* 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 @@
}
-static
-void
+static void
swap_sample_byte_order(void* buffer, uint32 format, size_t length)
{
type_code type = B_ANY_TYPE;
@@ -175,6 +143,48 @@
}
+// #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)
{
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.h
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.h
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioFormatConverter.h
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,25 +1,28 @@
/*
- * Copyright © 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
- * Copyright © 2008 Stephan Aßmus <superstippi@xxxxxx>
+ * Copyright 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
+ * Copyright 2008 Stephan Aßmus <superstippi@xxxxxx>
* 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;
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioProducer.cpp
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioProducer.cpp
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioProducer.cpp
2010-04-12 20:36:19 UTC (rev 36199)
@@ -37,13 +37,13 @@
// debugging
//#define TRACE_AUDIO_PRODUCER
#ifdef TRACE_AUDIO_PRODUCER
-# define TRACE(x...) printf(x)
-# define TRACE_BUFFER(x...)
-# define ERROR(x...) fprintf(stderr, x)
+# define TRACE(x...) printf(x)
+# define TRACE_BUFFER(x...)
+# define ERROR(x...) fprintf(stderr, x)
#else
-# define TRACE(x...)
-# define TRACE_BUFFER(x...)
-# define ERROR(x...) fprintf(stderr, x)
+# define TRACE(x...)
+# define TRACE_BUFFER(x...)
+# define ERROR(x...) fprintf(stderr, x)
#endif
@@ -85,24 +85,63 @@
#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;
+}
-// constructor
+
+// #pragma mark -
+
+
AudioProducer::AudioProducer(const char* name, AudioSupplier* supplier,
bool lowLatency)
- : BMediaNode(name),
- BBufferProducer(B_MEDIA_RAW_AUDIO),
- BMediaEventLooper(),
+ :
+ BMediaNode(name),
+ BBufferProducer(B_MEDIA_RAW_AUDIO),
+ BMediaEventLooper(),
- fBufferGroup(NULL),
- fLatency(0),
- fInternalLatency(0),
- fLowLatency(lowLatency),
- fOutputEnabled(true),
- fFramesSent(0),
- fStartTime(0),
- fSupplier(supplier),
+ fBufferGroup(NULL),
+ fLatency(0),
+ fInternalLatency(0),
+ fLastLateNotice(0),
+ fNextScheduledBuffer(0),
+ fLowLatency(lowLatency),
+ fOutputEnabled(true),
+ 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 @@
// 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 @@
}
-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 @@
// 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 @@
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 @@
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 @@
/ (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));
- media_timed_event nextBufferEvent(nextEvent,
+ /
double(fOutput.format.u.raw_audio.frame_rate));
+ media_timed_event
nextBufferEvent(fNextScheduledBuffer,
BTimedEventQueue::B_HANDLE_BUFFER);
EventQueue()->AddEvent(nextBufferEvent);
} else {
@@ -663,7 +677,8 @@
{
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 @@
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 @@
}
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 @@
if (!fLowLatency)
format->u.raw_audio.buffer_size *= 3;
-
}
return B_OK;
@@ -794,9 +809,9 @@
// number of sample in the buffer
// fill in the buffer header
- media_header* hdr = buffer->Header();
- hdr->type = B_MEDIA_RAW_AUDIO;
- hdr->time_source = TimeSource()->ID();
+ media_header* header = buffer->Header();
+ header->type = B_MEDIA_RAW_AUDIO;
+ 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 @@
// 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
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioProducer.h
===================================================================
--- haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioProducer.h
2010-04-12 20:34:04 UTC (rev 36198)
+++ haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioProducer.h
2010-04-12 20:36:19 UTC (rev 36199)
@@ -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 @@
MSG_PEAK_NOTIFICATION = 'pknt'
};
+
class AudioProducer : public BBufferProducer, public BMediaEventLooper {
public:
AudioProducer(const char* name,
@@ -116,6 +118,8 @@
bool fUsingOurBuffers;
bigtime_t fLatency;
bigtime_t fInternalLatency;
+ bigtime_t fLastLateNotice;
+ bigtime_t fNextScheduledBuffer;
bool fLowLatency;
media_output fOutput;
bool fOutputEnabled;
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioReader.h
===================================================================
--- haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioReader.h
2010-04-12 20:34:04 UTC (rev 36198)
+++ haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioReader.h
2010-04-12 20:36:19 UTC (rev 36199)
@@ -18,6 +18,7 @@
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);
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioResampler.cpp
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioResampler.cpp
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioResampler.cpp
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,7 +1,9 @@
/*
- * Copyright © 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
+ * Copyright 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
* 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 @@
}
+// #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 @@
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 @@
TRACE("AudioResampler::SetSource() - not host byte order\n");
return;
}
-
+
float frameRate = FrameRate();
// don't overwrite previous audio frame rate
fSource = source;
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioResampler.h
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioResampler.h
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioResampler.h
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,19 +1,21 @@
/*
- * Copyright © 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
+ * Copyright 2000-2006 Ingo Weinhold <ingo_weinhold@xxxxxx>
* 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
source data from 41.1 KHz to 96 KHz played backward twice as fast
(time scale = -2).
*/
-#ifndef AUDIO_RESAMPLER_H
-#define AUDIO_RESAMPLER_H
#include "AudioReader.h"
+
class AudioResampler : public AudioReader {
public:
AudioResampler();
@@ -21,6 +23,7 @@
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;
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.cpp
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.cpp
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.cpp
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,6 +1,8 @@
-/* Copyright (c) 2000-2008, Ingo Weinhold <ingo_weinhold@xxxxxx>,
+/* Copyright 2000-2008, Ingo Weinhold <ingo_weinhold@xxxxxx>,
* All Rights Reserved. Distributed under the terms of the MIT license.
*/
+
+
#include "AudioSupplier.h"
#include "AudioProducer.h"
@@ -26,6 +28,6 @@
status_t
AudioSupplier::InitCheck() const
{
- return (fAudioProducer ? B_OK : B_NO_INIT);
+ return fAudioProducer ? B_OK : B_NO_INIT;
}
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.h
===================================================================
--- haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.h
2010-04-12 20:34:04 UTC (rev 36198)
+++ haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioSupplier.h
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,11 +1,12 @@
-/* Copyright (c) 2000-2008, Ingo Weinhold <ingo_weinhold@xxxxxx>,
+/* Copyright 2000-2008, Ingo Weinhold <ingo_weinhold@xxxxxx>,
* 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;
};
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.cpp
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.cpp
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.cpp
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,8 +1,9 @@
/*
- * Copyright © 2008 Stephan Aßmus <superstippi@xxxxxx>
+ * Copyright 2008 Stephan Aßmus <superstippi@xxxxxx>
* 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 @@
}
+// #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)
{
Modified:
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.h
===================================================================
---
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.h
2010-04-12 20:34:04 UTC (rev 36198)
+++
haiku/trunk/src/apps/mediaplayer/media_node_framework/audio/AudioVolumeConverter.h
2010-04-12 20:36:19 UTC (rev 36199)
@@ -1,23 +1,26 @@
/*
- * Copyright © 2008 Stephan Aßmus <superstippi@xxxxxx>
+ * Copyright 2008 Stephan Aßmus <superstippi@xxxxxx>
[... truncated: 272 lines follow ...]
Other related posts:
- » [haiku-commits] r36199 - in haiku/trunk/src/apps/mediaplayer: media_node_framework/audio supplier - axeld
