* 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
This commit is contained in:
Axel Dörfler
2009-12-03 22:16:54 +00:00
parent e508a7fbd3
commit 4ed344af25
4 changed files with 332 additions and 352 deletions
+11 -9
View File
@@ -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 {
_playing_sound* next;
struct playing_sound {
playing_sound* next;
off_t current_offset;
BSound* sound;
play_id id;
@@ -168,8 +170,8 @@ private:
float volume;
};
struct _waiting_sound {
_waiting_sound* next;
struct waiting_sound {
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;
_waiting_sound* fWaitingSounds;
playing_sound* fPlayingSounds;
waiting_sound* fWaitingSounds;
BufferPlayerFunc fPlayBufferFunc;
EventNotifierFunc fNotifierFunc;
+183 -179
View File
@@ -8,9 +8,7 @@
*/
/*! This is the BBufferProducer, used internally by BSoundPlayer
This belongs into a private namespace, but isn't for
compatibility reasons.
/*! This is the BBufferProducer used internally by BSoundPlayer.
*/
@@ -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),
mInitCheckStatus(B_OK),
mOutputEnabled(true),
mBufferGroup(NULL),
mFramesSent(0),
mTooEarlyCount(0)
fPlayer(player),
fInitStatus(B_OK),
fOutputEnabled(true),
fBufferGroup(NULL),
fFramesSent(0),
fTooEarlyCount(0)
{
CALLED();
mOutput.format.type = B_MEDIA_RAW_AUDIO;
mOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
fOutput.format.type = B_MEDIA_RAW_AUDIO;
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;
mOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
mOutput.destination = media_destination::null;
mOutput.source.port = ControlPort();
mOutput.source.id = 0;
mOutput.node = Node();
strcpy(mOutput.name, Name());
fOutput.format.type = B_MEDIA_RAW_AUDIO;
fOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
fOutput.destination = media_destination::null;
fOutput.source.port = ControlPort();
fOutput.source.id = 0;
fOutput.node = Node();
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) {
TRACE("_SoundPlayNode::FormatProposal returning B_MEDIA_BAD_SOURCE\n");
if (output != fOutput.source) {
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) {
TRACE("_SoundPlayNode::SetBufferGroup returning B_MEDIA_BAD_SOURCE\n");
if (forSource != fOutput.source) {
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;
int32 count = int32(mLatency / BufferDuration() + 1 + 1);
size_t size = fOutput.format.u.raw_audio.buffer_size;
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) {
TRACE("_SoundPlayNode::PrepareToConnect returning "
if (what != fOutput.source) {
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;
mOutput.format = *format;
*_source = mOutput.source;
fOutput.destination = where;
fOutput.format = *format;
*_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) {
TRACE("_SoundPlayNode::Connect returning\n");
if (source != fOutput.source) {
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;
mOutput.format.type = B_MEDIA_RAW_AUDIO;
mOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
fOutput.destination = media_destination::null;
fOutput.format.type = B_MEDIA_RAW_AUDIO;
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;
mOutput.format = format;
fOutput.destination = destination;
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);
TRACE("_SoundPlayNode::Connect: downstream latency = %Ld\n", mLatency);
FindLatencyFor(fOutput.destination, &fLatency, &id);
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)
/ ((mOutput.format.u.raw_audio.format
bigtime_t duration = ((fOutput.format.u.raw_audio.buffer_size * 1000000LL)
/ ((fOutput.format.u.raw_audio.format
& media_raw_audio_format::B_AUDIO_SIZE_MASK)
* mOutput.format.u.raw_audio.channel_count))
/ (int32)mOutput.format.u.raw_audio.frame_rate;
* fOutput.format.u.raw_audio.channel_count))
/ (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;
TRACE("_SoundPlayNode::Connect: using %Ld as internal latency\n",
mInternalLatency);
SetEventLatency(mLatency + mInternalLatency);
fInternalLatency = (3 * BufferDuration()) / 4;
TRACE("SoundPlayNode::Connect: using %Ld as internal latency\n",
fInternalLatency);
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) {
TRACE("_SoundPlayNode::Disconnect returning\n");
if (what != fOutput.source) {
TRACE("SoundPlayNode::Disconnect returning\n");
return;
}
// Make sure that our connection is the one being disconnected
if (where == mOutput.destination && what == mOutput.source) {
mOutput.destination = media_destination::null;
mOutput.format.type = B_MEDIA_RAW_AUDIO;
mOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
delete mBufferGroup;
mBufferGroup = NULL;
if (where == fOutput.destination && what == fOutput.source) {
fOutput.destination = media_destination::null;
fOutput.format.type = B_MEDIA_RAW_AUDIO;
fOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
delete fBufferGroup;
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) {
TRACE("_SoundPlayNode::LateNoticeReceived returning\n");
if (what != fOutput.source) {
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
mInternalLatency = 30000;
if (fInternalLatency > 30000) // avoid getting a too high latency
fInternalLatency = 30000;
SetEventLatency(mLatency + mInternalLatency);
TRACE("_SoundPlayNode::LateNoticeReceived: increasing latency to %Ld\n", mLatency + mInternalLatency);
SetEventLatency(fLatency + fInternalLatency);
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
/ ((mOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK)
* mOutput.format.u.raw_audio.channel_count);
size_t nFrames = fOutput.format.u.raw_audio.buffer_size
/ ((fOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK)
* 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) {
fprintf(stderr, "_SoundPlayNode::EnableOutput returning\n");
if (what != fOutput.source) {
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) {
mLatency = newLatency;
SetEventLatency(mLatency + mInternalLatency);
if (source == fOutput.source && destination == fOutput.destination) {
fLatency = newLatency;
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) {
mInternalLatency -= how_early;
if (mInternalLatency < 500)
mInternalLatency = 500;
printf("_SoundPlayNode::SendNewBuffer setting internal latency to %Ld\n", mInternalLatency);
SetEventLatency(mLatency + mInternalLatency);
mTooEarlyCount = 0;
if (fTooEarlyCount++ == 5) {
fInternalLatency -= how_early;
if (fInternalLatency < 500)
fInternalLatency = 500;
printf("SoundPlayNode::SendNewBuffer setting internal latency to %Ld\n", fInternalLatency);
SetEventLatency(fLatency + fInternalLatency);
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
/ ((mOutput.format.u.raw_audio.format
size_t nFrames = fOutput.format.u.raw_audio.buffer_size
/ ((fOutput.format.u.raw_audio.format
& media_raw_audio_format::B_AUDIO_SIZE_MASK)
* mOutput.format.u.raw_audio.channel_count);
mFramesSent += nFrames;
* fOutput.format.u.raw_audio.channel_count);
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)
/ (int32)mOutput.format.u.raw_audio.frame_rate);
bigtime_t nextEvent = fStartTime + bigtime_t((1000000LL * fFramesSent)
/ (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;
mStartTime = event->event_time;
fFramesSent = 0;
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;
int32 count = int32(mLatency / BufferDuration() + 1 + 1);
size_t size = fOutput.format.u.raw_audio.buffer_size;
int32 count = int32(fLatency / BufferDuration() + 1 + 1);
TRACE("_SoundPlayNode::AllocateBuffers: latency = %Ld, buffer duration "
"= %Ld, count %ld\n", mLatency, BufferDuration(), count);
TRACE("SoundPlayNode::AllocateBuffers: latency = %Ld, buffer duration "
"= %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);
if (mBufferGroup->InitCheck() != B_OK) {
ERROR("_SoundPlayNode::AllocateBuffers: BufferGroup::InitCheck() "
fBufferGroup = new BBufferGroup(size, count);
if (fBufferGroup->InitCheck() != B_OK) {
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(
mOutput.format.u.raw_audio.buffer_size, BufferDuration() / 2);
BBuffer* buffer = fBufferGroup->RequestBuffer(
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) {
ERROR("_SoundPlayNode::FillNextBuffer: RequestBuffer failed\n");
if (buffer == NULL) {
ERROR("SoundPlayNode::FillNextBuffer: RequestBuffer failed\n");
return NULL;
}
if (mPlayer->HasData()) {
mPlayer->_PlayBuffer(buf->Data(),
mOutput.format.u.raw_audio.buffer_size, mOutput.format.u.raw_audio);
} else {
memset(buf->Data(), 0, mOutput.format.u.raw_audio.buffer_size);
}
if (fPlayer->HasData()) {
fPlayer->_PlayBuffer(buffer->Data(),
fOutput.format.u.raw_audio.buffer_size, fOutput.format.u.raw_audio);
} else
memset(buffer->Data(), 0, fOutput.format.u.raw_audio.buffer_size);
// fill in the buffer header
media_header* hdr = buf->Header();
hdr->type = B_MEDIA_RAW_AUDIO;
hdr->size_used = mOutput.format.u.raw_audio.buffer_size;
hdr->time_source = TimeSource()->ID();
hdr->start_time = event_time;
return buf;
media_header* header = buffer->Header();
header->type = B_MEDIA_RAW_AUDIO;
header->size_used = fOutput.format.u.raw_audio.buffer_size;
header->time_source = TimeSource()->ID();
header->start_time = eventTime;
return buffer;
}
} // namespace BPrivate
+127 -152
View File
@@ -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
: public BBufferProducer, public BMediaEventLooper
{
namespace BPrivate {
class SoundPlayNode : public BBufferProducer, public BMediaEventLooper {
public:
_SoundPlayNode(const char *name, BSoundPlayer *player);
~_SoundPlayNode();
bool IsPlaying();
bigtime_t CurrentTime();
/*************************/
/* begin from BMediaNode */
public:
virtual BMediaAddOn* AddOn(
int32 * internal_id) const; /* Who instantiated you -- or NULL for app class */
SoundPlayNode(const char* name,
BSoundPlayer* player);
virtual ~SoundPlayNode();
bool IsPlaying();
bigtime_t CurrentTime();
// BMediaNode methods
virtual BMediaAddOn* AddOn(int32* _internalID) const;
protected:
/* These don't return errors; instead, they use the global error condition reporter. */
/* 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);
virtual void Preroll();
public:
virtual status_t HandleMessage(
int32 message,
const void * data,
size_t size);
virtual status_t HandleMessage(int32 message, const void* data,
size_t size);
protected:
virtual void NodeRegistered(void); /* reserved 2 */
virtual status_t RequestCompleted(const media_request_info &info);
virtual void SetTimeSource(BTimeSource *timeSource);
virtual void SetRunMode(run_mode mode);
virtual void NodeRegistered();
virtual status_t RequestCompleted(
const media_request_info& info);
virtual void SetTimeSource(BTimeSource* timeSource);
virtual void SetRunMode(run_mode mode);
/* end from BMediaNode */
/***********************/
// BBufferProducer methods
/******************************/
/* begin from BBufferProducer */
virtual status_t FormatSuggestionRequested(media_type type,
int32 quality, media_format* format);
virtual status_t FormatSuggestionRequested( media_type type,
int32 quality,
media_format* format);
virtual status_t FormatProposal( const media_source& output,
media_format* format);
virtual status_t FormatChangeRequested( const media_source& source,
const media_destination& destination,
media_format* io_format,
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,
BBufferGroup* group);
virtual status_t GetLatency( bigtime_t * out_latency);
virtual status_t PrepareToConnect( const media_source& what,
const media_destination& where,
media_format* format,
media_source* out_source,
char* out_name);
virtual void Connect( status_t error,
const media_source& source,
const media_destination& destination,
const media_format& format,
char* io_name);
virtual void Disconnect( const media_source& what,
const media_destination& where);
virtual void LateNoticeReceived( const media_source& what,
bigtime_t how_much,
bigtime_t performance_time);
virtual void EnableOutput( const media_source & what,
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 */
/****************************/
virtual status_t FormatProposal(const media_source& output,
media_format* format);
/********************************/
/* start from BMediaEventLooper */
virtual status_t FormatChangeRequested(
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);
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);
virtual status_t SetBufferGroup(const media_source& forSource,
BBufferGroup* group);
virtual status_t GetLatency(bigtime_t* _latency);
/* end from BMediaEventLooper */
/******************************/
virtual status_t PrepareToConnect(const media_source& what,
const media_destination& where,
media_format* format, media_source* _source,
char* _name);
virtual void Connect(status_t error,
const media_source& source,
const media_destination& destination,
const media_format& format,
char* name);
virtual void Disconnect(const media_source& what,
const media_destination& where);
virtual void LateNoticeReceived(const media_source& what,
bigtime_t howMuch,
bigtime_t performanceTime);
virtual void EnableOutput(const media_source& what,
bool enabled, int32* _deprecated_);
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);
// BMediaEventLooper methods
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);
void AllocateBuffers();
BBuffer* FillNextBuffer(bigtime_t eventTime);
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);
void AllocateBuffers();
BBuffer* FillNextBuffer(bigtime_t event_time);
private:
BSoundPlayer *mPlayer;
status_t mInitCheckStatus;
bool mOutputEnabled;
media_output mOutput;
BBufferGroup *mBufferGroup;
media_format mFormat;
bigtime_t mLatency;
bigtime_t mInternalLatency;
bigtime_t mStartTime;
uint64 mFramesSent;
int32 mTooEarlyCount;
BSoundPlayer* fPlayer;
status_t fInitStatus;
bool fOutputEnabled;
media_output fOutput;
BBufferGroup* fBufferGroup;
bigtime_t fLatency;
bigtime_t fInternalLatency;
bigtime_t fStartTime;
uint64 fFramesSent;
int32 fTooEarlyCount;
};
#endif
} // namespace BPrivate
#endif // _SOUND_PLAY_NODE_H
+11 -12
View File
@@ -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* item = fPlayingSounds;
playing_sound** link = &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",