/*
* Copyright 2002-2014 Haiku, Inc. All rights reserved.
* Distributed under the terms of the MIT License.
*
* Authors:
* Christopher ML Zumwalt May (zummy@users.sf.net)
*/
/* A MediaKit producer node which mixes sound from the GameKit
and sends them to the audio mixer
*/
#include "GameProducer.h"
#include <string.h>
#include <stdio.h>
#include <Buffer.h>
#include <BufferGroup.h>
#include <ByteOrder.h>
#include <List.h>
#include <MediaDefs.h>
#include <TimeSource.h>
#include "GameSoundBuffer.h"
#include "GameSoundDevice.h"
#include "GSUtility.h"
struct _gs_play {
gs_id sound;
bool* hook;
_gs_play* next;
_gs_play* previous;
};
GameProducer::GameProducer(GameSoundBuffer* object,
const gs_audio_format* format)
:
BMediaNode("GameProducer.h"),
BBufferProducer(B_MEDIA_RAW_AUDIO),
BMediaEventLooper(),
fBufferGroup(NULL),
fLatency(0),
fInternalLatency(0),
fOutputEnabled(true)
{
// initialize our preferred format object
fPreferredFormat.type = B_MEDIA_RAW_AUDIO;
fPreferredFormat.u.raw_audio.format = format->format;
fPreferredFormat.u.raw_audio.channel_count = format->channel_count;
fPreferredFormat.u.raw_audio.frame_rate = format->frame_rate; // Hertz
fPreferredFormat.u.raw_audio.byte_order = format->byte_order;
// fPreferredFormat.u.raw_audio.channel_mask
// = B_CHANNEL_LEFT | B_CHANNEL_RIGHT;
// fPreferredFormat.u.raw_audio.valid_bits = 32;
// fPreferredFormat.u.raw_audio.matrix_mask = B_MATRIX_AMBISONIC_WXYZ;
// we'll use the consumer's preferred buffer size, if any
fPreferredFormat.u.raw_audio.buffer_size
= media_raw_audio_format::wildcard.buffer_size;
// we're not connected yet
fOutput.destination = media_destination::null;
fOutput.format = fPreferredFormat;
fFrameSize = get_sample_size(format->format) * format->channel_count;
fObject = object;
}
GameProducer::~GameProducer()
{
// Stop the BMediaEventLooper thread
Quit();
}
// BMediaNode methods
BMediaAddOn*
GameProducer::AddOn(int32* internal_id) const
{
return NULL;
}
// BBufferProducer methods
status_t
GameProducer::GetNextOutput(int32* cookie, media_output* _output)
{
// we currently support only one output
if (0 != *cookie)
return B_BAD_INDEX;
*_output = fOutput;
*cookie += 1;
return B_OK;
}
status_t
GameProducer::DisposeOutputCookie(int32 cookie)
{
// do nothing because our cookie is only an integer
return B_OK;
}
void
GameProducer::EnableOutput(const media_source& what, bool enabled,
int32* _deprecated_)
{
// If I had more than one output, I'd have to walk my list of output records
// to see which one matched the given source, and then enable/disable that
// one. But this node only has one output, so I just make sure the given
// source matches, then set the enable state accordingly.
if (what == fOutput.source)
{
fOutputEnabled = enabled;
}
}
status_t
GameProducer::FormatSuggestionRequested(media_type type, int32 /*quality*/,
media_format* format)
{
// insure that we received a format
if (!format)
return B_BAD_VALUE;
// returning our preferred format
*format = fPreferredFormat;
// our format is supported
if (type == B_MEDIA_UNKNOWN_TYPE)
return B_OK;
// we only support raw audo
return (type != B_MEDIA_RAW_AUDIO) ? B_MEDIA_BAD_FORMAT : B_OK;
}
status_t
GameProducer::FormatProposal(const media_source& output, media_format* format)
{
// doest the proposed output match our output?
if (output != fOutput.source)
return B_MEDIA_BAD_SOURCE;
// return our preferred format
*format = fPreferredFormat;
// we will reject the proposal if the format is not audio
media_type requestedType = format->type;
if ((requestedType != B_MEDIA_UNKNOWN_TYPE)
&& (requestedType != B_MEDIA_RAW_AUDIO)) {
return B_MEDIA_BAD_FORMAT;
}
return B_OK; // raw audio or wildcard type, either is okay by us
}
status_t
GameProducer::PrepareToConnect(const media_source& what,
const media_destination& where, media_format* format,
media_source* _source, char* out_name)
{
// The format has been processed by the consumer at this point. We need
// to insure the format is still acceptable and any wild care are filled in.
// trying to connect something that isn't our source?
if (what != fOutput.source)
return B_MEDIA_BAD_SOURCE;
// are we already connected?
if (fOutput.destination != media_destination::null)
return B_MEDIA_ALREADY_CONNECTED;
// the format may not yet be fully specialized (the consumer might have
// passed back some wildcards). Finish specializing it now, and return an
// error if we don't support the requested format.
if (format->type != B_MEDIA_RAW_AUDIO)
return B_MEDIA_BAD_FORMAT;
if (format->u.raw_audio.format != fPreferredFormat.u.raw_audio.format)
return B_MEDIA_BAD_FORMAT;
// check the buffer size, which may still be wildcarded
if (format->u.raw_audio.buffer_size
== media_raw_audio_format::wildcard.buffer_size) {
format->u.raw_audio.buffer_size = 4096;
// pick something comfortable to suggest
}
// Now reserve the connection, and return information about it
fOutput.destination = where;
fOutput.format = *format;
*_source = fOutput.source;
strlcpy(out_name, fOutput.name, B_MEDIA_NAME_LENGTH);
return B_OK;
}
void
GameProducer::Connect(status_t error, const media_source& source,
const media_destination& destination, const media_format& format,
char* ioName)
{
// If something earlier failed, Connect() might still be called, but with a
// non-zero error code. When that happens we simply unreserve the
// connection and do nothing else.
if (error) {
fOutput.destination = media_destination::null;
fOutput.format = fPreferredFormat;
return;
}
// Okay, the connection has been confirmed. Record the destination and
// format that we agreed on, and report our connection name again.
fOutput.destination = destination;
fOutput.format = format;
strlcpy(ioName, fOutput.name, B_MEDIA_NAME_LENGTH);
// 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(fOutput.destination, &fLatency, &id);
if (!fBufferGroup)
fBufferSize = fOutput.format.u.raw_audio.buffer_size;
// Have to set it before latency calculating
// Use a dry run to see how long it takes me to fill a buffer of data
// The first step to setup the buffer
bigtime_t start, produceLatency;
int32 frames = int32(fBufferSize / fFrameSize);
float* data = new float[frames * 2];
// Second, fill the buffer
start = ::system_time();
for (int32 i = 0; i < frames; i++) {
data[i * 2] = 0.8 * float(i / frames);
data[i * 2 + 1] = 0.8 * float(i / frames);
}
produceLatency = ::system_time();
// Third, calculate the latency
fInternalLatency = produceLatency - start;
SetEventLatency(fLatency + fInternalLatency);
// Finaily, clean up
delete [] data;
// reset our buffer duration, etc. to avoid later calculations
bigtime_t duration = bigtime_t(1000000) * frames
/ bigtime_t(fOutput.format.u.raw_audio.frame_rate);
SetBufferDuration(duration);
// Set up the buffer group for our connection, as long as nobody handed us a
// buffer group (via SetBufferGroup()) prior to this.
if (!fBufferGroup) {
int32 count = int32(fLatency / BufferDuration() + 2);
fBufferGroup = new BBufferGroup(fBufferSize, count);
}
}
void
GameProducer::Disconnect(const media_source& what,
const media_destination& where)
{
// Make sure that our connection is the one being disconnected
if ((where == fOutput.destination) && (what == fOutput.source)) {
fOutput.destination = media_destination::null;
fOutput.format = fPreferredFormat;
delete fBufferGroup;
fBufferGroup = NULL;
}
}
status_t
GameProducer::FormatChangeRequested(const media_source& source,
const media_destination& destination, media_format* io_format,
int32* _deprecated_)
{
// we don't support any other formats, so we just reject any format changes.
return B_ERROR;
}
status_t
GameProducer::SetBufferGroup(const media_source& forSource,
BBufferGroup* newGroup)
{
// verify that we didn't get bogus arguments before we proceed
if (forSource != fOutput.source)
return B_MEDIA_BAD_SOURCE;
// Are we being passed the buffer group we're already using?
if (newGroup == fBufferGroup)
return B_OK;
// Ahh, someone wants us to use a different buffer group. At this point we
// delete the one we are using and use the specified one instead. If the
// specified group is NULL, we need to recreate one ourselves, and 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 fBufferGroup; // waits for all buffers to recycle
if (newGroup != NULL) {
// we were given a valid group; just use that one from now on
fBufferGroup = newGroup;
// get buffer length from the first buffer
BBuffer* buffers[1];
if (newGroup->GetBufferList(1, buffers) != B_OK)
return B_BAD_VALUE;
fBufferSize = buffers[0]->SizeAvailable();
} else {
// we were passed a NULL group pointer; that means we construct
// our own buffer group to use from now on
fBufferSize = fOutput.format.u.raw_audio.buffer_size;
int32 count = int32(fLatency / BufferDuration() + 2);
fBufferGroup = new BBufferGroup(fBufferSize, count);
}
return B_OK;
}
status_t
GameProducer::GetLatency(bigtime_t* _latency)
{
// report our *total* latency: internal plus downstream plus scheduling
*_latency = EventLatency() + SchedulingLatency();
return B_OK;
}
void
GameProducer::LateNoticeReceived(const media_source& what, bigtime_t howMuch,
bigtime_t performanceDuration)
{
// If we're late, we need to catch up. Respond in a manner appropriate to
// our current run mode.
if (what == fOutput.source) {
if (RunMode() == B_RECORDING) {
// A hardware capture node can't adjust; it simply emits buffers at
// appropriate points. We (partially) simulate this by not
// adjusting our behavior upon receiving late notices -- after all,
// the hardware can't choose to capture "sooner"...
} else if (RunMode() == B_INCREASE_LATENCY) {
// We're late, and our run mode dictates that we try to produce
// buffers earlier in order to catch up. This argues that the
// downstream nodes are not properly reporting their latency, but
// there's not much we can do about that at the moment, so we try
// to start producing buffers earlier to compensate.
fInternalLatency += howMuch;
SetEventLatency(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 nSamples = fBufferSize / fFrameSize;
fFramesSent += nSamples;
}
}
}
void
GameProducer::LatencyChanged(const media_source& source,
const media_destination& destination, bigtime_t new_latency, uint32 flags)
{
// 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 == fOutput.source) && (destination == fOutput.destination)) {
fLatency = new_latency;
SetEventLatency(fLatency + fInternalLatency);
}
}
status_t
GameProducer::SetPlayRate(int32 numerator, int32 denominator)
{
// Play rates are weird. We don't support them
return B_ERROR;
}
status_t
GameProducer::HandleMessage(int32 message, const void* data, size_t size)
{
// We currently do not handle private messages
return B_ERROR;
}
void
GameProducer::AdditionalBufferRequested(const media_source& source,
media_buffer_id prev_buffer, bigtime_t prev_time,
const media_seek_tag* prev_tag)
{
// we don't support offline mode (yet...)
return;
}
// BMediaEventLooper methods
void
GameProducer::NodeRegistered()
{
// set up as much information about our output as we can
fOutput.source.port = ControlPort();
fOutput.source.id = 0;
fOutput.node = Node();
strlcpy(fOutput.name, "GameProducer Output", B_MEDIA_NAME_LENGTH);
// Start the BMediaEventLooper thread
SetPriority(B_REAL_TIME_PRIORITY);
Run();
}
void
GameProducer::SetRunMode(run_mode mode)
{
// We don't support offline run mode, so broadcast an error if we're set to
// B_OFFLINE. Unfortunately, we can't actually reject the mode change...
if (B_OFFLINE == mode) {
ReportError(B_NODE_FAILED_SET_RUN_MODE);
}
}
void
GameProducer::HandleEvent(const media_timed_event* event, bigtime_t lateness,
bool realTimeEvent)
{
// FPRINTF(stderr, "ToneProducer::HandleEvent\n");
switch (event->type)
{
case BTimedEventQueue::B_START:
// don't do anything if we're already running
if (RunState() != B_STARTED) {
// Going to start sending buffers so setup the needed bookkeeping
fFramesSent = 0;
fStartTime = event->event_time;
media_timed_event firstBufferEvent(fStartTime,
BTimedEventQueue::B_HANDLE_BUFFER);
// Alternatively, we could call HandleEvent() directly with this
// event, to avoid a trip through the event queue like this:
// this->HandleEvent(&firstBufferEvent, 0, false);
EventQueue()->AddEvent(firstBufferEvent);
}
break;
case BTimedEventQueue::B_STOP:
// When we handle a stop, we must ensure that downstream consumers don't
// get any more buffers from us. This means we have to flush any
// pending buffer-producing events from the queue.
EventQueue()->FlushEvents(0, BTimedEventQueue::B_ALWAYS, true,
BTimedEventQueue::B_HANDLE_BUFFER);
break;
case BTimedEventQueue::B_HANDLE_BUFFER:
{
// Ensure we're both started and connected before delivering buffer
if ((RunState() == BMediaEventLooper::B_STARTED)
&& (fOutput.destination != media_destination::null)) {
// Get the next buffer of data
BBuffer* buffer = FillNextBuffer(event->event_time);
if (buffer) {
// Send the buffer downstream if output is enabled
status_t err = B_ERROR;
if (fOutputEnabled) {
err = SendBuffer(buffer, fOutput.source,
fOutput.destination);
}
if (err) {
// we need to recycle the buffer ourselves if output is
// disabled or if the call to SendBuffer() fails
buffer->Recycle();
}
}
// track how much media we've delivered so far
size_t nFrames = fBufferSize / fFrameSize;
fFramesSent += nFrames;
// The buffer is on its way; now schedule the next one to go
bigtime_t nextEvent = fStartTime + bigtime_t(double(fFramesSent)
/ double(fOutput.format.u.raw_audio.frame_rate)
* 1000000.0);
media_timed_event nextBufferEvent(nextEvent,
BTimedEventQueue::B_HANDLE_BUFFER);
EventQueue()->AddEvent(nextBufferEvent);
}
}
break;
default:
break;
}
}
BBuffer*
GameProducer::FillNextBuffer(bigtime_t event_time)
{
// get a buffer from our buffer group
BBuffer* buf = fBufferGroup->RequestBuffer(fBufferSize, BufferDuration());
// 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)
return NULL;
// we need to discribe the buffer
int64 frames = int64(fBufferSize / fFrameSize);
memset(buf->Data(), 0, fBufferSize);
// now fill the buffer with data, continuing where the last buffer left off
fObject->Play(buf->Data(), frames);
// fill in the buffer header
media_header* hdr = buf->Header();
hdr->type = B_MEDIA_RAW_AUDIO;
hdr->size_used = fBufferSize;
hdr->time_source = TimeSource()->ID();
bigtime_t stamp;
if (RunMode() == B_RECORDING) {
// In B_RECORDING mode, we stamp with the capture time. We're not
// really a hardware capture node, but we simulate it by using the
// (precalculated) time at which this buffer "should" have been created.
stamp = event_time;
} else {
// okay, we're in one of the "live" performance run modes. in these
// modes, we stamp the buffer with the time at which the buffer should
// be rendered to the output, not with the capture time. fStartTime is
// the cached value of the first buffer's performance time; we calculate
// this buffer's performance time as an offset from that time, based on
// the amount of media we've created so far.
// Recalculating every buffer like this avoids accumulation of error.
stamp = fStartTime + bigtime_t(double(fFramesSent)
/ double(fOutput.format.u.raw_audio.frame_rate) * 1000000.0);
}
hdr->start_time = stamp;
return buf;
}
↑ V730 Not all members of a class are initialized inside the constructor. Consider inspecting: fStartTime, fFramesSent, fBufferSize.