void FlangerNode::Connect(
status_t status,
const media_source& source,
const media_destination& destination,
const media_format& format,
char* pioName) {
PRINT(("FlangerNode::Connect()\n"));
status_t err;
// connection failed?
if(status < B_OK) {
PRINT(("\tStatus: %s\n", strerror(status)));
// 'unreserve' the output
m_output.destination = media_destination::null;
return;
}
// connection established:
strncpy(pioName, m_output.name, B_MEDIA_NAME_LENGTH);
m_output.destination = destination;
m_format = format;
// figure downstream latency
media_node_id timeSource;
err = FindLatencyFor(m_output.destination, &m_downstreamLatency, &timeSource);
if(err < B_OK) {
PRINT(("\t!!! FindLatencyFor(): %s\n", strerror(err)));
}
PRINT(("\tdownstream latency = %Ld\n", m_downstreamLatency));
// prepare the filter
initFilter();
// figure processing time
m_processingLatency = calcProcessingLatency();
PRINT(("\tprocessing latency = %Ld\n", m_processingLatency));
// store summed latency
SetEventLatency(m_downstreamLatency + m_processingLatency);
if(m_input.source != media_source::null) {
// pass new latency upstream
err = SendLatencyChange(
m_input.source,
m_input.destination,
EventLatency() + SchedulingLatency());
if(err < B_OK)
PRINT(("\t!!! SendLatencyChange(): %s\n", strerror(err)));
}
// cache buffer duration
SetBufferDuration(
buffer_duration(
m_format.u.raw_audio));
}
void AudioFilterNode::Connect(
status_t status,
const media_source& source,
const media_destination& destination,
const media_format& format,
char* ioName) {
PRINT(("AudioFilterNode::Connect()\n"));
status_t err;
#if DEBUG
char formatStr[256];
string_for_format(format, formatStr, 255);
PRINT(("\tformat: %s\n", formatStr));
#endif
// connection failed?
if(status < B_OK) {
PRINT(("\tCONNECTION FAILED: Status '%s'\n", strerror(status)));
// 'unreserve' the output
m_output.destination = media_destination::null;
return;
}
// connection established:
strncpy(ioName, m_output.name, B_MEDIA_NAME_LENGTH);
m_output.destination = destination;
// figure downstream latency
media_node_id timeSource;
err = FindLatencyFor(m_output.destination, &m_downstreamLatency, &timeSource);
if(err < B_OK) {
PRINT(("\t!!! FindLatencyFor(): %s\n", strerror(err)));
}
PRINT(("\tdownstream latency = %Ld\n", m_downstreamLatency));
// // prepare the filter
// initFilter();
//
// // figure processing time
// m_processingLatency = calcProcessingLatency();
// PRINT(("\tprocessing latency = %Ld\n", m_processingLatency));
//
// // store summed latency
// SetEventLatency(m_downstreamLatency + m_processingLatency);
//
// if(m_input.source != media_source::null) {
// // pass new latency upstream
// err = SendLatencyChange(
// m_input.source,
// m_input.destination,
// EventLatency() + SchedulingLatency());
// if(err < B_OK)
// PRINT(("\t!!! SendLatencyChange(): %s\n", strerror(err)));
// }
// cache buffer duration
SetBufferDuration(
buffer_duration(
m_output.format.u.raw_audio));
// [re-]initialize operation
updateOperation();
// initialize buffer group if necessary
updateBufferGroup();
}
void
MixerCore::_MixThread()
{
// The broken BeOS R5 multiaudio node starts with time 0,
// then publishes negative times for about 50ms, publishes 0
// again until it finally reaches time values > 0
if (!LockFromMixThread())
return;
bigtime_t start = fTimeSource->Now();
Unlock();
while (start <= 0) {
TRACE("MixerCore: delaying _MixThread start, timesource is at %Ld\n",
start);
snooze(5000);
if (!LockFromMixThread())
return;
start = fTimeSource->Now();
Unlock();
}
if (!LockFromMixThread())
return;
bigtime_t latency = max((bigtime_t)3600, bigtime_t(0.4 * buffer_duration(
fOutput->MediaOutput().format.u.raw_audio)));
// TODO: when the format changes while running, everything is wrong!
bigtime_t bufferRequestTimeout = buffer_duration(
fOutput->MediaOutput().format.u.raw_audio) / 2;
TRACE("MixerCore: starting _MixThread at %Ld with latency %Ld and "
"downstream latency %Ld, bufferRequestTimeout %Ld\n", start, latency,
fDownstreamLatency, bufferRequestTimeout);
// We must read from the input buffer at a position (pos) that is always
// a multiple of fMixBufferFrameCount.
int64 temp = frames_for_duration(fMixBufferFrameRate, start);
int64 frameBase = ((temp / fMixBufferFrameCount) + 1)
* fMixBufferFrameCount;
bigtime_t timeBase = duration_for_frames(fMixBufferFrameRate, frameBase);
Unlock();
TRACE("MixerCore: starting _MixThread, start %Ld, timeBase %Ld, "
"frameBase %Ld\n", start, timeBase, frameBase);
ASSERT(fMixBufferFrameCount > 0);
#if DEBUG
uint64 bufferIndex = 0;
#endif
typedef RtList<chan_info> chan_info_list;
chan_info_list inputChanInfos[MAX_CHANNEL_TYPES];
BStackOrHeapArray<chan_info_list, 16> mixChanInfos(fMixBufferChannelCount);
// TODO: this does not support changing output channel count
bigtime_t eventTime = timeBase;
int64 framePos = 0;
for (;;) {
if (!LockFromMixThread())
return;
bigtime_t waitUntil = fTimeSource->RealTimeFor(eventTime, 0)
- latency - fDownstreamLatency;
Unlock();
status_t rv = acquire_sem_etc(fMixThreadWaitSem, 1, B_ABSOLUTE_TIMEOUT,
waitUntil);
if (rv == B_INTERRUPTED)
continue;
if (rv != B_TIMED_OUT && rv < B_OK)
return;
if (!LockWithTimeout(10000)) {
ERROR("MixerCore: LockWithTimeout failed\n");
continue;
}
// no inputs or output muted, skip further processing and just send an
// empty buffer
if (fInputs->IsEmpty() || fOutput->IsMuted()) {
int size = fOutput->MediaOutput().format.u.raw_audio.buffer_size;
BBuffer* buffer = fBufferGroup->RequestBuffer(size,
bufferRequestTimeout);
if (buffer != NULL) {
memset(buffer->Data(), 0, size);
// fill in the buffer header
media_header* hdr = buffer->Header();
hdr->type = B_MEDIA_RAW_AUDIO;
hdr->size_used = size;
hdr->time_source = fTimeSource->ID();
hdr->start_time = eventTime;
if (fNode->SendBuffer(buffer, fOutput) != B_OK) {
#if DEBUG
ERROR("MixerCore: SendBuffer failed for buffer %Ld\n",
bufferIndex);
#else
ERROR("MixerCore: SendBuffer failed\n");
#endif
buffer->Recycle();
}
} else {
#if DEBUG
ERROR("MixerCore: RequestBuffer failed for buffer %Ld\n",
bufferIndex);
#else
ERROR("MixerCore: RequestBuffer failed\n");
#endif
}
goto schedule_next_event;
}
int64 currentFramePos;
currentFramePos = frameBase + framePos;
// mix all data from all inputs into the mix buffer
ASSERT(currentFramePos % fMixBufferFrameCount == 0);
PRINT(4, "create new buffer event at %Ld, reading input frames at "
"%Ld\n", eventTime, currentFramePos);
// Init the channel information for each MixerInput.
for (int i = 0; MixerInput* input = Input(i); i++) {
int count = input->GetMixerChannelCount();
for (int channel = 0; channel < count; channel++) {
int type;
const float* base;
uint32 sampleOffset;
float gain;
if (!input->GetMixerChannelInfo(channel, currentFramePos,
eventTime, &base, &sampleOffset, &type, &gain)) {
continue;
}
if (type < 0 || type >= MAX_CHANNEL_TYPES)
continue;
chan_info* info = inputChanInfos[type].Create();
info->base = (const char*)base;
info->sample_offset = sampleOffset;
info->gain = gain;
}
}
for (int channel = 0; channel < fMixBufferChannelCount; channel++) {
int sourceCount = fOutput->GetOutputChannelSourceCount(channel);
for (int i = 0; i < sourceCount; i++) {
int type;
float gain;
fOutput->GetOutputChannelSourceInfoAt(channel, i, &type,
&gain);
if (type < 0 || type >= MAX_CHANNEL_TYPES)
continue;
int count = inputChanInfos[type].CountItems();
for (int j = 0; j < count; j++) {
chan_info* info = inputChanInfos[type].ItemAt(j);
chan_info* newInfo = mixChanInfos[channel].Create();
newInfo->base = info->base;
newInfo->sample_offset = info->sample_offset;
newInfo->gain = info->gain * gain;
}
}
}
memset(fMixBuffer, 0,
fMixBufferChannelCount * fMixBufferFrameCount * sizeof(float));
for (int channel = 0; channel < fMixBufferChannelCount; channel++) {
PRINT(5, "_MixThread: channel %d has %d sources\n", channel,
mixChanInfos[channel].CountItems());
int count = mixChanInfos[channel].CountItems();
for (int i = 0; i < count; i++) {
chan_info* info = mixChanInfos[channel].ItemAt(i);
PRINT(5, "_MixThread: base %p, sample-offset %2d, gain %.3f\n",
info->base, info->sample_offset, info->gain);
// This looks slightly ugly, but the current GCC will generate
// the fastest code this way.
// fMixBufferFrameCount is always > 0.
uint32 dstSampleOffset
= fMixBufferChannelCount * sizeof(float);
uint32 srcSampleOffset = info->sample_offset;
register char* dst = (char*)&fMixBuffer[channel];
register char* src = (char*)info->base;
register float gain = info->gain;
register int j = fMixBufferFrameCount;
do {
*(float*)dst += *(const float*)src * gain;
dst += dstSampleOffset;
src += srcSampleOffset;
} while (--j);
}
}
// request a buffer
BBuffer* buffer;
buffer = fBufferGroup->RequestBuffer(
fOutput->MediaOutput().format.u.raw_audio.buffer_size,
bufferRequestTimeout);
if (buffer != NULL) {
// copy data from mix buffer into output buffer
for (int i = 0; i < fMixBufferChannelCount; i++) {
fResampler[i]->Resample(
reinterpret_cast<char*>(fMixBuffer) + i * sizeof(float),
fMixBufferChannelCount * sizeof(float),
fMixBufferFrameCount,
reinterpret_cast<char*>(buffer->Data())
+ (i * bytes_per_sample(
fOutput->MediaOutput().format.u.raw_audio)),
bytes_per_frame(fOutput->MediaOutput().format.u.raw_audio),
frames_per_buffer(
fOutput->MediaOutput().format.u.raw_audio),
fOutputGain * fOutput->GetOutputChannelGain(i));
}
PRINT(4, "send buffer, inframes %ld, outframes %ld\n",
fMixBufferFrameCount,
frames_per_buffer(fOutput->MediaOutput().format.u.raw_audio));
// fill in the buffer header
media_header* hdr = buffer->Header();
hdr->type = B_MEDIA_RAW_AUDIO;
hdr->size_used
= fOutput->MediaOutput().format.u.raw_audio.buffer_size;
hdr->time_source = fTimeSource->ID();
hdr->start_time = eventTime;
// swap byte order if necessary
fOutput->AdjustByteOrder(buffer);
// send the buffer
status_t res = fNode->SendBuffer(buffer, fOutput);
if (res != B_OK) {
#if DEBUG
ERROR("MixerCore: SendBuffer failed for buffer %Ld\n",
bufferIndex);
#else
ERROR("MixerCore: SendBuffer failed\n");
#endif
buffer->Recycle();
}
} else {
#if DEBUG
ERROR("MixerCore: RequestBuffer failed for buffer %Ld\n",
bufferIndex);
#else
ERROR("MixerCore: RequestBuffer failed\n");
#endif
}
// make all lists empty
for (int i = 0; i < MAX_CHANNEL_TYPES; i++)
inputChanInfos[i].MakeEmpty();
for (int i = 0; i < fOutput->GetOutputChannelCount(); i++)
mixChanInfos[i].MakeEmpty();
schedule_next_event:
// schedule next event
framePos += fMixBufferFrameCount;
eventTime = timeBase + bigtime_t((1000000LL * framePos)
/ fMixBufferFrameRate);
Unlock();
#if DEBUG
bufferIndex++;
#endif
}
}
