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 } }