int64_t DSoundBuf::GetPosition() const
{
DWORD iCursor, iJunk;
HRESULT hr = m_pBuffer->GetCurrentPosition( &iCursor, &iJunk );
ASSERT_M( SUCCEEDED(hr), hr_ssprintf(hr, "GetCurrentPosition") );
/* This happens occasionally on "Realtek AC97 Audio". */
if( (int) iCursor == m_iBufferSize )
iCursor = 0;
ASSERT_M( (int) iCursor < m_iBufferSize, ssprintf("%i, %i", iCursor, m_iBufferSize) );
int iCursorFrames = int(iCursor) / bytes_per_frame();
int iWriteCursorFrames = m_iWriteCursor / bytes_per_frame();
int iFramesBehind = iWriteCursorFrames - iCursorFrames;
/* iFramesBehind will be 0 if we're called before the buffer starts playing:
* both iWriteCursorFrames and iCursorFrames will be 0. */
if( iFramesBehind < 0 )
iFramesBehind += buffersize_frames(); /* unwrap */
int64_t iRet = m_iWriteCursorPos - iFramesBehind;
/* Failsafe: never return a value smaller than we've already returned.
* This can happen once in a while in underrun conditions. */
iRet = max( m_iLastPosition, iRet );
m_iLastPosition = iRet;
return iRet;
}
/* Check to make sure that, given the current writeahead and chunksize, we're
* capable of filling the prefetch region entirely. If we aren't, increase
* the writeahead. If this happens, we're underruning. */
void DSoundBuf::CheckWriteahead( int iCursorStart, int iCursorEnd )
{
/* If we're in a recovering-from-underrun state, stop. */
if( m_iExtraWriteahead )
return;
/* If the driver is requesting an unreasonably large prefetch, ignore it entirely.
* Some drivers seem to give broken write cursors sporadically, requesting that
* almost the entire buffer be filled. There's no reason a driver should ever need
* more than 8k frames of writeahead. */
int iPrefetch = iCursorEnd - iCursorStart;
wrap( iPrefetch, m_iBufferSize );
if( iPrefetch >= 1024*32 )
{
static bool bLogged = false;
if( bLogged )
return;
bLogged = true;
LOG->Warn("Sound driver is requesting an overly large prefetch: wants %i (cursor at %i..%i), writeahead not adjusted",
iPrefetch / bytes_per_frame(), iCursorStart, iCursorEnd );
return;
}
if( m_iWriteAhead >= iPrefetch )
return;
/* We need to increase the writeahead. */
LOG->Trace("insufficient writeahead: wants %i (cursor at %i..%i), writeahead adjusted from %i to %i",
iPrefetch / bytes_per_frame(), iCursorStart, iCursorEnd, m_iWriteAhead, iPrefetch );
m_iWriteAhead = iPrefetch;
}
static void
cubeb_submit_buffer(cubeb_stream * stm, WAVEHDR * hdr)
{
long got;
MMRESULT r;
got = stm->data_callback(stm, stm->user_ptr, hdr->lpData,
hdr->dwBufferLength / bytes_per_frame(stm->params));
if (got < 0) {
/* XXX handle this case */
assert(0);
return;
} else if ((DWORD) got < hdr->dwBufferLength / bytes_per_frame(stm->params)) {
r = waveOutUnprepareHeader(stm->waveout, hdr, sizeof(*hdr));
assert(r == MMSYSERR_NOERROR);
hdr->dwBufferLength = got * bytes_per_frame(stm->params);
r = waveOutPrepareHeader(stm->waveout, hdr, sizeof(*hdr));
assert(r == MMSYSERR_NOERROR);
stm->draining = 1;
}
assert(hdr->dwFlags & WHDR_PREPARED);
r = waveOutWrite(stm->waveout, hdr, sizeof(*hdr));
assert(r == MMSYSERR_NOERROR);
}
static int
directsound_stream_get_position(cubeb_stream * stm, uint64_t * position)
{
EnterCriticalSection(&stm->lock);
DWORD play, write;
HRESULT rv = stm->buffer->GetCurrentPosition(&play, &write);
assert(rv == DS_OK);
// XXX upper limit on position is stm->written,
// XXX then adjust by overflow timer
// XXX then adjust by play positiong97
unsigned long writepos = stm->written % stm->buffer_size;
long space = play - writepos;
if (space <= 0) {
space += stm->buffer_size;
}
if (!stm->active) {
space = 0;
}
long delay = stm->buffer_size - space;
double pos = (double) ((stm->written - delay) / bytes_per_frame(stm->params)) / (double) stm->params.rate * 1000.0;
#if 1
fprintf(stderr, "w=%lu space=%ld delay=%ld pos=%.2f (p=%u w=%u)\n", stm->written, space, delay, pos, play, write);
#endif
*position = (stm->written - delay) / bytes_per_frame(stm->params);
LeaveCriticalSection(&stm->lock);
return CUBEB_OK;
}
static void
cubeb_refill_stream(cubeb_stream * stm)
{
WAVEHDR * hdr;
long got;
long wanted;
MMRESULT r;
EnterCriticalSection(&stm->lock);
stm->free_buffers += 1;
assert(stm->free_buffers > 0 && stm->free_buffers <= NBUFS);
if (stm->draining) {
LeaveCriticalSection(&stm->lock);
if (stm->free_buffers == NBUFS) {
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED);
}
SetEvent(stm->event);
return;
}
if (stm->shutdown) {
LeaveCriticalSection(&stm->lock);
SetEvent(stm->event);
return;
}
hdr = cubeb_get_next_buffer(stm);
wanted = (DWORD) stm->buffer_size / bytes_per_frame(stm->params);
/* It is assumed that the caller is holding this lock. It must be dropped
during the callback to avoid deadlocks. */
LeaveCriticalSection(&stm->lock);
got = stm->data_callback(stm, stm->user_ptr, hdr->lpData, wanted);
EnterCriticalSection(&stm->lock);
if (got < 0) {
LeaveCriticalSection(&stm->lock);
/* XXX handle this case */
assert(0);
return;
} else if (got < wanted) {
stm->draining = 1;
}
assert(hdr->dwFlags & WHDR_PREPARED);
hdr->dwBufferLength = got * bytes_per_frame(stm->params);
assert(hdr->dwBufferLength <= stm->buffer_size);
r = waveOutWrite(stm->waveout, hdr, sizeof(*hdr));
if (r != MMSYSERR_NOERROR) {
LeaveCriticalSection(&stm->lock);
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR);
return;
}
LeaveCriticalSection(&stm->lock);
}
int
frames_per_buffer(
const media_raw_audio_format & format)
{
// This will give us the number of full-sized frames that will fit
// in a buffer. (Remember, integer division automatically rounds
// down.)
int frames = 0;
if (bytes_per_frame(format) > 0) {
frames = format.buffer_size / bytes_per_frame(format);
}
return frames;
}
void AudioFilterNode::processBuffer(
BBuffer* inputBuffer,
BBuffer* outputBuffer) {
ASSERT(inputBuffer);
ASSERT(outputBuffer);
ASSERT(m_op);
// create wrapper objects
AudioBuffer input(m_input.format.u.raw_audio, inputBuffer);
AudioBuffer output(m_output.format.u.raw_audio, outputBuffer);
double sourceOffset = 0.0;
uint32 destinationOffset = 0L;
// when is the first frame due to be consumed?
bigtime_t startTime = outputBuffer->Header()->start_time;
// when is the next frame to be produced going to be consumed?
bigtime_t targetTime = startTime;
// when will the first frame of the next buffer be consumed?
bigtime_t endTime = startTime + BufferDuration();
uint32 framesRemaining = input.frames();
while(framesRemaining) {
// handle all events occurring before targetTime
// +++++
bigtime_t nextEventTime = endTime;
// look for next event occurring before endTime
// +++++
// process up to found event, if any, or to end of buffer
int64 toProcess = frames_for_duration(output.format(), nextEventTime - targetTime);
ASSERT(toProcess > 0);
uint32 processed = m_op->process(
input, output, sourceOffset, destinationOffset, (uint32)toProcess, targetTime);
if(processed < toProcess) {
// +++++ in offline mode this will have to request additional buffer(s), right?
PRINT((
"*** AudioFilterNode::processBuffer(): insufficient frames filled\n"));
}
if(toProcess > framesRemaining)
framesRemaining = 0;
else
framesRemaining -= toProcess;
// advance target time
targetTime = nextEventTime; // +++++ might this drift from the real frame offset?
}
outputBuffer->Header()->size_used = input.frames() * bytes_per_frame(m_output.format.u.raw_audio);
// PRINT(("### output size: %ld\n", outputBuffer->Header()->size_used));
}
bigtime_t
buffer_duration(
const media_raw_audio_format & format)
{
// Figuring out duration is easy. We take extra precaution to
// not divide by zero or return irrelevant results.
bigtime_t duration = 0;
if (format.buffer_size > 0 && format.frame_rate > 0 && bytes_per_frame(format) > 0) {
// In these kinds of calculations, it's always useful to double-check
// the unit conversions. (Anyone remember high school physics?)
// bytes/(bytes/frame) / frames/sec
// = frames * sec/frames
// = secs which is what we want.
duration = s_to_us((format.buffer_size / bytes_per_frame(format)) / format.frame_rate);
}
return duration;
}
status_t _AudioAdapterNode::validateProposedOutputFormat(
const media_format& preferredFormat,
media_format& ioProposedFormat) {
status_t err = _inherited::validateProposedOutputFormat(
preferredFormat, ioProposedFormat);
media_raw_audio_format& w = media_raw_audio_format::wildcard;
if(input().source != media_source::null) {
// an input connection exists; constrain the output format
// is there enough information to suggest a buffer size?
if(
ioProposedFormat.u.raw_audio.format != w.format &&
ioProposedFormat.u.raw_audio.channel_count != w.channel_count) {
size_t target_buffer_size =
bytes_per_frame(ioProposedFormat.u.raw_audio) *
frames_per_buffer(input().format.u.raw_audio);
if(ioProposedFormat.u.raw_audio.buffer_size != target_buffer_size) {
if(ioProposedFormat.u.raw_audio.buffer_size != w.buffer_size)
err = B_MEDIA_BAD_FORMAT;
ioProposedFormat.u.raw_audio.buffer_size = target_buffer_size;
}
}
// require same frame rate as input
if(ioProposedFormat.u.raw_audio.frame_rate != input().format.u.raw_audio.frame_rate) {
if(ioProposedFormat.u.raw_audio.frame_rate != w.frame_rate)
err = B_MEDIA_BAD_FORMAT;
ioProposedFormat.u.raw_audio.frame_rate = input().format.u.raw_audio.frame_rate;
}
}
char fmt_string[256];
string_for_format(ioProposedFormat, fmt_string, 255);
PRINT((
"### _AudioAdapterNode::validateProposedOutputFormat():\n"
" %s\n", fmt_string));
return err;
}
status_t _AudioAdapterNode::getPreferredOutputFormat(
media_format& ioFormat) {
status_t err = _inherited::getPreferredOutputFormat(ioFormat);
if(err < B_OK)
return err;
_AudioAdapterParams* p = dynamic_cast<_AudioAdapterParams*>(parameterSet());
ASSERT(p);
media_raw_audio_format& w = media_raw_audio_format::wildcard;
// copy user preferences
if(p->outputFormat.format != w.format)
ioFormat.u.raw_audio.format = p->outputFormat.format;
if(p->outputFormat.channel_count != w.channel_count)
ioFormat.u.raw_audio.channel_count = p->outputFormat.channel_count;
//// // if one end is connected, prefer not to do channel conversions [15sep99]
//// if(input().source != media_source::null)
//// ioFormat.u.raw_audio.channel_count = input().format.u.raw_audio.channel_count;
// if input connected, constrain:
// buffer_size
// frame_rate
if(input().source != media_source::null) {
// if the user doesn't care, default to the input's frame format
if(ioFormat.u.raw_audio.format == w.format)
ioFormat.u.raw_audio.format = input().format.u.raw_audio.format;
if(ioFormat.u.raw_audio.channel_count == w.channel_count)
ioFormat.u.raw_audio.channel_count = input().format.u.raw_audio.channel_count;
ioFormat.u.raw_audio.buffer_size =
bytes_per_frame(ioFormat.u.raw_audio) *
frames_per_buffer(input().format.u.raw_audio);
PRINT(("##### preferred output buffer_size: %ld (%x)\n", ioFormat.u.raw_audio.buffer_size, ioFormat.u.raw_audio.buffer_size));
ioFormat.u.raw_audio.frame_rate = input().format.u.raw_audio.frame_rate;
}
return B_OK;
}
status_t _AudioAdapterNode::getPreferredInputFormat(
media_format& ioFormat) {
status_t err = _inherited::getPreferredInputFormat(ioFormat);
if(err < B_OK)
return err;
_AudioAdapterParams* p = dynamic_cast<_AudioAdapterParams*>(parameterSet());
ASSERT(p);
media_raw_audio_format& f = ioFormat.u.raw_audio;
media_raw_audio_format& w = media_raw_audio_format::wildcard;
// copy user preferences
if(p->inputFormat.format != w.format)
f.format = p->inputFormat.format;
if(p->inputFormat.channel_count != w.channel_count)
f.channel_count = p->inputFormat.channel_count;
// // if one end is connected, prefer not to do channel conversions [15sep99]
// if(output().destination != media_destination::null)
// ioFormat.u.raw_audio.channel_count = output().format.u.raw_audio.channel_count;
// if output connected, constrain:
// buffer_size
// frame_rate
if(output().destination != media_destination::null) {
// if the user doesn't care, default to the output's frame format
if(f.format == w.format)
f.format = output().format.u.raw_audio.format;
if(f.channel_count == w.channel_count)
f.channel_count = output().format.u.raw_audio.channel_count;
f.buffer_size =
bytes_per_frame(f) *
frames_per_buffer(output().format.u.raw_audio);
f.frame_rate = output().format.u.raw_audio.frame_rate;
}
return B_OK;
}
int
cubeb_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name,
cubeb_stream_params stream_params, unsigned int latency,
cubeb_data_callback data_callback,
cubeb_state_callback state_callback,
void * user_ptr)
{
MMRESULT r;
WAVEFORMATEXTENSIBLE wfx;
cubeb_stream * stm;
int i;
size_t bufsz;
assert(context);
assert(stream);
*stream = NULL;
if (stream_params.rate < 1 || stream_params.rate > 192000 ||
stream_params.channels < 1 || stream_params.channels > 32 ||
latency < 1 || latency > 2000) {
return CUBEB_ERROR_INVALID_FORMAT;
}
memset(&wfx, 0, sizeof(wfx));
if (stream_params.channels > 2) {
wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
wfx.Format.cbSize = sizeof(wfx) - sizeof(wfx.Format);
} else {
wfx.Format.wFormatTag = WAVE_FORMAT_PCM;
if (stream_params.format == CUBEB_SAMPLE_FLOAT32LE) {
wfx.Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
}
wfx.Format.cbSize = 0;
}
wfx.Format.nChannels = stream_params.channels;
wfx.Format.nSamplesPerSec = stream_params.rate;
/* XXX fix channel mappings */
wfx.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
switch (stream_params.format) {
case CUBEB_SAMPLE_S16LE:
wfx.Format.wBitsPerSample = 16;
wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
break;
case CUBEB_SAMPLE_FLOAT32LE:
wfx.Format.wBitsPerSample = 32;
wfx.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
break;
default:
return CUBEB_ERROR_INVALID_FORMAT;
}
wfx.Format.nBlockAlign = (wfx.Format.wBitsPerSample * wfx.Format.nChannels) / 8;
wfx.Format.nAvgBytesPerSec = wfx.Format.nSamplesPerSec * wfx.Format.nBlockAlign;
wfx.Samples.wValidBitsPerSample = 0;
wfx.Samples.wSamplesPerBlock = 0;
wfx.Samples.wReserved = 0;
EnterCriticalSection(&context->lock);
/* CUBEB_STREAM_MAX is a horrible hack to avoid a situation where, when
many streams are active at once, a subset of them will not consume (via
playback) or release (via waveOutReset) their buffers. */
if (context->active_streams >= CUBEB_STREAM_MAX) {
LeaveCriticalSection(&context->lock);
return CUBEB_ERROR;
}
context->active_streams += 1;
LeaveCriticalSection(&context->lock);
stm = calloc(1, sizeof(*stm));
assert(stm);
stm->context = context;
stm->params = stream_params;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->user_ptr = user_ptr;
bufsz = (size_t) (stm->params.rate / 1000.0 * latency * bytes_per_frame(stm->params) / NBUFS);
if (bufsz % bytes_per_frame(stm->params) != 0) {
bufsz += bytes_per_frame(stm->params) - (bufsz % bytes_per_frame(stm->params));
}
assert(bufsz % bytes_per_frame(stm->params) == 0);
stm->buffer_size = bufsz;
InitializeCriticalSection(&stm->lock);
stm->event = CreateEvent(NULL, FALSE, FALSE, NULL);
if (!stm->event) {
cubeb_stream_destroy(stm);
return CUBEB_ERROR;
}
/* cubeb_buffer_callback will be called during waveOutOpen, so all
other initialization must be complete before calling it. */
r = waveOutOpen(&stm->waveout, WAVE_MAPPER, &wfx.Format,
(DWORD_PTR) cubeb_buffer_callback, (DWORD_PTR) stm,
CALLBACK_FUNCTION);
if (r != MMSYSERR_NOERROR) {
cubeb_stream_destroy(stm);
return CUBEB_ERROR;
}
r = waveOutPause(stm->waveout);
if (r != MMSYSERR_NOERROR) {
cubeb_stream_destroy(stm);
return CUBEB_ERROR;
}
for (i = 0; i < NBUFS; ++i) {
WAVEHDR * hdr = &stm->buffers[i];
hdr->lpData = calloc(1, bufsz);
assert(hdr->lpData);
hdr->dwBufferLength = bufsz;
hdr->dwFlags = 0;
r = waveOutPrepareHeader(stm->waveout, hdr, sizeof(*hdr));
if (r != MMSYSERR_NOERROR) {
cubeb_stream_destroy(stm);
return CUBEB_ERROR;
}
cubeb_refill_stream(stm);
}
*stream = stm;
return CUBEB_OK;
}
static int
winmm_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name,
cubeb_devid input_device,
cubeb_stream_params * input_stream_params,
cubeb_devid output_device,
cubeb_stream_params * output_stream_params,
unsigned int latency_frames,
cubeb_data_callback data_callback,
cubeb_state_callback state_callback,
void * user_ptr)
{
MMRESULT r;
WAVEFORMATEXTENSIBLE wfx;
cubeb_stream * stm;
int i;
size_t bufsz;
XASSERT(context);
XASSERT(stream);
if (input_stream_params) {
/* Capture support not yet implemented. */
return CUBEB_ERROR_NOT_SUPPORTED;
}
if (input_device || output_device) {
/* Device selection not yet implemented. */
return CUBEB_ERROR_DEVICE_UNAVAILABLE;
}
*stream = NULL;
memset(&wfx, 0, sizeof(wfx));
if (output_stream_params->channels > 2) {
wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
wfx.Format.cbSize = sizeof(wfx) - sizeof(wfx.Format);
} else {
wfx.Format.wFormatTag = WAVE_FORMAT_PCM;
if (output_stream_params->format == CUBEB_SAMPLE_FLOAT32LE) {
wfx.Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
}
wfx.Format.cbSize = 0;
}
wfx.Format.nChannels = output_stream_params->channels;
wfx.Format.nSamplesPerSec = output_stream_params->rate;
/* XXX fix channel mappings */
wfx.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
switch (output_stream_params->format) {
case CUBEB_SAMPLE_S16LE:
wfx.Format.wBitsPerSample = 16;
wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
break;
case CUBEB_SAMPLE_FLOAT32LE:
wfx.Format.wBitsPerSample = 32;
wfx.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
break;
default:
return CUBEB_ERROR_INVALID_FORMAT;
}
wfx.Format.nBlockAlign = (wfx.Format.wBitsPerSample * wfx.Format.nChannels) / 8;
wfx.Format.nAvgBytesPerSec = wfx.Format.nSamplesPerSec * wfx.Format.nBlockAlign;
wfx.Samples.wValidBitsPerSample = wfx.Format.wBitsPerSample;
EnterCriticalSection(&context->lock);
/* CUBEB_STREAM_MAX is a horrible hack to avoid a situation where, when
many streams are active at once, a subset of them will not consume (via
playback) or release (via waveOutReset) their buffers. */
if (context->active_streams >= CUBEB_STREAM_MAX) {
LeaveCriticalSection(&context->lock);
return CUBEB_ERROR;
}
context->active_streams += 1;
LeaveCriticalSection(&context->lock);
stm = calloc(1, sizeof(*stm));
XASSERT(stm);
stm->context = context;
stm->params = *output_stream_params;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->user_ptr = user_ptr;
stm->written = 0;
uint32_t latency_ms = latency_frames * 1000 / output_stream_params->rate;
if (latency_ms < context->minimum_latency_ms) {
latency_ms = context->minimum_latency_ms;
}
bufsz = (size_t) (stm->params.rate / 1000.0 * latency_ms * bytes_per_frame(stm->params) / NBUFS);
if (bufsz % bytes_per_frame(stm->params) != 0) {
bufsz += bytes_per_frame(stm->params) - (bufsz % bytes_per_frame(stm->params));
}
XASSERT(bufsz % bytes_per_frame(stm->params) == 0);
stm->buffer_size = bufsz;
InitializeCriticalSection(&stm->lock);
stm->event = CreateEvent(NULL, FALSE, FALSE, NULL);
if (!stm->event) {
winmm_stream_destroy(stm);
return CUBEB_ERROR;
}
stm->soft_volume = -1.0;
/* winmm_buffer_callback will be called during waveOutOpen, so all
other initialization must be complete before calling it. */
r = waveOutOpen(&stm->waveout, WAVE_MAPPER, &wfx.Format,
(DWORD_PTR) winmm_buffer_callback, (DWORD_PTR) stm,
CALLBACK_FUNCTION);
if (r != MMSYSERR_NOERROR) {
winmm_stream_destroy(stm);
return CUBEB_ERROR;
}
r = waveOutPause(stm->waveout);
if (r != MMSYSERR_NOERROR) {
winmm_stream_destroy(stm);
return CUBEB_ERROR;
}
for (i = 0; i < NBUFS; ++i) {
WAVEHDR * hdr = &stm->buffers[i];
hdr->lpData = calloc(1, bufsz);
XASSERT(hdr->lpData);
hdr->dwBufferLength = bufsz;
hdr->dwFlags = 0;
r = waveOutPrepareHeader(stm->waveout, hdr, sizeof(*hdr));
if (r != MMSYSERR_NOERROR) {
winmm_stream_destroy(stm);
return CUBEB_ERROR;
}
winmm_refill_stream(stm);
}
*stream = stm;
return CUBEB_OK;
}
static void
winmm_refill_stream(cubeb_stream * stm)
{
WAVEHDR * hdr;
long got;
long wanted;
MMRESULT r;
EnterCriticalSection(&stm->lock);
stm->free_buffers += 1;
XASSERT(stm->free_buffers > 0 && stm->free_buffers <= NBUFS);
if (stm->draining) {
LeaveCriticalSection(&stm->lock);
if (stm->free_buffers == NBUFS) {
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED);
}
SetEvent(stm->event);
return;
}
if (stm->shutdown) {
LeaveCriticalSection(&stm->lock);
SetEvent(stm->event);
return;
}
hdr = winmm_get_next_buffer(stm);
wanted = (DWORD) stm->buffer_size / bytes_per_frame(stm->params);
/* It is assumed that the caller is holding this lock. It must be dropped
during the callback to avoid deadlocks. */
LeaveCriticalSection(&stm->lock);
got = stm->data_callback(stm, stm->user_ptr, NULL, hdr->lpData, wanted);
EnterCriticalSection(&stm->lock);
if (got < 0) {
LeaveCriticalSection(&stm->lock);
/* XXX handle this case */
XASSERT(0);
return;
} else if (got < wanted) {
stm->draining = 1;
}
stm->written += got;
XASSERT(hdr->dwFlags & WHDR_PREPARED);
hdr->dwBufferLength = got * bytes_per_frame(stm->params);
XASSERT(hdr->dwBufferLength <= stm->buffer_size);
if (stm->soft_volume != -1.0) {
if (stm->params.format == CUBEB_SAMPLE_FLOAT32NE) {
float * b = (float *) hdr->lpData;
uint32_t i;
for (i = 0; i < got * stm->params.channels; i++) {
b[i] *= stm->soft_volume;
}
} else {
short * b = (short *) hdr->lpData;
uint32_t i;
for (i = 0; i < got * stm->params.channels; i++) {
b[i] = (short) (b[i] * stm->soft_volume);
}
}
}
r = waveOutWrite(stm->waveout, hdr, sizeof(*hdr));
if (r != MMSYSERR_NOERROR) {
LeaveCriticalSection(&stm->lock);
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR);
return;
}
LeaveCriticalSection(&stm->lock);
}
static void
refill_stream(cubeb_stream * stm, int prefill)
{
VOID * p1, * p2;
DWORD p1sz, p2sz;
HRESULT rv;
long dt;
/* calculate how much has played since last refill */
DWORD play, write;
rv = stm->buffer->GetCurrentPosition(&play, &write);
assert(rv == DS_OK);
long gap = write - play;
if (gap < 0) {
gap += stm->buffer_size;
}
#if 1
dt = GetTickCount() - stm->last_refill;
if (!prefill) {
double buflen = (double) (stm->buffer_size - gap) / bytes_per_frame(stm->params) / stm->params.rate * 1000;
if (dt > buflen) {
fprintf(stderr, "*** buffer wrap (%ld, %f, %f)***\n", dt, buflen, dt - buflen);
stm->slipped += (dt - buflen) / 1000.0 * bytes_per_frame(stm->params) * stm->params.rate;
}
}
#endif
unsigned long writepos = stm->written % stm->buffer_size;
long playsz = 0;
if (write < writepos) {
playsz = write + stm->buffer_size - writepos;
} else {
playsz = write - writepos;
}
/* can't write between play and write cursors */
playsz -= gap;
if (playsz < 0) {
#if 0
fprintf(stderr, "** negcapped, dt=%u real nwl=%ld p=%u w=%u g=%ld wo=%u **\n",
dt, playsz, play, write, gap, writepos);
#endif
return;
}
if (prefill) {
playsz = stm->buffer_size;
}
playsz -= bytes_per_frame(stm->params);
/* no space to refill */
if (playsz <= 0)
return;
/*assert(writepos >= write && ((writepos + playsz) % stm->buffer_size) < play);*/
/*
assumptions: buffer with w==p is full or empty
we know total writes is stm->written
so w==p and stm->written%stm->buffer_size==0 full or empty
need abs play pos to determine
rel play pos is (write + stm->buffer_size) - play
(0 + 10) - 0 -> 10 -> also assumes buffer is full
absplayed must be between stm->written-stm->buffer_size and stm->written.
XXX want prefill logic to work anytime as we will eventually call it from start()
*/
rv = stm->buffer->Lock(writepos, playsz, &p1, &p1sz, &p2, &p2sz, 0);
if (rv == DSERR_BUFFERLOST) {
stm->buffer->Restore();
rv = stm->buffer->Lock(writepos, playsz, &p1, &p1sz, &p2, &p2sz, 0);
}
assert(rv == DS_OK);
assert(p1sz % bytes_per_frame(stm->params) == 0);
assert(p2sz % bytes_per_frame(stm->params) == 0);
int r = stm->data_callback(stm, stm->user_ptr, p1,
p1sz / bytes_per_frame(stm->params));
if (p2 && r == CUBEB_OK) {
r = stm->data_callback(stm, stm->user_ptr, p2,
p2sz / bytes_per_frame(stm->params));
} else {
p2sz = 0;
}
#if 0
// XXX fix EOS/drain handling
if (r == CUBEB_EOS) {
LPDIRECTSOUNDNOTIFY notify;
rv = stm->buffer->QueryInterface(IID_IDirectSoundNotify, (LPVOID *) ¬ify);
assert(rv == DS_OK);
DSBPOSITIONNOTIFY note;
note.dwOffset = (writepos + p1sz + p2sz) % stm->buffer_size;
note.hEventNotify = stm->context->streams_event;
if (notify->SetNotificationPositions(1, ¬e) != DS_OK) {
/* XXX free resources */
assert(false);
}
notify->Release();
stm->draining = 1;
}
#endif
stm->last_refill = GetTickCount();
stm->written += p1sz + p2sz;
rv = stm->buffer->Unlock(p1, p1sz, p2, p2sz);
assert(rv == DS_OK);
}
CString DSoundBuf::Init( DSound &ds, DSoundBuf::hw hardware,
int iChannels, int iSampleRate, int iSampleBits, int iWriteAhead )
{
m_iChannels = iChannels;
m_iSampleRate = iSampleRate;
m_iSampleBits = iSampleBits;
m_iWriteAhead = iWriteAhead * bytes_per_frame();
m_iVolume = -1; /* unset */
m_bBufferLocked = false;
m_iWriteCursorPos = m_iWriteCursor = m_iBufferBytesFilled = 0;
m_iExtraWriteahead = 0;
m_iLastPosition = 0;
m_bPlaying = false;
ZERO( m_iLastCursors );
/* The size of the actual DSound buffer. This can be large; we generally
* won't fill it completely. */
m_iBufferSize = 1024*64;
m_iBufferSize = max( m_iBufferSize, m_iWriteAhead );
WAVEFORMATEX waveformat;
memset( &waveformat, 0, sizeof(waveformat) );
waveformat.cbSize = 0;
waveformat.wFormatTag = WAVE_FORMAT_PCM;
bool NeedCtrlFrequency = false;
if( m_iSampleRate == DYNAMIC_SAMPLERATE )
{
m_iSampleRate = 44100;
NeedCtrlFrequency = true;
}
int bytes = m_iSampleBits / 8;
waveformat.wBitsPerSample = WORD(m_iSampleBits);
waveformat.nChannels = WORD(m_iChannels);
waveformat.nSamplesPerSec = DWORD(m_iSampleRate);
waveformat.nBlockAlign = WORD(bytes*m_iChannels);
waveformat.nAvgBytesPerSec = m_iSampleRate * bytes*m_iChannels;
/* Try to create the secondary buffer */
DSBUFFERDESC format;
memset( &format, 0, sizeof(format) );
format.dwSize = sizeof(format);
#ifdef _XBOX
format.dwFlags = 0;
#else
format.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLVOLUME;
#endif
#ifndef _XBOX
/* Don't use DSBCAPS_STATIC. It's meant for static buffers, and we
* only use streaming buffers. */
if( hardware == HW_HARDWARE )
format.dwFlags |= DSBCAPS_LOCHARDWARE;
else
format.dwFlags |= DSBCAPS_LOCSOFTWARE;
#endif
if( NeedCtrlFrequency )
format.dwFlags |= DSBCAPS_CTRLFREQUENCY;
format.dwBufferBytes = m_iBufferSize;
#ifndef _XBOX
format.dwReserved = 0;
#else
DSMIXBINVOLUMEPAIR dsmbvp[8] =
{
{ DSMIXBIN_FRONT_LEFT, DSBVOLUME_MAX }, // left channel
{ DSMIXBIN_FRONT_RIGHT, DSBVOLUME_MAX }, // right channel
{ DSMIXBIN_FRONT_CENTER, DSBVOLUME_MAX }, // left channel
{ DSMIXBIN_FRONT_CENTER, DSBVOLUME_MAX }, // right channel
{ DSMIXBIN_BACK_LEFT, DSBVOLUME_MAX }, // left channel
{ DSMIXBIN_BACK_RIGHT, DSBVOLUME_MAX }, // right channel
{ DSMIXBIN_LOW_FREQUENCY, DSBVOLUME_MAX }, // left channel
{ DSMIXBIN_LOW_FREQUENCY, DSBVOLUME_MAX } // right channel
};
DSMIXBINS dsmb;
dsmb.dwMixBinCount = 8;
dsmb.lpMixBinVolumePairs = dsmbvp;
format.lpMixBins = &dsmb;
#endif
format.lpwfxFormat = &waveformat;
HRESULT hr = ds.GetDS()->CreateSoundBuffer( &format, &m_pBuffer, NULL );
if( FAILED(hr) )
return hr_ssprintf( hr, "CreateSoundBuffer failed" );
#ifndef _XBOX
/* I'm not sure this should ever be needed, but ... */
DSBCAPS bcaps;
bcaps.dwSize=sizeof(bcaps);
hr = m_pBuffer->GetCaps( &bcaps );
if( FAILED(hr) )
return hr_ssprintf( hr, "m_pBuffer->GetCaps" );
if( int(bcaps.dwBufferBytes) != m_iBufferSize )
{
LOG->Warn( "bcaps.dwBufferBytes (%i) != m_iBufferSize(%i); adjusting", bcaps.dwBufferBytes, m_iBufferSize );
m_iBufferSize = bcaps.dwBufferBytes;
m_iWriteAhead = min( m_iWriteAhead, m_iBufferSize );
}
if( !(bcaps.dwFlags & DSBCAPS_CTRLVOLUME) )
LOG->Warn( "Sound channel missing DSBCAPS_CTRLVOLUME" );
if( !(bcaps.dwFlags & DSBCAPS_GETCURRENTPOSITION2) )
LOG->Warn( "Sound channel missing DSBCAPS_GETCURRENTPOSITION2" );
DWORD got;
hr = m_pBuffer->GetFormat( &waveformat, sizeof(waveformat), &got );
if( FAILED(hr) )
LOG->Warn( hr_ssprintf(hr, "GetFormat on secondary buffer") );
else if( (int) waveformat.nSamplesPerSec != m_iSampleRate )
LOG->Warn( "Secondary buffer set to %i instead of %i", waveformat.nSamplesPerSec, m_iSampleRate );
#endif
m_pTempBuffer = new char[m_iBufferSize];
return "";
}
int S9xMovieOpen (const char* filename, bool8 read_only, uint8 sync_flags, uint8 sync_flags2)
{
FILE* fd;
STREAM stream;
int result;
int fn;
char movie_filename [_MAX_PATH];
#ifdef WIN32
_fullpath(movie_filename, filename, _MAX_PATH);
#else
strcpy(movie_filename, filename);
#endif
if(!(fd=fopen(movie_filename, "rb+")))
if(!(fd=fopen(movie_filename, "rb")))
return FILE_NOT_FOUND;
else
read_only = TRUE;
const bool8 wasPaused = Settings.Paused;
const uint32 prevFrameTime = Settings.FrameTime;
// stop current movie before opening
change_state(MOVIE_STATE_NONE);
// read header
if((result=read_movie_header(fd, &Movie))!=SUCCESS)
{
fclose(fd);
return result;
}
read_movie_extrarominfo(fd, &Movie);
fn=dup(fileno(fd));
fclose(fd);
// apparently this lseek is necessary
lseek(fn, Movie.SaveStateOffset, SEEK_SET);
if(!(stream=REOPEN_STREAM(fn, "rb")))
return FILE_NOT_FOUND;
// store previous, before changing to the movie's settings
store_previous_settings();
// store default
if (sync_flags & MOVIE_SYNC_DATA_EXISTS)
{
Settings.UseWIPAPUTiming = (sync_flags & MOVIE_SYNC_WIP1TIMING) ? TRUE : FALSE;
Settings.SoundEnvelopeHeightReading = (sync_flags & MOVIE_SYNC_VOLUMEENVX) ? TRUE : FALSE;
Settings.FakeMuteFix = (sync_flags & MOVIE_SYNC_FAKEMUTE) ? TRUE : FALSE;
Settings.UpAndDown = (sync_flags & MOVIE_SYNC_LEFTRIGHT) ? TRUE : FALSE; // doesn't actually affect synchronization
Settings.SoundSync = (sync_flags & MOVIE_SYNC_SYNCSOUND) ? TRUE : FALSE; // doesn't seem to affect synchronization
Settings.InitFastROMSetting = (sync_flags2 & MOVIE_SYNC2_INIT_FASTROM) ? TRUE : FALSE;
//Settings.ShutdownMaster = (sync_flags & MOVIE_SYNC_NOCPUSHUTDOWN) ? FALSE : TRUE;
}
// set from movie
restore_movie_settings();
if(Movie.Opts & MOVIE_OPT_FROM_RESET)
{
Movie.State = MOVIE_STATE_PLAY; // prevent NSRT controller switching (in S9xPostRomInit)
if(!Memory.LoadLastROM())
S9xReset();
Memory.ClearSRAM(false); // in case the SRAM read fails
Movie.State = MOVIE_STATE_NONE;
// save only SRAM for a from-reset snapshot
result=(READ_STREAM(Memory.SRAM, 0x20000, stream) == 0x20000) ? SUCCESS : WRONG_FORMAT;
}
else
{
result=S9xUnfreezeFromStream(stream);
}
CLOSE_STREAM(stream);
if(result!=SUCCESS)
{
return result;
}
if(!(fd=fopen(movie_filename, "rb+")))
if(!(fd=fopen(movie_filename, "rb")))
return FILE_NOT_FOUND;
else
read_only = TRUE;
if(fseek(fd, Movie.ControllerDataOffset, SEEK_SET))
return WRONG_FORMAT;
// read controller data
Movie.File=fd;
Movie.BytesPerFrame=bytes_per_frame();
Movie.InputBufferPtr=Movie.InputBuffer;
uint32 to_read=Movie.BytesPerFrame * (Movie.MaxFrame+1);
reserve_buffer_space(to_read);
fread(Movie.InputBufferPtr, 1, to_read, fd);
// read "baseline" controller data
if(Movie.MaxFrame)
read_frame_controller_data();
strncpy(Movie.Filename, movie_filename, _MAX_PATH);
Movie.Filename[_MAX_PATH-1]='\0';
Movie.CurrentFrame=0;
Movie.ReadOnly=read_only;
change_state(MOVIE_STATE_PLAY);
Settings.Paused = wasPaused;
Settings.FrameTime = prevFrameTime; // restore emulation speed
Movie.RecordedThisSession = false;
S9xUpdateFrameCounter(-1);
Movie.RequiresReset = false;
S9xMessage(S9X_INFO, S9X_MOVIE_INFO, MOVIE_INFO_REPLAY);
return SUCCESS;
}
bool DSoundBuf::get_output_buf( char **pBuffer, unsigned *pBufferSize, int iChunksize )
{
ASSERT( !m_bBufferLocked );
iChunksize *= bytes_per_frame();
DWORD iCursorStart, iCursorEnd;
HRESULT result;
/* It's easiest to think of the cursor as a block, starting and ending at
* the two values returned by GetCurrentPosition, that we can't write to. */
result = m_pBuffer->GetCurrentPosition( &iCursorStart, &iCursorEnd );
#ifndef _XBOX
if( result == DSERR_BUFFERLOST )
{
m_pBuffer->Restore();
result = m_pBuffer->GetCurrentPosition( &iCursorStart, &iCursorEnd );
}
if( result != DS_OK )
{
LOG->Warn( hr_ssprintf(result, "DirectSound::GetCurrentPosition failed") );
return false;
}
#endif
memmove( &m_iLastCursors[0][0], &m_iLastCursors[1][0], sizeof(int)*6 );
m_iLastCursors[3][0] = iCursorStart;
m_iLastCursors[3][1] = iCursorEnd;
/* Some cards (Creative AudioPCI) have a no-write area even when not playing. I'm not
* sure what that means, but it breaks the assumption that we can fill the whole writeahead
* when prebuffering. */
if( !m_bPlaying )
iCursorEnd = iCursorStart;
/*
* Some cards (Game Theater XP 7.1 hercwdm.sys 5.12.01.4101 [466688b, 01-10-2003])
* have odd behavior when starting a sound: the start/end cursors go:
*
* 0,0 end cursor forced equal to start above (normal)
* 4608, 1764 end cursor trailing the write cursor; except with old emulated
* WaveOut devices, this shouldn't happen; it indicates that the
* driver expects almost the whole buffer to be filled. Also, the
* play cursor is too far ahead from the last call for the amount
* of actual time passed.
* 704, XXX start cursor moves back to where it should be. I don't have an exact
* end cursor position, but in general from now on it stays about 5kb
* ahead of start (which is where it should be).
*
* The second call is completely wrong; both the start and end cursors are meaningless.
* Detect this: if the end cursor is close behind the start cursor, don't do anything.
* (We can't; we have no idea what the cursors actually are.)
*/
{
int iPrefetch = iCursorEnd - iCursorStart;
wrap( iPrefetch, m_iBufferSize );
if( m_iBufferSize - iPrefetch < 1024*4 )
{
LOG->Trace( "Strange DirectSound cursor ignored: %i..%i", iCursorStart, iCursorEnd );
return false;
}
}
/* Update m_iBufferBytesFilled. */
{
int iFirstByteFilled = m_iWriteCursor - m_iBufferBytesFilled;
wrap( iFirstByteFilled, m_iBufferSize );
/* The number of bytes that have been played since the last time we got here: */
int bytes_played = iCursorStart - iFirstByteFilled;
wrap( bytes_played, m_iBufferSize );
m_iBufferBytesFilled -= bytes_played;
m_iBufferBytesFilled = max( 0, m_iBufferBytesFilled );
if( m_iExtraWriteahead )
{
int used = min( m_iExtraWriteahead, bytes_played );
CString s = ssprintf("used %i of %i (%i..%i)", used, m_iExtraWriteahead, iCursorStart, iCursorEnd );
s += "; last: ";
for( int i = 0; i < 4; ++i )
s += ssprintf( "%i, %i; ", m_iLastCursors[i][0], m_iLastCursors[i][1] );
LOG->Trace("%s", s.c_str());
m_iWriteAhead -= used;
m_iExtraWriteahead -= used;
}
}
CheckWriteahead( iCursorStart, iCursorEnd );
CheckUnderrun( iCursorStart, iCursorEnd );
/* If we already have enough bytes written ahead, stop. */
if( m_iBufferBytesFilled > m_iWriteAhead )
return false;
int iNumBytesEmpty = m_iWriteAhead - m_iBufferBytesFilled;
/* num_bytes_empty is the amount of free buffer space. If it's
* too small, come back later. */
if( iNumBytesEmpty < iChunksize )
return false;
// LOG->Trace("gave %i at %i (%i, %i) %i filled", iNumBytesEmpty, m_iWriteCursor, cursor, write, m_iBufferBytesFilled);
/* Lock the audio buffer. */
result = m_pBuffer->Lock( m_iWriteCursor, iNumBytesEmpty, (LPVOID *) &m_pLockedBuf1, (DWORD *) &m_iLockedSize1, (LPVOID *) &m_pLockedBuf2, (DWORD *) &m_iLockedSize2, 0 );
#ifndef _XBOX
if( result == DSERR_BUFFERLOST )
{
m_pBuffer->Restore();
result = m_pBuffer->Lock( m_iWriteCursor, iNumBytesEmpty, (LPVOID *) &m_pLockedBuf1, (DWORD *) &m_iLockedSize1, (LPVOID *) &m_pLockedBuf2, (DWORD *) &m_iLockedSize2, 0 );
}
#endif
if( result != DS_OK )
{
LOG->Warn( hr_ssprintf(result, "Couldn't lock the DirectSound buffer.") );
return false;
}
*pBuffer = m_pTempBuffer;
*pBufferSize = m_iLockedSize1 + m_iLockedSize2;
m_iWriteCursor += iNumBytesEmpty;
if( m_iWriteCursor >= m_iBufferSize )
m_iWriteCursor -= m_iBufferSize;
m_iBufferBytesFilled += iNumBytesEmpty;
m_iWriteCursorPos += iNumBytesEmpty / bytes_per_frame();
m_bBufferLocked = true;
return true;
}
// create or discard buffer group if necessary
void AudioFilterNode::updateBufferGroup() {
status_t err;
size_t inputSize = bytes_per_frame(m_input.format.u.raw_audio);
size_t outputSize = bytes_per_frame(m_output.format.u.raw_audio);
if(m_input.source == media_source::null ||
m_output.destination == media_destination::null ||
inputSize >= outputSize) {
PRINT(("###### NO BUFFER GROUP NEEDED\n"));
// no internal buffer group needed
if(m_bufferGroup) {
// does this block? +++++
delete m_bufferGroup;
m_bufferGroup = 0;
}
return;
}
int32 bufferCount = EventLatency() / BufferDuration() + 1 + 1;
// +++++
// [e.moon 27sep99] this is a reasonable number of buffers,
// but it fails with looped file-player node in BeOS 4.5.2.
//
if(bufferCount < 5)
bufferCount = 5;
// if(bufferCount < 3)
// bufferCount = 3;
if(m_bufferGroup) {
// is the current group sufficient?
int32 curBufferCount;
err = m_bufferGroup->CountBuffers(&curBufferCount);
if(err == B_OK && curBufferCount >= bufferCount) {
BBuffer* buf = m_bufferGroup->RequestBuffer(
outputSize, -1);
if(buf) {
// yup
buf->Recycle();
return;
}
}
// nope, delete it to make way for the new one
delete m_bufferGroup;
m_bufferGroup = 0;
}
// create buffer group
PRINT((
"##### AudioFilterNode::updateBufferGroup():\n"
"##### creating %ld buffers of size %ld\n",
bufferCount, m_output.format.u.raw_audio.buffer_size));
m_bufferGroup = new BBufferGroup(
m_output.format.u.raw_audio.buffer_size,
bufferCount);
}
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
}
}
void
MixerInput::BufferReceived(BBuffer* buffer)
{
void* data;
size_t size;
bigtime_t start;
bigtime_t buffer_duration;
if (!fMixBuffer) {
ERROR("MixerInput::BufferReceived: dropped incoming buffer as we "
"don't have a mix buffer\n");
return;
}
data = buffer->Data();
size = buffer->SizeUsed();
start = buffer->Header()->start_time;
buffer_duration = duration_for_frames(fInput.format.u.raw_audio.frame_rate,
size / bytes_per_frame(fInput.format.u.raw_audio));
if (start < 0) {
ERROR("MixerInput::BufferReceived: buffer with negative start time of "
"%Ld dropped\n", start);
return;
}
// swap the byte order of this buffer, if necessary
if (fInputByteSwap)
fInputByteSwap->Swap(data, size);
int offset = frames_for_duration(fMixBufferFrameRate, start)
% fMixBufferFrameCount;
PRINT(4, "MixerInput::BufferReceived: buffer start %10Ld, offset %6d\n",
start, offset);
int in_frames = size / bytes_per_frame(fInput.format.u.raw_audio);
double frames = ((double)in_frames * fMixBufferFrameRate)
/ fInput.format.u.raw_audio.frame_rate;
int out_frames = int(frames);
fFractionalFrames += frames - double(out_frames);
if (fFractionalFrames >= 1.0) {
fFractionalFrames -= 1.0;
out_frames++;
}
// if fLastDataFrameWritten != -1, then we have a valid last position
// and can do glitch compensation
if (fLastDataFrameWritten >= 0) {
int expected_frame = (fLastDataFrameWritten + 1)
% fMixBufferFrameCount;
if (offset != expected_frame) {
// due to rounding and other errors, offset might be off by +/- 1
// this is not really a bad glitch, we just adjust the position
if (offset == fLastDataFrameWritten) {
// printf("MixerInput::BufferReceived: -1 frame GLITCH! last "
// "frame was %ld, expected frame was %d, new frame is %d\n",
// fLastDataFrameWritten, expected_frame, offset);
offset = expected_frame;
} else if (offset == ((fLastDataFrameWritten + 2)
% fMixBufferFrameCount)) {
// printf("MixerInput::BufferReceived: +1 frame GLITCH! last "
// "frame was %ld, expected frame was %d, new frame is %d\n",
// fLastDataFrameWritten, expected_frame, offset);
offset = expected_frame;
} else {
printf("MixerInput::BufferReceived: GLITCH! last frame was "
"%4ld, expected frame was %4d, new frame is %4d\n",
fLastDataFrameWritten, expected_frame, offset);
if (start > fLastDataAvailableTime) {
if ((start - fLastDataAvailableTime)
< (buffer_duration / 10)) {
// buffer is less than 10% of buffer duration too late
printf("short glitch, buffer too late, time delta "
"%Ld\n", start - fLastDataAvailableTime);
offset = expected_frame;
out_frames++;
} else {
// buffer more than 10% of buffer duration too late
// TODO: zerofill buffer
printf("MAJOR glitch, buffer too late, time delta "
"%Ld\n", start - fLastDataAvailableTime);
}
} else { // start <= fLastDataAvailableTime
// the new buffer is too early
if ((fLastDataAvailableTime - start)
< (buffer_duration / 10)) {
// buffer is less than 10% of buffer duration too early
printf("short glitch, buffer too early, time delta "
"%Ld\n", fLastDataAvailableTime - start);
offset = expected_frame;
out_frames--;
if (out_frames < 1)
out_frames = 1;
} else {
// buffer more than 10% of buffer duration too early
// TODO: zerofill buffer
printf("MAJOR glitch, buffer too early, time delta "
"%Ld\n", fLastDataAvailableTime - start);
}
}
}
}
}
// printf("data arrived for %10Ld to %10Ld, storing at frames %ld to %ld\n",
// start,
// start + duration_for_frames(fInput.format.u.raw_audio.frame_rate,
// frames_per_buffer(fInput.format.u.raw_audio)), offset,
// offset + out_frames);
if (offset + out_frames > fMixBufferFrameCount) {
int out_frames1 = fMixBufferFrameCount - offset;
int out_frames2 = out_frames - out_frames1;
int in_frames1 = (out_frames1 * in_frames) / out_frames;
int in_frames2 = in_frames - in_frames1;
// printf("at %10Ld, data arrived for %10Ld to %10Ld, storing at "
// "frames %ld to %ld and %ld to %ld\n", fCore->fTimeSource->Now(),
// start,
// start + duration_for_frames(fInput.format.u.raw_audio.frame_rate,
// frames_per_buffer(fInput.format.u.raw_audio)), offset,
// offset + out_frames1 - 1, 0, out_frames2 - 1);
PRINT(3, "at %10Ld, data arrived for %10Ld to %10Ld, storing at "
"frames %ld to %ld and %ld to %ld\n", fCore->fTimeSource->Now(),
start,
start + duration_for_frames(fInput.format.u.raw_audio.frame_rate,
frames_per_buffer(fInput.format.u.raw_audio)), offset,
offset + out_frames1 - 1, 0, out_frames2 - 1);
PRINT(5, " in_frames %5d, out_frames %5d, in_frames1 %5d, "
"out_frames1 %5d, in_frames2 %5d, out_frames2 %5d\n",
in_frames, out_frames, in_frames1, out_frames1, in_frames2,
out_frames2);
fLastDataFrameWritten = out_frames2 - 1;
// convert offset from frames into bytes
offset *= sizeof(float) * fInputChannelCount;
for (int i = 0; i < fInputChannelCount; i++) {
fResampler[i]->Resample(
reinterpret_cast<char*>(data)
+ i * bytes_per_sample(fInput.format.u.raw_audio),
bytes_per_frame(fInput.format.u.raw_audio), in_frames1,
reinterpret_cast<char*>(fInputChannelInfo[i].buffer_base)
+ offset, fInputChannelCount * sizeof(float), out_frames1,
fInputChannelInfo[i].gain);
fResampler[i]->Resample(
reinterpret_cast<char*>(data)
+ i * bytes_per_sample(fInput.format.u.raw_audio)
+ in_frames1 * bytes_per_frame(fInput.format.u.raw_audio),
bytes_per_frame(fInput.format.u.raw_audio), in_frames2,
reinterpret_cast<char*>(fInputChannelInfo[i].buffer_base),
fInputChannelCount * sizeof(float), out_frames2,
fInputChannelInfo[i].gain);
}
} else {
// printf("at %10Ld, data arrived for %10Ld to %10Ld, storing at "
// "frames %ld to %ld\n", fCore->fTimeSource->Now(), start,
// start + duration_for_frames(fInput.format.u.raw_audio.frame_rate,
// frames_per_buffer(fInput.format.u.raw_audio)), offset,
// offset + out_frames - 1);
PRINT(3, "at %10Ld, data arrived for %10Ld to %10Ld, storing at "
"frames %ld to %ld\n", fCore->fTimeSource->Now(), start,
start + duration_for_frames(fInput.format.u.raw_audio.frame_rate,
frames_per_buffer(fInput.format.u.raw_audio)), offset,
offset + out_frames - 1);
PRINT(5, " in_frames %5d, out_frames %5d\n", in_frames, out_frames);
fLastDataFrameWritten = offset + out_frames - 1;
// convert offset from frames into bytes
offset *= sizeof(float) * fInputChannelCount;
for (int i = 0; i < fInputChannelCount; i++) {
fResampler[i]->Resample(
reinterpret_cast<char*>(data)
+ i * bytes_per_sample(fInput.format.u.raw_audio),
bytes_per_frame(fInput.format.u.raw_audio), in_frames,
reinterpret_cast<char*>(fInputChannelInfo[i].buffer_base)
+ offset, fInputChannelCount * sizeof(float),
out_frames, fInputChannelInfo[i].gain);
}
}
fLastDataAvailableTime = start + buffer_duration;
}
int S9xMovieOpen (const char* filename, bool8 read_only)
{
FILE* fd;
STREAM stream;
int result;
int fn;
if(!(fd=fopen(filename, read_only ? "rb" : "rb+")))
return FILE_NOT_FOUND;
// stop current movie before opening
change_state(MOVIE_STATE_NONE);
// read header
if((result=read_movie_header(fd, &Movie))!=SUCCESS)
{
fclose(fd);
return result;
}
fn=dup(fileno(fd));
fclose(fd);
// apparently this lseek is necessary
lseek(fn, Movie.SaveStateOffset, SEEK_SET);
if(!(stream=REOPEN_STREAM(fn, "rb")))
return FILE_NOT_FOUND;
if(Movie.Opts & MOVIE_OPT_FROM_RESET)
{
S9xReset();
// save only SRAM for a from-reset snapshot
result=(READ_STREAM(SRAM, 0x20000, stream) == 0x20000) ? SUCCESS : WRONG_FORMAT;
}
else
{
result=S9xUnfreezeFromStream(stream);
}
CLOSE_STREAM(stream);
if(result!=SUCCESS)
{
return result;
}
if(!(fd=fopen(filename, read_only ? "rb" : "rb+")))
return FILE_NOT_FOUND;
if(fseek(fd, Movie.ControllerDataOffset, SEEK_SET))
return WRONG_FORMAT;
// read controller data
Movie.File=fd;
Movie.BytesPerFrame=bytes_per_frame();
Movie.InputBufferPtr=Movie.InputBuffer;
uint32 to_read=Movie.BytesPerFrame * (Movie.MaxFrame+1);
reserve_buffer_space(to_read);
fread(Movie.InputBufferPtr, 1, to_read, fd);
// read "baseline" controller data
read_frame_controller_data();
strncpy(Movie.Filename, filename, _MAX_PATH);
Movie.Filename[_MAX_PATH-1]='\0';
Movie.CurrentFrame=0;
Movie.ReadOnly=read_only;
change_state(MOVIE_STATE_PLAY);
S9xMessage(S9X_INFO, S9X_MOVIE_INFO, MOVIE_INFO_REPLAY);
return SUCCESS;
}
static int
directsound_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name,
cubeb_stream_params stream_params, unsigned int latency,
cubeb_data_callback data_callback,
cubeb_state_callback state_callback,
void * user_ptr)
{
struct cubeb_list_node * node;
assert(context);
*stream = NULL;
/*
create primary buffer
*/
DSBUFFERDESC bd;
bd.dwSize = sizeof(DSBUFFERDESC);
bd.dwFlags = DSBCAPS_PRIMARYBUFFER;
bd.dwBufferBytes = 0;
bd.dwReserved = 0;
bd.lpwfxFormat = NULL;
bd.guid3DAlgorithm = DS3DALG_DEFAULT;
LPDIRECTSOUNDBUFFER primary;
if (FAILED(context->dsound->CreateSoundBuffer(&bd, &primary, NULL))) {
return 1;
}
WAVEFORMATEXTENSIBLE wfx;
wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
wfx.Format.nChannels = stream_params.channels;
wfx.Format.nSamplesPerSec = stream_params.rate;
wfx.Format.cbSize = sizeof(wfx) - sizeof(wfx.Format);
/* XXX fix channel mappings */
wfx.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
switch (stream_params.format) {
case CUBEB_SAMPLE_S16LE:
wfx.Format.wBitsPerSample = 16;
wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
break;
case CUBEB_SAMPLE_FLOAT32LE:
wfx.Format.wBitsPerSample = 32;
wfx.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
break;
default:
return CUBEB_ERROR_INVALID_FORMAT;
}
wfx.Format.nBlockAlign = (wfx.Format.wBitsPerSample * wfx.Format.nChannels) / 8;
wfx.Format.nAvgBytesPerSec = wfx.Format.nSamplesPerSec * wfx.Format.nBlockAlign;
wfx.Samples.wValidBitsPerSample = wfx.Format.wBitsPerSample;
if (FAILED(primary->SetFormat((LPWAVEFORMATEX) &wfx))) {
/* XXX free primary */
return CUBEB_ERROR;
}
primary->Release();
cubeb_stream * stm = (cubeb_stream *) calloc(1, sizeof(*stm));
assert(stm);
stm->context = context;
stm->params = stream_params;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->user_ptr = user_ptr;
InitializeCriticalSection(&stm->lock);
/*
create secondary buffer
*/
bd.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLVOLUME | DSBCAPS_CTRLPOSITIONNOTIFY;
bd.dwBufferBytes = (DWORD) (wfx.Format.nSamplesPerSec / 1000.0 * latency * bytes_per_frame(stream_params));
if (bd.dwBufferBytes % bytes_per_frame(stream_params) != 0) {
bd.dwBufferBytes += bytes_per_frame(stream_params) - (bd.dwBufferBytes % bytes_per_frame(stream_params));
}
bd.lpwfxFormat = (LPWAVEFORMATEX) &wfx;
if (FAILED(context->dsound->CreateSoundBuffer(&bd, &stm->buffer, NULL))) {
return CUBEB_ERROR;
}
stm->buffer_size = bd.dwBufferBytes;
LPDIRECTSOUNDNOTIFY notify;
if (stm->buffer->QueryInterface(IID_IDirectSoundNotify, (LPVOID *) ¬ify) != DS_OK) {
/* XXX free resources */
return CUBEB_ERROR;
}
DSBPOSITIONNOTIFY note[3];
for (int i = 0; i < 3; ++i) {
note[i].dwOffset = (stm->buffer_size / 4) * i;
note[i].hEventNotify = context->streams_event;
}
if (notify->SetNotificationPositions(3, note) != DS_OK) {
/* XXX free resources */
return CUBEB_ERROR;
}
notify->Release();
refill_stream(stm, 1);
/* XXX remove this, just a test that double refill does not overwrite existing data */
refill_stream(stm, 0);
uint64_t pos;
cubeb_stream_get_position(stm, &pos);
stm->node = (struct cubeb_list_node *) calloc(1, sizeof(*node));
stm->node->stream = stm;
EnterCriticalSection(&context->lock);
if (!context->streams) {
context->streams = stm->node;
} else {
node = context->streams;
while (node->next) {
node = node->next;
}
node->next = stm->node;
stm->node->prev = node;
}
LeaveCriticalSection(&context->lock);
SetEvent(context->streams_event);
*stream = stm;
return CUBEB_OK;
}