void UseSeg (const SegDef* D)
/* Use the segment with the given name */
{
Segment* Seg = SegmentList;
while (Seg) {
if (strcmp (Seg->Def->Name, D->Name) == 0) {
/* We found this segment. Check if the type is identical */
if (D->AddrSize != ADDR_SIZE_DEFAULT &&
Seg->Def->AddrSize != D->AddrSize) {
Error ("Segment attribute mismatch");
/* Use the new attribute to avoid errors */
Seg->Def->AddrSize = D->AddrSize;
}
ActiveSeg = Seg;
return;
}
/* Check next segment */
Seg = Seg->List;
}
/* Segment is not in list, create a new one */
if (D->AddrSize == ADDR_SIZE_DEFAULT) {
Seg = NewSegment (D->Name, ADDR_SIZE_ABS);
} else {
Seg = NewSegment (D->Name, D->AddrSize);
}
ActiveSeg = Seg;
}
void UseSeg (const SegDef* D)
/* Use the segment with the given name */
{
unsigned I;
for (I = 0; I < CollCount (&SegmentList); ++I) {
Segment* Seg = CollAtUnchecked (&SegmentList, I);
if (strcmp (Seg->Def->Name, D->Name) == 0) {
/* We found this segment. Check if the type is identical */
if (D->AddrSize != ADDR_SIZE_DEFAULT &&
Seg->Def->AddrSize != D->AddrSize) {
Error ("Segment attribute mismatch");
/* Use the new attribute to avoid errors */
Seg->Def->AddrSize = D->AddrSize;
}
ActiveSeg = Seg;
return;
}
}
/* Segment is not in list, create a new one */
if (D->AddrSize == ADDR_SIZE_DEFAULT) {
ActiveSeg = NewSegment (D->Name, ADDR_SIZE_ABS);
} else {
ActiveSeg = NewSegment (D->Name, D->AddrSize);
}
}
static already_AddRefed<SharedMemory>
ReadSegment(const IPC::Message& aDescriptor, Shmem::id_t* aId, size_t* aNBytes, size_t aExtraSize)
{
if (SHMEM_CREATED_MESSAGE_TYPE != aDescriptor.type()) {
NS_ERROR("expected 'shmem created' message");
return nullptr;
}
SharedMemory::SharedMemoryType type;
PickleIterator iter(aDescriptor);
if (!ShmemCreated::ReadInfo(&aDescriptor, &iter, aId, aNBytes, &type)) {
return nullptr;
}
RefPtr<SharedMemory> segment = NewSegment(type);
if (!segment) {
return nullptr;
}
if (!segment->ReadHandle(&aDescriptor, &iter)) {
NS_ERROR("trying to open invalid handle");
return nullptr;
}
aDescriptor.EndRead(iter);
size_t size = SharedMemory::PageAlignedSize(*aNBytes + aExtraSize);
if (!segment->Map(size)) {
return nullptr;
}
// close the handle to the segment after it is mapped
segment->CloseHandle();
return segment.forget();
}
void CDXVADecoderMpeg2::Init()
{
TRACE(_T("CDXVADecoderMpeg2::Init()\n"));
memset (&m_PictureParams, 0, sizeof(m_PictureParams));
memset (&m_SliceInfo, 0, sizeof(m_SliceInfo));
memset (&m_QMatrixData, 0, sizeof(m_QMatrixData));
m_PictureParams.bMacroblockWidthMinus1 = 15; // This is equal to "15" for MPEG-1, MPEG-2, H.263, and MPEG-4
m_PictureParams.bMacroblockHeightMinus1 = 15; // This is equal to "15" for MPEG-1, MPEG-2, H.261, H.263, and MPEG-4
m_PictureParams.bBlockWidthMinus1 = 7; // This is equal to "7" for MPEG-1, MPEG-2, H.261, H.263, and MPEG-4
m_PictureParams.bBlockHeightMinus1 = 7; // This is equal to "7" for MPEG-1, MPEG-2, H.261, H.263, and MPEG-4
m_PictureParams.bBPPminus1 = 7; // It is equal to "7" for MPEG-1, MPEG-2, H.261, and H.263
m_PictureParams.bChromaFormat = 0x01; // For MPEG-1, MPEG-2 "Main Profile," H.261 and H.263 bitstreams, this value shall always be set to "01", indicating "4:2:0" format
m_nMaxWaiting = 5;
m_wRefPictureIndex[0] = NO_REF_FRAME;
m_wRefPictureIndex[1] = NO_REF_FRAME;
m_nSliceCount = 0;
switch (GetMode()) {
case MPEG2_VLD :
AllocExecuteParams (4);
break;
default :
ASSERT(FALSE);
}
m_pMPEG2Buffer = NULL;
m_nMPEG2BufferSize = 0;
NewSegment();
}
Segment* GetSegment (unsigned Name, unsigned char AddrSize, const char* ObjName)
/* Search for a segment and return an existing one. If the segment does not
** exist, create a new one and return that. ObjName is only used for the error
** message and may be NULL if the segment is linker generated.
*/
{
/* Try to locate the segment in the table */
Segment* S = SegFind (Name);
/* If we don't have that segment already, allocate it using the type of
** the first section.
*/
if (S == 0) {
/* Create a new segment */
S = NewSegment (Name, AddrSize);
} else {
/* Check if the existing segment has the requested address size */
if (S->AddrSize != AddrSize) {
/* Allow an empty object name */
if (ObjName == 0) {
ObjName = "[linker generated]";
}
Error ("Module '%s': Type mismatch for segment '%s'", ObjName,
GetString (Name));
}
}
/* Return the segment */
return S;
}
//this is where the list of segments is created
//returns t which is an object whose head points to the first segment of the track
track *NewTrack(int maxx, int maxy){
int i, random;
track *t = (track *) malloc(sizeof(track));
t->maxx = maxx;
t->maxy = maxy;
t->tail = NewSegment(10, t, maxx, maxy);
drawSegment(t->tail);
for(i = 0; i < maxy-2; i++){
t->head = NewSegment(10, t, maxx, maxy);
//drawSegment(t->head);
}
drawSegment(t->head);
refresh();
return t;
}
static void registerSegment( orl_sec_handle o_shnd )
//**************************************************
{
orl_sec_flags sec_flags;
orl_sec_handle reloc_section;
orl_sec_alignment alignment;
char * content;
int ctr;
segment *seg;
seg = NewSegment();
seg->name = ORLSecGetName( o_shnd );
seg->size = ORLSecGetSize( o_shnd );
seg->start = 0;
seg->use_32 = 1; // only 32-bit object files use ORL
seg->attr = ( 2 << 2 ); // (?) combine public
alignment = ORLSecGetAlignment( o_shnd );
// FIXME: Need better alignment translation.
switch( alignment ) {
case 0:
seg->attr |= ( 1 << 5 ); break;
case 1:
seg->attr |= ( 2 << 5 ); break;
case 3:
case 4:
seg->attr |= ( 3 << 5 ); break;
case 8:
seg->attr |= ( 4 << 5 ); break;
case 2:
seg->attr |= ( 5 << 5 ); break;
case 12:
seg->attr |= ( 6 << 5 ); break;
default:
// fprintf( stderr, "NOTE! 'Strange' alignment (%d) found. Using byte alignment.\n", alignment );
seg->attr |= ( 1 << 5 ); break;
}
sec_flags = ORLSecGetFlags( o_shnd );
if( !( sec_flags & ORL_SEC_FLAG_EXEC ) ) {
seg->data_seg = true;
}
if( seg->size > 0 && ORLSecGetContents( o_shnd, &content ) == ORL_OKAY ) {
Segment = seg;
// Putting contents into segment struct.
for( ctr = 0; ctr < seg->size; ctr++ ) {
PutSegByte( ctr, content[ctr] );
}
}
if( !HashTableInsert( SectionToSegmentTable, (hash_value)o_shnd, (hash_data)seg ) ) {
SysError( ERR_OUT_OF_MEM, false );
}
reloc_section = ORLSecGetRelocTable( o_shnd );
if( reloc_section ) {
if( !addRelocSection( reloc_section ) ) {
SysError( ERR_OUT_OF_MEM, false );
}
}
}
void Synchronizer::NewSegment() {
if(m_output_format->m_audio_enabled) {
AudioLock audiolock(&m_audio_data);
InitAudioSegment(audiolock.get());
}
SharedLock lock(&m_shared_data);
NewSegment(lock.get());
}
void Synchronizer::NewSegment() {
{
AudioLock audiolock(&m_audio_data);
InitAudioSegment(audiolock.get());
}
SharedLock lock(&m_shared_data);
NewSegment(lock.get());
}
void Synchronizer::ReadAudioHole() {
Q_ASSERT(m_audio_encoder != NULL);
SharedLock lock(&m_shared_data);
// if the audio has not been started, ignore it
if(!lock->m_segment_audio_started)
return;
Logger::LogWarning("[Synchronizer::ReadAudioSamples] Warning: Received hole in audio stream, starting new segment to keep the audio in sync with the video (some video and/or audio may be lost).");
NewSegment(lock.get());
}
static already_AddRefed<SharedMemory>
CreateSegment(SharedMemory::SharedMemoryType aType, size_t aNBytes, size_t aExtraSize)
{
RefPtr<SharedMemory> segment = NewSegment(aType);
if (!segment) {
return nullptr;
}
size_t size = SharedMemory::PageAlignedSize(aNBytes + aExtraSize);
if (!segment->Create(size) || !segment->Map(size)) {
return nullptr;
}
return segment.forget();
}
//----------------------------------------------------------------------------
//! @brief フィルタをポーズする
//! @return エラーコード
//----------------------------------------------------------------------------
STDMETHODIMP CDemuxSource::Pause()
{
{
CAutoLock lock(&m_crtFilterLock);
if( m_State == State_Stopped )
{
HRESULT hr = Reader()->OnStart();
if( FAILED(hr) )
return hr;
NewSegment();
}
}
return CSource::Pause();
}
//--------------------------------------------------------------------------------------------------
static dstr_Ref_t NewOrFirstSegmentRef
(
dstr_Ref_t headRef ///< [IN] The head of the string.
)
//--------------------------------------------------------------------------------------------------
{
dstr_Ref_t segmentRef = FirstSegmentRef(headRef);
if (segmentRef != NULL)
{
return segmentRef;
}
segmentRef = NewSegment();
le_sls_Stack(&headRef->head.list, &segmentRef->body.link);
return segmentRef;
}
//--------------------------------------------------------------------------------------------------
static dstr_Ref_t NewOrNextSegmentRef
(
dstr_Ref_t headRef, ///< [IN] The head of the string.
dstr_Ref_t currentPtr ///< [IN] The current sub-section of the string.
)
//--------------------------------------------------------------------------------------------------
{
dstr_Ref_t segmentRef = NextSegmentRef(headRef, currentPtr);
if (segmentRef != NULL)
{
return segmentRef;
}
segmentRef = NewSegment();
le_sls_AddAfter(&headRef->head.list, ¤tPtr->body.link, &segmentRef->body.link);
return segmentRef;
}
HRESULT CBufferFilterOutputPin::DeliverNewSegment(REFERENCE_TIME tStart, REFERENCE_TIME tStop, double dRate)
{
CallQueue(NewSegment(tStart, tStop, dRate));
}
void Synchronizer::ReadAudioSamples(unsigned int sample_rate, unsigned int channels, unsigned int sample_count, const uint8_t* data, AVSampleFormat format, int64_t timestamp) {
Q_ASSERT(m_audio_encoder != NULL);
SharedLock lock(&m_shared_data);
if(lock->m_sync_diagram != NULL) {
lock->m_sync_diagram->AddBlock(1, (double) timestamp * 1.0e-6, (double) timestamp * 1.0e-6 + (double) sample_count / (double) m_audio_sample_rate, QColor(0, 255, 0));
}
if(sample_count == 0)
return;
// check the timestamp
if(lock->m_segment_audio_started && timestamp < lock->m_segment_audio_last_timestamp) {
if(timestamp < lock->m_segment_audio_last_timestamp - 10000)
Logger::LogWarning("[Synchronizer::ReadAudioSamples] Warning: Received audio samples with non-monotonic timestamp.");
timestamp = lock->m_segment_audio_last_timestamp;
}
Q_ASSERT(sample_rate == m_audio_sample_rate); // resampling isn't supported
Q_ASSERT(channels == m_audio_channels); // remixing isn't supported
Q_ASSERT(format == AV_SAMPLE_FMT_S16); // only S16 is currently supported
// avoid memory problems by limiting the audio buffer size
if(lock->m_audio_buffer.GetSize() / m_audio_sample_size >= MAX_AUDIO_SAMPLES_BUFFERED) {
if(lock->m_segment_video_started) {
Logger::LogWarning("[Synchronizer::ReadAudioSamples] Warning: Audio buffer overflow, starting new segment to keep the audio in sync with the video "
"(some video and/or audio may be lost). The video input seems to be too slow.");
NewSegment(lock.get());
} else {
// If the video hasn't started yet, it makes more sense to drop the oldest samples.
// Shifting the start time like this isn't completely accurate, but this shouldn't happen often anyway.
// The number of samples dropped is calculated so that the buffer will be 90% full after this.
size_t n = lock->m_audio_buffer.GetSize() / m_audio_sample_size - (MAX_AUDIO_SAMPLES_BUFFERED - MAX_AUDIO_SAMPLES_BUFFERED / 10);
lock->m_audio_buffer.Drop(n * m_audio_sample_size);
lock->m_segment_audio_start_time += (int64_t) round((double) n / (double) m_audio_sample_rate * 1.0e6);
}
}
// do speed correction (i.e. do the calculations so the video can synchronize to it)
// The point of speed correction is to keep video and audio in sync even when the clocks are not running at exactly the same speed.
// This can happen because the sample rate of the sound card is not always 100% accurate. Even a 0.1% error will result in audio that is
// seconds too early or too late at the end of a one hour video. This problem doesn't occur on all computers though (I'm not sure why).
// Another cause of desynchronization is problems/glitches with PulseAudio (e.g. jumps in time when switching between sources).
if(lock->m_segment_audio_started) {
double sample_length = (double) (lock->m_segment_audio_samples_read + lock->m_audio_buffer.GetSize() / m_audio_sample_size) / (double) m_audio_sample_rate;
double time_length = (double) (timestamp - lock->m_segment_audio_start_time) * 1.0e-6;
double current_error = (sample_length - time_length) - lock->m_av_desync;
if(fabs(current_error) > DESYNC_ERROR_THRESHOLD) {
Logger::LogWarning("[Synchronizer::ReadAudioSamples] Warning: Desynchronization is too high, starting new segment to keep the audio "
"in sync with the video (some video and/or audio may be lost).");
NewSegment(lock.get());
} else {
double dt = std::min((double) (timestamp - lock->m_segment_audio_last_timestamp) * 1.0e-6, 0.5);
lock->m_av_desync_i = clamp(lock->m_av_desync_i + DESYNC_CORRECTION_I * current_error * dt, -1.0, 1.0);
lock->m_av_desync += (DESYNC_CORRECTION_P * current_error + lock->m_av_desync_i) * dt;
if(lock->m_av_desync_i < -0.05 && lock->m_warn_desync) {
lock->m_warn_desync = false;
Logger::LogWarning("[Synchronizer::ReadAudioSamples] Warning: Audio input is more than 5% too slow!");
}
if(lock->m_av_desync_i > 0.05 && lock->m_warn_desync) {
lock->m_warn_desync = false;
Logger::LogWarning("[Synchronizer::ReadAudioSamples] Warning: Audio input is more than 5% too fast!");
}
}
}
// start audio
if(!lock->m_segment_audio_started) {
lock->m_segment_audio_started = true;
lock->m_segment_audio_start_time = timestamp;
lock->m_segment_audio_stop_time = timestamp;
}
// store the samples
lock->m_segment_audio_last_timestamp = timestamp;
lock->m_audio_buffer.Write((const char*) data, sample_count * m_audio_sample_size);
// increase segment stop time
double sample_length = (double) (lock->m_segment_audio_samples_read + lock->m_audio_buffer.GetSize() / m_audio_sample_size) / (double) m_audio_sample_rate;
lock->m_segment_audio_stop_time = lock->m_segment_audio_start_time + (int64_t) round(sample_length * 1.0e6);
//Logger::LogInfo("[Synchronizer::ReadAudioSamples] Added audio samples at " + QString::number(timestamp) + ".");
}
void Synchronizer::NewSegment() {
SharedLock lock(&m_shared_data);
NewSegment(lock.get());
}
void Synchronizer::ReadAudioSamples(unsigned int channels, unsigned int sample_rate, AVSampleFormat format, unsigned int sample_count, const uint8_t* data, int64_t timestamp) {
assert(m_output_format->m_audio_enabled);
// sanity check
if(sample_count == 0)
return;
// add new block to sync diagram
if(m_sync_diagram != NULL)
m_sync_diagram->AddBlock(1, (double) timestamp * 1.0e-6, (double) timestamp * 1.0e-6 + (double) sample_count / (double) sample_rate, QColor(0, 255, 0));
AudioLock audiolock(&m_audio_data);
// check the timestamp
if(timestamp < audiolock->m_last_timestamp) {
if(timestamp < audiolock->m_last_timestamp - 10000)
Logger::LogWarning("[Synchronizer::ReadAudioSamples] " + Logger::tr("Warning: Received audio samples with non-monotonic timestamp."));
timestamp = audiolock->m_last_timestamp;
}
// update the timestamps
int64_t previous_timestamp;
if(audiolock->m_first_timestamp == (int64_t) AV_NOPTS_VALUE) {
audiolock->m_filtered_timestamp = timestamp;
audiolock->m_first_timestamp = timestamp;
previous_timestamp = timestamp;
} else {
previous_timestamp = audiolock->m_last_timestamp;
}
audiolock->m_last_timestamp = timestamp;
// filter the timestamp
int64_t timestamp_delta = (int64_t) sample_count * (int64_t) 1000000 / (int64_t) sample_rate;
audiolock->m_filtered_timestamp += (timestamp - audiolock->m_filtered_timestamp) / AUDIO_TIMESTAMP_FILTER;
// calculate drift
double current_drift = GetAudioDrift(audiolock.get());
// if there are too many audio samples, drop some of them (unlikely unless you use PulseAudio)
if(current_drift > DRIFT_ERROR_THRESHOLD && !audiolock->m_drop_samples) {
audiolock->m_drop_samples = true;
Logger::LogWarning("[Synchronizer::ReadAudioSamples] " + Logger::tr("Warning: Too many audio samples, dropping samples to keep the audio in sync with the video."));
}
// if there are not enough audio samples, insert zeros
if(current_drift < -DRIFT_ERROR_THRESHOLD && !audiolock->m_insert_samples) {
audiolock->m_insert_samples = true;
Logger::LogWarning("[Synchronizer::ReadAudioSamples] " + Logger::tr("Warning: Not enough audio samples, inserting silence to keep the audio in sync with the video."));
}
// reset filter and recalculate drift if necessary
if(audiolock->m_drop_samples || audiolock->m_insert_samples) {
audiolock->m_filtered_timestamp = timestamp;
current_drift = GetAudioDrift(audiolock.get());
}
// drop samples
if(audiolock->m_drop_samples) {
audiolock->m_drop_samples = false;
// drop samples
int n = (int) round(current_drift * (double) sample_rate);
if(n > 0) {
if(n >= (int) sample_count) {
audiolock->m_drop_samples = true;
return; // drop all samples
}
if(format == AV_SAMPLE_FMT_FLT) {
data += n * channels * sizeof(float);
} else if(format == AV_SAMPLE_FMT_S16) {
data += n * channels * sizeof(int16_t);
} else {
assert(false);
}
sample_count -= n;
}
}
// insert zeros
unsigned int sample_count_out = 0;
if(audiolock->m_insert_samples) {
audiolock->m_insert_samples = false;
// how many samples should be inserted?
int n = (int) round(-current_drift * (double) sample_rate);
if(n > 0) {
// insert zeros
audiolock->m_temp_input_buffer.Alloc(n * m_output_format->m_audio_channels);
std::fill_n(audiolock->m_temp_input_buffer.GetData(), n * m_output_format->m_audio_channels, 0.0f);
sample_count_out = audiolock->m_fast_resampler->Resample((double) sample_rate / (double) m_output_format->m_audio_sample_rate, 1.0,
audiolock->m_temp_input_buffer.GetData(), n, &audiolock->m_temp_output_buffer, sample_count_out);
// recalculate drift
current_drift = GetAudioDrift(audiolock.get(), sample_count_out);
}
}
// increase filtered timestamp
audiolock->m_filtered_timestamp += timestamp_delta;
// do drift correction
// The point of drift correction is to keep video and audio in sync even when the clocks are not running at exactly the same speed.
// This can happen because the sample rate of the sound card is not always 100% accurate. Even a 0.1% error will result in audio that is
// seconds too early or too late at the end of a one hour video. This problem doesn't occur on all computers though (I'm not sure why).
// Another cause of desynchronization is problems/glitches with PulseAudio (e.g. jumps in time when switching between sources).
double drift_correction_dt = fmin((double) (timestamp - previous_timestamp) * 1.0e-6, DRIFT_MAX_BLOCK);
audiolock->m_average_drift = clamp(audiolock->m_average_drift + DRIFT_CORRECTION_I * current_drift * drift_correction_dt, -0.5, 0.5);
if(audiolock->m_average_drift < -0.02 && audiolock->m_warn_desync) {
audiolock->m_warn_desync = false;
Logger::LogWarning("[Synchronizer::ReadAudioSamples] " + Logger::tr("Warning: Audio input is more than 2% too slow!"));
}
if(audiolock->m_average_drift > 0.02 && audiolock->m_warn_desync) {
audiolock->m_warn_desync = false;
Logger::LogWarning("[Synchronizer::ReadAudioSamples] " + Logger::tr("Warning: Audio input is more than 2% too fast!"));
}
double length = (double) sample_count / (double) sample_rate;
double drift_correction = clamp(DRIFT_CORRECTION_P * current_drift + audiolock->m_average_drift, -0.5, 0.5) * fmin(1.0, DRIFT_MAX_BLOCK / length);
//qDebug() << "current_drift" << current_drift << "average_drift" << audiolock->m_average_drift << "drift_correction" << drift_correction;
// convert the samples
const float *data_float = NULL; // to keep GCC happy
if(format == AV_SAMPLE_FMT_FLT) {
if(channels == m_output_format->m_audio_channels) {
data_float = (const float*) data;
} else {
audiolock->m_temp_input_buffer.Alloc(sample_count * m_output_format->m_audio_channels);
data_float = audiolock->m_temp_input_buffer.GetData();
SampleChannelRemap(sample_count, (const float*) data, channels, audiolock->m_temp_input_buffer.GetData(), m_output_format->m_audio_channels);
}
} else if(format == AV_SAMPLE_FMT_S16) {
audiolock->m_temp_input_buffer.Alloc(sample_count * m_output_format->m_audio_channels);
data_float = audiolock->m_temp_input_buffer.GetData();
SampleChannelRemap(sample_count, (const int16_t*) data, channels, audiolock->m_temp_input_buffer.GetData(), m_output_format->m_audio_channels);
} else {
assert(false);
}
// resample
sample_count_out = audiolock->m_fast_resampler->Resample((double) sample_rate / (double) m_output_format->m_audio_sample_rate, 1.0 / (1.0 - drift_correction),
data_float, sample_count, &audiolock->m_temp_output_buffer, sample_count_out);
audiolock->m_samples_written += sample_count_out;
SharedLock lock(&m_shared_data);
// avoid memory problems by limiting the audio buffer size
if(lock->m_audio_buffer.GetSize() / m_output_format->m_audio_channels >= MAX_AUDIO_SAMPLES_BUFFERED) {
if(lock->m_segment_video_started) {
Logger::LogWarning("[Synchronizer::ReadAudioSamples] " + Logger::tr("Warning: Audio buffer overflow, starting new segment to keep the audio in sync with the video "
"(some video and/or audio may be lost). The video input seems to be too slow."));
NewSegment(lock.get());
} else {
// If the video hasn't started yet, it makes more sense to drop the oldest samples.
// Shifting the start time like this isn't completely accurate, but this shouldn't happen often anyway.
// The number of samples dropped is calculated so that the buffer will be 90% full after this.
size_t n = lock->m_audio_buffer.GetSize() / m_output_format->m_audio_channels - MAX_AUDIO_SAMPLES_BUFFERED * 9 / 10;
lock->m_audio_buffer.Pop(n * m_output_format->m_audio_channels);
lock->m_segment_audio_start_time += (int64_t) round((double) n / (double) m_output_format->m_audio_sample_rate * 1.0e6);
}
}
// start audio
if(!lock->m_segment_audio_started) {
lock->m_segment_audio_started = true;
lock->m_segment_audio_start_time = timestamp;
lock->m_segment_audio_stop_time = timestamp;
}
// store the samples
lock->m_audio_buffer.Push(audiolock->m_temp_output_buffer.GetData(), sample_count_out * m_output_format->m_audio_channels);
// increase segment stop time
double new_sample_length = (double) (lock->m_segment_audio_samples_read + lock->m_audio_buffer.GetSize() / m_output_format->m_audio_channels) / (double) m_output_format->m_audio_sample_rate;
lock->m_segment_audio_stop_time = lock->m_segment_audio_start_time + (int64_t) round(new_sample_length * 1.0e6);
}
CDXVADecoderMpeg2::~CDXVADecoderMpeg2(void)
{
TRACE(_T("CDXVADecoderMpeg2::Destroy()\n"));
NewSegment();
}
