LRESULT Twindow::TwidgetSubclassChb::onContextMenu(HWND hwnd,UINT uMsg,WPARAM wParam,LPARAM lParam)
{
if (self->tr->translateMode) {
return TwidgetSubclass::onContextMenu(hwnd,uMsg,wParam,lParam);
}
static const char_t *menu[]= {_l("Reset"),NULL};
if (self->selectFromMenu(menu,0)==0)
if (self->deci->resetParam(ff_abs(bind->idff))!=S_OK) {
return 0;
} else if (bind->onClick) {
(self->*bind->onClick)();
} else {
self->cfg2dlg();
}
return 0;
}
template<class sample_t> void TaudioFilterVolume::volume(sample_t* const samples, size_t numsamples, const TsampleFormat &fmt, const TvolumeSettings *cfg)
{
if (cfg->is) {
if (!cfg->normalize) {
if (isVol) {
typedef void (*TprocessVol)(sample_t * const samples, size_t numsamples, const int * volumes);
static const TprocessVol processVol[9] = {
NULL,
&Tmultiply<1>::processVol,
&Tmultiply<2>::processVol,
&Tmultiply<3>::processVol,
&Tmultiply<4>::processVol,
&Tmultiply<5>::processVol,
&Tmultiply<6>::processVol,
&Tmultiply<7>::processVol,
&Tmultiply<8>::processVol
};
processVol[fmt.nchannels](samples, numsamples, volumes);
}
} else {
size_t len = numsamples * fmt.nchannels;
float max = 0.0f;
for (size_t i = 0; i < len; i++) {
float tmp = float(samples[i]);
//max = max(max, tmp)
max = max > tmp ? max : tmp;
}
//upper = min(mul, cfg->normalizeMax/100.0f)
float upper = mul > (cfg->normalizeMax / 100.0f) ? (cfg->normalizeMax / 100.0f) : mul;
// Evaluate an adequate 'mul' coefficient based on previous state, current samples level, etc
if (mul > TsampleFormatInfo<sample_t>::max() / max || mul > cfg->normalizeMax / 100.0f) {
mul = limit((float)(TsampleFormatInfo<sample_t>::max() / max), MUL_MIN, upper);
} else {
if (cfg->normalizeRegainVolume) {
// here we make sure that the current sound level (max * mul) will not rise above 'RegainThreshold'.
if (max < (TsampleFormatInfo<sample_t>::max() / mul) * RegainThreshold) {
// note that in one second, in average, the sum of ((float)numsamples / fmt.freq) will be 1.
float step = MUL_STEP * ((float)numsamples / fmt.freq);
if (mul + step <= cfg->normalizeMax / 100.0f) {
mul += step;
} else { // mul + step > cfg->normalizeMax/100.0f
// make sure that the last increment will be performed, even if it's smaller than 'step'
// otherwise, current amplification could be displayed as 399% instead of 400%, for example.
mul = cfg->normalizeMax / 100.0f;
}
}
}
}
// Scale & clamp the samples
typedef void (*TprocessMul)(sample_t * const samples, size_t numsamples, const int * volumes, float mul);
static const TprocessMul processMul[9] = {
NULL,
&Tmultiply<1>::processMul,
&Tmultiply<2>::processMul,
&Tmultiply<3>::processMul,
&Tmultiply<4>::processMul,
&Tmultiply<5>::processMul,
&Tmultiply<6>::processMul,
&Tmultiply<7>::processMul,
&Tmultiply<8>::processMul
};
processMul[fmt.nchannels](samples, numsamples, volumes, mul);
}
if (numsamples > 0 && deci->getParam2(IDFF_showCurrentVolume) && deci->getCfgDlgHwnd()) {
int64_t sum[8];
memset(sum, 0, sizeof(sum));
for (size_t i = 0; i < numsamples * fmt.nchannels;) {
for (unsigned int ch = 0; ch < fmt.nchannels; ch++, i++) {
sum[ch] += int64_t((int64_t)65536 * ff_abs(samples[i]));
}
}
CAutoLock lock(&csVolumes);
for (unsigned int i = 0; i < fmt.nchannels; i++) {
storedvolumes.volumes[i] = int((sum[i] / numsamples) / int64_t(TsampleFormatInfo<sample_t>::max()));
}
storedvolumes.have = true;
}
}
}
STDMETHODIMP TffdshowDecAudioInputPin::Receive(IMediaSample* pIn)
{
if (this != filter->inpin) {
//DPRINTF(_l("TffdshowDecAudioInputPin::Receive Not right pin : this = %u, filter inpin = %u"), this, filter->inpin);
if (m_useBlock) {
m_evBlock.Wait();
} else {
return S_FALSE;
}
}
if (!isActive()) {
//DPRINTF(_l("TffdshowDecAudioInputPin::Receive Pin unlocked : this = %u, filter inpin = %u"), this, filter->inpin);
if (this != filter->inpin) {
return S_FALSE;
}
return E_FAIL;
}
//CAutoLock cAutoLock(&m_csReceive);
if (filter->IsStopped()) {
return S_FALSE;
}
HRESULT hr = TinputPin::Receive(pIn);
if (hr != S_OK) {
return hr;
}
AM_SAMPLE2_PROPERTIES* const pProps = SampleProps();
if (pProps->dwStreamId != AM_STREAM_MEDIA) {
return filter->m_pOutput->Deliver(pIn);
}
AM_MEDIA_TYPE *pmt = NULL;
if (SUCCEEDED(pIn->GetMediaType(&pmt)) && pmt) {
CMediaType mt(*pmt);
SetMediaType(&mt);
DeleteMediaType(pmt);
pmt = NULL;
}
BYTE *src = NULL;
if (FAILED(hr = pIn->GetPointer(&src))) {
return hr;
}
long srclen = pIn->GetActualDataLength();
if (strippacket) {
StripPacket(src, srclen);
}
REFERENCE_TIME rtStart = _I64_MIN, rtStop = _I64_MIN;
hr = pIn->GetTime(&rtStart, &rtStop);
if (hr == S_OK) {
insample_rtStart = rtStart;
insample_rtStop = rtStop;
//DPRINTF(_l("TffdshowDecAudioInputPin::Receive audio sample start duration %I64i %I64i"),rtStart,rtStop-rtStart);
}
if (pIn->IsDiscontinuity() == S_OK) {
filter->discontinuity = true;
buf.clear();
if (FAILED(hr)) {
return S_OK;
}
filter->m_rtStartDec = filter->m_rtStartProc = rtStart;
}
if (SUCCEEDED(hr)) {
REFERENCE_TIME j = filter->m_rtStartDec - rtStart;
jitter = int(j);
if ((uint64_t)ff_abs(j) > 100 * (REF_SECOND_MULT / 1000) // +-100ms jitter is allowed for now
&& codecId != AV_CODEC_ID_FLAC
&& codecId != AV_CODEC_ID_TTA
&& codecId != AV_CODEC_ID_WAVPACK
&& codecId != AV_CODEC_ID_TRUEHD
&& codecId != AV_CODEC_ID_MLP
&& codecId != AV_CODEC_ID_COOK
&& !bitstream_codec(codecId)
&& filter->getParam2(IDFF_audio_decoder_JitterCorrection)) {
DPRINTF(_l("jitter correction"));
buf.clear();
newSrcBuffer.clear();
filter->m_rtStartDec = filter->m_rtStartProc = rtStart;
if (audioParser) {
audioParser->NewSegment();
}
}
}
buf.append(src, srclen);
buf.reserve(buf.size() + 32);
AVCodecID newCodecId = codecId;
TaudioParserData audioParserData;
// Before sending data to the decoder, we parse it
switch (codecId) {
case AV_CODEC_ID_DTS:
case CODEC_ID_LIBDTS:
case CODEC_ID_SPDIF_DTS:
case AV_CODEC_ID_AC3:
case AV_CODEC_ID_EAC3:
case AV_CODEC_ID_MLP:
case AV_CODEC_ID_TRUEHD:
case CODEC_ID_LIBA52:
case CODEC_ID_SPDIF_AC3:
case CODEC_ID_PCM:
case CODEC_ID_BITSTREAM_TRUEHD:
case CODEC_ID_BITSTREAM_DTSHD:
// Search for DTS in Wav only if option is checked
if (codecId == CODEC_ID_PCM && !searchdts) {
break;
}
// Do not search for DTS in PCM in next frames (otherwise DTS syncword maybe wrongly detected)
searchdts = false;
newCodecId = audioParser->parseStream(buf.size() ? &buf[0] : NULL, (int)buf.size(), &newSrcBuffer);
if (newCodecId == AV_CODEC_ID_NONE) {
newSrcBuffer.clear();
break;
}
// Get updated data from the parser
audioParserData = audioParser->getParserData();
// Clear input buffer (if 2 source buffers are coexisting, sound will be garbled)
buf.clear();
if (codecId != newCodecId) {
DPRINTF(_l("TffdshowDecAudioInputPin : switching codec from %s to %s"), getCodecName(codecId), getCodecName(newCodecId));
codecId = newCodecId;
// Update input sample format from (corrected or updated) parser data
if (audioParserData.channels != 0) {
filter->insf.setChannels(audioParserData.channels);
}
if (audioParserData.sample_rate != 0) {
filter->insf.freq = audioParserData.sample_rate;
}
if (audioParserData.sample_format != 0) {
filter->insf.sf = audioParserData.sample_format;
}
filter->insf.alternateSF = audioParserData.alternateSampleFormat;
// Sample format from audio parser data
TsampleFormat fmt = TsampleFormat(
(audioParserData.sample_format != 0) ? audioParserData.sample_format : TsampleFormat::SF_PCM16,
audioParserData.sample_rate, audioParserData.channels);
// Reinitialize the audio codec according to the new codecId
DPRINTF(_l("TffdshowDecAudioInputPin::Receive : Initialize audio codec %s"), getCodecName(codecId));
if (audio) {
delete audio;
codec = audio = NULL;
}
codec = audio = TaudioCodec::initSource(filter, this, codecId, filter->insf, filter->insf.toCMediaType());
if (!audio) {
return false;
}
jitter = 0;
}
// Update libavcodec context with correct channels and bitrate read from parser
if (lavc_codec(codecId)) {
TaudioCodecLibavcodec *audioCodecLibavcodec = (TaudioCodecLibavcodec*)audio;
if (audioCodecLibavcodec) {
if (audioParserData.channels != 0) {
audioCodecLibavcodec->avctx->channels = audioParserData.channels;
}
if (audioParserData.bit_rate != 0) {
audioCodecLibavcodec->avctx->bit_rate = audioParserData.bit_rate;
}
if (audioParserData.sample_rate != 0) {
audioCodecLibavcodec->avctx->sample_rate = audioParserData.sample_rate;
}
}
}
if (audioParserData.channels != 0) {
filter->insf.setChannels(audioParserData.channels);
}
if (audioParserData.sample_rate != 0) {
filter->insf.freq = audioParserData.sample_rate;
}
if (audioParserData.sample_format != 0) {
filter->insf.sf = audioParserData.sample_format;
}
newSrcBuffer.reserve(newSrcBuffer.size() + 32);
hr = audio->decode(newSrcBuffer);
if (hr == S_FALSE) {
return S_OK;
} else if (hr != S_OK) {
DPRINTF(_l("TffdshowDecAudioInputPin::Receive decode failed pin %u (%lx)"), this, hr);
}
return hr;
break;
default:
// Decode data
hr = audio->decode(buf);
if (hr == S_FALSE) {
return S_OK;
} else if (hr != S_OK) {
DPRINTF(_l("TffdshowDecAudioInputPin::Receive decode failed pin %u (%lx)"), this, hr);
}
return hr;
break;
}
hr = audio->decode(buf);
if (hr == S_FALSE) {
return S_OK;
} else if (hr != S_OK) {
DPRINTF(_l("TffdshowDecAudioInputPin::Receive decode failed pin %u (%lx)"), this, hr);
}
return hr;
}