Added basic sample rate control in the DirectSound module, removed chunk and buffer settings as they must be computed from rate.
git-svn-id: http://yate.null.ro/svn/yate/trunk@3164 acf43c95-373e-0410-b603-e72c3f656dc1
This commit is contained in:
paulc
parent
8e07545b79
commit
f4537ba165
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@ -4,17 +4,5 @@
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; primary: boolean: Use the primary playback sound buffer instead of a secondary
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;primary=yes
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; There is not much tweaking possible with these settings.
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; Some sanity is enforced in code but actual working limits depend on the sound drivers
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; chunk: int: Number of bytes in the chunk of samples transferred at once
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;chunk=320
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; minsize: int: Number of bytes in buffer before we start playing back
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;minsize=640
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; bufsize: int: Bytes allocated for secondary playback and the record buffers
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;bufsize=1280
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; maxsize: int: Number of buffered bytes when we start dropping chunks
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;maxsize=1600
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; rate: int: Default sampling rate used, can be 8000, 16000, 32000
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;rate=8000
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@ -43,18 +43,14 @@ namespace { // anonymous
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// we should use the primary sound buffer else we will lose sound while we have no input focus
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static bool s_primary = true;
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// 20ms minimum chunk, 100ms buffer
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#define CHUNK_SIZE 320
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static unsigned int s_chunk = CHUNK_SIZE;
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static unsigned int s_minsize = 2*CHUNK_SIZE;
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static unsigned int s_bufsize = 4*CHUNK_SIZE;
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static unsigned int s_maxsize = 5*CHUNK_SIZE;
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// default sampling rate
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static int s_rate = 8000;
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class DSoundSource : public DataSource
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{
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friend class DSoundRec;
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public:
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DSoundSource();
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DSoundSource(int rate = 8000);
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~DSoundSource();
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bool control(NamedList& msg);
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private:
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@ -65,7 +61,7 @@ class DSoundConsumer : public DataConsumer
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{
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friend class DSoundPlay;
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public:
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DSoundConsumer(bool stereo = false);
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DSoundConsumer(int rate = 8000, bool stereo = false);
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~DSoundConsumer();
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virtual unsigned long Consume(const DataBlock &data, unsigned long tStamp, unsigned long flags);
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bool control(NamedList& msg);
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@ -78,7 +74,7 @@ private:
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class DSoundPlay : public Thread, public Mutex
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{
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public:
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DSoundPlay(DSoundConsumer* owner, LPGUID device = 0);
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DSoundPlay(DSoundConsumer* owner, DWORD rate, LPGUID device = 0);
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virtual ~DSoundPlay();
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virtual void run();
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virtual void cleanup();
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@ -93,10 +89,12 @@ public:
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bool control(NamedList& msg);
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private:
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DSoundConsumer* m_owner;
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DWORD m_rate;
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LPGUID m_device;
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LPDIRECTSOUND m_ds;
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LPDIRECTSOUNDBUFFER m_dsb;
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DWORD m_buffSize;
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DWORD m_chunk;
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DataBlock m_buf;
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u_int64_t m_start;
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u_int64_t m_total;
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@ -106,7 +104,7 @@ private:
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class DSoundRec : public Thread
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{
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public:
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DSoundRec(DSoundSource* owner, LPGUID device = 0);
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DSoundRec(DSoundSource* owner, DWORD rate, LPGUID device = 0);
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virtual ~DSoundRec();
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virtual void run();
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virtual void cleanup();
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@ -120,6 +118,7 @@ public:
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bool control(NamedList& msg);
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private:
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DSoundSource* m_owner;
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DWORD m_rate;
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LPGUID m_device;
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LPDIRECTSOUNDCAPTURE m_ds;
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LPDIRECTSOUNDCAPTUREBUFFER m_dsb;
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@ -133,7 +132,7 @@ private:
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class DSoundChan : public Channel
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{
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public:
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DSoundChan();
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DSoundChan(int rate = 8000);
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virtual ~DSoundChan();
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};
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@ -162,10 +161,10 @@ private:
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INIT_PLUGIN(SoundDriver);
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DSoundPlay::DSoundPlay(DSoundConsumer* owner, LPGUID device)
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DSoundPlay::DSoundPlay(DSoundConsumer* owner, DWORD rate, LPGUID device)
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: Thread("DirectSound Play",High), Mutex(false,"DSoundPlay"),
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m_owner(0), m_device(device), m_ds(0), m_dsb(0),
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m_buffSize(0), m_start(0), m_total(0)
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m_owner(0), m_rate(rate), m_device(device), m_ds(0), m_dsb(0),
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m_buffSize(0), m_chunk(320), m_start(0), m_total(0)
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{
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if (owner && owner->ref())
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m_owner = owner;
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@ -205,18 +204,19 @@ bool DSoundPlay::init()
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// Set channel number depending data
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WORD nChannels = 1;
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DWORD nAvgBytesPerSec = 16000; // nSamplesPerSec * nBlockAlign.
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DWORD nAvgBytesPerSec = 2 * m_rate; // nSamplesPerSec * nBlockAlign.
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WORD nBlockAlign = 2; // nChannels * wBitsPerSample / 8
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if (m_owner && m_owner->m_stereo) {
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nChannels = 2;
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nAvgBytesPerSec = 32000;
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nAvgBytesPerSec = 4 * m_rate;
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nBlockAlign = 4;
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}
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m_chunk = nChannels * m_rate / 25; // 20ms of 2-byte samples
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WAVEFORMATEX fmt;
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fmt.wFormatTag = WAVE_FORMAT_PCM;
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fmt.nChannels = nChannels;
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fmt.nSamplesPerSec = 8000;
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fmt.nSamplesPerSec = m_rate;
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fmt.nAvgBytesPerSec = nAvgBytesPerSec;
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fmt.nBlockAlign = nBlockAlign;
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fmt.wBitsPerSample = 16;
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@ -230,7 +230,7 @@ bool DSoundPlay::init()
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else {
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bdesc.dwFlags |= DSBCAPS_GLOBALFOCUS;
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// we have to set format when creating secondary buffers
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bdesc.dwBufferBytes = s_bufsize;
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bdesc.dwBufferBytes = 4*m_chunk;
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bdesc.lpwfxFormat = &fmt;
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}
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if (FAILED(hr = m_ds->CreateSoundBuffer(&bdesc, &m_dsb, NULL)) || !m_dsb) {
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@ -248,10 +248,10 @@ bool DSoundPlay::init()
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}
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if ((fmt.wFormatTag != WAVE_FORMAT_PCM) ||
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(fmt.nChannels != nChannels) ||
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(fmt.nSamplesPerSec != 8000) ||
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(fmt.nSamplesPerSec != m_rate) ||
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(fmt.wBitsPerSample != 16)) {
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Debug(DebugGoOn,"DirectSound does not support 8000Hz 16bit %s PCM format, "
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"got fmt=%u, chans=%d samp=%d size=%u",nChannels == 1 ? "mono" : "stereo",
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Debug(DebugGoOn,"DirectSound does not support %dHz 16bit %s PCM format, "
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"got fmt=%u, chans=%d samp=%d size=%u",m_rate,nChannels == 1 ? "mono" : "stereo",
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fmt.wFormatTag,fmt.nChannels,fmt.nSamplesPerSec,fmt.wBitsPerSample);
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return false;
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}
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@ -283,12 +283,12 @@ void DSoundPlay::run()
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else
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return;
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DWORD writeOffs = 0;
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DWORD margin = s_chunk/4;
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DWORD margin = m_chunk/4;
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bool first = true;
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while (m_owner->refcount() > 1) {
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msleep(1,true);
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if (first) {
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if ((m_buf.length() < s_minsize) || !m_dsb)
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if ((m_buf.length() < 2*m_chunk) || !m_dsb)
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continue;
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first = false;
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m_dsb->GetCurrentPosition(NULL,&writeOffs);
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@ -316,10 +316,10 @@ void DSoundPlay::run()
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writeOffs,adjOffs,playPos,writePos);
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writeOffs = adjOffs;
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}
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bool hasData = (m_buf.length() >= s_chunk);
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bool hasData = (m_buf.length() >= m_chunk);
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if (!(adjust || hasData)) {
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// don't fill the buffer if we still have at least one chunk until underflow
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if ((m_buffSize + writeOffs - writePos) % m_buffSize >= s_chunk)
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if ((m_buffSize + writeOffs - writePos) % m_buffSize >= m_chunk)
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break;
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}
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void* buf = 0;
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@ -327,13 +327,13 @@ void DSoundPlay::run()
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DWORD len = 0;
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DWORD len2 = 0;
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// locking will prevent us to skip ahead and overwrite the play position
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HRESULT hr = m_dsb->Lock(writeOffs,s_chunk,&buf,&len,&buf2,&len2,0);
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HRESULT hr = m_dsb->Lock(writeOffs,m_chunk,&buf,&len,&buf2,&len2,0);
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if (FAILED(hr)) {
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writeOffs = 0;
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if ((hr == DSERR_BUFFERLOST) && SUCCEEDED(m_dsb->Restore())) {
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m_dsb->Play(0,0,DSBPLAY_LOOPING);
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m_dsb->GetCurrentPosition(NULL,&writeOffs);
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writeOffs = (s_chunk/4 + writeOffs) % m_buffSize;
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writeOffs = (margin + writeOffs) % m_buffSize;
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Debug(&__plugin,DebugAll,"DirectSound buffer lost and restored, playing at %u",
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writeOffs);
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}
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@ -356,15 +356,15 @@ void DSoundPlay::run()
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::memset(buf2,0,len2);
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}
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m_dsb->Unlock(buf,len,buf2,len2);
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m_total += s_chunk;
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m_buf.cut(-(int)s_chunk);
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m_total += m_chunk;
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m_buf.cut(-(int)m_chunk);
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unlock();
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#ifdef DEBUG
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if (!hasData)
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Debug(&__plugin,DebugInfo,"Underflow, filled %u bytes at %u, p=%u w=%u",
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s_chunk,writeOffs,playPos,writePos);
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m_chunk,writeOffs,playPos,writePos);
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#endif
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writeOffs += s_chunk;
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writeOffs += m_chunk;
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if (writeOffs >= m_buffSize)
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writeOffs -= m_buffSize;
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XDebug(&__plugin,DebugAll,"Locked %p,%d %p,%d",buf,len,buf2,len2);
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@ -420,7 +420,7 @@ void DSoundPlay::put(const DataBlock& data)
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if (!m_dsb)
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return;
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lock();
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if (m_buf.length() + data.length() <= s_maxsize)
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if (m_buf.length() + data.length() <= m_buffSize + m_chunk)
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m_buf += data;
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else
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Debug(&__plugin,DebugMild,"DSoundPlay skipped %u bytes, buffer is full",data.length());
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@ -428,9 +428,9 @@ void DSoundPlay::put(const DataBlock& data)
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}
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DSoundRec::DSoundRec(DSoundSource* owner, LPGUID device)
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DSoundRec::DSoundRec(DSoundSource* owner, DWORD rate, LPGUID device)
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: Thread("DirectSound Rec",High),
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m_owner(0), m_device(device), m_ds(0), m_dsb(0),
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m_owner(0), m_rate(rate), m_device(device), m_ds(0), m_dsb(0),
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m_buffSize(0), m_readPos(0), m_start(0), m_total(0), m_rshift(0)
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{
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if (owner && owner->ref())
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@ -464,8 +464,8 @@ bool DSoundRec::init()
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WAVEFORMATEX fmt;
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fmt.wFormatTag = WAVE_FORMAT_PCM;
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fmt.nChannels = 1;
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fmt.nSamplesPerSec = 8000;
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fmt.nAvgBytesPerSec = 16000;
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fmt.nSamplesPerSec = m_rate;
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fmt.nAvgBytesPerSec = 2 * m_rate;
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fmt.nBlockAlign = 2;
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fmt.wBitsPerSample = 16;
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fmt.cbSize = 0;
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@ -473,7 +473,7 @@ bool DSoundRec::init()
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ZeroMemory(&bdesc, sizeof(bdesc));
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bdesc.dwSize = sizeof(bdesc);
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bdesc.dwFlags = DSCBCAPS_WAVEMAPPED;
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bdesc.dwBufferBytes = s_bufsize;
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bdesc.dwBufferBytes = 4 * m_rate / 25;
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bdesc.lpwfxFormat = &fmt;
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if (FAILED(hr = m_ds->CreateCaptureBuffer(&bdesc, &m_dsb, NULL)) || !m_dsb) {
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Debug(DebugGoOn,"Could not create the DirectSoundCapture buffer, code 0x%X",hr);
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@ -485,10 +485,10 @@ bool DSoundRec::init()
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}
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if ((fmt.wFormatTag != WAVE_FORMAT_PCM) ||
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(fmt.nChannels != 1) ||
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(fmt.nSamplesPerSec != 8000) ||
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(fmt.nSamplesPerSec != m_rate) ||
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(fmt.wBitsPerSample != 16)) {
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Debug(DebugGoOn,"DirectSoundCapture does not support 8000Hz 16bit mono PCM format, "
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"got fmt=%u, chans=%d samp=%d size=%u",
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Debug(DebugGoOn,"DirectSoundCapture does not support %dHz 16bit mono PCM format, "
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"got fmt=%u, chans=%d samp=%d size=%u",m_rate,
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fmt.wFormatTag,fmt.nChannels,fmt.nSamplesPerSec,fmt.wBitsPerSample);
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return false;
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}
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@ -512,6 +512,7 @@ void DSoundRec::run()
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if (!init())
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return;
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Debug(&__plugin,DebugInfo,"DSoundRec is initialized and running");
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DWORD chunk = m_rate / 25; // 20ms of 2-byte samples
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m_start = Time::now();
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if (m_owner)
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m_owner->m_dsound = this;
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@ -526,13 +527,13 @@ void DSoundRec::run()
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if (pos < m_readPos)
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pos += m_buffSize;
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pos -= m_readPos;
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if (pos < s_chunk)
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if (pos < chunk)
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continue;
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void* buf = 0;
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void* buf2 = 0;
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DWORD len = 0;
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DWORD len2 = 0;
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if (FAILED(m_dsb->Lock(m_readPos,s_chunk,&buf,&len,&buf2,&len2,0)))
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if (FAILED(m_dsb->Lock(m_readPos,chunk,&buf,&len,&buf2,&len2,0)))
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continue;
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DataBlock data(0,len+len2);
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::memcpy(data.data(),buf,len);
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@ -592,10 +593,12 @@ bool DSoundRec::control(TelEngine::NamedList &msg)
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}
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DSoundSource::DSoundSource()
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DSoundSource::DSoundSource(int rate)
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: m_dsound(0)
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{
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DSoundRec* ds = new DSoundRec(this);
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if (rate != 8000)
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m_format << "/" << rate;
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DSoundRec* ds = new DSoundRec(this,rate);
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ds->startup();
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}
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@ -613,11 +616,13 @@ bool DSoundSource::control(NamedList& msg)
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}
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DSoundConsumer::DSoundConsumer(bool stereo)
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DSoundConsumer::DSoundConsumer(int rate, bool stereo)
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: DataConsumer(stereo ? "2*slin" : "slin"),
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m_dsound(0), m_stereo(stereo)
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{
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DSoundPlay* ds = new DSoundPlay(this);
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if (rate != 8000)
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m_format << "/" << rate;
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DSoundPlay* ds = new DSoundPlay(this,rate);
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ds->startup();
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}
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@ -644,15 +649,15 @@ bool DSoundConsumer::control(NamedList& msg)
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}
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DSoundChan::DSoundChan()
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DSoundChan::DSoundChan(int rate)
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: Channel(__plugin)
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{
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Debug(this,DebugAll,"DSoundChan::DSoundChan() [%p]",this);
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Debug(this,DebugAll,"DSoundChan::DSoundChan(%d) [%p]",rate,this);
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setConsumer(new DSoundConsumer);
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setConsumer(new DSoundConsumer(rate));
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getConsumer()->deref();
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Thread::msleep(50);
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setSource(new DSoundSource);
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setSource(new DSoundSource(rate));
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getSource()->deref();
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Thread::msleep(50);
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}
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@ -693,15 +698,17 @@ bool AttachHandler::received(Message &msg)
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return false;
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}
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int rate = msg.getIntValue("rate",s_rate);
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if (cons) {
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DSoundConsumer* c = new DSoundConsumer(msg.getBoolValue("stereo"));
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DSoundConsumer* c = new DSoundConsumer(rate,msg.getBoolValue("stereo"));
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dd->setConsumer(c);
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c->deref();
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Thread::msleep(50);
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}
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if (src) {
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DSoundSource* s = new DSoundSource;
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DSoundSource* s = new DSoundSource(rate);
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dd->setSource(s);
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s->deref();
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Thread::msleep(50);
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@ -717,7 +724,7 @@ bool SoundDriver::msgExecute(Message& msg, String& dest)
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{
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CallEndpoint* ch = static_cast<CallEndpoint*>(msg.userData());
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if (ch) {
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DSoundChan *ds = new DSoundChan;
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DSoundChan *ds = new DSoundChan(msg.getIntValue("rate",s_rate));
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if (ch->connect(ds,msg.getValue("reason"))) {
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msg.setParam("peerid",ds->id());
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ds->deref();
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@ -751,7 +758,7 @@ bool SoundDriver::msgExecute(Message& msg, String& dest)
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}
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m = "call.execute";
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m.addParam("callto",callto);
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DSoundChan *ds = new DSoundChan;
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DSoundChan *ds = new DSoundChan(msg.getIntValue("rate",8000));
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m.setParam("targetid",ds->id());
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m.userData(ds);
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if (Engine::dispatch(m)) {
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@ -783,33 +790,11 @@ void SoundDriver::initialize()
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Output("Initializing module DirectSound");
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setup(0,true); // no need to install notifications
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Driver::initialize();
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Configuration cfg(Engine::configFile("dsoundchan"));
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s_rate = cfg.getIntValue("general","rate",8000);
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// prefer primary buffer as we try to retain control of audio board
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s_primary = cfg.getBoolValue("general","primary",true);
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if (!m_handler) {
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Configuration cfg(Engine::configFile("dsoundchan"));
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s_chunk = cfg.getIntValue("general","chunk",CHUNK_SIZE);
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// make sure the chunk is even sized and has some decent limits (20-50 ms)
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s_chunk &= ~1;
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if (s_chunk < 320)
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s_chunk = 320;
|
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if (s_chunk > 800)
|
||||
s_chunk = 800;
|
||||
s_minsize = cfg.getIntValue("general","minsize",2*s_chunk);
|
||||
s_bufsize = cfg.getIntValue("general","bufsize",4*s_chunk);
|
||||
s_maxsize = cfg.getIntValue("general","maxsize",5*s_chunk);
|
||||
// the buffer MUST hold at least one chunk and about 15ms of audio - we allow 30
|
||||
if (s_bufsize < s_chunk + 480)
|
||||
s_bufsize = s_chunk + 480;
|
||||
// also keep it under 2s and even sized
|
||||
if (s_bufsize > 32000)
|
||||
s_bufsize = 32000;
|
||||
s_bufsize &= ~1;
|
||||
// make sure playback can ever start
|
||||
if (s_minsize > s_bufsize - s_chunk)
|
||||
s_minsize = s_bufsize - s_chunk;
|
||||
// and that we don't do stupid drops
|
||||
if (s_maxsize < s_bufsize + s_chunk)
|
||||
s_maxsize = s_bufsize + s_chunk;
|
||||
// prefer primary buffer as we try to retain control of audio board
|
||||
s_primary = cfg.getBoolValue("general","primary",true);
|
||||
m_handler = new AttachHandler;
|
||||
Engine::install(m_handler);
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in New Issue