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windows-nt/Source/XPSP1/NT/net/tapi/skywalker/confmsp/confaud.cpp

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/*++
Copyright (c) 1997 Microsoft Corporation
Module Name:
confaud.cpp
Abstract:
This module contains implementation of the audio send and receive
stream implementations.
Author:
Mu Han (muhan) 15-September-1999
--*/
#include "stdafx.h"
#include "common.h"
#include <initguid.h>
#include <amrtpnet.h> // rtp guids
#include <amrtpdmx.h> // demux guid
#include <amrtpuid.h> // AMRTP media types
#include <amrtpss.h> // for silence suppression filter
#include <irtprph.h> // for IRTPRPHFilter
#include <irtpsph.h> // for IRTPSPHFilter
#include <mixflter.h> // audio mixer
#include <g711uids.h> // for G711 codec CLSID
#include <g723uids.h> // for G723 codec CLSID
//#define DISABLE_MIXER 1
/////////////////////////////////////////////////////////////////////////////
//
// CStreamAudioRecv
//
/////////////////////////////////////////////////////////////////////////////
CStreamAudioRecv::CStreamAudioRecv()
: CIPConfMSPStream(),
m_pWaveFormatEx(NULL),
m_dwSizeWaveFormatEx(0),
m_fUseACM(FALSE),
m_dwMaxPacketSize(0),
m_dwAudioSampleRate(0)
{
m_szName = L"AudioRecv";
}
void CStreamAudioRecv::FinalRelease()
{
CIPConfMSPStream::FinalRelease();
if (m_pWaveFormatEx)
{
free(m_pWaveFormatEx);
}
}
HRESULT CStreamAudioRecv::Configure(
IN STREAMSETTINGS &StreamSettings
)
/*++
Routine Description:
Configure the settings of this stream.
Arguments:
StreamSettings - The setting structure got from the SDP blob.
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioRecv Configure entered."));
CLock lock(m_lock);
_ASSERTE(m_fIsConfigured == FALSE);
switch (StreamSettings.dwPayloadType)
{
case PAYLOAD_G711U:
// The mixer can convert them, no codec needed.
m_pClsidCodecFilter = &GUID_NULL;
m_pRPHInputMinorType = &MEDIASUBTYPE_RTP_Payload_G711U;
m_pClsidPHFilter = &CLSID_INTEL_RPHAUD;
m_dwMaxPacketSize = g_dwMaxG711PacketSize;
m_dwAudioSampleRate = g_dwG711AudioSampleRate;
break;
case PAYLOAD_G711A:
m_pClsidCodecFilter = &CLSID_G711Codec;
m_pRPHInputMinorType = &MEDIASUBTYPE_RTP_Payload_G711A;
m_pClsidPHFilter = &CLSID_INTEL_RPHAUD;
m_dwMaxPacketSize = g_dwMaxG711PacketSize;
m_dwAudioSampleRate = g_dwG711AudioSampleRate;
break;
case PAYLOAD_GSM:
m_fUseACM = TRUE;
m_pClsidCodecFilter = &CLSID_ACMWrapper;
m_pRPHInputMinorType = &MEDIASUBTYPE_RTP_Payload_ANY;
m_pClsidPHFilter = &CLSID_INTEL_RPHGENA;
m_dwMaxPacketSize = g_dwMaxGSMPacketSize;
m_dwAudioSampleRate = g_dwGSMAudioSampleRate;
{
GSM610WAVEFORMAT * pWaveFormat =
(GSM610WAVEFORMAT *)malloc(sizeof GSM610WAVEFORMAT);
if (pWaveFormat == NULL)
{
return E_OUTOFMEMORY;
}
pWaveFormat->wfx.wFormatTag = WAVE_FORMAT_GSM610;
pWaveFormat->wfx.wBitsPerSample = 0;
pWaveFormat->wfx.nChannels = g_wAudioChannels;
pWaveFormat->wfx.nSamplesPerSec = m_dwAudioSampleRate;
pWaveFormat->wfx.nAvgBytesPerSec = g_dwGSMBytesPerSecond;
pWaveFormat->wfx.nBlockAlign = g_wGSMBlockAlignment;
pWaveFormat->wfx.cbSize =
sizeof GSM610WAVEFORMAT - sizeof WAVEFORMATEX;
pWaveFormat->wSamplesPerBlock = g_wGSMSamplesPerBlock;
m_pWaveFormatEx = (BYTE *)pWaveFormat;
m_dwSizeWaveFormatEx = sizeof GSM610WAVEFORMAT;
}
break;
// This is a test of the MSAudio wideband codec
case PAYLOAD_MSAUDIO:
m_fUseACM = TRUE;
m_pClsidCodecFilter = &CLSID_ACMWrapper;
m_pRPHInputMinorType = &MEDIASUBTYPE_RTP_Payload_ANY;
m_pClsidPHFilter = &CLSID_INTEL_RPHGENA;
m_dwMaxPacketSize = g_dwMaxMSAudioPacketSize;
m_dwAudioSampleRate = g_dwMSAudioSampleRate;
{
MSAUDIO1WAVEFORMAT * pWaveFormat =
(MSAUDIO1WAVEFORMAT *)malloc(sizeof MSAUDIO1WAVEFORMAT);
if (pWaveFormat == NULL)
{
return E_OUTOFMEMORY;
}
pWaveFormat->wfx.wFormatTag = WAVE_FORMAT_MSAUDIO1;
pWaveFormat->wfx.wBitsPerSample = MSAUDIO1_BITS_PER_SAMPLE;
pWaveFormat->wfx.nChannels = MSAUDIO1_MAX_CHANNELS;
pWaveFormat->wfx.nSamplesPerSec = m_dwAudioSampleRate;
pWaveFormat->wfx.nAvgBytesPerSec = g_dwMSAudioBytesPerSecond;
pWaveFormat->wfx.nBlockAlign = g_wMSAudioBlockAlignment;
pWaveFormat->wfx.cbSize =
sizeof MSAUDIO1WAVEFORMAT - sizeof WAVEFORMATEX;
pWaveFormat->wSamplesPerBlock = g_wMSAudioSamplesPerBlock;
m_pWaveFormatEx = (BYTE *)pWaveFormat;
m_dwSizeWaveFormatEx = sizeof MSAUDIO1WAVEFORMAT;
}
break;
#ifdef DVI
case PAYLOAD_DVI4_8:
m_fUseACM = TRUE;
m_pClsidCodecFilter = &CLSID_ACMWrapper;
m_pRPHInputMinorType = &MEDIASUBTYPE_RTP_Payload_ANY;
m_pClsidPHFilter = &CLSID_INTEL_RPHGENA;
m_dwMaxPacketSize = g_dwMaxDVI4PacketSize;
m_dwAudioSampleRate = g_dwDVI4AudioSampleRate;
{
IMAADPCMWAVEFORMAT * pWaveFormat =
(IMAADPCMWAVEFORMAT *)malloc(sizeof IMAADPCMWAVEFORMAT);
if (pWaveFormat == NULL)
{
return E_OUTOFMEMORY;
}
pWaveFormat->wfx.wFormatTag = WAVE_FORMAT_IMA_ADPCM;
pWaveFormat->wfx.wBitsPerSample = g_wDVI4BitsPerSample;
pWaveFormat->wfx.nChannels = g_wAudioChannels;
pWaveFormat->wfx.nSamplesPerSec = m_dwAudioSampleRate;
pWaveFormat->wfx.nAvgBytesPerSec = g_dwDVI4BytesPerSecond;
pWaveFormat->wfx.nBlockAlign = g_wDVI4BlockAlignment;
pWaveFormat->wfx.cbSize =
sizeof IMAADPCMWAVEFORMAT - sizeof WAVEFORMATEX;
pWaveFormat->wSamplesPerBlock = g_wDVI4SamplesPerBlock;
m_pWaveFormatEx = (BYTE *)pWaveFormat;
m_dwSizeWaveFormatEx = sizeof IMAADPCMWAVEFORMAT;
}
break;
#endif
default:
LOG((MSP_ERROR, "unknown payload type, %x", StreamSettings.dwPayloadType));
return E_FAIL;
}
m_Settings = StreamSettings;
m_fIsConfigured = TRUE;
return InternalConfigure();
}
HRESULT CStreamAudioRecv::ConfigureRTPFilter(
IN IBaseFilter * pIBaseFilter
)
/*++
Routine Description:
Configure the source RTP filter. Including set the address, port, TTL,
QOS, thread priority, clcokrate, etc.
Arguments:
pIBaseFilter - The source RTP Filter.
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioRecv ConfigureRTPFilter"));
HRESULT hr;
// Get the IRTPStream interface pointer on the filter.
CComQIPtr<IRTPStream, &IID_IRTPStream> pIRTPStream(pIBaseFilter);
if (pIRTPStream == NULL)
{
LOG((MSP_ERROR, "get RTP Stream interface"));
return E_NOINTERFACE;
}
LOG((MSP_INFO, "set remote Address:%x, port:%d",
m_Settings.dwIPRemote, m_Settings.wRTPPortRemote));
// Set the address and port used in the filter.
if (FAILED(hr = pIRTPStream->SetAddress(
htons(m_Settings.wRTPPortRemote), // local port to listen on.
0, // remote port.
htonl(m_Settings.dwIPRemote) // remote address.
)))
{
LOG((MSP_ERROR, "set remote Address, hr:%x", hr));
return hr;
}
// Set the TTL used in the filter.
if (FAILED(hr = pIRTPStream->SetMulticastScope(m_Settings.dwTTL)))
{
LOG((MSP_ERROR, "set TTL. %x", hr));
return hr;
}
if (m_Settings.dwIPLocal != INADDR_ANY)
{
// set the local interface that the socket should bind to
LOG((MSP_INFO, "set locol Address:%x", m_Settings.dwIPLocal));
if (FAILED(hr = pIRTPStream->SelectLocalIPAddress(
htonl(m_Settings.dwIPLocal)
)))
{
LOG((MSP_ERROR, "set locol Address, hr:%x", hr));
return hr;
}
}
// Set the priority of the session
if (FAILED(hr = pIRTPStream->SetSessionClassPriority(
RTP_CLASS_AUDIO,
g_dwAudioThreadPriority
)))
{
LOG((MSP_WARN, "set session class and priority. %x", hr));
}
// Set the sample rate of the session
LOG((MSP_INFO, "setting session sample rate to %d", m_dwAudioSampleRate));
if (FAILED(hr = pIRTPStream->SetDataClock(m_dwAudioSampleRate)))
{
LOG((MSP_WARN, "set session sample rate. %x", hr));
}
// Enable the RTCP events
if (FAILED(hr = ::EnableRTCPEvents(pIBaseFilter)))
{
LOG((MSP_WARN, "can not enable RTCP events %x", hr));
}
DWORD dwLoopback = 0;
if (TRUE == ::GetRegValue(gszMSPLoopback, &dwLoopback)
&& dwLoopback != 0)
{
// Loopback is required.
if (FAILED(hr = ::SetLoopbackOption(pIBaseFilter, dwLoopback)))
{
LOG((MSP_ERROR, "set loopback option. %x", hr));
return hr;
}
}
if (m_Settings.dwQOSLevel != QSL_BEST_EFFORT)
{
if (FAILED(hr = ::SetQOSOption(
pIBaseFilter,
m_Settings.dwPayloadType, // payload
-1, // use the default bitrate
(m_Settings.dwQOSLevel == QSL_NEEDED), // fail if no qos.
TRUE, // receive stream.
g_wAudioDemuxPins // number of streams reserved.
)))
{
LOG((MSP_ERROR, "set QOS option. %x", hr));
return hr;
}
}
SetLocalInfoOnRTPFilter(pIBaseFilter);
return S_OK;
}
HRESULT CStreamAudioRecv::ConnectTerminal(
IN ITTerminal * pITTerminal
)
/*++
Routine Description:
connect the mixer to the audio render terminal.
Arguments:
pITTerminal - The terminal to be connected.
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioRecv.ConnectTerminal, pITTerminal %p", pITTerminal));
HRESULT hr;
// if our filters have not been contructed, do it now.
if (m_pEdgeFilter == NULL)
{
hr = SetUpInternalFilters();
if (FAILED(hr))
{
LOG((MSP_ERROR, "Set up internal filter failed, %x", hr));
CleanUpFilters();
return hr;
}
}
// get the terminal control interface.
CComQIPtr<ITTerminalControl, &IID_ITTerminalControl>
pTerminal(pITTerminal);
if (pTerminal == NULL)
{
LOG((MSP_ERROR, "can't get Terminal Control interface"));
SendStreamEvent(
CALL_TERMINAL_FAIL,
CALL_CAUSE_BAD_DEVICE,
E_NOINTERFACE,
pITTerminal
);
return E_NOINTERFACE;
}
const DWORD MAXPINS = 8;
DWORD dwNumPins = MAXPINS;
IPin * Pins[MAXPINS];
// Get the pins.
hr = pTerminal->ConnectTerminal(
m_pIGraphBuilder, 0, &dwNumPins, Pins
);
if (FAILED(hr))
{
LOG((MSP_ERROR, "can't connect to terminal, %x", hr));
SendStreamEvent(
CALL_TERMINAL_FAIL,
CALL_CAUSE_BAD_DEVICE,
hr,
pITTerminal
);
return hr;
}
// the pin count should never be 0.
if (dwNumPins == 0)
{
LOG((MSP_ERROR, "terminal has no pins."));
SendStreamEvent(
CALL_TERMINAL_FAIL,
CALL_CAUSE_BAD_DEVICE,
hr,
pITTerminal
);
pTerminal->DisconnectTerminal(m_pIGraphBuilder, 0);
return E_UNEXPECTED;
}
// Connect the mixer filter to the audio render terminal.
hr = ::ConnectFilters(
m_pIGraphBuilder,
(IBaseFilter *)m_pEdgeFilter,
(IPin *)Pins[0]
);
// release the refcounts on the pins.
for (DWORD i = 0; i < dwNumPins; i ++)
{
Pins[i]->Release();
}
if (FAILED(hr))
{
LOG((MSP_ERROR, "connect to the mixer filter. %x", hr));
pTerminal->DisconnectTerminal(m_pIGraphBuilder, 0);
return hr;
}
//
// Now we are actually connected. Update our state and perform postconnection
// (ignore postconnection error code).
//
pTerminal->CompleteConnectTerminal();
return hr;
}
HRESULT CStreamAudioRecv::SetUpInternalFilters()
/*++
Routine Description:
set up the filters used in the stream.
RTP->Demux->RPH(->DECODER)->Mixer
Arguments:
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioRecv.SetUpInternalFilters"));
CComPtr<IBaseFilter> pSourceFilter;
HRESULT hr;
// create and add the source fitler.
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
CLSID_RTPSourceFilter,
L"RtpSource",
&pSourceFilter)))
{
LOG((MSP_ERROR, "adding source filter. %x", hr));
return hr;
}
if (FAILED(hr = ConfigureRTPFilter(pSourceFilter)))
{
LOG((MSP_ERROR, "configure RTP source filter. %x", hr));
return hr;
}
CComPtr<IBaseFilter> pDemuxFilter;
// create and add the demux fitler.
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
CLSID_IntelRTPDemux,
L"RtpDemux",
&pDemuxFilter)))
{
LOG((MSP_ERROR, "adding demux filter. %x", hr));
return hr;
}
// Connect the source filter and the demux filter.
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
(IBaseFilter *)pSourceFilter,
(IBaseFilter *)pDemuxFilter)))
{
LOG((MSP_ERROR, "connect source filter and demux filter. %x", hr));
return hr;
}
// Get the IRTPDemuxFilter interface used in configuring the demux filter.
CComQIPtr<IRTPDemuxFilter, &IID_IRTPDemuxFilter> pIRTPDemux(pDemuxFilter);
if (pIRTPDemux == NULL)
{
LOG((MSP_ERROR, "get RTP Demux interface"));
return E_NOINTERFACE;
}
// Set the number of output pins on the demux filter based on the number
// of channels needed.
if (FAILED(hr = pIRTPDemux->SetPinCount(
g_wAudioDemuxPins
)))
{
LOG((MSP_ERROR, "set demux output pin count"));
return hr;
}
LOG((MSP_INFO,
"set demux output pin count to %d",
g_wAudioDemuxPins
));
// Get the enumerator of pins on the demux filter.
CComPtr<IEnumPins> pIEnumPins;
if (FAILED(hr = pDemuxFilter->EnumPins(&pIEnumPins)))
{
LOG((MSP_ERROR, "enumerate pins on the demux filter %x", hr));
return hr;
}
#ifndef DISABLE_MIXER
// Create and add the mixer filter into the filtergraph.
CComPtr<IBaseFilter> pIMixerFilter;
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
CLSID_AudioMixFilter,
L"Mixer",
&pIMixerFilter
)))
{
LOG((MSP_ERROR, "add Mixer filter. %x", hr));
return hr;
}
LOG((MSP_INFO, "Added Mixer filter"));
// currently we support only one format for each stream.
#endif
#ifndef DISABLE_MIXER
for (DWORD i = 0; i < g_wAudioDemuxPins; i++)
#else
CComPtr<IBaseFilter> pIFilter;
for (DWORD i = 0; i < 1; i++)
#endif
{
// Find the next output pin on the demux fitler.
CComPtr<IPin> pIPinOutput;
for (;;)
{
if ((hr = pIEnumPins->Next(1, &pIPinOutput, NULL)) != S_OK)
{
LOG((MSP_ERROR, "find output pin on demux."));
break;
}
PIN_DIRECTION dir;
if (FAILED(hr = pIPinOutput->QueryDirection(&dir)))
{
LOG((MSP_ERROR, "query pin direction. %x", hr));
pIPinOutput.Release();
break;
}
if (PINDIR_OUTPUT == dir)
{
break;
}
pIPinOutput.Release();
}
if (hr != S_OK)
{
// There is no more output pin on the demux filter.
// This should never happen.
hr = E_UNEXPECTED;
break;
}
// Set the media type on this output pin.
if (FAILED(hr = pIRTPDemux->SetPinTypeInfo(
pIPinOutput,
(BYTE)m_Settings.dwPayloadType,
*m_pRPHInputMinorType
)))
{
LOG((MSP_ERROR, "set demux output pin type info"));
break;
}
LOG((MSP_INFO,
"set demux output pin payload type to %d",
m_Settings.dwPayloadType
));
// Create and add the payload handler into the filtergraph.
CComPtr<IBaseFilter> pIRPHFilter;
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
*m_pClsidPHFilter,
L"RPH",
&pIRPHFilter
)))
{
LOG((MSP_ERROR, "add RPH filter. %x", hr));
break;
}
// Connect the payload handler to the output pin on the demux.
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
(IPin *)pIPinOutput,
(IBaseFilter *)pIRPHFilter
)))
{
LOG((MSP_ERROR, "connect demux and RPH filter. %x", hr));
break;
}
// Get the IRTPRPHFilter interface.
CComQIPtr<IRTPRPHFilter, &IID_IRTPRPHFilter>pIRTPRPHFilter(pIRPHFilter);
if (pIRTPRPHFilter == NULL)
{
LOG((MSP_ERROR, "get IRTPRPHFilter interface"));
break;
}
// set the media buffer size so that the receive buffers are of the
// right size. Note, G723 needs smaller buffers than G711.
if (FAILED(hr = pIRTPRPHFilter->SetMediaBufferSize(
m_dwMaxPacketSize
)))
{
LOG((MSP_ERROR, "Set media buffer size. %x", hr));
break;
}
LOG((MSP_INFO, "Set RPH media buffer size to %d", m_dwMaxPacketSize));
if (m_fUseACM)
{
// We are using the ACM codec, so we have to set the media types
AM_MEDIA_TYPE mt;
mt.majortype = MEDIATYPE_Audio;
mt.subtype = MEDIASUBTYPE_NULL;
mt.bFixedSizeSamples = TRUE;
mt.bTemporalCompression = FALSE;
mt.lSampleSize = 0;
mt.formattype = FORMAT_WaveFormatEx;
mt.pUnk = NULL;
mt.cbFormat = m_dwSizeWaveFormatEx;
mt.pbFormat = m_pWaveFormatEx;
if (FAILED(hr = pIRTPRPHFilter->SetOutputPinMediaType(&mt)))
{
LOG((MSP_ERROR, "Set RPHGENA output pin media type. %x", hr));
return FALSE;
}
if (FAILED(hr = pIRTPRPHFilter->OverridePayloadType(
(BYTE)m_Settings.dwPayloadType
)))
{
LOG((MSP_ERROR, "Set RPHGENA output pin media type. %x", hr));
return FALSE;
}
}
#ifndef DISABLE_MIXER
CComPtr<IBaseFilter> pIFilter;
#endif
// connect the codec filter if it is needed.
if (*m_pClsidCodecFilter != GUID_NULL)
{
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
*m_pClsidCodecFilter,
L"codec",
&pIFilter
)))
{
LOG((MSP_ERROR, "add Codec filter. %x", hr));
break;
}
// Connect the payload handler to the output pin on the demux.
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
(IBaseFilter *)pIRPHFilter,
(IBaseFilter *)pIFilter
)))
{
LOG((MSP_ERROR, "connect RPH filter and codec. %x", hr));
break;
}
}
else
{
pIFilter = pIRPHFilter;
}
#ifndef DISABLE_MIXER
// Connect the payload handler or the codec filter to the mixer filter.
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
(IBaseFilter *)pIFilter,
(IBaseFilter *)pIMixerFilter
)))
{
LOG((MSP_ERROR, "connect to the mixer filter. %x", hr));
break;
}
#endif
}
if (SUCCEEDED(hr))
{
// keep a reference to the last filter so that the change of terminal
// will not require a recreating of all the filters.
#ifndef DISABLE_MIXER
m_pEdgeFilter = pIMixerFilter;
#else
m_pEdgeFilter = pIFilter;
#endif
m_pEdgeFilter->AddRef();
// Get the IRTPParticipant interface pointer on the RTP filter.
CComQIPtr<IRTPParticipant,
&IID_IRTPParticipant> pIRTPParticipant(pSourceFilter);
if (pIRTPParticipant == NULL)
{
LOG((MSP_WARN, "can't get RTP participant interface"));
}
else
{
m_pRTPFilter = pIRTPParticipant;
m_pRTPFilter->AddRef();
}
}
return hr;
}
HRESULT CStreamAudioRecv::SetUpFilters()
/*++
Routine Description:
Insert filters into the graph and connect to the terminals.
Arguments:
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioRecv SetupFilters entered."));
HRESULT hr;
// we only support one terminal for this stream.
if (m_Terminals.GetSize() != 1)
{
return E_UNEXPECTED;
}
// Connect the mixer to the terminal.
if (FAILED(hr = ConnectTerminal(
m_Terminals[0]
)))
{
LOG((MSP_ERROR, "connect the mixer filter to terminal. %x", hr));
return hr;
}
return hr;
}
HRESULT CStreamAudioRecv::ProcessSSRCMappedEvent(
IN DWORD dwSSRC
)
/*++
Routine Description:
a SSRC is active, file a participant active event.
Arguments:
dwSSRC - the SSRC of the participant.
Return Value:
S_OK,
E_UNEXPECTED
--*/
{
LOG((MSP_TRACE, "%ls Processes pin mapped event, pIPin: %p", m_szName, dwSSRC));
CLock lock(m_lock);
ITParticipant * pITParticipant = NULL;
// find the SSRC in our participant list.
for (int i = 0; i < m_Participants.GetSize(); i ++)
{
if (((CParticipant *)m_Participants[i])->
HasSSRC((ITStream *)this, dwSSRC))
{
pITParticipant = m_Participants[i];
}
}
// if the participant is not there yet, put the event in a queue and it
// will be fired when we have the CName fo the participant.
if (!pITParticipant)
{
LOG((MSP_INFO, "can't find a participant that has SSRC %x", dwSSRC));
m_PendingSSRCs.Add(dwSSRC);
LOG((MSP_INFO, "added the event to pending list, new list size:%d",
m_PendingSSRCs.GetSize()));
return S_OK;
}
((CIPConfMSPCall *)m_pMSPCall)->SendParticipantEvent(
PE_PARTICIPANT_ACTIVE,
pITParticipant
);
return S_OK;
}
HRESULT CStreamAudioRecv::NewParticipantPostProcess(
IN DWORD dwSSRC,
IN ITParticipant *pITParticipant
)
/*++
Routine Description:
A mapped event happended when we didn't have the participant's name so
it was queued in a list. Now that we have a new participant, let's check
if this is the same participant. If it is, we complete the mapped event
by sending the app an notification.
Arguments:
dwSSRC - the SSRC of the participant.
pITParticipant - the participant object.
Return Value:
S_OK,
E_UNEXPECTED
--*/
{
LOG((MSP_TRACE, "%ls Check pending mapped event, dwSSRC: %x", m_szName, dwSSRC));
// look at the pending SSRC list and find out if this report
// fits in the list.
int i = m_PendingSSRCs.Find(dwSSRC);
if (i < 0)
{
// the SSRC is not in the list of pending PinMappedEvents.
LOG((MSP_TRACE, "the SSRC %x is not in the pending list", dwSSRC));
return S_OK;
}
// get rid of the peding SSRC.
m_PendingSSRCs.RemoveAt(i);
// complete the event.
((CIPConfMSPCall *)m_pMSPCall)->SendParticipantEvent(
PE_PARTICIPANT_ACTIVE,
pITParticipant
);
return S_OK;
}
HRESULT CStreamAudioRecv::ProcessSSRCUnmapEvent(
IN DWORD dwSSRC
)
/*++
Routine Description:
A SSRC just got unmapped by the demux. Notify the app that a participant
becomes inactive.
Arguments:
dwSSRC - the SSRC of the participant.
Return Value:
S_OK,
E_UNEXPECTED
--*/
{
LOG((MSP_TRACE, "%ls Processes SSRC unmapped event, pIPin: %p", m_szName, dwSSRC));
CLock lock(m_lock);
// look at the pending SSRC list and find out if it is in the pending list.
int i = m_PendingSSRCs.Find(dwSSRC);
// if the SSRC is in the pending list, just remove it.
if (i >= 0)
{
m_PendingSSRCs.RemoveAt(i);
return S_OK;
}
ITParticipant *pITParticipant = NULL;
// find the SSRC in our participant list.
for (i = 0; i < m_Participants.GetSize(); i ++)
{
if (((CParticipant *)m_Participants[i])->
HasSSRC((ITStream *)this, dwSSRC))
{
pITParticipant = m_Participants[i];
}
}
if (pITParticipant)
{
// fire an event to tell the app that the participant is inactive.
((CIPConfMSPCall *)m_pMSPCall)->SendParticipantEvent(
PE_PARTICIPANT_INACTIVE,
pITParticipant
);
}
return S_OK;
}
HRESULT CStreamAudioRecv::ProcessParticipantLeave(
IN DWORD dwSSRC
)
/*++
Routine Description:
When participant left the session, remove the stream from the participant
object's list of streams. If all streams are removed, remove the
participant from the call object's list too.
Arguments:
dwSSRC - the SSRC of the participant left.
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "%ls ProcessParticipantLeave, SSRC: %x", m_szName, dwSSRC));
CLock lock(m_lock);
// look at the pending SSRC list and find out if it is in the pending list.
int i = m_PendingSSRCs.Find(dwSSRC);
// if the SSRC is in the pending list, remove it.
if (i >= 0)
{
m_PendingSSRCs.RemoveAt(i);
}
CParticipant *pParticipant;
BOOL fLast = FALSE;
HRESULT hr = E_FAIL;
// first try to find the SSRC in our participant list.
for (i = 0; i < m_Participants.GetSize(); i ++)
{
pParticipant = (CParticipant *)m_Participants[i];
hr = pParticipant->RemoveStream(
(ITStream *)this,
dwSSRC,
&fLast
);
if (SUCCEEDED(hr))
{
break;
}
}
// if the participant is not found
if (FAILED(hr))
{
LOG((MSP_WARN, "%ws, can't find the SSRC %x", m_szName, dwSSRC));
return hr;
}
ITParticipant *pITParticipant = m_Participants[i];
// fire an event to tell the app that the participant is in active.
((CIPConfMSPCall *)m_pMSPCall)->SendParticipantEvent(
PE_PARTICIPANT_INACTIVE,
pITParticipant
);
m_Participants.RemoveAt(i);
// if this stream is the last stream that the participant is on,
// tell the call object to remove it from its list.
if (fLast)
{
((CIPConfMSPCall *)m_pMSPCall)->ParticipantLeft(pITParticipant);
}
pITParticipant->Release();
return S_OK;
}
HRESULT CStreamAudioRecv::ProcessGraphEvent(
IN long lEventCode,
IN long lParam1,
IN long lParam2
)
{
LOG((MSP_TRACE, "%ws ProcessGraphEvent %d", m_szName, lEventCode));
switch (lEventCode)
{
case RTPDMX_EVENTBASE + RTPDEMUX_SSRC_MAPPED:
LOG((MSP_INFO, "handling SSRC mapped event, SSRC%x", lParam1));
ProcessSSRCMappedEvent((DWORD)lParam1);
break;
case RTPDMX_EVENTBASE + RTPDEMUX_SSRC_UNMAPPED:
LOG((MSP_INFO, "handling SSRC unmap event, SSRC%x", lParam1));
ProcessSSRCUnmapEvent((DWORD)lParam1);
break;
default:
return CIPConfMSPStream::ProcessGraphEvent(
lEventCode, lParam1, lParam2
);
}
return S_OK;
}
/////////////////////////////////////////////////////////////////////////////
//
// CStreamAudioSend
//
/////////////////////////////////////////////////////////////////////////////
CStreamAudioSend::CStreamAudioSend()
: CIPConfMSPStream(),
m_iACMID(0),
m_dwMSPerPacket(0),
m_fUseACM(FALSE),
m_dwMaxPacketSize(0),
m_dwAudioSampleRate(0)
{
m_szName = L"AudioSend";
}
HRESULT CStreamAudioSend::Configure(
IN STREAMSETTINGS &StreamSettings
)
/*++
Routine Description:
Configure the settings of this stream.
Arguments:
StreamSettings - The setting structure got from the SDP blob.
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioSend Configure entered."));
CLock lock(m_lock);
_ASSERTE(m_fIsConfigured == FALSE);
switch (StreamSettings.dwPayloadType)
{
case PAYLOAD_G711U:
case PAYLOAD_G711A:
m_pClsidCodecFilter = &CLSID_G711Codec;
m_pClsidPHFilter = &CLSID_INTEL_SPHAUD;
m_dwMSPerPacket = g_dwG711MSPerPacket;
m_dwMaxPacketSize = g_dwG711BytesPerPacket + g_dwRTPHeaderSize;
m_dwAudioSampleRate = g_dwG711AudioSampleRate;
if (StreamSettings.dwMSPerPacket != 0)
{
m_dwMSPerPacket = StreamSettings.dwMSPerPacket;
m_dwMaxPacketSize = m_dwMSPerPacket * m_dwAudioSampleRate / 1000
+ g_dwRTPHeaderSize;
}
break;
#ifdef DVI
case PAYLOAD_DVI4_8:
m_fUseACM = TRUE;
m_iACMID = WAVE_FORMAT_IMA_ADPCM;
m_pClsidCodecFilter = &CLSID_ACMWrapper;
m_pRPHInputMinorType = &MEDIASUBTYPE_RTP_Payload_ANY;
m_pClsidPHFilter = &CLSID_INTEL_SPHGENA;
m_dwMSPerPacket = g_dwDVI4MSPerPacket;
m_dwMaxPacketSize = g_dwDVI4BytesPerPacket + g_dwRTPHeaderSize;
m_dwAudioSampleRate = g_dwDVI4AudioSampleRate;
break;
#endif
case PAYLOAD_GSM:
m_fUseACM = TRUE;
m_iACMID = WAVE_FORMAT_GSM610;
m_pClsidCodecFilter = &CLSID_ACMWrapper;
m_pRPHInputMinorType = &MEDIASUBTYPE_RTP_Payload_ANY;
m_pClsidPHFilter = &CLSID_INTEL_SPHGENA;
m_dwMSPerPacket = g_dwGSMMSPerPacket;
m_dwMaxPacketSize = g_dwGSMBytesPerPacket + g_dwRTPHeaderSize;
m_dwAudioSampleRate = g_dwGSMAudioSampleRate;
break;
case PAYLOAD_MSAUDIO:
m_fUseACM = TRUE;
m_iACMID = WAVE_FORMAT_MSAUDIO1;
m_pClsidCodecFilter = &CLSID_ACMWrapper;
m_pRPHInputMinorType = &MEDIASUBTYPE_RTP_Payload_ANY;
m_pClsidPHFilter = &CLSID_INTEL_SPHGENA;
m_dwMSPerPacket = g_dwMSAudioMSPerPacket;
m_dwMaxPacketSize = g_dwMaxMSAudioPacketSize;
m_dwAudioSampleRate = g_dwMSAudioSampleRate;
break;
default:
LOG((MSP_ERROR,
"unknow payload type, %x", StreamSettings.dwPayloadType));
return E_FAIL;
}
m_Settings = StreamSettings;
m_fIsConfigured = TRUE;
return InternalConfigure();
}
HRESULT CStreamAudioSend::ConfigureAudioCaptureTerminal(
IN ITTerminalControl * pTerminal,
OUT IPin ** ppIPin
)
/*++
Routine Description:
Configure the audio capture terminal. This function gets a output pin from
the capture terminal and the configure the audio format and media type.
Arguments:
pTerminal - An audio capture terminal.
ppIPin - the address to hold the returned pointer to IPin interface.
Return Value:
HRESULT
--*/
{
LOG((MSP_TRACE, "AudioSend configure audio capture terminal."));
const DWORD MAXPINS = 8;
DWORD dwNumPins = MAXPINS;
IPin * Pins[MAXPINS];
// Get the pins from the first terminal because we only use on terminal
// on this stream.
HRESULT hr = pTerminal->ConnectTerminal(
m_pIGraphBuilder, 0, &dwNumPins, Pins
);
if (FAILED(hr))
{
LOG((MSP_ERROR, "can't connect to terminal, %x", hr));
return hr;
}
// The number of pins should never be 0.
if (dwNumPins == 0)
{
LOG((MSP_ERROR, "terminal has no pins."));
return E_UNEXPECTED;
}
// Save the first pin and release the others.
CComPtr <IPin> pIPin = Pins[0];
for (DWORD i = 0; i < dwNumPins; i ++)
{
Pins[i]->Release();
}
// Set the format of the audio to 8KHZ, 16Bit/Sample, MONO.
hr = SetAudioFormat(
pIPin,
g_wAudioCaptureBitPerSample,
m_dwAudioSampleRate
);
if (FAILED(hr))
{
LOG((MSP_ERROR, "can't set audio format, %x", hr));
return hr;
}
// Set the capture buffer size.
hr = SetAudioBufferSize(
pIPin,
g_dwAudioCaptureNumBufffers,
AudioCaptureBufferSize(m_dwMSPerPacket, m_dwAudioSampleRate)
);
if (FAILED(hr))
{
LOG((MSP_ERROR, "can't set aduio capture buffer size, %x", hr));
return hr;
}
pIPin->AddRef();
*ppIPin = pIPin;
return hr;
}
HRESULT CStreamAudioSend::ConnectTerminal(
IN ITTerminal * pITTerminal
)
/*++
Routine Description:
connect the audio capture terminal to the stream.
Arguments:
pITTerminal - The terminal to be connected.
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioSend ConnectTerminal, pITTerminal %p", pITTerminal));
CComQIPtr<ITTerminalControl, &IID_ITTerminalControl>
pTerminal(pITTerminal);
if (pTerminal == NULL)
{
LOG((MSP_ERROR, "can't get Terminal Control interface"));
SendStreamEvent(
CALL_TERMINAL_FAIL,
CALL_CAUSE_BAD_DEVICE,
E_NOINTERFACE,
pITTerminal
);
return E_NOINTERFACE;
}
// configure the terminal.
CComPtr<IPin> pIPin;
HRESULT hr = ConfigureAudioCaptureTerminal(pTerminal, &pIPin);
if (FAILED(hr))
{
LOG((MSP_ERROR, "configure audio capture termianl failed. %x", hr));
SendStreamEvent(
CALL_TERMINAL_FAIL,
CALL_CAUSE_BAD_DEVICE,
hr,
pITTerminal
);
return hr;
}
// Create other filters to be use in the stream.
hr = CreateSendFilters(pIPin);
if (FAILED(hr))
{
LOG((MSP_ERROR, "Create audio send filters failed. %x", hr));
pTerminal->DisconnectTerminal(m_pIGraphBuilder, 0);
// clean up internal filters as well.
CleanUpFilters();
return hr;
}
//
// Now we are actually connected. Update our state and perform postconnection
// (ignore postconnection error code).
//
pTerminal->CompleteConnectTerminal();
return hr;
}
HRESULT CStreamAudioSend::SetUpFilters()
/*++
Routine Description:
Insert filters into the graph and connect to the terminals.
Arguments:
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioSend SetUpFilters"));
// we only support one terminal for this stream.
if (m_Terminals.GetSize() != 1)
{
return E_UNEXPECTED;
}
HRESULT hr;
// Connect the terminal to the rest of the stream.
if (FAILED(hr = ConnectTerminal(
m_Terminals[0]
)))
{
LOG((MSP_ERROR, "connect the terminal to the filters. %x", hr));
return hr;
}
return hr;
}
HRESULT CStreamAudioSend::ConfigureRTPFilter(
IN IBaseFilter * pIBaseFilter
)
/*++
Routine Description:
Configure the source RTP filter. Including set the address, port, TTL,
QOS, thread priority, clcokrate, etc.
Arguments:
pIBaseFilter - The source RTP Filter.
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioSend ConfigureRTPFilter"));
HRESULT hr;
// Get the IRTPStream interface pointer on the filter.
CComQIPtr<IRTPStream, &IID_IRTPStream> pIRTPStream(pIBaseFilter);
if (pIRTPStream == NULL)
{
LOG((MSP_ERROR, "get IRTPStream interface"));
return E_NOINTERFACE;
}
LOG((MSP_INFO, "set remote Address:%x, port:%d, TTL:%d",
m_Settings.dwIPRemote, m_Settings.wRTPPortRemote, m_Settings.dwTTL));
// Set the remote address and port used in the filter.
if (FAILED(hr = pIRTPStream->SetAddress(
0, // local port.
htons(m_Settings.wRTPPortRemote), // remote port.
htonl(m_Settings.dwIPRemote)
)))
{
LOG((MSP_ERROR, "set remote Address, hr:%x", hr));
return hr;
}
// Set the TTL used in the filter.
if (FAILED(hr = pIRTPStream->SetMulticastScope(m_Settings.dwTTL)))
{
LOG((MSP_ERROR, "set TTL. %x", hr));
return hr;
}
if (m_Settings.dwIPLocal != INADDR_ANY)
{
// set the local interface that the socket should bind to
LOG((MSP_INFO, "set locol Address:%x", m_Settings.dwIPLocal));
if (FAILED(hr = pIRTPStream->SelectLocalIPAddress(
htonl(m_Settings.dwIPLocal)
)))
{
LOG((MSP_ERROR, "set local Address, hr:%x", hr));
return hr;
}
}
// Set the priority of the session
if (FAILED(hr = pIRTPStream->SetSessionClassPriority(
RTP_CLASS_AUDIO,
g_dwAudioThreadPriority
)))
{
LOG((MSP_WARN, "set session class and priority. %x", hr));
}
// Set the sample rate of the session
LOG((MSP_INFO, "setting session sample rate to %d", m_dwAudioSampleRate));
if (FAILED(hr = pIRTPStream->SetDataClock(m_dwAudioSampleRate)))
{
LOG((MSP_WARN, "set session sample rate. %x", hr));
}
// Enable the RTCP events
if (FAILED(hr = ::EnableRTCPEvents(pIBaseFilter)))
{
LOG((MSP_WARN, "can not enable RTCP events %x", hr));
}
if (m_Settings.dwQOSLevel != QSL_BEST_EFFORT)
{
if (FAILED(hr = ::SetQOSOption(
pIBaseFilter,
m_Settings.dwPayloadType, // payload
-1, // use the default bitrate
(m_Settings.dwQOSLevel == QSL_NEEDED) // fail if no qos.
)))
{
LOG((MSP_ERROR, "set QOS option. %x", hr));
return hr;
}
}
SetLocalInfoOnRTPFilter(pIBaseFilter);
return S_OK;
}
HRESULT CStreamAudioSend::CreateSendFilters(
IN IPin *pPin
)
/*++
Routine Description:
Insert filters into the graph and connect to the capture pin.
Capturepin->SilenceSuppressor->Encoder->SPH->RTPRender
Arguments:
pPin - The output pin on the capture filter.
Return Value:
HRESULT.
--*/
{
LOG((MSP_TRACE, "AudioSend.CreateSendFilters"));
HRESULT hr;
// if the the internal filters have been created before, just
// connect the terminal to the first filter in the chain.
if (m_pEdgeFilter != NULL)
{
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
pPin,
(IBaseFilter *)m_pEdgeFilter
)))
{
LOG((MSP_ERROR, "connect capture and ss %x", hr));
return hr;
}
return hr;
}
// Create the silence suppression filter and add it into the graph.
CComPtr<IBaseFilter> pISSFilter;
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
CLSID_SilenceSuppressionFilter,
L"SS",
&pISSFilter
)))
{
LOG((MSP_ERROR, "can't add SS filter, %x", hr));
return hr;
}
DWORD dwAGC = 0;
if (FALSE == ::GetRegValue(L"AGC", &dwAGC) || dwAGC != 0)
{
// AGC is not disabled, just do it.
CComQIPtr<ISilenceSuppressor, &IID_ISilenceSuppressor>
pISilcnecSuppressor(pISSFilter);
if (pISilcnecSuppressor != NULL)
{
hr = pISilcnecSuppressor->EnableEvents(
(1 << AGC_INCREASE_GAIN) |
(1 << AGC_DECREASE_GAIN) |
(1 << AGC_TALKING) |
(1 << AGC_SILENCE),
2000 // no more that an event every two seconds.
);
if (FAILED(hr))
{
LOG((MSP_WARN, "can't enable AGC events, %x", hr));
}
}
}
// connect the capture pin with the SS filter.
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
pPin,
(IBaseFilter *)pISSFilter
)))
{
LOG((MSP_ERROR, "connect capture and ss %x", hr));
return hr;
}
// Create the codec filter and add it into the graph.
CComPtr<IBaseFilter> pICodecFilter;
if (m_fUseACM)
{
if (S_OK != (hr = ::FindACMAudioCodec(
m_Settings.dwPayloadType,
&pICodecFilter
)))
{
LOG((MSP_ERROR, "Find Codec filter. %x", hr));
return hr;
}
if (FAILED(hr = m_pIGraphBuilder->AddFilter(
pICodecFilter, L"AudioCodec"
)))
{
LOG((MSP_ERROR, "add codec filter. %x", hr));
return hr;
}
}
else
{
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
*m_pClsidCodecFilter,
L"Encoder",
&pICodecFilter)))
{
LOG((MSP_ERROR, "add Codec filter. %x", hr));
return hr;
}
}
// connect the SS filter and the Codec filter.
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
(IBaseFilter *)pISSFilter,
(IBaseFilter *)pICodecFilter
)))
{
LOG((MSP_ERROR, "connect ss filter and codec filter. %x", hr));
return hr;
}
// Create the send payload handler and add it into the graph.
CComPtr<IBaseFilter> pISPHFilter;
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
*m_pClsidPHFilter,
L"SPH",
&pISPHFilter
)))
{
LOG((MSP_ERROR, "add SPH filter. %x", hr));
return hr;
}
// Get the IRTPSPHFilter interface.
CComQIPtr<IRTPSPHFilter,
&IID_IRTPSPHFilter> pIRTPSPHFilter(pISPHFilter);
if (pIRTPSPHFilter == NULL)
{
LOG((MSP_ERROR, "get IRTPSPHFilter interface"));
return E_NOINTERFACE;
}
// Set the packetSize.
if (FAILED(hr= pIRTPSPHFilter->SetMaxPacketSize(m_dwMaxPacketSize)))
{
LOG((MSP_ERROR, "set SPH filter Max packet size: %d hr: %x",
m_dwMaxPacketSize, hr));
return hr;
}
if (FAILED(hr = pIRTPSPHFilter->OverridePayloadType(
(BYTE)m_Settings.dwPayloadType
)))
{
LOG((MSP_ERROR, "Set SPHGENA payload type. %x", hr));
return hr;
}
// Connect the Codec filter with the SPH filter .
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
(IBaseFilter *)pICodecFilter,
(IBaseFilter *)pISPHFilter
)))
{
LOG((MSP_ERROR, "connect codec filter and SPH filter. %x", hr));
return hr;
}
// Create the RTP render filter and add it into the graph.
CComPtr<IBaseFilter> pRenderFilter;
if (FAILED(hr = ::AddFilter(
m_pIGraphBuilder,
CLSID_RTPRenderFilter,
L"RtpRender",
&pRenderFilter)))
{
LOG((MSP_ERROR, "adding render filter. %x", hr));
return hr;
}
// Set the address for the render fitler.
if (FAILED(hr = ConfigureRTPFilter(pRenderFilter)))
{
LOG((MSP_ERROR, "set destination address. %x", hr));
return hr;
}
// Connect the SPH filter with the RTP Render filter.
if (FAILED(hr = ::ConnectFilters(
m_pIGraphBuilder,
(IBaseFilter *)pISPHFilter,
(IBaseFilter *)pRenderFilter
)))
{
LOG((MSP_ERROR, "connect SPH filter and Render filter. %x", hr));
return hr;
}
// remember the first filter after the terminal
m_pEdgeFilter = pISSFilter;
m_pEdgeFilter->AddRef();
// Get the IRTPParticipant interface pointer on the RTP filter.
CComQIPtr<IRTPParticipant,
&IID_IRTPParticipant> pIRTPParticipant(pRenderFilter);
if (pIRTPParticipant == NULL)
{
LOG((MSP_WARN, "can't get RTP participant interface"));
}
else
{
m_pRTPFilter = pIRTPParticipant;
m_pRTPFilter->AddRef();
}
return S_OK;
}
HRESULT AdjustGain(
IN IUnknown * pIUnknown,
IN long lPercent
)
/*++
Routine Description:
This function uses IAMAudioInputMixer interface to adjust the gain.
Arguments:
pIUnknown - the object that supports IAMAudioInputMixer
lPercent - the adjustment, a negative value means decrease.
Return Value:
S_OK,
E_NOINTERFACE,
E_UNEXPECTED
--*/
{
CComPtr <IAMAudioInputMixer> pMixer;
HRESULT hr = pIUnknown->QueryInterface(
IID_IAMAudioInputMixer, (void **)&pMixer
);
if (FAILED(hr))
{
LOG((MSP_ERROR, "can't get IAMAudioInputMixer interface."));
return hr;
}
BOOL fEnabled;
hr = pMixer->get_Enable(&fEnabled);
if (SUCCEEDED(hr) && !fEnabled)
{
return S_OK;
}
double MixLevel;
hr = pMixer->get_MixLevel(&MixLevel);
if (FAILED(hr))
{
LOG((MSP_ERROR, "get_MixLevel returned %d", hr));
return hr;
}
LOG((MSP_INFO, "get_MixLevel returned %d", hr));
MixLevel = MixLevel * (100 + lPercent) / 100;
hr = pMixer->put_MixLevel(MixLevel);
if (FAILED(hr))
{
LOG((MSP_ERROR, "put_MixLevel returned %d", hr));
return hr;
}
return S_OK;
}
HRESULT CStreamAudioSend::ProcessAGCEvent(
IN AGC_EVENT Event,
IN long lPercent
)
/*++
Routine Description:
The filters fire AGC events to requste a change in the microphone gain.
This function finds the capture terminal and adjust the gain on it.
Arguments:
Event - either AGC_INCREASE_GAIN or AGC_DECREASE_GAIN.
Return Value:
S_OK,
E_UNEXPECTED
--*/
{
LOG((MSP_TRACE, "ProcessAGCEvent %s %d percent",
(Event == AGC_INCREASE_GAIN) ? "Increase" : "Decrease",
lPercent
));
_ASSERTE(lPercent > 0 && lPercent <= 100);
CLock lock(m_lock);
if (m_pEdgeFilter == NULL)
{
LOG((MSP_ERROR, "No filter to adjust gain."));
return E_UNEXPECTED;
}
CComPtr<IPin> pMyPin, pCapturePin;
// find the first pin in the stream
HRESULT hr = ::FindPin(m_pEdgeFilter, &pMyPin, PINDIR_INPUT, FALSE);
if (FAILED(hr))
{
LOG((MSP_ERROR, "can't get find the first pin the stream, %x", hr));
return hr;
}
// find the capture pin that connects to our first pin.
hr = pMyPin->ConnectedTo(&pCapturePin);
if (FAILED(hr))
{
LOG((MSP_ERROR, "can't find the capture pin, %x", hr));
return hr;
}
// find the filter that has the capture pin.
PIN_INFO PinInfo;
hr = pCapturePin->QueryPinInfo(&PinInfo);
if (FAILED(hr))
{
LOG((MSP_ERROR, "can't find the capture filter, %x", hr));
return hr;
}
// save the filter pointer.
CComPtr<IBaseFilter> pICaptureFilter = PinInfo.pFilter;
PinInfo.pFilter->Release();
// get the amount to adjust.
if (Event == AGC_DECREASE_GAIN)
{
lPercent = -lPercent;
}
AdjustGain(pICaptureFilter, lPercent);
// Get the enumerator of pins on the filter.
CComPtr<IEnumPins> pIEnumPins;
if (FAILED(hr = pICaptureFilter->EnumPins(&pIEnumPins)))
{
LOG((MSP_ERROR, "enumerate pins on the filter %x", hr));
return hr;
}
// Enumerate all the pins and adjust gains on each active one.
for (;;)
{
CComPtr<IPin> pIPin;
DWORD dwFeched;
if (pIEnumPins->Next(1, &pIPin, &dwFeched) != S_OK)
{
LOG((MSP_ERROR, "find pin on filter."));
break;
}
AdjustGain(pIPin, lPercent);
}
return hr;
}
HRESULT CStreamAudioSend::ProcessGraphEvent(
IN long lEventCode,
IN long lParam1,
IN long lParam2
)
{
LOG((MSP_TRACE, "%ws ProcessGraphEvent %d", m_szName, lEventCode));
switch (lEventCode)
{
case AGC_EVENTBASE + AGC_INCREASE_GAIN:
ProcessAGCEvent(AGC_INCREASE_GAIN, lParam1);
break;
case AGC_EVENTBASE + AGC_DECREASE_GAIN:
ProcessAGCEvent(AGC_DECREASE_GAIN, lParam1);
break;
case AGC_EVENTBASE + AGC_TALKING:
m_lock.Lock();
if (m_pMSPCall != NULL)
{
((CIPConfMSPCall *)m_pMSPCall)->SendParticipantEvent(
PE_LOCAL_TALKING,
NULL
);
}
m_lock.Unlock();
break;
case AGC_EVENTBASE + AGC_SILENCE:
m_lock.Lock();
if (m_pMSPCall != NULL)
{
((CIPConfMSPCall *)m_pMSPCall)->SendParticipantEvent(
PE_LOCAL_SILENT,
NULL
);
}
m_lock.Unlock();
break;
default:
return CIPConfMSPStream::ProcessGraphEvent(
lEventCode, lParam1, lParam2
);
}
return S_OK;
}
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