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chromium / external / webrtc / stable / src / 0755971db657c273c3e704413440a2281399c974 / . / voice_engine / channel.cc
blob: b4889e29ec016049ebd92a9dedb266b3092335a6 [file] [log] [blame]
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "channel.h"
#include "audio_device.h"
#include "audio_frame_operations.h"
#include "audio_processing.h"
#include "critical_section_wrapper.h"
#include "output_mixer.h"
#include "process_thread.h"
#include "rtp_dump.h"
#include "statistics.h"
#include "trace.h"
#include "transmit_mixer.h"
#include "utility.h"
#include "voe_base.h"
#include "voe_external_media.h"
#include "voe_rtp_rtcp.h"
#if defined(_WIN32)
#include <Qos.h>
#endif
namespace webrtc
{
namespace voe
{
WebRtc_Word32
Channel::SendData(FrameType frameType,
WebRtc_UWord8 payloadType,
WebRtc_UWord32 timeStamp,
const WebRtc_UWord8* payloadData,
WebRtc_UWord16 payloadSize,
const RTPFragmentationHeader* fragmentation)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
" payloadSize=%u, fragmentation=0x%x)",
frameType, payloadType, timeStamp, payloadSize, fragmentation);
if (_includeAudioLevelIndication)
{
assert(_rtpAudioProc.get() != NULL);
// Store current audio level in the RTP/RTCP module.
// The level will be used in combination with voice-activity state
// (frameType) to add an RTP header extension
_rtpRtcpModule->SetAudioLevel(_rtpAudioProc->level_estimator()->RMS());
}
// Push data from ACM to RTP/RTCP-module to deliver audio frame for
// packetization.
// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
if (_rtpRtcpModule->SendOutgoingData((FrameType&)frameType,
payloadType,
timeStamp,
// Leaving the time when this frame was
// received from the capture device as
// undefined for voice for now.
-1,
payloadData,
payloadSize,
fragmentation) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
"Channel::SendData() failed to send data to RTP/RTCP module");
return -1;
}
_lastLocalTimeStamp = timeStamp;
_lastPayloadType = payloadType;
return 0;
}
WebRtc_Word32
Channel::InFrameType(WebRtc_Word16 frameType)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::InFrameType(frameType=%d)", frameType);
CriticalSectionScoped cs(&_callbackCritSect);
// 1 indicates speech
_sendFrameType = (frameType == 1) ? 1 : 0;
return 0;
}
#ifdef WEBRTC_DTMF_DETECTION
int
Channel::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::IncomingDtmf(digitDtmf=%u, end=%d)",
digitDtmf, end);
if (digitDtmf != 999)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_telephoneEventDetectionPtr)
{
_telephoneEventDetectionPtr->OnReceivedTelephoneEventInband(
_channelId, digitDtmf, end);
}
}
return 0;
}
#endif
WebRtc_Word32
Channel::OnRxVadDetected(const int vadDecision)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnRxVadDetected(vadDecision=%d)", vadDecision);
CriticalSectionScoped cs(&_callbackCritSect);
if (_rxVadObserverPtr)
{
_rxVadObserverPtr->OnRxVad(_channelId, vadDecision);
}
return 0;
}
int
Channel::SendPacket(int channel, const void *data, int len)
{
channel = VoEChannelId(channel);
assert(channel == _channelId);
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SendPacket(channel=%d, len=%d)", channel, len);
if (_transportPtr == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SendPacket() failed to send RTP packet due to"
" invalid transport object");
return -1;
}
// Insert extra RTP packet using if user has called the InsertExtraRTPPacket
// API
if (_insertExtraRTPPacket)
{
WebRtc_UWord8* rtpHdr = (WebRtc_UWord8*)data;
WebRtc_UWord8 M_PT(0);
if (_extraMarkerBit)
{
M_PT = 0x80; // set the M-bit
}
M_PT += _extraPayloadType; // set the payload type
*(++rtpHdr) = M_PT; // modify the M|PT-byte within the RTP header
_insertExtraRTPPacket = false; // insert one packet only
}
WebRtc_UWord8* bufferToSendPtr = (WebRtc_UWord8*)data;
WebRtc_Word32 bufferLength = len;
// Dump the RTP packet to a file (if RTP dump is enabled).
if (_rtpDumpOut.DumpPacket((const WebRtc_UWord8*)data, len) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTP dump to output file failed");
}
// SRTP or External encryption
if (_encrypting)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_encryptionPtr)
{
if (!_encryptionRTPBufferPtr)
{
// Allocate memory for encryption buffer one time only
_encryptionRTPBufferPtr =
new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
}
// Perform encryption (SRTP or external)
WebRtc_Word32 encryptedBufferLength = 0;
_encryptionPtr->encrypt(_channelId,
bufferToSendPtr,
_encryptionRTPBufferPtr,
bufferLength,
(int*)&encryptedBufferLength);
if (encryptedBufferLength <= 0)
{
_engineStatisticsPtr->SetLastError(
VE_ENCRYPTION_FAILED,
kTraceError, "Channel::SendPacket() encryption failed");
return -1;
}
// Replace default data buffer with encrypted buffer
bufferToSendPtr = _encryptionRTPBufferPtr;
bufferLength = encryptedBufferLength;
}
}
// Packet transmission using WebRtc socket transport
if (!_externalTransport)
{
int n = _transportPtr->SendPacket(channel, bufferToSendPtr,
bufferLength);
if (n < 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTP transmission using WebRtc"
" sockets failed");
return -1;
}
return n;
}
// Packet transmission using external transport transport
{
CriticalSectionScoped cs(&_callbackCritSect);
int n = _transportPtr->SendPacket(channel,
bufferToSendPtr,
bufferLength);
if (n < 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTP transmission using external"
" transport failed");
return -1;
}
return n;
}
}
int
Channel::SendRTCPPacket(int channel, const void *data, int len)
{
channel = VoEChannelId(channel);
assert(channel == _channelId);
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SendRTCPPacket(channel=%d, len=%d)", channel, len);
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_transportPtr == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendRTCPPacket() failed to send RTCP packet"
" due to invalid transport object");
return -1;
}
}
WebRtc_UWord8* bufferToSendPtr = (WebRtc_UWord8*)data;
WebRtc_Word32 bufferLength = len;
// Dump the RTCP packet to a file (if RTP dump is enabled).
if (_rtpDumpOut.DumpPacket((const WebRtc_UWord8*)data, len) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTCP dump to output file failed");
}
// SRTP or External encryption
if (_encrypting)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_encryptionPtr)
{
if (!_encryptionRTCPBufferPtr)
{
// Allocate memory for encryption buffer one time only
_encryptionRTCPBufferPtr =
new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
}
// Perform encryption (SRTP or external).
WebRtc_Word32 encryptedBufferLength = 0;
_encryptionPtr->encrypt_rtcp(_channelId,
bufferToSendPtr,
_encryptionRTCPBufferPtr,
bufferLength,
(int*)&encryptedBufferLength);
if (encryptedBufferLength <= 0)
{
_engineStatisticsPtr->SetLastError(
VE_ENCRYPTION_FAILED, kTraceError,
"Channel::SendRTCPPacket() encryption failed");
return -1;
}
// Replace default data buffer with encrypted buffer
bufferToSendPtr = _encryptionRTCPBufferPtr;
bufferLength = encryptedBufferLength;
}
}
// Packet transmission using WebRtc socket transport
if (!_externalTransport)
{
int n = _transportPtr->SendRTCPPacket(channel,
bufferToSendPtr,
bufferLength);
if (n < 0)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendRTCPPacket() transmission using WebRtc"
" sockets failed");
return -1;
}
return n;
}
// Packet transmission using external transport transport
{
CriticalSectionScoped cs(&_callbackCritSect);
int n = _transportPtr->SendRTCPPacket(channel,
bufferToSendPtr,
bufferLength);
if (n < 0)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendRTCPPacket() transmission using external"
" transport failed");
return -1;
}
return n;
}
return len;
}
void
Channel::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
const WebRtc_Word32 rtpPacketLength,
const char* fromIP,
const WebRtc_UWord16 fromPort)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::IncomingRTPPacket(rtpPacketLength=%d,"
" fromIP=%s, fromPort=%u)",
rtpPacketLength, fromIP, fromPort);
// Store playout timestamp for the received RTP packet
// to be used for upcoming delay estimations
WebRtc_UWord32 playoutTimestamp(0);
if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
{
_playoutTimeStampRTP = playoutTimestamp;
}
WebRtc_UWord8* rtpBufferPtr = (WebRtc_UWord8*)incomingRtpPacket;
WebRtc_Word32 rtpBufferLength = rtpPacketLength;
// SRTP or External decryption
if (_decrypting)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_encryptionPtr)
{
if (!_decryptionRTPBufferPtr)
{
// Allocate memory for decryption buffer one time only
_decryptionRTPBufferPtr =
new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
}
// Perform decryption (SRTP or external)
WebRtc_Word32 decryptedBufferLength = 0;
_encryptionPtr->decrypt(_channelId,
rtpBufferPtr,
_decryptionRTPBufferPtr,
rtpBufferLength,
(int*)&decryptedBufferLength);
if (decryptedBufferLength <= 0)
{
_engineStatisticsPtr->SetLastError(
VE_DECRYPTION_FAILED, kTraceError,
"Channel::IncomingRTPPacket() decryption failed");
return;
}
// Replace default data buffer with decrypted buffer
rtpBufferPtr = _decryptionRTPBufferPtr;
rtpBufferLength = decryptedBufferLength;
}
}
// Dump the RTP packet to a file (if RTP dump is enabled).
if (_rtpDumpIn.DumpPacket(rtpBufferPtr,
(WebRtc_UWord16)rtpBufferLength) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTP dump to input file failed");
}
// Deliver RTP packet to RTP/RTCP module for parsing
// The packet will be pushed back to the channel thru the
// OnReceivedPayloadData callback so we don't push it to the ACM here
if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)rtpBufferPtr,
(WebRtc_UWord16)rtpBufferLength) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
"Channel::IncomingRTPPacket() RTP packet is invalid");
return;
}
}
void
Channel::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
const WebRtc_Word32 rtcpPacketLength,
const char* fromIP,
const WebRtc_UWord16 fromPort)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::IncomingRTCPPacket(rtcpPacketLength=%d, fromIP=%s,"
" fromPort=%u)",
rtcpPacketLength, fromIP, fromPort);
// Temporary buffer pointer and size for decryption
WebRtc_UWord8* rtcpBufferPtr = (WebRtc_UWord8*)incomingRtcpPacket;
WebRtc_Word32 rtcpBufferLength = rtcpPacketLength;
// Store playout timestamp for the received RTCP packet
// which will be read by the GetRemoteRTCPData API
WebRtc_UWord32 playoutTimestamp(0);
if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
{
_playoutTimeStampRTCP = playoutTimestamp;
}
// SRTP or External decryption
if (_decrypting)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_encryptionPtr)
{
if (!_decryptionRTCPBufferPtr)
{
// Allocate memory for decryption buffer one time only
_decryptionRTCPBufferPtr =
new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
}
// Perform decryption (SRTP or external).
WebRtc_Word32 decryptedBufferLength = 0;
_encryptionPtr->decrypt_rtcp(_channelId,
rtcpBufferPtr,
_decryptionRTCPBufferPtr,
rtcpBufferLength,
(int*)&decryptedBufferLength);
if (decryptedBufferLength <= 0)
{
_engineStatisticsPtr->SetLastError(
VE_DECRYPTION_FAILED, kTraceError,
"Channel::IncomingRTCPPacket() decryption failed");
return;
}
// Replace default data buffer with decrypted buffer
rtcpBufferPtr = _decryptionRTCPBufferPtr;
rtcpBufferLength = decryptedBufferLength;
}
}
// Dump the RTCP packet to a file (if RTP dump is enabled).
if (_rtpDumpIn.DumpPacket(rtcpBufferPtr,
(WebRtc_UWord16)rtcpBufferLength) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTCP dump to input file failed");
}
// Deliver RTCP packet to RTP/RTCP module for parsing
if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)rtcpBufferPtr,
(WebRtc_UWord16)rtcpBufferLength) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
"Channel::IncomingRTPPacket() RTCP packet is invalid");
return;
}
}
void
Channel::OnReceivedTelephoneEvent(const WebRtc_Word32 id,
const WebRtc_UWord8 event,
const bool endOfEvent)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnReceivedTelephoneEvent(id=%d, event=%u,"
" endOfEvent=%d)", id, event, endOfEvent);
#ifdef WEBRTC_DTMF_DETECTION
if (_outOfBandTelephoneEventDetecion)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_telephoneEventDetectionPtr)
{
_telephoneEventDetectionPtr->OnReceivedTelephoneEventOutOfBand(
_channelId, event, endOfEvent);
}
}
#endif
}
void
Channel::OnPlayTelephoneEvent(const WebRtc_Word32 id,
const WebRtc_UWord8 event,
const WebRtc_UWord16 lengthMs,
const WebRtc_UWord8 volume)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u,"
" volume=%u)", id, event, lengthMs, volume);
if (!_playOutbandDtmfEvent || (event > 15))
{
// Ignore callback since feedback is disabled or event is not a
// Dtmf tone event.
return;
}
assert(_outputMixerPtr != NULL);
// Start playing out the Dtmf tone (if playout is enabled).
// Reduce length of tone with 80ms to the reduce risk of echo.
_outputMixerPtr->PlayDtmfTone(event, lengthMs - 80, volume);
}
void
Channel::OnIncomingSSRCChanged(const WebRtc_Word32 id,
const WebRtc_UWord32 SSRC)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
id, SSRC);
WebRtc_Word32 channel = VoEChannelId(id);
assert(channel == _channelId);
// Reset RTP-module counters since a new incoming RTP stream is detected
_rtpRtcpModule->ResetReceiveDataCountersRTP();
_rtpRtcpModule->ResetStatisticsRTP();
if (_rtpObserver)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_rtpObserverPtr)
{
// Send new SSRC to registered observer using callback
_rtpObserverPtr->OnIncomingSSRCChanged(channel, SSRC);
}
}
}
void Channel::OnIncomingCSRCChanged(const WebRtc_Word32 id,
const WebRtc_UWord32 CSRC,
const bool added)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
id, CSRC, added);
WebRtc_Word32 channel = VoEChannelId(id);
assert(channel == _channelId);
if (_rtpObserver)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_rtpObserverPtr)
{
_rtpObserverPtr->OnIncomingCSRCChanged(channel, CSRC, added);
}
}
}
void
Channel::OnApplicationDataReceived(const WebRtc_Word32 id,
const WebRtc_UWord8 subType,
const WebRtc_UWord32 name,
const WebRtc_UWord16 length,
const WebRtc_UWord8* data)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnApplicationDataReceived(id=%d, subType=%u,"
" name=%u, length=%u)",
id, subType, name, length);
WebRtc_Word32 channel = VoEChannelId(id);
assert(channel == _channelId);
if (_rtcpObserver)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_rtcpObserverPtr)
{
_rtcpObserverPtr->OnApplicationDataReceived(channel,
subType,
name,
data,
length);
}
}
}
WebRtc_Word32
Channel::OnInitializeDecoder(
const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnInitializeDecoder(id=%d, payloadType=%d, "
"payloadName=%s, frequency=%u, channels=%u, rate=%u)",
id, payloadType, payloadName, frequency, channels, rate);
assert(VoEChannelId(id) == _channelId);
CodecInst receiveCodec = {0};
CodecInst dummyCodec = {0};
receiveCodec.pltype = payloadType;
receiveCodec.plfreq = frequency;
receiveCodec.channels = channels;
receiveCodec.rate = rate;
strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
_audioCodingModule.Codec(payloadName, dummyCodec, frequency, channels);
receiveCodec.pacsize = dummyCodec.pacsize;
// Register the new codec to the ACM
if (_audioCodingModule.RegisterReceiveCodec(receiveCodec) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::OnInitializeDecoder() invalid codec ("
"pt=%d, name=%s) received - 1", payloadType, payloadName);
_engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
return -1;
}
return 0;
}
void
Channel::OnPacketTimeout(const WebRtc_Word32 id)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnPacketTimeout(id=%d)", id);
CriticalSectionScoped cs(_callbackCritSectPtr);
if (_voiceEngineObserverPtr)
{
if (_receiving || _externalTransport)
{
WebRtc_Word32 channel = VoEChannelId(id);
assert(channel == _channelId);
// Ensure that next OnReceivedPacket() callback will trigger
// a VE_PACKET_RECEIPT_RESTARTED callback.
_rtpPacketTimedOut = true;
// Deliver callback to the observer
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::OnPacketTimeout() => "
"CallbackOnError(VE_RECEIVE_PACKET_TIMEOUT)");
_voiceEngineObserverPtr->CallbackOnError(channel,
VE_RECEIVE_PACKET_TIMEOUT);
}
}
}
void
Channel::OnReceivedPacket(const WebRtc_Word32 id,
const RtpRtcpPacketType packetType)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnReceivedPacket(id=%d, packetType=%d)",
id, packetType);
assert(VoEChannelId(id) == _channelId);
// Notify only for the case when we have restarted an RTP session.
if (_rtpPacketTimedOut && (kPacketRtp == packetType))
{
CriticalSectionScoped cs(_callbackCritSectPtr);
if (_voiceEngineObserverPtr)
{
WebRtc_Word32 channel = VoEChannelId(id);
assert(channel == _channelId);
// Reset timeout mechanism
_rtpPacketTimedOut = false;
// Deliver callback to the observer
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::OnPacketTimeout() =>"
" CallbackOnError(VE_PACKET_RECEIPT_RESTARTED)");
_voiceEngineObserverPtr->CallbackOnError(
channel,
VE_PACKET_RECEIPT_RESTARTED);
}
}
}
void
Channel::OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
const RTPAliveType alive)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnPeriodicDeadOrAlive(id=%d, alive=%d)", id, alive);
if (!_connectionObserver)
return;
WebRtc_Word32 channel = VoEChannelId(id);
assert(channel == _channelId);
// Use Alive as default to limit risk of false Dead detections
bool isAlive(true);
// Always mark the connection as Dead when the module reports kRtpDead
if (kRtpDead == alive)
{
isAlive = false;
}
// It is possible that the connection is alive even if no RTP packet has
// been received for a long time since the other side might use VAD/DTX
// and a low SID-packet update rate.
if ((kRtpNoRtp == alive) && _playing)
{
// Detect Alive for all NetEQ states except for the case when we are
// in PLC_CNG state.
// PLC_CNG <=> background noise only due to long expand or error.
// Note that, the case where the other side stops sending during CNG
// state will be detected as Alive. Dead is is not set until after
// missing RTCP packets for at least twelve seconds (handled
// internally by the RTP/RTCP module).
isAlive = (_outputSpeechType != AudioFrame::kPLCCNG);
}
UpdateDeadOrAliveCounters(isAlive);
// Send callback to the registered observer
if (_connectionObserver)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_connectionObserverPtr)
{
_connectionObserverPtr->OnPeriodicDeadOrAlive(channel, isAlive);
}
}
}
WebRtc_Word32
Channel::OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
const WebRtcRTPHeader* rtpHeader)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::OnReceivedPayloadData(payloadSize=%d,"
" payloadType=%u, audioChannel=%u)",
payloadSize,
rtpHeader->header.payloadType,
rtpHeader->type.Audio.channel);
if (!_playing)
{
// Avoid inserting into NetEQ when we are not playing. Count the
// packet as discarded.
WEBRTC_TRACE(kTraceStream, kTraceVoice,
VoEId(_instanceId, _channelId),
"received packet is discarded since playing is not"
" activated");
_numberOfDiscardedPackets++;
return 0;
}
// Push the incoming payload (parsed and ready for decoding) into the ACM
if (_audioCodingModule.IncomingPacket(payloadData,
payloadSize,
*rtpHeader) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
"Channel::OnReceivedPayloadData() unable to push data to the ACM");
return -1;
}
// Update the packet delay
UpdatePacketDelay(rtpHeader->header.timestamp,
rtpHeader->header.sequenceNumber);
return 0;
}
WebRtc_Word32 Channel::GetAudioFrame(const WebRtc_Word32 id,
AudioFrame& audioFrame)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetAudioFrame(id=%d)", id);
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
if (_audioCodingModule.PlayoutData10Ms(audioFrame.sample_rate_hz_,
audioFrame) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::GetAudioFrame() PlayoutData10Ms() failed!");
// In all likelihood, the audio in this frame is garbage. We return an
// error so that the audio mixer module doesn't add it to the mix. As
// a result, it won't be played out and the actions skipped here are
// irrelevant.
return -1;
}
if (_RxVadDetection)
{
UpdateRxVadDetection(audioFrame);
}
// Convert module ID to internal VoE channel ID
audioFrame.id_ = VoEChannelId(audioFrame.id_);
// Store speech type for dead-or-alive detection
_outputSpeechType = audioFrame.speech_type_;
// Perform far-end AudioProcessing module processing on the received signal
if (_rxApmIsEnabled)
{
ApmProcessRx(audioFrame);
}
// Output volume scaling
if (_outputGain < 0.99f || _outputGain > 1.01f)
{
AudioFrameOperations::ScaleWithSat(_outputGain, audioFrame);
}
// Scale left and/or right channel(s) if stereo and master balance is
// active
if (_panLeft != 1.0f || _panRight != 1.0f)
{
if (audioFrame.num_channels_ == 1)
{
// Emulate stereo mode since panning is active.
// The mono signal is copied to both left and right channels here.
AudioFrameOperations::MonoToStereo(&audioFrame);
}
// For true stereo mode (when we are receiving a stereo signal), no
// action is needed.
// Do the panning operation (the audio frame contains stereo at this
// stage)
AudioFrameOperations::Scale(_panLeft, _panRight, audioFrame);
}
// Mix decoded PCM output with file if file mixing is enabled
if (_outputFilePlaying)
{
MixAudioWithFile(audioFrame, audioFrame.sample_rate_hz_);
}
// Place channel in on-hold state (~muted) if on-hold is activated
if (_outputIsOnHold)
{
AudioFrameOperations::Mute(audioFrame);
}
// External media
if (_outputExternalMedia)
{
CriticalSectionScoped cs(&_callbackCritSect);
const bool isStereo = (audioFrame.num_channels_ == 2);
if (_outputExternalMediaCallbackPtr)
{
_outputExternalMediaCallbackPtr->Process(
_channelId,
kPlaybackPerChannel,
(WebRtc_Word16*)audioFrame.data_,
audioFrame.samples_per_channel_,
audioFrame.sample_rate_hz_,
isStereo);
}
}
// Record playout if enabled
{
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFileRecording && _outputFileRecorderPtr)
{
_outputFileRecorderPtr->RecordAudioToFile(audioFrame);
}
}
// Measure audio level (0-9)
_outputAudioLevel.ComputeLevel(audioFrame);
return 0;
}
WebRtc_Word32
Channel::NeededFrequency(const WebRtc_Word32 id)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::NeededFrequency(id=%d)", id);
int highestNeeded = 0;
// Determine highest needed receive frequency
WebRtc_Word32 receiveFrequency = _audioCodingModule.ReceiveFrequency();
// Return the bigger of playout and receive frequency in the ACM.
if (_audioCodingModule.PlayoutFrequency() > receiveFrequency)
{
highestNeeded = _audioCodingModule.PlayoutFrequency();
}
else
{
highestNeeded = receiveFrequency;
}
// Special case, if we're playing a file on the playout side
// we take that frequency into consideration as well
// This is not needed on sending side, since the codec will
// limit the spectrum anyway.
if (_outputFilePlaying)
{
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFilePlayerPtr && _outputFilePlaying)
{
if(_outputFilePlayerPtr->Frequency()>highestNeeded)
{
highestNeeded=_outputFilePlayerPtr->Frequency();
}
}
}
return(highestNeeded);
}
WebRtc_Word32
Channel::CreateChannel(Channel*& channel,
const WebRtc_Word32 channelId,
const WebRtc_UWord32 instanceId)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)",
channelId, instanceId);
channel = new Channel(channelId, instanceId);
if (channel == NULL)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice,
VoEId(instanceId,channelId),
"Channel::CreateChannel() unable to allocate memory for"
" channel");
return -1;
}
return 0;
}
void
Channel::PlayNotification(const WebRtc_Word32 id,
const WebRtc_UWord32 durationMs)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::PlayNotification(id=%d, durationMs=%d)",
id, durationMs);
// Not implement yet
}
void
Channel::RecordNotification(const WebRtc_Word32 id,
const WebRtc_UWord32 durationMs)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RecordNotification(id=%d, durationMs=%d)",
id, durationMs);
// Not implement yet
}
void
Channel::PlayFileEnded(const WebRtc_Word32 id)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::PlayFileEnded(id=%d)", id);
if (id == _inputFilePlayerId)
{
CriticalSectionScoped cs(&_fileCritSect);
_inputFilePlaying = false;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::PlayFileEnded() => input file player module is"
" shutdown");
}
else if (id == _outputFilePlayerId)
{
CriticalSectionScoped cs(&_fileCritSect);
_outputFilePlaying = false;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::PlayFileEnded() => output file player module is"
" shutdown");
}
}
void
Channel::RecordFileEnded(const WebRtc_Word32 id)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RecordFileEnded(id=%d)", id);
assert(id == _outputFileRecorderId);
CriticalSectionScoped cs(&_fileCritSect);
_outputFileRecording = false;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::RecordFileEnded() => output file recorder module is"
" shutdown");
}
Channel::Channel(const WebRtc_Word32 channelId,
const WebRtc_UWord32 instanceId) :
_fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_instanceId(instanceId),
_channelId(channelId),
_audioCodingModule(*AudioCodingModule::Create(
VoEModuleId(instanceId, channelId))),
#ifndef WEBRTC_EXTERNAL_TRANSPORT
_numSocketThreads(KNumSocketThreads),
_socketTransportModule(*UdpTransport::Create(
VoEModuleId(instanceId, channelId), _numSocketThreads)),
#endif
#ifdef WEBRTC_SRTP
_srtpModule(*SrtpModule::CreateSrtpModule(VoEModuleId(instanceId,
channelId))),
#endif
_rtpDumpIn(*RtpDump::CreateRtpDump()),
_rtpDumpOut(*RtpDump::CreateRtpDump()),
_outputAudioLevel(),
_externalTransport(false),
_inputFilePlayerPtr(NULL),
_outputFilePlayerPtr(NULL),
_outputFileRecorderPtr(NULL),
// Avoid conflict with other channels by adding 1024 - 1026,
// won't use as much as 1024 channels.
_inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
_outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
_outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
_inputFilePlaying(false),
_outputFilePlaying(false),
_outputFileRecording(false),
_inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
_inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
_inputExternalMedia(false),
_outputExternalMedia(false),
_inputExternalMediaCallbackPtr(NULL),
_outputExternalMediaCallbackPtr(NULL),
_encryptionRTPBufferPtr(NULL),
_decryptionRTPBufferPtr(NULL),
_encryptionRTCPBufferPtr(NULL),
_decryptionRTCPBufferPtr(NULL),
_timeStamp(0), // This is just an offset, RTP module will add it's own random offset
_sendTelephoneEventPayloadType(106),
_playoutTimeStampRTP(0),
_playoutTimeStampRTCP(0),
_numberOfDiscardedPackets(0),
_engineStatisticsPtr(NULL),
_outputMixerPtr(NULL),
_transmitMixerPtr(NULL),
_moduleProcessThreadPtr(NULL),
_audioDeviceModulePtr(NULL),
_voiceEngineObserverPtr(NULL),
_callbackCritSectPtr(NULL),
_transportPtr(NULL),
_encryptionPtr(NULL),
_rtpAudioProc(NULL),
_rxAudioProcessingModulePtr(NULL),
#ifdef WEBRTC_DTMF_DETECTION
_telephoneEventDetectionPtr(NULL),
#endif
_rxVadObserverPtr(NULL),
_oldVadDecision(-1),
_sendFrameType(0),
_rtpObserverPtr(NULL),
_rtcpObserverPtr(NULL),
_outputIsOnHold(false),
_externalPlayout(false),
_inputIsOnHold(false),
_playing(false),
_sending(false),
_receiving(false),
_mixFileWithMicrophone(false),
_rtpObserver(false),
_rtcpObserver(false),
_mute(false),
_panLeft(1.0f),
_panRight(1.0f),
_outputGain(1.0f),
_encrypting(false),
_decrypting(false),
_playOutbandDtmfEvent(false),
_playInbandDtmfEvent(false),
_inbandTelephoneEventDetection(false),
_outOfBandTelephoneEventDetecion(false),
_extraPayloadType(0),
_insertExtraRTPPacket(false),
_extraMarkerBit(false),
_lastLocalTimeStamp(0),
_lastPayloadType(0),
_includeAudioLevelIndication(false),
_rtpPacketTimedOut(false),
_rtpPacketTimeOutIsEnabled(false),
_rtpTimeOutSeconds(0),
_connectionObserver(false),
_connectionObserverPtr(NULL),
_countAliveDetections(0),
_countDeadDetections(0),
_outputSpeechType(AudioFrame::kNormalSpeech),
_averageDelayMs(0),
_previousSequenceNumber(0),
_previousTimestamp(0),
_recPacketDelayMs(20),
_RxVadDetection(false),
_rxApmIsEnabled(false),
_rxAgcIsEnabled(false),
_rxNsIsEnabled(false)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::Channel() - ctor");
_inbandDtmfQueue.ResetDtmf();
_inbandDtmfGenerator.Init();
_outputAudioLevel.Clear();
RtpRtcp::Configuration configuration;
configuration.id = VoEModuleId(instanceId, channelId);
configuration.audio = true;
configuration.incoming_data = this;
configuration.incoming_messages = this;
configuration.outgoing_transport = this;
configuration.rtcp_feedback = this;
configuration.audio_messages = this;
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
// Create far end AudioProcessing Module
_rxAudioProcessingModulePtr = AudioProcessing::Create(
VoEModuleId(instanceId, channelId));
}
Channel::~Channel()
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::~Channel() - dtor");
if (_outputExternalMedia)
{
DeRegisterExternalMediaProcessing(kPlaybackPerChannel);
}
if (_inputExternalMedia)
{
DeRegisterExternalMediaProcessing(kRecordingPerChannel);
}
StopSend();
#ifndef WEBRTC_EXTERNAL_TRANSPORT
StopReceiving();
// De-register packet callback to ensure we're not in a callback when
// deleting channel state, avoids race condition and deadlock.
if (_socketTransportModule.InitializeReceiveSockets(NULL, 0, NULL, NULL, 0)
!= 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"~Channel() failed to de-register receive callback");
}
#endif
StopPlayout();
{
CriticalSectionScoped cs(&_fileCritSect);
if (_inputFilePlayerPtr)
{
_inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
_inputFilePlayerPtr->StopPlayingFile();
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
_inputFilePlayerPtr = NULL;
}
if (_outputFilePlayerPtr)
{
_outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
_outputFilePlayerPtr->StopPlayingFile();
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
_outputFilePlayerPtr = NULL;
}
if (_outputFileRecorderPtr)
{
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
_outputFileRecorderPtr->StopRecording();
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
}
}
// The order to safely shutdown modules in a channel is:
// 1. De-register callbacks in modules
// 2. De-register modules in process thread
// 3. Destroy modules
if (_audioCodingModule.RegisterTransportCallback(NULL) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"~Channel() failed to de-register transport callback"
" (Audio coding module)");
}
if (_audioCodingModule.RegisterVADCallback(NULL) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"~Channel() failed to de-register VAD callback"
" (Audio coding module)");
}
#ifdef WEBRTC_DTMF_DETECTION
if (_audioCodingModule.RegisterIncomingMessagesCallback(NULL) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"~Channel() failed to de-register incoming messages "
"callback (Audio coding module)");
}
#endif
// De-register modules in process thread
#ifndef WEBRTC_EXTERNAL_TRANSPORT
if (_moduleProcessThreadPtr->DeRegisterModule(&_socketTransportModule)
== -1)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"~Channel() failed to deregister socket module");
}
#endif
if (_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()) == -1)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"~Channel() failed to deregister RTP/RTCP module");
}
// Destroy modules
#ifndef WEBRTC_EXTERNAL_TRANSPORT
UdpTransport::Destroy(
&_socketTransportModule);
#endif
AudioCodingModule::Destroy(&_audioCodingModule);
#ifdef WEBRTC_SRTP
SrtpModule::DestroySrtpModule(&_srtpModule);
#endif
if (_rxAudioProcessingModulePtr != NULL)
{
AudioProcessing::Destroy(_rxAudioProcessingModulePtr); // far end APM
_rxAudioProcessingModulePtr = NULL;
}
// End of modules shutdown
// Delete other objects
RtpDump::DestroyRtpDump(&_rtpDumpIn);
RtpDump::DestroyRtpDump(&_rtpDumpOut);
delete [] _encryptionRTPBufferPtr;
delete [] _decryptionRTPBufferPtr;
delete [] _encryptionRTCPBufferPtr;
delete [] _decryptionRTCPBufferPtr;
delete &_callbackCritSect;
delete &_fileCritSect;
}
WebRtc_Word32
Channel::Init()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::Init()");
// --- Initial sanity
if ((_engineStatisticsPtr == NULL) ||
(_moduleProcessThreadPtr == NULL))
{
WEBRTC_TRACE(kTraceError, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::Init() must call SetEngineInformation() first");
return -1;
}
// --- Add modules to process thread (for periodic schedulation)
const bool processThreadFail =
((_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get()) != 0) ||
#ifndef WEBRTC_EXTERNAL_TRANSPORT
(_moduleProcessThreadPtr->RegisterModule(
&_socketTransportModule) != 0));
#else
false);
#endif
if (processThreadFail)
{
_engineStatisticsPtr->SetLastError(
VE_CANNOT_INIT_CHANNEL, kTraceError,
"Channel::Init() modules not registered");
return -1;
}
// --- ACM initialization
if ((_audioCodingModule.InitializeReceiver() == -1) ||
#ifdef WEBRTC_CODEC_AVT
// out-of-band Dtmf tones are played out by default
(_audioCodingModule.SetDtmfPlayoutStatus(true) == -1) ||
#endif
(_audioCodingModule.InitializeSender() == -1))
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"Channel::Init() unable to initialize the ACM - 1");
return -1;
}
// --- RTP/RTCP module initialization
// Ensure that RTCP is enabled by default for the created channel.
// Note that, the module will keep generating RTCP until it is explicitly
// disabled by the user.
// After StopListen (when no sockets exists), RTCP packets will no longer
// be transmitted since the Transport object will then be invalid.
const bool rtpRtcpFail =
((_rtpRtcpModule->SetTelephoneEventStatus(false, true, true) == -1) ||
// RTCP is enabled by default
(_rtpRtcpModule->SetRTCPStatus(kRtcpCompound) == -1));
if (rtpRtcpFail)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"Channel::Init() RTP/RTCP module not initialized");
return -1;
}
// --- Register all permanent callbacks
const bool fail =
(_audioCodingModule.RegisterTransportCallback(this) == -1) ||
(_audioCodingModule.RegisterVADCallback(this) == -1);
if (fail)
{
_engineStatisticsPtr->SetLastError(
VE_CANNOT_INIT_CHANNEL, kTraceError,
"Channel::Init() callbacks not registered");
return -1;
}
// --- Register all supported codecs to the receiving side of the
// RTP/RTCP module
CodecInst codec;
const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
for (int idx = 0; idx < nSupportedCodecs; idx++)
{
// Open up the RTP/RTCP receiver for all supported codecs
if ((_audioCodingModule.Codec(idx, codec) == -1) ||
(_rtpRtcpModule->RegisterReceivePayload(codec) == -1))
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::Init() unable to register %s (%d/%d/%d/%d) "
"to RTP/RTCP receiver",
codec.plname, codec.pltype, codec.plfreq,
codec.channels, codec.rate);
}
else
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::Init() %s (%d/%d/%d/%d) has been added to "
"the RTP/RTCP receiver",
codec.plname, codec.pltype, codec.plfreq,
codec.channels, codec.rate);
}
// Ensure that PCMU is used as default codec on the sending side
if (!STR_CASE_CMP(codec.plname, "PCMU") && (codec.channels == 1))
{
SetSendCodec(codec);
}
// Register default PT for outband 'telephone-event'
if (!STR_CASE_CMP(codec.plname, "telephone-event"))
{
if ((_rtpRtcpModule->RegisterSendPayload(codec) == -1) ||
(_audioCodingModule.RegisterReceiveCodec(codec) == -1))
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::Init() failed to register outband "
"'telephone-event' (%d/%d) correctly",
codec.pltype, codec.plfreq);
}
}
if (!STR_CASE_CMP(codec.plname, "CN"))
{
if ((_audioCodingModule.RegisterSendCodec(codec) == -1) ||
(_audioCodingModule.RegisterReceiveCodec(codec) == -1) ||
(_rtpRtcpModule->RegisterSendPayload(codec) == -1))
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::Init() failed to register CN (%d/%d) "
"correctly - 1",
codec.pltype, codec.plfreq);
}
}
#ifdef WEBRTC_CODEC_RED
// Register RED to the receiving side of the ACM.
// We will not receive an OnInitializeDecoder() callback for RED.
if (!STR_CASE_CMP(codec.plname, "RED"))
{
if (_audioCodingModule.RegisterReceiveCodec(codec) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::Init() failed to register RED (%d/%d) "
"correctly",
codec.pltype, codec.plfreq);
}
}
#endif
}
#ifndef WEBRTC_EXTERNAL_TRANSPORT
// Ensure that the WebRtcSocketTransport implementation is used as
// Transport on the sending side
{
// A lock is needed here since users can call
// RegisterExternalTransport() at the same time.
CriticalSectionScoped cs(&_callbackCritSect);
_transportPtr = &_socketTransportModule;
}
#endif
// Initialize the far end AP module
// Using 8 kHz as initial Fs, the same as in transmission. Might be
// changed at the first receiving audio.
if (_rxAudioProcessingModulePtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_NO_MEMORY, kTraceCritical,
"Channel::Init() failed to create the far-end AudioProcessing"
" module");
return -1;
}
if (_rxAudioProcessingModulePtr->set_sample_rate_hz(8000))
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceWarning,
"Channel::Init() failed to set the sample rate to 8K for"
" far-end AP module");
}
if (_rxAudioProcessingModulePtr->set_num_channels(1, 1) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set channels for the primary audio stream");
}
if (_rxAudioProcessingModulePtr->high_pass_filter()->Enable(
WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceWarning,
"Channel::Init() failed to set the high-pass filter for"
" far-end AP module");
}
if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(
(NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceWarning,
"Init() failed to set noise reduction level for far-end"
" AP module");
}
if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(
WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceWarning,
"Init() failed to set noise reduction state for far-end"
" AP module");
}
if (_rxAudioProcessingModulePtr->gain_control()->set_mode(
(GainControl::Mode)WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceWarning,
"Init() failed to set AGC mode for far-end AP module");
}
if (_rxAudioProcessingModulePtr->gain_control()->Enable(
WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceWarning,
"Init() failed to set AGC state for far-end AP module");
}
return 0;
}
WebRtc_Word32
Channel::SetEngineInformation(Statistics& engineStatistics,
OutputMixer& outputMixer,
voe::TransmitMixer& transmitMixer,
ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
VoiceEngineObserver* voiceEngineObserver,
CriticalSectionWrapper* callbackCritSect)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetEngineInformation()");
_engineStatisticsPtr = &engineStatistics;
_outputMixerPtr = &outputMixer;
_transmitMixerPtr = &transmitMixer,
_moduleProcessThreadPtr = &moduleProcessThread;
_audioDeviceModulePtr = &audioDeviceModule;
_voiceEngineObserverPtr = voiceEngineObserver;
_callbackCritSectPtr = callbackCritSect;
return 0;
}
WebRtc_Word32
Channel::UpdateLocalTimeStamp()
{
_timeStamp += _audioFrame.samples_per_channel_;
return 0;
}
WebRtc_Word32
Channel::StartPlayout()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartPlayout()");
if (_playing)
{
return 0;
}
// Add participant as candidates for mixing.
if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
"StartPlayout() failed to add participant to mixer");
return -1;
}
_playing = true;
if (RegisterFilePlayingToMixer() != 0)
return -1;
return 0;
}
WebRtc_Word32
Channel::StopPlayout()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StopPlayout()");
if (!_playing)
{
return 0;
}
// Remove participant as candidates for mixing
if (_outputMixerPtr->SetMixabilityStatus(*this, false) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
"StartPlayout() failed to remove participant from mixer");
return -1;
}
_playing = false;
_outputAudioLevel.Clear();
return 0;
}
WebRtc_Word32
Channel::StartSend()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartSend()");
{
// A lock is needed because |_sending| can be accessed or modified by
// another thread at the same time.
CriticalSectionScoped cs(&_callbackCritSect);
if (_sending)
{
return 0;
}
_sending = true;
}
if (_rtpRtcpModule->SetSendingStatus(true) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"StartSend() RTP/RTCP failed to start sending");
CriticalSectionScoped cs(&_callbackCritSect);
_sending = false;
return -1;
}
return 0;
}
WebRtc_Word32
Channel::StopSend()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StopSend()");
{
// A lock is needed because |_sending| can be accessed or modified by
// another thread at the same time.
CriticalSectionScoped cs(&_callbackCritSect);
if (!_sending)
{
return 0;
}
_sending = false;
}
// Reset sending SSRC and sequence number and triggers direct transmission
// of RTCP BYE
if (_rtpRtcpModule->SetSendingStatus(false) == -1 ||
_rtpRtcpModule->ResetSendDataCountersRTP() == -1)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
"StartSend() RTP/RTCP failed to stop sending");
}
return 0;
}
WebRtc_Word32
Channel::StartReceiving()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartReceiving()");
if (_receiving)
{
return 0;
}
// If external transport is used, we will only initialize/set the variables
// after this section, since we are not using the WebRtc transport but
// still need to keep track of e.g. if we are receiving.
#ifndef WEBRTC_EXTERNAL_TRANSPORT
if (!_externalTransport)
{
if (!_socketTransportModule.ReceiveSocketsInitialized())
{
_engineStatisticsPtr->SetLastError(
VE_SOCKETS_NOT_INITED, kTraceError,
"StartReceive() must set local receiver first");
return -1;
}
if (_socketTransportModule.StartReceiving(KNumberOfSocketBuffers) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
"StartReceiving() failed to start receiving");
return -1;
}
}
#endif
_receiving = true;
_numberOfDiscardedPackets = 0;
return 0;
}
WebRtc_Word32
Channel::StopReceiving()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StopReceiving()");
if (!_receiving)
{
return 0;
}
#ifndef WEBRTC_EXTERNAL_TRANSPORT
if (!_externalTransport &&
_socketTransportModule.ReceiveSocketsInitialized())
{
if (_socketTransportModule.StopReceiving() != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
"StopReceiving() failed to stop receiving.");
return -1;
}
}
#endif
bool dtmfDetection = _rtpRtcpModule->TelephoneEvent();
// Recover DTMF detection status.
WebRtc_Word32 ret = _rtpRtcpModule->SetTelephoneEventStatus(dtmfDetection,
true, true);
if (ret != 0) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"StopReceiving() failed to restore telephone-event status.");
}
RegisterReceiveCodecsToRTPModule();
_receiving = false;
return 0;
}
#ifndef WEBRTC_EXTERNAL_TRANSPORT
WebRtc_Word32
Channel::SetLocalReceiver(const WebRtc_UWord16 rtpPort,
const WebRtc_UWord16 rtcpPort,
const char ipAddr[64],
const char multicastIpAddr[64])
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetLocalReceiver()");
if (_externalTransport)
{
_engineStatisticsPtr->SetLastError(
VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
"SetLocalReceiver() conflict with external transport");
return -1;
}
if (_sending)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_SENDING, kTraceError,
"SetLocalReceiver() already sending");
return -1;
}
if (_receiving)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_LISTENING, kTraceError,
"SetLocalReceiver() already receiving");
return -1;
}
if (_socketTransportModule.InitializeReceiveSockets(this,
rtpPort,
ipAddr,
multicastIpAddr,
rtcpPort) != 0)
{
UdpTransport::ErrorCode lastSockError(
_socketTransportModule.LastError());
switch (lastSockError)
{
case UdpTransport::kIpAddressInvalid:
_engineStatisticsPtr->SetLastError(
VE_INVALID_IP_ADDRESS, kTraceError,
"SetLocalReceiver() invalid IP address");
break;
case UdpTransport::kSocketInvalid:
_engineStatisticsPtr->SetLastError(
VE_SOCKET_ERROR, kTraceError,
"SetLocalReceiver() invalid socket");
break;
case UdpTransport::kPortInvalid:
_engineStatisticsPtr->SetLastError(
VE_INVALID_PORT_NMBR, kTraceError,
"SetLocalReceiver() invalid port");
break;
case UdpTransport::kFailedToBindPort:
_engineStatisticsPtr->SetLastError(
VE_BINDING_SOCKET_TO_LOCAL_ADDRESS_FAILED, kTraceError,
"SetLocalReceiver() binding failed");
break;
default:
_engineStatisticsPtr->SetLastError(
VE_SOCKET_ERROR, kTraceError,
"SetLocalReceiver() undefined socket error");
break;
}
return -1;
}
return 0;
}
#endif
#ifndef WEBRTC_EXTERNAL_TRANSPORT
WebRtc_Word32
Channel::GetLocalReceiver(int& port, int& RTCPport, char ipAddr[64])
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetLocalReceiver()");
if (_externalTransport)
{
_engineStatisticsPtr->SetLastError(
VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
"SetLocalReceiver() conflict with external transport");
return -1;
}
char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
WebRtc_UWord16 rtpPort(0);
WebRtc_UWord16 rtcpPort(0);
char multicastIpAddr[UdpTransport::kIpAddressVersion6Length] = {0};
// Acquire socket information from the socket module
if (_socketTransportModule.ReceiveSocketInformation(ipAddrTmp,
rtpPort,
rtcpPort,
multicastIpAddr) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_CANNOT_GET_SOCKET_INFO, kTraceError,
"GetLocalReceiver() unable to retrieve socket information");
return -1;
}
// Deliver valid results to the user
port = static_cast<int> (rtpPort);
RTCPport = static_cast<int> (rtcpPort);
if (ipAddr != NULL)
{
strcpy(ipAddr, ipAddrTmp);
}
return 0;
}
#endif
#ifndef WEBRTC_EXTERNAL_TRANSPORT
WebRtc_Word32
Channel::SetSendDestination(const WebRtc_UWord16 rtpPort,
const char ipAddr[64],
const int sourcePort,
const WebRtc_UWord16 rtcpPort)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetSendDestination()");
if (_externalTransport)
{
_engineStatisticsPtr->SetLastError(
VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
"SetSendDestination() conflict with external transport");
return -1;
}
// Initialize ports and IP address for the remote (destination) side.
// By default, the sockets used for receiving are used for transmission as
// well, hence the source ports for outgoing packets are the same as the
// receiving ports specified in SetLocalReceiver.
// If an extra send socket has been created, it will be utilized until a
// new source port is specified or until the channel has been deleted and
// recreated. If no socket exists, sockets will be created when the first
// RTP and RTCP packets shall be transmitted (see e.g.
// UdpTransportImpl::SendPacket()).
//
// NOTE: this function does not require that sockets exists; all it does is
// to build send structures to be used with the sockets when they exist.
// It is therefore possible to call this method before SetLocalReceiver.
// However, sockets must exist if a multi-cast address is given as input.
// Build send structures and enable QoS (if enabled and supported)
if (_socketTransportModule.InitializeSendSockets(
ipAddr, rtpPort, rtcpPort) != UdpTransport::kNoSocketError)
{
UdpTransport::ErrorCode lastSockError(
_socketTransportModule.LastError());
switch (lastSockError)
{
case UdpTransport::kIpAddressInvalid:
_engineStatisticsPtr->SetLastError(
VE_INVALID_IP_ADDRESS, kTraceError,
"SetSendDestination() invalid IP address 1");
break;
case UdpTransport::kSocketInvalid:
_engineStatisticsPtr->SetLastError(
VE_SOCKET_ERROR, kTraceError,
"SetSendDestination() invalid socket 1");
break;
case UdpTransport::kQosError:
_engineStatisticsPtr->SetLastError(
VE_GQOS_ERROR, kTraceError,
"SetSendDestination() failed to set QoS");
break;
case UdpTransport::kMulticastAddressInvalid:
_engineStatisticsPtr->SetLastError(
VE_INVALID_MULTICAST_ADDRESS, kTraceError,
"SetSendDestination() invalid multicast address");
break;
default:
_engineStatisticsPtr->SetLastError(
VE_SOCKET_ERROR, kTraceError,
"SetSendDestination() undefined socket error 1");
break;
}
return -1;
}
// Check if the user has specified a non-default source port different from
// the local receive port.
// If so, an extra local socket will be created unless the source port is
// not unique.
if (sourcePort != kVoEDefault)
{
WebRtc_UWord16 receiverRtpPort(0);
WebRtc_UWord16 rtcpNA(0);
if (_socketTransportModule.ReceiveSocketInformation(NULL,
receiverRtpPort,
rtcpNA,
NULL) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_CANNOT_GET_SOCKET_INFO, kTraceError,
"SetSendDestination() failed to retrieve socket information");
return -1;
}
WebRtc_UWord16 sourcePortUW16 =
static_cast<WebRtc_UWord16> (sourcePort);
// An extra socket will only be created if the specified source port
// differs from the local receive port.
if (sourcePortUW16 != receiverRtpPort)
{
// Initialize extra local socket to get a different source port
// than the local
// receiver port. Always use default source for RTCP.
// Note that, this calls UdpTransport::CloseSendSockets().
if (_socketTransportModule.InitializeSourcePorts(
sourcePortUW16,
sourcePortUW16+1) != 0)
{
UdpTransport::ErrorCode lastSockError(
_socketTransportModule.LastError());
switch (lastSockError)
{
case UdpTransport::kIpAddressInvalid:
_engineStatisticsPtr->SetLastError(
VE_INVALID_IP_ADDRESS, kTraceError,
"SetSendDestination() invalid IP address 2");
break;
case UdpTransport::kSocketInvalid:
_engineStatisticsPtr->SetLastError(
VE_SOCKET_ERROR, kTraceError,
"SetSendDestination() invalid socket 2");
break;
default:
_engineStatisticsPtr->SetLastError(
VE_SOCKET_ERROR, kTraceError,
"SetSendDestination() undefined socket error 2");
break;
}
return -1;
}
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"SetSendDestination() extra local socket is created"
" to facilitate unique source port");
}
else
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"SetSendDestination() sourcePort equals the local"
" receive port => no extra socket is created");
}
}
return 0;
}
#endif
#ifndef WEBRTC_EXTERNAL_TRANSPORT
WebRtc_Word32
Channel::GetSendDestination(int& port,
char ipAddr[64],
int& sourcePort,
int& RTCPport)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetSendDestination()");
if (_externalTransport)
{
_engineStatisticsPtr->SetLastError(
VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
"GetSendDestination() conflict with external transport");
return -1;
}
char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
WebRtc_UWord16 rtpPort(0);
WebRtc_UWord16 rtcpPort(0);
WebRtc_UWord16 rtpSourcePort(0);
WebRtc_UWord16 rtcpSourcePort(0);
// Acquire sending socket information from the socket module
_socketTransportModule.SendSocketInformation(ipAddrTmp, rtpPort, rtcpPort);
_socketTransportModule.SourcePorts(rtpSourcePort, rtcpSourcePort);
// Deliver valid results to the user
port = static_cast<int> (rtpPort);
RTCPport = static_cast<int> (rtcpPort);
sourcePort = static_cast<int> (rtpSourcePort);
if (ipAddr != NULL)
{
strcpy(ipAddr, ipAddrTmp);
}
return 0;
}
#endif
WebRtc_Word32
Channel::SetNetEQPlayoutMode(NetEqModes mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetNetEQPlayoutMode()");
AudioPlayoutMode playoutMode(voice);
switch (mode)
{
case kNetEqDefault:
playoutMode = voice;
break;
case kNetEqStreaming:
playoutMode = streaming;
break;
case kNetEqFax:
playoutMode = fax;
break;
}
if (_audioCodingModule.SetPlayoutMode(playoutMode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetNetEQPlayoutMode() failed to set playout mode");
return -1;
}
return 0;
}
WebRtc_Word32
Channel::GetNetEQPlayoutMode(NetEqModes& mode)
{
const AudioPlayoutMode playoutMode = _audioCodingModule.PlayoutMode();
switch (playoutMode)
{
case voice:
mode = kNetEqDefault;
break;
case streaming:
mode = kNetEqStreaming;
break;
case fax:
mode = kNetEqFax;
break;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::GetNetEQPlayoutMode() => mode=%u", mode);
return 0;
}
WebRtc_Word32
Channel::SetNetEQBGNMode(NetEqBgnModes mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetNetEQPlayoutMode()");
ACMBackgroundNoiseMode noiseMode(On);
switch (mode)
{
case kBgnOn:
noiseMode = On;
break;
case kBgnFade:
noiseMode = Fade;
break;
case kBgnOff:
noiseMode = Off;
break;
}
if (_audioCodingModule.SetBackgroundNoiseMode(noiseMode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetBackgroundNoiseMode() failed to set noise mode");
return -1;
}
return 0;
}
WebRtc_Word32
Channel::SetOnHoldStatus(bool enable, OnHoldModes mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetOnHoldStatus()");
if (mode == kHoldSendAndPlay)
{
_outputIsOnHold = enable;
_inputIsOnHold = enable;
}
else if (mode == kHoldPlayOnly)
{
_outputIsOnHold = enable;
}
if (mode == kHoldSendOnly)
{
_inputIsOnHold = enable;
}
return 0;
}
WebRtc_Word32
Channel::GetOnHoldStatus(bool& enabled, OnHoldModes& mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetOnHoldStatus()");
enabled = (_outputIsOnHold || _inputIsOnHold);
if (_outputIsOnHold && _inputIsOnHold)
{
mode = kHoldSendAndPlay;
}
else if (_outputIsOnHold && !_inputIsOnHold)
{
mode = kHoldPlayOnly;
}
else if (!_outputIsOnHold && _inputIsOnHold)
{
mode = kHoldSendOnly;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetOnHoldStatus() => enabled=%d, mode=%d",
enabled, mode);
return 0;
}
WebRtc_Word32
Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RegisterVoiceEngineObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_voiceEngineObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"RegisterVoiceEngineObserver() observer already enabled");
return -1;
}
_voiceEngineObserverPtr = &observer;
return 0;
}
WebRtc_Word32
Channel::DeRegisterVoiceEngineObserver()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DeRegisterVoiceEngineObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_voiceEngineObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterVoiceEngineObserver() observer already disabled");
return 0;
}
_voiceEngineObserverPtr = NULL;
return 0;
}
WebRtc_Word32
Channel::GetNetEQBGNMode(NetEqBgnModes& mode)
{
ACMBackgroundNoiseMode noiseMode(On);
_audioCodingModule.BackgroundNoiseMode(noiseMode);
switch (noiseMode)
{
case On:
mode = kBgnOn;
break;
case Fade:
mode = kBgnFade;
break;
case Off:
mode = kBgnOff;
break;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetNetEQBGNMode() => mode=%u", mode);
return 0;
}
WebRtc_Word32
Channel::GetSendCodec(CodecInst& codec)
{
return (_audioCodingModule.SendCodec(codec));
}
WebRtc_Word32
Channel::GetRecCodec(CodecInst& codec)
{
return (_audioCodingModule.ReceiveCodec(codec));
}
WebRtc_Word32
Channel::SetSendCodec(const CodecInst& codec)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetSendCodec()");
if (_audioCodingModule.RegisterSendCodec(codec) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"SetSendCodec() failed to register codec to ACM");
return -1;
}
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
{
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
{
WEBRTC_TRACE(
kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"SetSendCodec() failed to register codec to"
" RTP/RTCP module");
return -1;
}
}
if (_rtpRtcpModule->SetAudioPacketSize(codec.pacsize) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"SetSendCodec() failed to set audio packet size");
return -1;
}
return 0;
}
WebRtc_Word32
Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetVADStatus(mode=%d)", mode);
// To disable VAD, DTX must be disabled too
disableDTX = ((enableVAD == false) ? true : disableDTX);
if (_audioCodingModule.SetVAD(!disableDTX, enableVAD, mode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetVADStatus() failed to set VAD");
return -1;
}
return 0;
}
WebRtc_Word32
Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetVADStatus");
if (_audioCodingModule.VAD(disabledDTX, enabledVAD, mode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"GetVADStatus() failed to get VAD status");
return -1;
}
disabledDTX = !disabledDTX;
return 0;
}
WebRtc_Word32
Channel::SetRecPayloadType(const CodecInst& codec)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetRecPayloadType()");
if (_playing)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_PLAYING, kTraceError,
"SetRecPayloadType() unable to set PT while playing");
return -1;
}
if (_receiving)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_LISTENING, kTraceError,
"SetRecPayloadType() unable to set PT while listening");
return -1;
}
if (codec.pltype == -1)
{
// De-register the selected codec (RTP/RTCP module and ACM)
WebRtc_Word8 pltype(-1);
CodecInst rxCodec = codec;
// Get payload type for the given codec
_rtpRtcpModule->ReceivePayloadType(rxCodec, &pltype);
rxCodec.pltype = pltype;
if (_rtpRtcpModule->DeRegisterReceivePayload(pltype) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR,
kTraceError,
"SetRecPayloadType() RTP/RTCP-module deregistration "
"failed");
return -1;
}
if (_audioCodingModule.UnregisterReceiveCodec(rxCodec.pltype) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetRecPayloadType() ACM deregistration failed - 1");
return -1;
}
return 0;
}
if (_rtpRtcpModule->RegisterReceivePayload(codec) != 0)
{
// First attempt to register failed => de-register and try again
_rtpRtcpModule->DeRegisterReceivePayload(codec.pltype);
if (_rtpRtcpModule->RegisterReceivePayload(codec) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetRecPayloadType() RTP/RTCP-module registration failed");
return -1;
}
}
if (_audioCodingModule.RegisterReceiveCodec(codec) != 0)
{
_audioCodingModule.UnregisterReceiveCodec(codec.pltype);
if (_audioCodingModule.RegisterReceiveCodec(codec) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetRecPayloadType() ACM registration failed - 1");
return -1;
}
}
return 0;
}
WebRtc_Word32
Channel::GetRecPayloadType(CodecInst& codec)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRecPayloadType()");
WebRtc_Word8 payloadType(-1);
if (_rtpRtcpModule->ReceivePayloadType(codec, &payloadType) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
"GetRecPayloadType() failed to retrieve RX payload type");
return -1;
}
codec.pltype = payloadType;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRecPayloadType() => pltype=%u", codec.pltype);
return 0;
}
WebRtc_Word32
Channel::SetAMREncFormat(AmrMode mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetAMREncFormat()");
// ACM doesn't support AMR
return -1;
}
WebRtc_Word32
Channel::SetAMRDecFormat(AmrMode mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetAMRDecFormat()");
// ACM doesn't support AMR
return -1;
}
WebRtc_Word32
Channel::SetAMRWbEncFormat(AmrMode mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetAMRWbEncFormat()");
// ACM doesn't support AMR
return -1;
}
WebRtc_Word32
Channel::SetAMRWbDecFormat(AmrMode mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetAMRWbDecFormat()");
// ACM doesn't support AMR
return -1;
}
WebRtc_Word32
Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetSendCNPayloadType()");
CodecInst codec;
WebRtc_Word32 samplingFreqHz(-1);
const int kMono = 1;
if (frequency == kFreq32000Hz)
samplingFreqHz = 32000;
else if (frequency == kFreq16000Hz)
samplingFreqHz = 16000;
if (_audioCodingModule.Codec("CN", codec, samplingFreqHz, kMono) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetSendCNPayloadType() failed to retrieve default CN codec "
"settings");
return -1;
}
// Modify the payload type (must be set to dynamic range)
codec.pltype = type;
if (_audioCodingModule.RegisterSendCodec(codec) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetSendCNPayloadType() failed to register CN to ACM");
return -1;
}
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
{
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetSendCNPayloadType() failed to register CN to RTP/RTCP "
"module");
return -1;
}
}
return 0;
}
WebRtc_Word32
Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetISACInitTargetRate()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
"SetISACInitTargetRate() failed to retrieve send codec");
return -1;
}
if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
{
// This API is only valid if iSAC is setup to run in channel-adaptive
// mode.
// We do not validate the adaptive mode here. It is done later in the
// ConfigISACBandwidthEstimator() API.
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
"SetISACInitTargetRate() send codec is not iSAC");
return -1;
}
WebRtc_UWord8 initFrameSizeMsec(0);
if (16000 == sendCodec.plfreq)
{
// Note that 0 is a valid and corresponds to "use default
if ((rateBps != 0 &&
rateBps < kVoiceEngineMinIsacInitTargetRateBpsWb) ||
(rateBps > kVoiceEngineMaxIsacInitTargetRateBpsWb))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetISACInitTargetRate() invalid target rate - 1");
return -1;
}
// 30 or 60ms
initFrameSizeMsec = (WebRtc_UWord8)(sendCodec.pacsize / 16);
}
else if (32000 == sendCodec.plfreq)
{
if ((rateBps != 0 &&
rateBps < kVoiceEngineMinIsacInitTargetRateBpsSwb) ||
(rateBps > kVoiceEngineMaxIsacInitTargetRateBpsSwb))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetISACInitTargetRate() invalid target rate - 2");
return -1;
}
initFrameSizeMsec = (WebRtc_UWord8)(sendCodec.pacsize / 32); // 30ms
}
if (_audioCodingModule.ConfigISACBandwidthEstimator(
initFrameSizeMsec, rateBps, useFixedFrameSize) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetISACInitTargetRate() iSAC BWE config failed");
return -1;
}
return 0;
}
WebRtc_Word32
Channel::SetISACMaxRate(int rateBps)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetISACMaxRate()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
"SetISACMaxRate() failed to retrieve send codec");
return -1;
}
if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
{
// This API is only valid if iSAC is selected as sending codec.
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
"SetISACMaxRate() send codec is not iSAC");
return -1;
}
if (16000 == sendCodec.plfreq)
{
if ((rateBps < kVoiceEngineMinIsacMaxRateBpsWb) ||
(rateBps > kVoiceEngineMaxIsacMaxRateBpsWb))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetISACMaxRate() invalid max rate - 1");
return -1;
}
}
else if (32000 == sendCodec.plfreq)
{
if ((rateBps < kVoiceEngineMinIsacMaxRateBpsSwb) ||
(rateBps > kVoiceEngineMaxIsacMaxRateBpsSwb))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetISACMaxRate() invalid max rate - 2");
return -1;
}
}
if (_sending)
{
_engineStatisticsPtr->SetLastError(
VE_SENDING, kTraceError,
"SetISACMaxRate() unable to set max rate while sending");
return -1;
}
// Set the maximum instantaneous rate of iSAC (works for both adaptive
// and non-adaptive mode)
if (_audioCodingModule.SetISACMaxRate(rateBps) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetISACMaxRate() failed to set max rate");
return -1;
}
return 0;
}
WebRtc_Word32
Channel::SetISACMaxPayloadSize(int sizeBytes)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetISACMaxPayloadSize()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
"SetISACMaxPayloadSize() failed to retrieve send codec");
return -1;
}
if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
"SetISACMaxPayloadSize() send codec is not iSAC");
return -1;
}
if (16000 == sendCodec.plfreq)
{
if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesWb) ||
(sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesWb))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetISACMaxPayloadSize() invalid max payload - 1");
return -1;
}
}
else if (32000 == sendCodec.plfreq)
{
if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesSwb) ||
(sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetISACMaxPayloadSize() invalid max payload - 2");
return -1;
}
}
if (_sending)
{
_engineStatisticsPtr->SetLastError(
VE_SENDING, kTraceError,
"SetISACMaxPayloadSize() unable to set max rate while sending");
return -1;
}
if (_audioCodingModule.SetISACMaxPayloadSize(sizeBytes) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetISACMaxPayloadSize() failed to set max payload size");
return -1;
}
return 0;
}
WebRtc_Word32 Channel::RegisterExternalTransport(Transport& transport)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterExternalTransport()");
CriticalSectionScoped cs(&_callbackCritSect);
#ifndef WEBRTC_EXTERNAL_TRANSPORT
// Sanity checks for default (non external transport) to avoid conflict with
// WebRtc sockets.
if (_socketTransportModule.SendSocketsInitialized())
{
_engineStatisticsPtr->SetLastError(VE_SEND_SOCKETS_CONFLICT,
kTraceError,
"RegisterExternalTransport() send sockets already initialized");
return -1;
}
if (_socketTransportModule.ReceiveSocketsInitialized())
{
_engineStatisticsPtr->SetLastError(VE_RECEIVE_SOCKETS_CONFLICT,
kTraceError,
"RegisterExternalTransport() receive sockets already initialized");
return -1;
}
#endif
if (_externalTransport)
{
_engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION,
kTraceError,
"RegisterExternalTransport() external transport already enabled");
return -1;
}
_externalTransport = true;
_transportPtr = &transport;
return 0;
}
WebRtc_Word32
Channel::DeRegisterExternalTransport()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DeRegisterExternalTransport()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_transportPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterExternalTransport() external transport already "
"disabled");
return 0;
}
_externalTransport = false;
#ifdef WEBRTC_EXTERNAL_TRANSPORT
_transportPtr = NULL;
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"DeRegisterExternalTransport() all transport is disabled");
#else
_transportPtr = &_socketTransportModule;
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"DeRegisterExternalTransport() internal Transport is enabled");
#endif
return 0;
}
WebRtc_Word32
Channel::ReceivedRTPPacket(const WebRtc_Word8* data, WebRtc_Word32 length)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::ReceivedRTPPacket()");
const char dummyIP[] = "127.0.0.1";
IncomingRTPPacket(data, length, dummyIP, 0);
return 0;
}
WebRtc_Word32
Channel::ReceivedRTCPPacket(const WebRtc_Word8* data, WebRtc_Word32 length)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::ReceivedRTCPPacket()");
const char dummyIP[] = "127.0.0.1";
IncomingRTCPPacket(data, length, dummyIP, 0);
return 0;
}
#ifndef WEBRTC_EXTERNAL_TRANSPORT
WebRtc_Word32
Channel::GetSourceInfo(int& rtpPort, int& rtcpPort, char ipAddr[64])
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetSourceInfo()");
WebRtc_UWord16 rtpPortModule;
WebRtc_UWord16 rtcpPortModule;
char ipaddr[UdpTransport::kIpAddressVersion6Length] = {0};
if (_socketTransportModule.RemoteSocketInformation(ipaddr,
rtpPortModule,
rtcpPortModule) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
"GetSourceInfo() failed to retrieve remote socket information");
return -1;
}
strcpy(ipAddr, ipaddr);
rtpPort = rtpPortModule;
rtcpPort = rtcpPortModule;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"GetSourceInfo() => rtpPort=%d, rtcpPort=%d, ipAddr=%s",
rtpPort, rtcpPort, ipAddr);
return 0;
}
WebRtc_Word32
Channel::EnableIPv6()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EnableIPv6()");
if (_socketTransportModule.ReceiveSocketsInitialized() ||
_socketTransportModule.SendSocketsInitialized())
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"EnableIPv6() socket layer is already initialized");
return -1;
}
if (_socketTransportModule.EnableIpV6() != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_ERROR, kTraceError,
"EnableIPv6() failed to enable IPv6");
const UdpTransport::ErrorCode lastError =
_socketTransportModule.LastError();
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"UdpTransport::LastError() => %d", lastError);
return -1;
}
return 0;
}
bool
Channel::IPv6IsEnabled() const
{
bool isEnabled = _socketTransportModule.IpV6Enabled();
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"IPv6IsEnabled() => %d", isEnabled);
return isEnabled;
}
WebRtc_Word32
Channel::SetSourceFilter(int rtpPort, int rtcpPort, const char ipAddr[64])
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetSourceFilter()");
if (_socketTransportModule.SetFilterPorts(
static_cast<WebRtc_UWord16>(rtpPort),
static_cast<WebRtc_UWord16>(rtcpPort)) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
"SetSourceFilter() failed to set filter ports");
const UdpTransport::ErrorCode lastError =
_socketTransportModule.LastError();
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"UdpTransport::LastError() => %d",
lastError);
return -1;
}
const char* filterIpAddress = ipAddr;
if (_socketTransportModule.SetFilterIP(filterIpAddress) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_IP_ADDRESS, kTraceError,
"SetSourceFilter() failed to set filter IP address");
const UdpTransport::ErrorCode lastError =
_socketTransportModule.LastError();
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"UdpTransport::LastError() => %d", lastError);
return -1;
}
return 0;
}
WebRtc_Word32
Channel::GetSourceFilter(int& rtpPort, int& rtcpPort, char ipAddr[64])
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetSourceFilter()");
WebRtc_UWord16 rtpFilterPort(0);
WebRtc_UWord16 rtcpFilterPort(0);
if (_socketTransportModule.FilterPorts(rtpFilterPort, rtcpFilterPort) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
"GetSourceFilter() failed to retrieve filter ports");
}
char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
if (_socketTransportModule.FilterIP(ipAddrTmp) != 0)
{
// no filter has been configured (not seen as an error)
memset(ipAddrTmp,
0, UdpTransport::kIpAddressVersion6Length);
}
rtpPort = static_cast<int> (rtpFilterPort);
rtcpPort = static_cast<int> (rtcpFilterPort);
strcpy(ipAddr, ipAddrTmp);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"GetSourceFilter() => rtpPort=%d, rtcpPort=%d, ipAddr=%s",
rtpPort, rtcpPort, ipAddr);
return 0;
}
WebRtc_Word32
Channel::SetSendTOS(int DSCP, int priority, bool useSetSockopt)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetSendTOS(DSCP=%d, useSetSockopt=%d)",
DSCP, (int)useSetSockopt);
// Set TOS value and possibly try to force usage of setsockopt()
if (_socketTransportModule.SetToS(DSCP, useSetSockopt) != 0)
{
UdpTransport::ErrorCode lastSockError(
_socketTransportModule.LastError());
switch (lastSockError)
{
case UdpTransport::kTosError:
_engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
"SetSendTOS() TOS error");
break;
case UdpTransport::kQosError:
_engineStatisticsPtr->SetLastError(
VE_TOS_GQOS_CONFLICT, kTraceError,
"SetSendTOS() GQOS error");
break;
case UdpTransport::kTosInvalid:
// can't switch SetSockOpt method without disabling TOS first, or
// SetSockopt() call failed
_engineStatisticsPtr->SetLastError(VE_TOS_INVALID, kTraceError,
"SetSendTOS() invalid TOS");
break;
case UdpTransport::kSocketInvalid:
_engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError,
"SetSendTOS() invalid Socket");
break;
default:
_engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
"SetSendTOS() TOS error");
break;
}
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"UdpTransport => lastError = %d",
lastSockError);
return -1;
}
// Set priority (PCP) value, -1 means don't change
if (-1 != priority)
{
if (_socketTransportModule.SetPCP(priority) != 0)
{
UdpTransport::ErrorCode lastSockError(
_socketTransportModule.LastError());
switch (lastSockError)
{
case UdpTransport::kPcpError:
_engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
"SetSendTOS() PCP error");
break;
case UdpTransport::kQosError:
_engineStatisticsPtr->SetLastError(
VE_TOS_GQOS_CONFLICT, kTraceError,
"SetSendTOS() GQOS conflict");
break;
case UdpTransport::kSocketInvalid:
_engineStatisticsPtr->SetLastError(
VE_SOCKET_ERROR, kTraceError,
"SetSendTOS() invalid Socket");
break;
default:
_engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
"SetSendTOS() PCP error");
break;
}
WEBRTC_TRACE(kTraceError, kTraceVoice,
VoEId(_instanceId,_channelId),
"UdpTransport => lastError = %d",
lastSockError);
return -1;
}
}
return 0;
}
WebRtc_Word32
Channel::GetSendTOS(int &DSCP, int& priority, bool &useSetSockopt)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetSendTOS(DSCP=?, useSetSockopt=?)");
WebRtc_Word32 dscp(0), prio(0);
bool setSockopt(false);
if (_socketTransportModule.ToS(dscp, setSockopt) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
"GetSendTOS() failed to get TOS info");
return -1;
}
if (_socketTransportModule.PCP(prio) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
"GetSendTOS() failed to get PCP info");
return -1;
}
DSCP = static_cast<int> (dscp);
priority = static_cast<int> (prio);
useSetSockopt = setSockopt;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
"GetSendTOS() => DSCP=%d, priority=%d, useSetSockopt=%d",
DSCP, priority, (int)useSetSockopt);
return 0;
}
#if defined(_WIN32)
WebRtc_Word32
Channel::SetSendGQoS(bool enable, int serviceType, int overrideDSCP)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetSendGQoS(enable=%d, serviceType=%d, "
"overrideDSCP=%d)",
(int)enable, serviceType, overrideDSCP);
if(!_socketTransportModule.ReceiveSocketsInitialized())
{
_engineStatisticsPtr->SetLastError(
VE_SOCKETS_NOT_INITED, kTraceError,
"SetSendGQoS() GQoS state must be set after sockets are created");
return -1;
}
if(!_socketTransportModule.SendSocketsInitialized())
{
_engineStatisticsPtr->SetLastError(
VE_DESTINATION_NOT_INITED, kTraceError,
"SetSendGQoS() GQoS state must be set after sending side is "
"initialized");
return -1;
}
if (enable &&
(serviceType != SERVICETYPE_BESTEFFORT) &&
(serviceType != SERVICETYPE_CONTROLLEDLOAD) &&
(serviceType != SERVICETYPE_GUARANTEED) &&
(serviceType != SERVICETYPE_QUALITATIVE))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetSendGQoS() Invalid service type");
return -1;
}
if (enable && ((overrideDSCP < 0) || (overrideDSCP > 63)))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetSendGQoS() Invalid overrideDSCP value");
return -1;
}
// Avoid GQoS/ToS conflict when user wants to override the default DSCP
// mapping
bool QoS(false);
WebRtc_Word32 sType(0);
WebRtc_Word32 ovrDSCP(0);
if (_socketTransportModule.QoS(QoS, sType, ovrDSCP))
{
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
"SetSendGQoS() failed to get QOS info");
return -1;
}
if (QoS && ovrDSCP == 0 && overrideDSCP != 0)
{
_engineStatisticsPtr->SetLastError(
VE_TOS_GQOS_CONFLICT, kTraceError,
"SetSendGQoS() QOS is already enabled and overrideDSCP differs,"
" not allowed");
return -1;
}
const WebRtc_Word32 maxBitrate(0);
if (_socketTransportModule.SetQoS(enable,
static_cast<WebRtc_Word32>(serviceType),
maxBitrate,
static_cast<WebRtc_Word32>(overrideDSCP),
true))
{
UdpTransport::ErrorCode lastSockError(
_socketTransportModule.LastError());
switch (lastSockError)
{
case UdpTransport::kQosError:
_engineStatisticsPtr->SetLastError(VE_GQOS_ERROR, kTraceError,
"SetSendGQoS() QOS error");
break;
default:
_engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError,
"SetSendGQoS() Socket error");
break;
}
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"UdpTransport() => lastError = %d",
lastSockError);
return -1;
}
return 0;
}
#endif
#if defined(_WIN32)
WebRtc_Word32
Channel::GetSendGQoS(bool &enabled, int &serviceType, int &overrideDSCP)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetSendGQoS(enable=?, serviceType=?, "
"overrideDSCP=?)");
bool QoS(false);
WebRtc_Word32 serviceTypeModule(0);
WebRtc_Word32 overrideDSCPModule(0);
_socketTransportModule.QoS(QoS, serviceTypeModule, overrideDSCPModule);
enabled = QoS;
serviceType = static_cast<int> (serviceTypeModule);
overrideDSCP = static_cast<int> (overrideDSCPModule);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"GetSendGQoS() => enabled=%d, serviceType=%d, overrideDSCP=%d",
(int)enabled, serviceType, overrideDSCP);
return 0;
}
#endif
#endif
WebRtc_Word32
Channel::SetPacketTimeoutNotification(bool enable, int timeoutSeconds)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetPacketTimeoutNotification()");
if (enable)
{
const WebRtc_UWord32 RTPtimeoutMS = 1000*timeoutSeconds;
const WebRtc_UWord32 RTCPtimeoutMS = 0;
_rtpRtcpModule->SetPacketTimeout(RTPtimeoutMS, RTCPtimeoutMS);
_rtpPacketTimeOutIsEnabled = true;
_rtpTimeOutSeconds = timeoutSeconds;
}
else
{
_rtpRtcpModule->SetPacketTimeout(0, 0);
_rtpPacketTimeOutIsEnabled = false;
_rtpTimeOutSeconds = 0;
}
return 0;
}
WebRtc_Word32
Channel::GetPacketTimeoutNotification(bool& enabled, int& timeoutSeconds)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetPacketTimeoutNotification()");
enabled = _rtpPacketTimeOutIsEnabled;
if (enabled)
{
timeoutSeconds = _rtpTimeOutSeconds;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
"GetPacketTimeoutNotification() => enabled=%d,"
" timeoutSeconds=%d",
enabled, timeoutSeconds);
return 0;
}
WebRtc_Word32
Channel::RegisterDeadOrAliveObserver(VoEConnectionObserver& observer)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RegisterDeadOrAliveObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_connectionObserverPtr)
{
_engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION, kTraceError,
"RegisterDeadOrAliveObserver() observer already enabled");
return -1;
}
_connectionObserverPtr = &observer;
_connectionObserver = true;
return 0;
}
WebRtc_Word32
Channel::DeRegisterDeadOrAliveObserver()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DeRegisterDeadOrAliveObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_connectionObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterDeadOrAliveObserver() observer already disabled");
return 0;
}
_connectionObserver = false;
_connectionObserverPtr = NULL;
return 0;
}
WebRtc_Word32
Channel::SetPeriodicDeadOrAliveStatus(bool enable, int sampleTimeSeconds)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetPeriodicDeadOrAliveStatus()");
if (!_connectionObserverPtr)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"SetPeriodicDeadOrAliveStatus() connection observer has"
" not been registered");
}
if (enable)
{
ResetDeadOrAliveCounters();
}
bool enabled(false);
WebRtc_UWord8 currentSampleTimeSec(0);
// Store last state (will be used later if dead-or-alive is disabled).
_rtpRtcpModule->PeriodicDeadOrAliveStatus(enabled, currentSampleTimeSec);
// Update the dead-or-alive state.
if (_rtpRtcpModule->SetPeriodicDeadOrAliveStatus(
enable, (WebRtc_UWord8)sampleTimeSeconds) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR,
kTraceError,
"SetPeriodicDeadOrAliveStatus() failed to set dead-or-alive "
"status");
return -1;
}
if (!enable)
{
// Restore last utilized sample time.
// Without this, the sample time would always be reset to default
// (2 sec), each time dead-or-alived was disabled without sample-time
// parameter.
_rtpRtcpModule->SetPeriodicDeadOrAliveStatus(enable,
currentSampleTimeSec);
}
return 0;
}
WebRtc_Word32
Channel::GetPeriodicDeadOrAliveStatus(bool& enabled, int& sampleTimeSeconds)
{
_rtpRtcpModule->PeriodicDeadOrAliveStatus(
enabled,
(WebRtc_UWord8&)sampleTimeSeconds);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
"GetPeriodicDeadOrAliveStatus() => enabled=%d,"
" sampleTimeSeconds=%d",
enabled, sampleTimeSeconds);
return 0;
}
WebRtc_Word32
Channel::SendUDPPacket(const void* data,
unsigned int length,
int& transmittedBytes,
bool useRtcpSocket)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SendUDPPacket()");
if (_externalTransport)
{
_engineStatisticsPtr->SetLastError(
VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
"SendUDPPacket() external transport is enabled");
return -1;
}
if (useRtcpSocket && !_rtpRtcpModule->RTCP())
{
_engineStatisticsPtr->SetLastError(
VE_RTCP_ERROR, kTraceError,
"SendUDPPacket() RTCP is disabled");
return -1;
}
if (!_sending)
{
_engineStatisticsPtr->SetLastError(
VE_NOT_SENDING, kTraceError,
"SendUDPPacket() not sending");
return -1;
}
char* dataC = new char[length];
if (NULL == dataC)
{
_engineStatisticsPtr->SetLastError(
VE_NO_MEMORY, kTraceError,
"SendUDPPacket() memory allocation failed");
return -1;
}
memcpy(dataC, data, length);
transmittedBytes = SendPacketRaw(dataC, length, useRtcpSocket);
delete [] dataC;
dataC = NULL;
if (transmittedBytes <= 0)
{
_engineStatisticsPtr->SetLastError(
VE_SEND_ERROR, kTraceError,
"SendUDPPacket() transmission failed");
transmittedBytes = 0;
return -1;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"SendUDPPacket() => transmittedBytes=%d", transmittedBytes);
return 0;
}
int Channel::StartPlayingFileLocally(const char* fileName,
const bool loop,
const FileFormats format,
const int startPosition,
const float volumeScaling,
const int stopPosition,
const CodecInst* codecInst)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartPlayingFileLocally(fileNameUTF8[]=%s, loop=%d,"
" format=%d, volumeScaling=%5.3f, startPosition=%d, "
"stopPosition=%d)", fileName, loop, format, volumeScaling,
startPosition, stopPosition);
if (_outputFilePlaying)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_PLAYING, kTraceError,
"StartPlayingFileLocally() is already playing");
return -1;
}
{
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFilePlayerPtr)
{
_outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
_outputFilePlayerPtr = NULL;
}
_outputFilePlayerPtr = FilePlayer::CreateFilePlayer(
_outputFilePlayerId, (const FileFormats)format);
if (_outputFilePlayerPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartPlayingFileLocally() filePlayer format is not correct");
return -1;
}
const WebRtc_UWord32 notificationTime(0);
if (_outputFilePlayerPtr->StartPlayingFile(
fileName,
loop,
startPosition,
volumeScaling,
notificationTime,
stopPosition,
(const CodecInst*)codecInst) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartPlayingFile() failed to start file playout");
_outputFilePlayerPtr->StopPlayingFile();
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
_outputFilePlayerPtr = NULL;
return -1;
}
_outputFilePlayerPtr->RegisterModuleFileCallback(this);
_outputFilePlaying = true;
}
if (RegisterFilePlayingToMixer() != 0)
return -1;
return 0;
}
int Channel::StartPlayingFileLocally(InStream* stream,
const FileFormats format,
const int startPosition,
const float volumeScaling,
const int stopPosition,
const CodecInst* codecInst)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartPlayingFileLocally(format=%d,"
" volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
format, volumeScaling, startPosition, stopPosition);
if(stream == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartPlayingFileLocally() NULL as input stream");
return -1;
}
if (_outputFilePlaying)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_PLAYING, kTraceError,
"StartPlayingFileLocally() is already playing");
return -1;
}
{
CriticalSectionScoped cs(&_fileCritSect);
// Destroy the old instance
if (_outputFilePlayerPtr)
{
_outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
_outputFilePlayerPtr = NULL;
}
// Create the instance
_outputFilePlayerPtr = FilePlayer::CreateFilePlayer(
_outputFilePlayerId,
(const FileFormats)format);
if (_outputFilePlayerPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartPlayingFileLocally() filePlayer format isnot correct");
return -1;
}
const WebRtc_UWord32 notificationTime(0);
if (_outputFilePlayerPtr->StartPlayingFile(*stream, startPosition,
volumeScaling,
notificationTime,
stopPosition, codecInst) != 0)
{
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
"StartPlayingFile() failed to "
"start file playout");
_outputFilePlayerPtr->StopPlayingFile();
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
_outputFilePlayerPtr = NULL;
return -1;
}
_outputFilePlayerPtr->RegisterModuleFileCallback(this);
_outputFilePlaying = true;
}
if (RegisterFilePlayingToMixer() != 0)
return -1;
return 0;
}
int Channel::StopPlayingFileLocally()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StopPlayingFileLocally()");
if (!_outputFilePlaying)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"StopPlayingFileLocally() isnot playing");
return 0;
}
{
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFilePlayerPtr->StopPlayingFile() != 0)
{
_engineStatisticsPtr->SetLastError(
VE_STOP_RECORDING_FAILED, kTraceError,
"StopPlayingFile() could not stop playing");
return -1;
}
_outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
_outputFilePlayerPtr = NULL;
_outputFilePlaying = false;
}
// _fileCritSect cannot be taken while calling
// SetAnonymousMixibilityStatus. Refer to comments in
// StartPlayingFileLocally(const char* ...) for more details.
if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, false) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
"StopPlayingFile() failed to stop participant from playing as"
"file in the mixer");
return -1;
}
return 0;
}
int Channel::IsPlayingFileLocally() const
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::IsPlayingFileLocally()");
return (WebRtc_Word32)_outputFilePlaying;
}
int Channel::RegisterFilePlayingToMixer()
{
// Return success for not registering for file playing to mixer if:
// 1. playing file before playout is started on that channel.
// 2. starting playout without file playing on that channel.
if (!_playing || !_outputFilePlaying)
{
return 0;
}
// |_fileCritSect| cannot be taken while calling
// SetAnonymousMixabilityStatus() since as soon as the participant is added
// frames can be pulled by the mixer. Since the frames are generated from
// the file, _fileCritSect will be taken. This would result in a deadlock.
if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
{
CriticalSectionScoped cs(&_fileCritSect);
_outputFilePlaying = false;
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
"StartPlayingFile() failed to add participant as file to mixer");
_outputFilePlayerPtr->StopPlayingFile();
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
_outputFilePlayerPtr = NULL;
return -1;
}
return 0;
}
int Channel::ScaleLocalFilePlayout(const float scale)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::ScaleLocalFilePlayout(scale=%5.3f)", scale);
CriticalSectionScoped cs(&_fileCritSect);
if (!_outputFilePlaying)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"ScaleLocalFilePlayout() isnot playing");
return -1;
}
if ((_outputFilePlayerPtr == NULL) ||
(_outputFilePlayerPtr->SetAudioScaling(scale) != 0))
{
_engineStatisticsPtr->SetLastError(
VE_BAD_ARGUMENT, kTraceError,
"SetAudioScaling() failed to scale the playout");
return -1;
}
return 0;
}
int Channel::GetLocalPlayoutPosition(int& positionMs)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetLocalPlayoutPosition(position=?)");
WebRtc_UWord32 position;
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFilePlayerPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"GetLocalPlayoutPosition() filePlayer instance doesnot exist");
return -1;
}
if (_outputFilePlayerPtr->GetPlayoutPosition(position) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"GetLocalPlayoutPosition() failed");
return -1;
}
positionMs = position;
return 0;
}
int Channel::StartPlayingFileAsMicrophone(const char* fileName,
const bool loop,
const FileFormats format,
const int startPosition,
const float volumeScaling,
const int stopPosition,
const CodecInst* codecInst)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartPlayingFileAsMicrophone(fileNameUTF8[]=%s, "
"loop=%d, format=%d, volumeScaling=%5.3f, startPosition=%d, "
"stopPosition=%d)", fileName, loop, format, volumeScaling,
startPosition, stopPosition);
if (_inputFilePlaying)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_PLAYING, kTraceWarning,
"StartPlayingFileAsMicrophone() filePlayer is playing");
return 0;
}
CriticalSectionScoped cs(&_fileCritSect);
// Destroy the old instance
if (_inputFilePlayerPtr)
{
_inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
_inputFilePlayerPtr = NULL;
}
// Create the instance
_inputFilePlayerPtr = FilePlayer::CreateFilePlayer(
_inputFilePlayerId, (const FileFormats)format);
if (_inputFilePlayerPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartPlayingFileAsMicrophone() filePlayer format isnot correct");
return -1;
}
const WebRtc_UWord32 notificationTime(0);
if (_inputFilePlayerPtr->StartPlayingFile(
fileName,
loop,
startPosition,
volumeScaling,
notificationTime,
stopPosition,
(const CodecInst*)codecInst) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartPlayingFile() failed to start file playout");
_inputFilePlayerPtr->StopPlayingFile();
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
_inputFilePlayerPtr = NULL;
return -1;
}
_inputFilePlayerPtr->RegisterModuleFileCallback(this);
_inputFilePlaying = true;
return 0;
}
int Channel::StartPlayingFileAsMicrophone(InStream* stream,
const FileFormats format,
const int startPosition,
const float volumeScaling,
const int stopPosition,
const CodecInst* codecInst)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartPlayingFileAsMicrophone(format=%d, "
"volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
format, volumeScaling, startPosition, stopPosition);
if(stream == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartPlayingFileAsMicrophone NULL as input stream");
return -1;
}
if (_inputFilePlaying)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_PLAYING, kTraceWarning,
"StartPlayingFileAsMicrophone() is playing");
return 0;
}
CriticalSectionScoped cs(&_fileCritSect);
// Destroy the old instance
if (_inputFilePlayerPtr)
{
_inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
_inputFilePlayerPtr = NULL;
}
// Create the instance
_inputFilePlayerPtr = FilePlayer::CreateFilePlayer(
_inputFilePlayerId, (const FileFormats)format);
if (_inputFilePlayerPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartPlayingInputFile() filePlayer format isnot correct");
return -1;
}
const WebRtc_UWord32 notificationTime(0);
if (_inputFilePlayerPtr->StartPlayingFile(*stream, startPosition,
volumeScaling, notificationTime,
stopPosition, codecInst) != 0)
{
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
"StartPlayingFile() failed to start "
"file playout");
_inputFilePlayerPtr->StopPlayingFile();
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
_inputFilePlayerPtr = NULL;
return -1;
}
_inputFilePlayerPtr->RegisterModuleFileCallback(this);
_inputFilePlaying = true;
return 0;
}
int Channel::StopPlayingFileAsMicrophone()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StopPlayingFileAsMicrophone()");
if (!_inputFilePlaying)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"StopPlayingFileAsMicrophone() isnot playing");
return 0;
}
CriticalSectionScoped cs(&_fileCritSect);
if (_inputFilePlayerPtr->StopPlayingFile() != 0)
{
_engineStatisticsPtr->SetLastError(
VE_STOP_RECORDING_FAILED, kTraceError,
"StopPlayingFile() could not stop playing");
return -1;
}
_inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
_inputFilePlayerPtr = NULL;
_inputFilePlaying = false;
return 0;
}
int Channel::IsPlayingFileAsMicrophone() const
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::IsPlayingFileAsMicrophone()");
return _inputFilePlaying;
}
int Channel::ScaleFileAsMicrophonePlayout(const float scale)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::ScaleFileAsMicrophonePlayout(scale=%5.3f)", scale);
CriticalSectionScoped cs(&_fileCritSect);
if (!_inputFilePlaying)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"ScaleFileAsMicrophonePlayout() isnot playing");
return -1;
}
if ((_inputFilePlayerPtr == NULL) ||
(_inputFilePlayerPtr->SetAudioScaling(scale) != 0))
{
_engineStatisticsPtr->SetLastError(
VE_BAD_ARGUMENT, kTraceError,
"SetAudioScaling() failed to scale playout");
return -1;
}
return 0;
}
int Channel::StartRecordingPlayout(const char* fileName,
const CodecInst* codecInst)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartRecordingPlayout(fileName=%s)", fileName);
if (_outputFileRecording)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
"StartRecordingPlayout() is already recording");
return 0;
}
FileFormats format;
const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
CodecInst dummyCodec={100,"L16",16000,320,1,320000};
if ((codecInst != NULL) &&
((codecInst->channels < 1) || (codecInst->channels > 2)))
{
_engineStatisticsPtr->SetLastError(
VE_BAD_ARGUMENT, kTraceError,
"StartRecordingPlayout() invalid compression");
return(-1);
}
if(codecInst == NULL)
{
format = kFileFormatPcm16kHzFile;
codecInst=&dummyCodec;
}
else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
(STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
(STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
{
format = kFileFormatWavFile;
}
else
{
format = kFileFormatCompressedFile;
}
CriticalSectionScoped cs(&_fileCritSect);
// Destroy the old instance
if (_outputFileRecorderPtr)
{
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
}
_outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
_outputFileRecorderId, (const FileFormats)format);
if (_outputFileRecorderPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartRecordingPlayout() fileRecorder format isnot correct");
return -1;
}
if (_outputFileRecorderPtr->StartRecordingAudioFile(
fileName, (const CodecInst&)*codecInst, notificationTime) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartRecordingAudioFile() failed to start file recording");
_outputFileRecorderPtr->StopRecording();
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
return -1;
}
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
_outputFileRecording = true;
return 0;
}
int Channel::StartRecordingPlayout(OutStream* stream,
const CodecInst* codecInst)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::StartRecordingPlayout()");
if (_outputFileRecording)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
"StartRecordingPlayout() is already recording");
return 0;
}
FileFormats format;
const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
CodecInst dummyCodec={100,"L16",16000,320,1,320000};
if (codecInst != NULL && codecInst->channels != 1)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_ARGUMENT, kTraceError,
"StartRecordingPlayout() invalid compression");
return(-1);
}
if(codecInst == NULL)
{
format = kFileFormatPcm16kHzFile;
codecInst=&dummyCodec;
}
else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
(STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
(STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
{
format = kFileFormatWavFile;
}
else
{
format = kFileFormatCompressedFile;
}
CriticalSectionScoped cs(&_fileCritSect);
// Destroy the old instance
if (_outputFileRecorderPtr)
{
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
}
_outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
_outputFileRecorderId, (const FileFormats)format);
if (_outputFileRecorderPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartRecordingPlayout() fileRecorder format isnot correct");
return -1;
}
if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst,
notificationTime) != 0)
{
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
"StartRecordingPlayout() failed to "
"start file recording");
_outputFileRecorderPtr->StopRecording();
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
return -1;
}
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
_outputFileRecording = true;
return 0;
}
int Channel::StopRecordingPlayout()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"Channel::StopRecordingPlayout()");
if (!_outputFileRecording)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
"StopRecordingPlayout() isnot recording");
return -1;
}
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFileRecorderPtr->StopRecording() != 0)
{
_engineStatisticsPtr->SetLastError(
VE_STOP_RECORDING_FAILED, kTraceError,
"StopRecording() could not stop recording");
return(-1);
}
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
_outputFileRecording = false;
return 0;
}
void
Channel::SetMixWithMicStatus(bool mix)
{
_mixFileWithMicrophone=mix;
}
int
Channel::GetSpeechOutputLevel(WebRtc_UWord32& level) const
{
WebRtc_Word8 currentLevel = _outputAudioLevel.Level();
level = static_cast<WebRtc_Word32> (currentLevel);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetSpeechOutputLevel() => level=%u", level);
return 0;
}
int
Channel::GetSpeechOutputLevelFullRange(WebRtc_UWord32& level) const
{
WebRtc_Word16 currentLevel = _outputAudioLevel.LevelFullRange();
level = static_cast<WebRtc_Word32> (currentLevel);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetSpeechOutputLevelFullRange() => level=%u", level);
return 0;
}
int
Channel::SetMute(bool enable)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetMute(enable=%d)", enable);
_mute = enable;
return 0;
}
bool
Channel::Mute() const
{
return _mute;
}
int
Channel::SetOutputVolumePan(float left, float right)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetOutputVolumePan()");
_panLeft = left;
_panRight = right;
return 0;
}
int
Channel::GetOutputVolumePan(float& left, float& right) const
{
left = _panLeft;
right = _panRight;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetOutputVolumePan() => left=%3.2f, right=%3.2f", left, right);
return 0;
}
int
Channel::SetChannelOutputVolumeScaling(float scaling)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetChannelOutputVolumeScaling()");
_outputGain = scaling;
return 0;
}
int
Channel::GetChannelOutputVolumeScaling(float& scaling) const
{
scaling = _outputGain;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetChannelOutputVolumeScaling() => scaling=%3.2f", scaling);
return 0;
}
#ifdef WEBRTC_SRTP
int
Channel::EnableSRTPSend(
CipherTypes cipherType,
int cipherKeyLength,
AuthenticationTypes authType,
int authKeyLength,
int authTagLength,
SecurityLevels level,
const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
bool useForRTCP)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EnableSRTPSend()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_encrypting)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"EnableSRTPSend() encryption already enabled");
return -1;
}
if (key == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceWarning,
"EnableSRTPSend() invalid key string");
return -1;
}
if (((kEncryption == level ||
kEncryptionAndAuthentication == level) &&
(cipherKeyLength < kVoiceEngineMinSrtpEncryptLength ||
cipherKeyLength > kVoiceEngineMaxSrtpEncryptLength)) ||
((kAuthentication == level ||
kEncryptionAndAuthentication == level) &&
kAuthHmacSha1 == authType &&
(authKeyLength > kVoiceEngineMaxSrtpAuthSha1Length ||
authTagLength > kVoiceEngineMaxSrtpAuthSha1Length)) ||
((kAuthentication == level ||
kEncryptionAndAuthentication == level) &&
kAuthNull == authType &&
(authKeyLength > kVoiceEngineMaxSrtpKeyAuthNullLength ||
authTagLength > kVoiceEngineMaxSrtpTagAuthNullLength)))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"EnableSRTPSend() invalid key length(s)");
return -1;
}
if (_srtpModule.EnableSRTPEncrypt(
!useForRTCP,
(SrtpModule::CipherTypes)cipherType,
cipherKeyLength,
(SrtpModule::AuthenticationTypes)authType,
authKeyLength, authTagLength,
(SrtpModule::SecurityLevels)level,
key) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_SRTP_ERROR, kTraceError,
"EnableSRTPSend() failed to enable SRTP encryption");
return -1;
}
if (_encryptionPtr == NULL)
{
_encryptionPtr = &_srtpModule;
}
_encrypting = true;
return 0;
}
int
Channel::DisableSRTPSend()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DisableSRTPSend()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_encrypting)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DisableSRTPSend() SRTP encryption already disabled");
return 0;
}
_encrypting = false;
if (_srtpModule.DisableSRTPEncrypt() == -1)
{
_engineStatisticsPtr->SetLastError(
VE_SRTP_ERROR, kTraceError,
"DisableSRTPSend() failed to disable SRTP encryption");
return -1;
}
if (!_srtpModule.SRTPDecrypt() && !_srtpModule.SRTPEncrypt())
{
// Both directions are disabled
_encryptionPtr = NULL;
}
return 0;
}
int
Channel::EnableSRTPReceive(
CipherTypes cipherType,
int cipherKeyLength,
AuthenticationTypes authType,
int authKeyLength,
int authTagLength,
SecurityLevels level,
const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
bool useForRTCP)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EnableSRTPReceive()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_decrypting)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"EnableSRTPReceive() SRTP decryption already enabled");
return -1;
}
if (key == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceWarning,
"EnableSRTPReceive() invalid key string");
return -1;
}
if ((((kEncryption == level) ||
(kEncryptionAndAuthentication == level)) &&
((cipherKeyLength < kVoiceEngineMinSrtpEncryptLength) ||
(cipherKeyLength > kVoiceEngineMaxSrtpEncryptLength))) ||
(((kAuthentication == level) ||
(kEncryptionAndAuthentication == level)) &&
(kAuthHmacSha1 == authType) &&
((authKeyLength > kVoiceEngineMaxSrtpAuthSha1Length) ||
(authTagLength > kVoiceEngineMaxSrtpAuthSha1Length))) ||
(((kAuthentication == level) ||
(kEncryptionAndAuthentication == level)) &&
(kAuthNull == authType) &&
((authKeyLength > kVoiceEngineMaxSrtpKeyAuthNullLength) ||
(authTagLength > kVoiceEngineMaxSrtpTagAuthNullLength))))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"EnableSRTPReceive() invalid key length(s)");
return -1;
}
if (_srtpModule.EnableSRTPDecrypt(
!useForRTCP,
(SrtpModule::CipherTypes)cipherType,
cipherKeyLength,
(SrtpModule::AuthenticationTypes)authType,
authKeyLength,
authTagLength,
(SrtpModule::SecurityLevels)level,
key) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_SRTP_ERROR, kTraceError,
"EnableSRTPReceive() failed to enable SRTP decryption");
return -1;
}
if (_encryptionPtr == NULL)
{
_encryptionPtr = &_srtpModule;
}
_decrypting = true;
return 0;
}
int
Channel::DisableSRTPReceive()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DisableSRTPReceive()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_decrypting)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DisableSRTPReceive() SRTP decryption already disabled");
return 0;
}
_decrypting = false;
if (_srtpModule.DisableSRTPDecrypt() == -1)
{
_engineStatisticsPtr->SetLastError(
VE_SRTP_ERROR, kTraceError,
"DisableSRTPReceive() failed to disable SRTP decryption");
return -1;
}
if (!_srtpModule.SRTPDecrypt() && !_srtpModule.SRTPEncrypt())
{
_encryptionPtr = NULL;
}
return 0;
}
#endif
int
Channel::RegisterExternalEncryption(Encryption& encryption)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RegisterExternalEncryption()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_encryptionPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"RegisterExternalEncryption() encryption already enabled");
return -1;
}
_encryptionPtr = &encryption;
_decrypting = true;
_encrypting = true;
return 0;
}
int
Channel::DeRegisterExternalEncryption()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DeRegisterExternalEncryption()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_encryptionPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterExternalEncryption() encryption already disabled");
return 0;
}
_decrypting = false;
_encrypting = false;
_encryptionPtr = NULL;
return 0;
}
int Channel::SendTelephoneEventOutband(unsigned char eventCode,
int lengthMs, int attenuationDb,
bool playDtmfEvent)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
playDtmfEvent);
_playOutbandDtmfEvent = playDtmfEvent;
if (_rtpRtcpModule->SendTelephoneEventOutband(eventCode, lengthMs,
attenuationDb) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SEND_DTMF_FAILED,
kTraceWarning,
"SendTelephoneEventOutband() failed to send event");
return -1;
}
return 0;
}
int Channel::SendTelephoneEventInband(unsigned char eventCode,
int lengthMs,
int attenuationDb,
bool playDtmfEvent)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendTelephoneEventInband(..., playDtmfEvent=%d)",
playDtmfEvent);
_playInbandDtmfEvent = playDtmfEvent;
_inbandDtmfQueue.AddDtmf(eventCode, lengthMs, attenuationDb);
return 0;
}
int
Channel::SetDtmfPlayoutStatus(bool enable)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetDtmfPlayoutStatus()");
if (_audioCodingModule.SetDtmfPlayoutStatus(enable) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
"SetDtmfPlayoutStatus() failed to set Dtmf playout");
return -1;
}
return 0;
}
bool
Channel::DtmfPlayoutStatus() const
{
return _audioCodingModule.DtmfPlayoutStatus();
}
int
Channel::SetSendTelephoneEventPayloadType(unsigned char type)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetSendTelephoneEventPayloadType()");
if (type > 127)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetSendTelephoneEventPayloadType() invalid type");
return -1;
}
CodecInst codec;
codec.plfreq = 8000;
codec.pltype = type;
memcpy(codec.plname, "telephone-event", 16);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetSendTelephoneEventPayloadType() failed to register send"
"payload type");
return -1;
}
_sendTelephoneEventPayloadType = type;
return 0;
}
int
Channel::GetSendTelephoneEventPayloadType(unsigned char& type)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetSendTelephoneEventPayloadType()");
type = _sendTelephoneEventPayloadType;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetSendTelephoneEventPayloadType() => type=%u", type);
return 0;
}
#ifdef WEBRTC_DTMF_DETECTION
WebRtc_Word32
Channel::RegisterTelephoneEventDetection(
TelephoneEventDetectionMethods detectionMethod,
VoETelephoneEventObserver& observer)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RegisterTelephoneEventDetection()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_telephoneEventDetectionPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"RegisterTelephoneEventDetection() detection already enabled");
return -1;
}
_telephoneEventDetectionPtr = &observer;
switch (detectionMethod)
{
case kInBand:
_inbandTelephoneEventDetection = true;
_outOfBandTelephoneEventDetecion = false;
break;
case kOutOfBand:
_inbandTelephoneEventDetection = false;
_outOfBandTelephoneEventDetecion = true;
break;
case kInAndOutOfBand:
_inbandTelephoneEventDetection = true;
_outOfBandTelephoneEventDetecion = true;
break;
default:
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"RegisterTelephoneEventDetection() invalid detection method");
return -1;
}
if (_inbandTelephoneEventDetection)
{
// Enable in-band Dtmf detectin in the ACM.
if (_audioCodingModule.RegisterIncomingMessagesCallback(this) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"RegisterTelephoneEventDetection() failed to enable Dtmf "
"detection");
}
}
// Enable/disable out-of-band detection of received telephone-events.
// When enabled, RtpAudioFeedback::OnReceivedTelephoneEvent() will be
// called two times by the RTP/RTCP module (start & end).
const bool forwardToDecoder =
_rtpRtcpModule->TelephoneEventForwardToDecoder();
const bool detectEndOfTone = true;
_rtpRtcpModule->SetTelephoneEventStatus(_outOfBandTelephoneEventDetecion,
forwardToDecoder,
detectEndOfTone);
return 0;
}
int
Channel::DeRegisterTelephoneEventDetection()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::DeRegisterTelephoneEventDetection()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_telephoneEventDetectionPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION,
kTraceWarning,
"DeRegisterTelephoneEventDetection() detection already disabled");
return 0;
}
// Disable out-of-band event detection
const bool forwardToDecoder =
_rtpRtcpModule->TelephoneEventForwardToDecoder();
_rtpRtcpModule->SetTelephoneEventStatus(false, forwardToDecoder);
// Disable in-band Dtmf detection
_audioCodingModule.RegisterIncomingMessagesCallback(NULL);
_inbandTelephoneEventDetection = false;
_outOfBandTelephoneEventDetecion = false;
_telephoneEventDetectionPtr = NULL;
return 0;
}
int
Channel::GetTelephoneEventDetectionStatus(
bool& enabled,
TelephoneEventDetectionMethods& detectionMethod)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::GetTelephoneEventDetectionStatus()");
{
CriticalSectionScoped cs(&_callbackCritSect);
enabled = (_telephoneEventDetectionPtr != NULL);
}
if (enabled)
{
if (_inbandTelephoneEventDetection && !_outOfBandTelephoneEventDetecion)
detectionMethod = kInBand;
else if (!_inbandTelephoneEventDetection
&& _outOfBandTelephoneEventDetecion)
detectionMethod = kOutOfBand;
else if (_inbandTelephoneEventDetection
&& _outOfBandTelephoneEventDetecion)
detectionMethod = kInAndOutOfBand;
else
{
assert(false);
return -1;
}
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetTelephoneEventDetectionStatus() => enabled=%d,"
"detectionMethod=%d", enabled, detectionMethod);
return 0;
}
#endif // #ifdef WEBRTC_DTMF_DETECTION
int
Channel::UpdateRxVadDetection(AudioFrame& audioFrame)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::UpdateRxVadDetection()");
int vadDecision = 1;
vadDecision = (audioFrame.vad_activity_ == AudioFrame::kVadActive)? 1 : 0;
if ((vadDecision != _oldVadDecision) && _rxVadObserverPtr)
{
OnRxVadDetected(vadDecision);
_oldVadDecision = vadDecision;
}
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::UpdateRxVadDetection() => vadDecision=%d",
vadDecision);
return 0;
}
int
Channel::RegisterRxVadObserver(VoERxVadCallback &observer)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RegisterRxVadObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_rxVadObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"RegisterRxVadObserver() observer already enabled");
return -1;
}
_rxVadObserverPtr = &observer;
_RxVadDetection = true;
return 0;
}
int
Channel::DeRegisterRxVadObserver()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DeRegisterRxVadObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_rxVadObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterRxVadObserver() observer already disabled");
return 0;
}
_rxVadObserverPtr = NULL;
_RxVadDetection = false;
return 0;
}
int
Channel::VoiceActivityIndicator(int &activity)
{
activity = _sendFrameType;
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::VoiceActivityIndicator(indicator=%d)", activity);
return 0;
}
#ifdef WEBRTC_VOICE_ENGINE_AGC
int
Channel::SetRxAgcStatus(const bool enable, const AgcModes mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetRxAgcStatus(enable=%d, mode=%d)",
(int)enable, (int)mode);
GainControl::Mode agcMode(GainControl::kFixedDigital);
switch (mode)
{
case kAgcDefault:
agcMode = GainControl::kAdaptiveDigital;
break;
case kAgcUnchanged:
agcMode = _rxAudioProcessingModulePtr->gain_control()->mode();
break;
case kAgcFixedDigital:
agcMode = GainControl::kFixedDigital;
break;
case kAgcAdaptiveDigital:
agcMode =GainControl::kAdaptiveDigital;
break;
default:
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetRxAgcStatus() invalid Agc mode");
return -1;
}
if (_rxAudioProcessingModulePtr->gain_control()->set_mode(agcMode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceError,
"SetRxAgcStatus() failed to set Agc mode");
return -1;
}
if (_rxAudioProcessingModulePtr->gain_control()->Enable(enable) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceError,
"SetRxAgcStatus() failed to set Agc state");
return -1;
}
_rxAgcIsEnabled = enable;
_rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true));
return 0;
}
int
Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRxAgcStatus(enable=?, mode=?)");
bool enable = _rxAudioProcessingModulePtr->gain_control()->is_enabled();
GainControl::Mode agcMode =
_rxAudioProcessingModulePtr->gain_control()->mode();
enabled = enable;
switch (agcMode)
{
case GainControl::kFixedDigital:
mode = kAgcFixedDigital;
break;
case GainControl::kAdaptiveDigital:
mode = kAgcAdaptiveDigital;
break;
default:
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceError,
"GetRxAgcStatus() invalid Agc mode");
return -1;
}
return 0;
}
int
Channel::SetRxAgcConfig(const AgcConfig config)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetRxAgcConfig()");
if (_rxAudioProcessingModulePtr->gain_control()->set_target_level_dbfs(
config.targetLeveldBOv) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceError,
"SetRxAgcConfig() failed to set target peak |level|"
"(or envelope) of the Agc");
return -1;
}
if (_rxAudioProcessingModulePtr->gain_control()->set_compression_gain_db(
config.digitalCompressionGaindB) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceError,
"SetRxAgcConfig() failed to set the range in |gain| the"
" digital compression stage may apply");
return -1;
}
if (_rxAudioProcessingModulePtr->gain_control()->enable_limiter(
config.limiterEnable) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceError,
"SetRxAgcConfig() failed to set hard limiter to the signal");
return -1;
}
return 0;
}
int
Channel::GetRxAgcConfig(AgcConfig& config)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRxAgcConfig(config=%?)");
config.targetLeveldBOv =
_rxAudioProcessingModulePtr->gain_control()->target_level_dbfs();
config.digitalCompressionGaindB =
_rxAudioProcessingModulePtr->gain_control()->compression_gain_db();
config.limiterEnable =
_rxAudioProcessingModulePtr->gain_control()->is_limiter_enabled();
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId), "GetRxAgcConfig() => "
"targetLeveldBOv=%u, digitalCompressionGaindB=%u,"
" limiterEnable=%d",
config.targetLeveldBOv,
config.digitalCompressionGaindB,
config.limiterEnable);
return 0;
}
#endif // #ifdef WEBRTC_VOICE_ENGINE_AGC
#ifdef WEBRTC_VOICE_ENGINE_NR
int
Channel::SetRxNsStatus(const bool enable, const NsModes mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetRxNsStatus(enable=%d, mode=%d)",
(int)enable, (int)mode);
NoiseSuppression::Level nsLevel(
(NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE);
switch (mode)
{
case kNsDefault:
nsLevel = (NoiseSuppression::Level)
WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE;
break;
case kNsUnchanged:
nsLevel = _rxAudioProcessingModulePtr->noise_suppression()->level();
break;
case kNsConference:
nsLevel = NoiseSuppression::kHigh;
break;
case kNsLowSuppression:
nsLevel = NoiseSuppression::kLow;
break;
case kNsModerateSuppression:
nsLevel = NoiseSuppression::kModerate;
break;
case kNsHighSuppression:
nsLevel = NoiseSuppression::kHigh;
break;
case kNsVeryHighSuppression:
nsLevel = NoiseSuppression::kVeryHigh;
break;
}
if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(nsLevel)
!= 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceError,
"SetRxAgcStatus() failed to set Ns level");
return -1;
}
if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(enable) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_APM_ERROR, kTraceError,
"SetRxAgcStatus() failed to set Agc state");
return -1;
}
_rxNsIsEnabled = enable;
_rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true));
return 0;
}
int
Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRxNsStatus(enable=?, mode=?)");
bool enable =
_rxAudioProcessingModulePtr->noise_suppression()->is_enabled();
NoiseSuppression::Level ncLevel =
_rxAudioProcessingModulePtr->noise_suppression()->level();
enabled = enable;
switch (ncLevel)
{
case NoiseSuppression::kLow:
mode = kNsLowSuppression;
break;
case NoiseSuppression::kModerate:
mode = kNsModerateSuppression;
break;
case NoiseSuppression::kHigh:
mode = kNsHighSuppression;
break;
case NoiseSuppression::kVeryHigh:
mode = kNsVeryHighSuppression;
break;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetRxNsStatus() => enabled=%d, mode=%d", enabled, mode);
return 0;
}
#endif // #ifdef WEBRTC_VOICE_ENGINE_NR
int
Channel::RegisterRTPObserver(VoERTPObserver& observer)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterRTPObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_rtpObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"RegisterRTPObserver() observer already enabled");
return -1;
}
_rtpObserverPtr = &observer;
_rtpObserver = true;
return 0;
}
int
Channel::DeRegisterRTPObserver()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DeRegisterRTPObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_rtpObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterRTPObserver() observer already disabled");
return 0;
}
_rtpObserver = false;
_rtpObserverPtr = NULL;
return 0;
}
int
Channel::RegisterRTCPObserver(VoERTCPObserver& observer)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RegisterRTCPObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (_rtcpObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"RegisterRTCPObserver() observer already enabled");
return -1;
}
_rtcpObserverPtr = &observer;
_rtcpObserver = true;
return 0;
}
int
Channel::DeRegisterRTCPObserver()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::DeRegisterRTCPObserver()");
CriticalSectionScoped cs(&_callbackCritSect);
if (!_rtcpObserverPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterRTCPObserver() observer already disabled");
return 0;
}
_rtcpObserver = false;
_rtcpObserverPtr = NULL;
return 0;
}
int
Channel::SetLocalSSRC(unsigned int ssrc)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetLocalSSRC()");
if (_sending)
{
_engineStatisticsPtr->SetLastError(
VE_ALREADY_SENDING, kTraceError,
"SetLocalSSRC() already sending");
return -1;
}
if (_rtpRtcpModule->SetSSRC(ssrc) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetLocalSSRC() failed to set SSRC");
return -1;
}
return 0;
}
int
Channel::GetLocalSSRC(unsigned int& ssrc)
{
ssrc = _rtpRtcpModule->SSRC();
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetLocalSSRC() => ssrc=%lu", ssrc);
return 0;
}
int
Channel::GetRemoteSSRC(unsigned int& ssrc)
{
ssrc = _rtpRtcpModule->RemoteSSRC();
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetRemoteSSRC() => ssrc=%lu", ssrc);
return 0;
}
int
Channel::GetRemoteCSRCs(unsigned int arrCSRC[15])
{
if (arrCSRC == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"GetRemoteCSRCs() invalid array argument");
return -1;
}
WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize];
WebRtc_Word32 CSRCs(0);
CSRCs = _rtpRtcpModule->CSRCs(arrOfCSRC);
if (CSRCs > 0)
{
memcpy(arrCSRC, arrOfCSRC, CSRCs * sizeof(WebRtc_UWord32));
for (int i = 0; i < (int) CSRCs; i++)
{
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRemoteCSRCs() => arrCSRC[%d]=%lu", i, arrCSRC[i]);
}
} else
{
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRemoteCSRCs() => list is empty!");
}
return CSRCs;
}
int
Channel::SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID)
{
if (_rtpAudioProc.get() == NULL)
{
_rtpAudioProc.reset(AudioProcessing::Create(VoEModuleId(_instanceId,
_channelId)));
if (_rtpAudioProc.get() == NULL)
{
_engineStatisticsPtr->SetLastError(VE_NO_MEMORY, kTraceCritical,
"Failed to create AudioProcessing");
return -1;
}
}
if (_rtpAudioProc->level_estimator()->Enable(enable) !=
AudioProcessing::kNoError)
{
_engineStatisticsPtr->SetLastError(VE_APM_ERROR, kTraceWarning,
"Failed to enable AudioProcessing::level_estimator()");
}
_includeAudioLevelIndication = enable;
return _rtpRtcpModule->SetRTPAudioLevelIndicationStatus(enable, ID);
}
int
Channel::GetRTPAudioLevelIndicationStatus(bool& enabled, unsigned char& ID)
{
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetRTPAudioLevelIndicationStatus() => enabled=%d, ID=%u",
enabled, ID);
return _rtpRtcpModule->GetRTPAudioLevelIndicationStatus(enabled, ID);
}
int
Channel::SetRTCPStatus(bool enable)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetRTCPStatus()");
if (_rtpRtcpModule->SetRTCPStatus(enable ?
kRtcpCompound : kRtcpOff) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetRTCPStatus() failed to set RTCP status");
return -1;
}
return 0;
}
int
Channel::GetRTCPStatus(bool& enabled)
{
RTCPMethod method = _rtpRtcpModule->RTCP();
enabled = (method != kRtcpOff);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetRTCPStatus() => enabled=%d", enabled);
return 0;
}
int
Channel::SetRTCP_CNAME(const char cName[256])
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCP_CNAME()");
if (_rtpRtcpModule->SetCNAME(cName) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetRTCP_CNAME() failed to set RTCP CNAME");
return -1;
}
return 0;
}
int
Channel::GetRTCP_CNAME(char cName[256])
{
if (_rtpRtcpModule->CNAME(cName) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"GetRTCP_CNAME() failed to retrieve RTCP CNAME");
return -1;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTCP_CNAME() => cName=%s", cName);
return 0;
}
int
Channel::GetRemoteRTCP_CNAME(char cName[256])
{
if (cName == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"GetRemoteRTCP_CNAME() invalid CNAME input buffer");
return -1;
}
char cname[RTCP_CNAME_SIZE];
const WebRtc_UWord32 remoteSSRC = _rtpRtcpModule->RemoteSSRC();
if (_rtpRtcpModule->RemoteCNAME(remoteSSRC, cname) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_CNAME, kTraceError,
"GetRemoteRTCP_CNAME() failed to retrieve remote RTCP CNAME");
return -1;
}
strcpy(cName, cname);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRemoteRTCP_CNAME() => cName=%s", cName);
return 0;
}
int
Channel::GetRemoteRTCPData(
unsigned int& NTPHigh,
unsigned int& NTPLow,
unsigned int& timestamp,
unsigned int& playoutTimestamp,
unsigned int* jitter,
unsigned short* fractionLost)
{
// --- Information from sender info in received Sender Reports
RTCPSenderInfo senderInfo;
if (_rtpRtcpModule->RemoteRTCPStat(&senderInfo) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"GetRemoteRTCPData() failed to retrieve sender info for remote "
"side");
return -1;
}
// We only utilize 12 out of 20 bytes in the sender info (ignores packet
// and octet count)
NTPHigh = senderInfo.NTPseconds;
NTPLow = senderInfo.NTPfraction;
timestamp = senderInfo.RTPtimeStamp;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRemoteRTCPData() => NTPHigh=%lu, NTPLow=%lu, "
"timestamp=%lu",
NTPHigh, NTPLow, timestamp);
// --- Locally derived information
// This value is updated on each incoming RTCP packet (0 when no packet
// has been received)
playoutTimestamp = _playoutTimeStampRTCP;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRemoteRTCPData() => playoutTimestamp=%lu",
_playoutTimeStampRTCP);
if (NULL != jitter || NULL != fractionLost)
{
// Get all RTCP receiver report blocks that have been received on this
// channel. If we receive RTP packets from a remote source we know the
// remote SSRC and use the report block from him.
// Otherwise use the first report block.
std::vector<RTCPReportBlock> remote_stats;
if (_rtpRtcpModule->RemoteRTCPStat(&remote_stats) != 0 ||
remote_stats.empty()) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRemoteRTCPData() failed to measure statistics due"
" to lack of received RTP and/or RTCP packets");
return -1;
}
WebRtc_UWord32 remoteSSRC = _rtpRtcpModule->RemoteSSRC();
std::vector<RTCPReportBlock>::const_iterator it = remote_stats.begin();
for (; it != remote_stats.end(); ++it) {
if (it->remoteSSRC == remoteSSRC)
break;
}
if (it == remote_stats.end()) {
// If we have not received any RTCP packets from this SSRC it probably
// means that we have not received any RTP packets.
// Use the first received report block instead.
it = remote_stats.begin();
remoteSSRC = it->remoteSSRC;
}
if (jitter) {
*jitter = it->jitter;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRemoteRTCPData() => jitter = %lu", *jitter);
}
if (fractionLost) {
*fractionLost = it->fractionLost;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRemoteRTCPData() => fractionLost = %lu",
*fractionLost);
}
}
return 0;
}
int
Channel::SendApplicationDefinedRTCPPacket(const unsigned char subType,
unsigned int name,
const char* data,
unsigned short dataLengthInBytes)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendApplicationDefinedRTCPPacket()");
if (!_sending)
{
_engineStatisticsPtr->SetLastError(
VE_NOT_SENDING, kTraceError,
"SendApplicationDefinedRTCPPacket() not sending");
return -1;
}
if (NULL == data)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SendApplicationDefinedRTCPPacket() invalid data value");
return -1;
}
if (dataLengthInBytes % 4 != 0)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SendApplicationDefinedRTCPPacket() invalid length value");
return -1;
}
RTCPMethod status = _rtpRtcpModule->RTCP();
if (status == kRtcpOff)
{
_engineStatisticsPtr->SetLastError(
VE_RTCP_ERROR, kTraceError,
"SendApplicationDefinedRTCPPacket() RTCP is disabled");
return -1;
}
// Create and schedule the RTCP APP packet for transmission
if (_rtpRtcpModule->SetRTCPApplicationSpecificData(
subType,
name,
(const unsigned char*) data,
dataLengthInBytes) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_SEND_ERROR, kTraceError,
"SendApplicationDefinedRTCPPacket() failed to send RTCP packet");
return -1;
}
return 0;
}
int
Channel::GetRTPStatistics(
unsigned int& averageJitterMs,
unsigned int& maxJitterMs,
unsigned int& discardedPackets)
{
WebRtc_UWord8 fraction_lost(0);
WebRtc_UWord32 cum_lost(0);
WebRtc_UWord32 ext_max(0);
WebRtc_UWord32 jitter(0);
WebRtc_UWord32 max_jitter(0);
// The jitter statistics is updated for each received RTP packet and is
// based on received packets.
if (_rtpRtcpModule->StatisticsRTP(&fraction_lost,
&cum_lost,
&ext_max,
&jitter,
&max_jitter) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
"GetRTPStatistics() failed to read RTP statistics from the "
"RTP/RTCP module");
}
const WebRtc_Word32 playoutFrequency =
_audioCodingModule.PlayoutFrequency();
if (playoutFrequency > 0)
{
// Scale RTP statistics given the current playout frequency
maxJitterMs = max_jitter / (playoutFrequency / 1000);
averageJitterMs = jitter / (playoutFrequency / 1000);
}
discardedPackets = _numberOfDiscardedPackets;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTPStatistics() => averageJitterMs = %lu, maxJitterMs = %lu,"
" discardedPackets = %lu)",
averageJitterMs, maxJitterMs, discardedPackets);
return 0;
}
int Channel::GetRemoteRTCPSenderInfo(SenderInfo* sender_info) {
if (sender_info == NULL) {
_engineStatisticsPtr->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"GetRemoteRTCPSenderInfo() invalid sender_info.");
return -1;
}
// Get the sender info from the latest received RTCP Sender Report.
RTCPSenderInfo rtcp_sender_info;
if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_sender_info) != 0) {
_engineStatisticsPtr->SetLastError(VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"GetRemoteRTCPSenderInfo() failed to read RTCP SR sender info.");
return -1;
}
sender_info->NTP_timestamp_high = rtcp_sender_info.NTPseconds;
sender_info->NTP_timestamp_low = rtcp_sender_info.NTPfraction;
sender_info->RTP_timestamp = rtcp_sender_info.RTPtimeStamp;
sender_info->sender_packet_count = rtcp_sender_info.sendPacketCount;
sender_info->sender_octet_count = rtcp_sender_info.sendOctetCount;
return 0;
}
int Channel::GetRemoteRTCPReportBlocks(
std::vector<ReportBlock>* report_blocks) {
if (report_blocks == NULL) {
_engineStatisticsPtr->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"GetRemoteRTCPReportBlock()s invalid report_blocks.");
return -1;
}
// Get the report blocks from the latest received RTCP Sender or Receiver
// Report. Each element in the vector contains the sender's SSRC and a
// report block according to RFC 3550.
std::vector<RTCPReportBlock> rtcp_report_blocks;
if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) {
_engineStatisticsPtr->SetLastError(VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"GetRemoteRTCPReportBlocks() failed to read RTCP SR/RR report block.");
return -1;
}
if (rtcp_report_blocks.empty())
return 0;
std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin();
for (; it != rtcp_report_blocks.end(); ++it) {
ReportBlock report_block;
report_block.sender_SSRC = it->remoteSSRC;
report_block.source_SSRC = it->sourceSSRC;
report_block.fraction_lost = it->fractionLost;
report_block.cumulative_num_packets_lost = it->cumulativeLost;
report_block.extended_highest_sequence_number = it->extendedHighSeqNum;
report_block.interarrival_jitter = it->jitter;
report_block.last_SR_timestamp = it->lastSR;
report_block.delay_since_last_SR = it->delaySinceLastSR;
report_blocks->push_back(report_block);
}
return 0;
}
int
Channel::GetRTPStatistics(CallStatistics& stats)
{
WebRtc_UWord8 fraction_lost(0);
WebRtc_UWord32 cum_lost(0);
WebRtc_UWord32 ext_max(0);
WebRtc_UWord32 jitter(0);
WebRtc_UWord32 max_jitter(0);
// --- Part one of the final structure (four values)
// The jitter statistics is updated for each received RTP packet and is
// based on received packets.
if (_rtpRtcpModule->StatisticsRTP(&fraction_lost,
&cum_lost,
&ext_max,
&jitter,
&max_jitter) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
"GetRTPStatistics() failed to read RTP statistics from the "
"RTP/RTCP module");
}
stats.fractionLost = fraction_lost;
stats.cumulativeLost = cum_lost;
stats.extendedMax = ext_max;
stats.jitterSamples = jitter;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTPStatistics() => fractionLost=%lu, cumulativeLost=%lu,"
" extendedMax=%lu, jitterSamples=%li)",
stats.fractionLost, stats.cumulativeLost, stats.extendedMax,
stats.jitterSamples);
// --- Part two of the final structure (one value)
WebRtc_UWord16 RTT(0);
RTCPMethod method = _rtpRtcpModule->RTCP();
if (method == kRtcpOff)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTPStatistics() RTCP is disabled => valid RTT "
"measurements cannot be retrieved");
} else
{
// The remote SSRC will be zero if no RTP packet has been received.
WebRtc_UWord32 remoteSSRC = _rtpRtcpModule->RemoteSSRC();
if (remoteSSRC > 0)
{
WebRtc_UWord16 avgRTT(0);
WebRtc_UWord16 maxRTT(0);
WebRtc_UWord16 minRTT(0);
if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT, &maxRTT)
!= 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTPStatistics() failed to retrieve RTT from "
"the RTP/RTCP module");
}
} else
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTPStatistics() failed to measure RTT since no "
"RTP packets have been received yet");
}
}
stats.rttMs = static_cast<int> (RTT);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTPStatistics() => rttMs=%d", stats.rttMs);
// --- Part three of the final structure (four values)
WebRtc_UWord32 bytesSent(0);
WebRtc_UWord32 packetsSent(0);
WebRtc_UWord32 bytesReceived(0);
WebRtc_UWord32 packetsReceived(0);
if (_rtpRtcpModule->DataCountersRTP(&bytesSent,
&packetsSent,
&bytesReceived,
&packetsReceived) != 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTPStatistics() failed to retrieve RTP datacounters =>"
" output will not be complete");
}
stats.bytesSent = bytesSent;
stats.packetsSent = packetsSent;
stats.bytesReceived = bytesReceived;
stats.packetsReceived = packetsReceived;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetRTPStatistics() => bytesSent=%d, packetsSent=%d,"
" bytesReceived=%d, packetsReceived=%d)",
stats.bytesSent, stats.packetsSent, stats.bytesReceived,
stats.packetsReceived);
return 0;
}
int
Channel::SetFECStatus(bool enable, int redPayloadtype)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetFECStatus()");
CodecInst codec;
// Get default RED settings from the ACM database
bool foundRED(false);
const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
for (int idx = 0; (!foundRED && idx < nSupportedCodecs); idx++)
{
_audioCodingModule.Codec(idx, codec);
if (!STR_CASE_CMP(codec.plname, "RED"))
{
foundRED = true;
}
}
if (!foundRED)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
"SetFECStatus() RED is not supported");
return -1;
}
if (redPayloadtype != -1)
{
codec.pltype = redPayloadtype;
}
if (_audioCodingModule.RegisterSendCodec(codec) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetFECStatus() RED registration in ACM module failed");
return -1;
}
if (_rtpRtcpModule->SetSendREDPayloadType(codec.pltype) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetFECStatus() RED registration in RTP/RTCP module failed");
return -1;
}
if (_audioCodingModule.SetFECStatus(enable) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetFECStatus() failed to set FEC state in the ACM");
return -1;
}
return 0;
}
int
Channel::GetFECStatus(bool& enabled, int& redPayloadtype)
{
enabled = _audioCodingModule.FECStatus();
if (enabled)
{
WebRtc_Word8 payloadType(0);
if (_rtpRtcpModule->SendREDPayloadType(payloadType) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"GetFECStatus() failed to retrieve RED PT from RTP/RTCP "
"module");
return -1;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetFECStatus() => enabled=%d, redPayloadtype=%d",
enabled, redPayloadtype);
return 0;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId, _channelId),
"GetFECStatus() => enabled=%d", enabled);
return 0;
}
int
Channel::StartRTPDump(const char fileNameUTF8[1024],
RTPDirections direction)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StartRTPDump()");
if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartRTPDump() invalid RTP direction");
return -1;
}
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
&_rtpDumpIn : &_rtpDumpOut;
if (rtpDumpPtr == NULL)
{
assert(false);
return -1;
}
if (rtpDumpPtr->IsActive())
{
rtpDumpPtr->Stop();
}
if (rtpDumpPtr->Start(fileNameUTF8) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartRTPDump() failed to create file");
return -1;
}
return 0;
}
int
Channel::StopRTPDump(RTPDirections direction)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StopRTPDump()");
if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StopRTPDump() invalid RTP direction");
return -1;
}
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
&_rtpDumpIn : &_rtpDumpOut;
if (rtpDumpPtr == NULL)
{
assert(false);
return -1;
}
if (!rtpDumpPtr->IsActive())
{
return 0;
}
return rtpDumpPtr->Stop();
}
bool
Channel::RTPDumpIsActive(RTPDirections direction)
{
if ((direction != kRtpIncoming) &&
(direction != kRtpOutgoing))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"RTPDumpIsActive() invalid RTP direction");
return false;
}
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
&_rtpDumpIn : &_rtpDumpOut;
return rtpDumpPtr->IsActive();
}
int
Channel::InsertExtraRTPPacket(unsigned char payloadType,
bool markerBit,
const char* payloadData,
unsigned short payloadSize)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::InsertExtraRTPPacket()");
if (payloadType > 127)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_PLTYPE, kTraceError,
"InsertExtraRTPPacket() invalid payload type");
return -1;
}
if (payloadData == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"InsertExtraRTPPacket() invalid payload data");
return -1;
}
if (payloadSize > _rtpRtcpModule->MaxDataPayloadLength())
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"InsertExtraRTPPacket() invalid payload size");
return -1;
}
if (!_sending)
{
_engineStatisticsPtr->SetLastError(
VE_NOT_SENDING, kTraceError,
"InsertExtraRTPPacket() not sending");
return -1;
}
// Create extra RTP packet by calling RtpRtcp::SendOutgoingData().
// Transport::SendPacket() will be called by the module when the RTP packet
// is created.
// The call to SendOutgoingData() does *not* modify the timestamp and
// payloadtype to ensure that the RTP module generates a valid RTP packet
// (user might utilize a non-registered payload type).
// The marker bit and payload type will be replaced just before the actual
// transmission, i.e., the actual modification is done *after* the RTP
// module has delivered its RTP packet back to the VoE.
// We will use the stored values above when the packet is modified
// (see Channel::SendPacket()).
_extraPayloadType = payloadType;
_extraMarkerBit = markerBit;
_insertExtraRTPPacket = true;
if (_rtpRtcpModule->SendOutgoingData(kAudioFrameSpeech,
_lastPayloadType,
_lastLocalTimeStamp,
// Leaving the time when this frame was
// received from the capture device as
// undefined for voice for now.
-1,
(const WebRtc_UWord8*) payloadData,
payloadSize) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"InsertExtraRTPPacket() failed to send extra RTP packet");
return -1;
}
return 0;
}
WebRtc_UWord32
Channel::Demultiplex(const AudioFrame& audioFrame)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::Demultiplex()");
_audioFrame = audioFrame;
_audioFrame.id_ = _channelId;
return 0;
}
WebRtc_UWord32
Channel::PrepareEncodeAndSend(int mixingFrequency)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::PrepareEncodeAndSend()");
if (_audioFrame.samples_per_channel_ == 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::PrepareEncodeAndSend() invalid audio frame");
return -1;
}
if (_inputFilePlaying)
{
MixOrReplaceAudioWithFile(mixingFrequency);
}
if (_mute)
{
AudioFrameOperations::Mute(_audioFrame);
}
if (_inputExternalMedia)
{
CriticalSectionScoped cs(&_callbackCritSect);
const bool isStereo = (_audioFrame.num_channels_ == 2);
if (_inputExternalMediaCallbackPtr)
{
_inputExternalMediaCallbackPtr->Process(
_channelId,
kRecordingPerChannel,
(WebRtc_Word16*)_audioFrame.data_,
_audioFrame.samples_per_channel_,
_audioFrame.sample_rate_hz_,
isStereo);
}
}
InsertInbandDtmfTone();
if (_includeAudioLevelIndication)
{
assert(_rtpAudioProc.get() != NULL);
// Check if settings need to be updated.
if (_rtpAudioProc->sample_rate_hz() != _audioFrame.sample_rate_hz_)
{
if (_rtpAudioProc->set_sample_rate_hz(_audioFrame.sample_rate_hz_) !=
AudioProcessing::kNoError)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Error setting AudioProcessing sample rate");
return -1;
}
}
if (_rtpAudioProc->num_input_channels() != _audioFrame.num_channels_)
{
if (_rtpAudioProc->set_num_channels(_audioFrame.num_channels_,
_audioFrame.num_channels_)
!= AudioProcessing::kNoError)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Error setting AudioProcessing channels");
return -1;
}
}
// Performs level analysis only; does not affect the signal.
_rtpAudioProc->ProcessStream(&_audioFrame);
}
return 0;
}
WebRtc_UWord32
Channel::EncodeAndSend()
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EncodeAndSend()");
assert(_audioFrame.num_channels_ <= 2);
if (_audioFrame.samples_per_channel_ == 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EncodeAndSend() invalid audio frame");
return -1;
}
_audioFrame.id_ = _channelId;
// --- Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
// The ACM resamples internally.
_audioFrame.timestamp_ = _timeStamp;
if (_audioCodingModule.Add10MsData((AudioFrame&)_audioFrame) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EncodeAndSend() ACM encoding failed");
return -1;
}
_timeStamp += _audioFrame.samples_per_channel_;
// --- Encode if complete frame is ready
// This call will trigger AudioPacketizationCallback::SendData if encoding
// is done and payload is ready for packetization and transmission.
return _audioCodingModule.Process();
}
int Channel::RegisterExternalMediaProcessing(
ProcessingTypes type,
VoEMediaProcess& processObject)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RegisterExternalMediaProcessing()");
CriticalSectionScoped cs(&_callbackCritSect);
if (kPlaybackPerChannel == type)
{
if (_outputExternalMediaCallbackPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"Channel::RegisterExternalMediaProcessing() "
"output external media already enabled");
return -1;
}
_outputExternalMediaCallbackPtr = &processObject;
_outputExternalMedia = true;
}
else if (kRecordingPerChannel == type)
{
if (_inputExternalMediaCallbackPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"Channel::RegisterExternalMediaProcessing() "
"output external media already enabled");
return -1;
}
_inputExternalMediaCallbackPtr = &processObject;
_inputExternalMedia = true;
}
return 0;
}
int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::DeRegisterExternalMediaProcessing()");
CriticalSectionScoped cs(&_callbackCritSect);
if (kPlaybackPerChannel == type)
{
if (!_outputExternalMediaCallbackPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"Channel::DeRegisterExternalMediaProcessing() "
"output external media already disabled");
return 0;
}
_outputExternalMedia = false;
_outputExternalMediaCallbackPtr = NULL;
}
else if (kRecordingPerChannel == type)
{
if (!_inputExternalMediaCallbackPtr)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"Channel::DeRegisterExternalMediaProcessing() "
"input external media already disabled");
return 0;
}
_inputExternalMedia = false;
_inputExternalMediaCallbackPtr = NULL;
}
return 0;
}
int
Channel::ResetRTCPStatistics()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::ResetRTCPStatistics()");
WebRtc_UWord32 remoteSSRC(0);
remoteSSRC = _rtpRtcpModule->RemoteSSRC();
return _rtpRtcpModule->ResetRTT(remoteSSRC);
}
int
Channel::GetRoundTripTimeSummary(StatVal& delaysMs) const
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRoundTripTimeSummary()");
// Override default module outputs for the case when RTCP is disabled.
// This is done to ensure that we are backward compatible with the
// VoiceEngine where we did not use RTP/RTCP module.
if (!_rtpRtcpModule->RTCP())
{
delaysMs.min = -1;
delaysMs.max = -1;
delaysMs.average = -1;
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRoundTripTimeSummary() RTCP is disabled =>"
" valid RTT measurements cannot be retrieved");
return 0;
}
WebRtc_UWord32 remoteSSRC;
WebRtc_UWord16 RTT;
WebRtc_UWord16 avgRTT;
WebRtc_UWord16 maxRTT;
WebRtc_UWord16 minRTT;
// The remote SSRC will be zero if no RTP packet has been received.
remoteSSRC = _rtpRtcpModule->RemoteSSRC();
if (remoteSSRC == 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRoundTripTimeSummary() unable to measure RTT"
" since no RTP packet has been received yet");
}
// Retrieve RTT statistics from the RTP/RTCP module for the specified
// channel and SSRC. The SSRC is required to parse out the correct source
// in conference scenarios.
if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT,&maxRTT) != 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"GetRoundTripTimeSummary unable to retrieve RTT values"
" from the RTCP layer");
delaysMs.min = -1; delaysMs.max = -1; delaysMs.average = -1;
}
else
{
delaysMs.min = minRTT;
delaysMs.max = maxRTT;
delaysMs.average = avgRTT;
}
return 0;
}
int
Channel::GetNetworkStatistics(NetworkStatistics& stats)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetNetworkStatistics()");
return _audioCodingModule.NetworkStatistics(
(ACMNetworkStatistics &)stats);
}
int
Channel::GetDelayEstimate(int& delayMs) const
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetDelayEstimate()");
delayMs = (_averageDelayMs + 5) / 10 + _recPacketDelayMs;
return 0;
}
int
Channel::SetMinimumPlayoutDelay(int delayMs)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetMinimumPlayoutDelay()");
if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
(delayMs > kVoiceEngineMaxMinPlayoutDelayMs))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetMinimumPlayoutDelay() invalid min delay");
return -1;
}
if (_audioCodingModule.SetMinimumPlayoutDelay(delayMs) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetMinimumPlayoutDelay() failed to set min playout delay");
return -1;
}
return 0;
}
int
Channel::GetPlayoutTimestamp(unsigned int& timestamp)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetPlayoutTimestamp()");
WebRtc_UWord32 playoutTimestamp(0);
if (GetPlayoutTimeStamp(playoutTimestamp) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_VALUE, kTraceError,
"GetPlayoutTimestamp() failed to retrieve timestamp");
return -1;
}
timestamp = playoutTimestamp;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
"GetPlayoutTimestamp() => timestamp=%u", timestamp);
return 0;
}
int
Channel::SetInitTimestamp(unsigned int timestamp)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetInitTimestamp()");
if (_sending)
{
_engineStatisticsPtr->SetLastError(
VE_SENDING, kTraceError, "SetInitTimestamp() already sending");
return -1;
}
if (_rtpRtcpModule->SetStartTimestamp(timestamp) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetInitTimestamp() failed to set timestamp");
return -1;
}
return 0;
}
int
Channel::SetInitSequenceNumber(short sequenceNumber)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetInitSequenceNumber()");
if (_sending)
{
_engineStatisticsPtr->SetLastError(
VE_SENDING, kTraceError,
"SetInitSequenceNumber() already sending");
return -1;
}
if (_rtpRtcpModule->SetSequenceNumber(sequenceNumber) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetInitSequenceNumber() failed to set sequence number");
return -1;
}
return 0;
}
int
Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetRtpRtcp()");
rtpRtcpModule = _rtpRtcpModule.get();
return 0;
}
// TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
// a shared helper.
WebRtc_Word32
Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
{
scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]);
int fileSamples(0);
{
CriticalSectionScoped cs(&_fileCritSect);
if (_inputFilePlayerPtr == NULL)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::MixOrReplaceAudioWithFile() fileplayer"
" doesnt exist");
return -1;
}
if (_inputFilePlayerPtr->Get10msAudioFromFile(fileBuffer.get(),
fileSamples,
mixingFrequency) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::MixOrReplaceAudioWithFile() file mixing "
"failed");
return -1;
}
if (fileSamples == 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::MixOrReplaceAudioWithFile() file is ended");
return 0;
}
}
assert(_audioFrame.samples_per_channel_ == fileSamples);
if (_mixFileWithMicrophone)
{
// Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo.
Utility::MixWithSat(_audioFrame.data_,
_audioFrame.num_channels_,
fileBuffer.get(),
1,
fileSamples);
}
else
{
// Replace ACM audio with file.
// Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo.
_audioFrame.UpdateFrame(_channelId,
-1,
fileBuffer.get(),
fileSamples,
mixingFrequency,
AudioFrame::kNormalSpeech,
AudioFrame::kVadUnknown,
1);
}
return 0;
}
WebRtc_Word32
Channel::MixAudioWithFile(AudioFrame& audioFrame,
const int mixingFrequency)
{
assert(mixingFrequency <= 32000);
scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]);
int fileSamples(0);
{
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFilePlayerPtr == NULL)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::MixAudioWithFile() file mixing failed");
return -1;
}
// We should get the frequency we ask for.
if (_outputFilePlayerPtr->Get10msAudioFromFile(fileBuffer.get(),
fileSamples,
mixingFrequency) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::MixAudioWithFile() file mixing failed");
return -1;
}
}
if (audioFrame.samples_per_channel_ == fileSamples)
{
// Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo.
Utility::MixWithSat(audioFrame.data_,
audioFrame.num_channels_,
fileBuffer.get(),
1,
fileSamples);
}
else
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::MixAudioWithFile() samples_per_channel_(%d) != "
"fileSamples(%d)",
audioFrame.samples_per_channel_, fileSamples);
return -1;
}
return 0;
}
int
Channel::InsertInbandDtmfTone()
{
// Check if we should start a new tone.
if (_inbandDtmfQueue.PendingDtmf() &&
!_inbandDtmfGenerator.IsAddingTone() &&
_inbandDtmfGenerator.DelaySinceLastTone() >
kMinTelephoneEventSeparationMs)
{
WebRtc_Word8 eventCode(0);
WebRtc_UWord16 lengthMs(0);
WebRtc_UWord8 attenuationDb(0);
eventCode = _inbandDtmfQueue.NextDtmf(&lengthMs, &attenuationDb);
_inbandDtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb);
if (_playInbandDtmfEvent)
{
// Add tone to output mixer using a reduced length to minimize
// risk of echo.
_outputMixerPtr->PlayDtmfTone(eventCode, lengthMs - 80,
attenuationDb);
}
}
if (_inbandDtmfGenerator.IsAddingTone())
{
WebRtc_UWord16 frequency(0);
_inbandDtmfGenerator.GetSampleRate(frequency);
if (frequency != _audioFrame.sample_rate_hz_)
{
// Update sample rate of Dtmf tone since the mixing frequency
// has changed.
_inbandDtmfGenerator.SetSampleRate(
(WebRtc_UWord16) (_audioFrame.sample_rate_hz_));
// Reset the tone to be added taking the new sample rate into
// account.
_inbandDtmfGenerator.ResetTone();
}
WebRtc_Word16 toneBuffer[320];
WebRtc_UWord16 toneSamples(0);
// Get 10ms tone segment and set time since last tone to zero
if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::EncodeAndSend() inserting Dtmf failed");
return -1;
}
// Replace mixed audio with DTMF tone.
for (int sample = 0;
sample < _audioFrame.samples_per_channel_;
sample++)
{
for (int channel = 0;
channel < _audioFrame.num_channels_;
channel++)
{
_audioFrame.data_[sample * _audioFrame.num_channels_ + channel] =
toneBuffer[sample];
}
}
assert(_audioFrame.samples_per_channel_ == toneSamples);
} else
{
// Add 10ms to "delay-since-last-tone" counter
_inbandDtmfGenerator.UpdateDelaySinceLastTone();
}
return 0;
}
WebRtc_Word32
Channel::GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp)
{
WebRtc_UWord32 timestamp(0);
CodecInst currRecCodec;
if (_audioCodingModule.PlayoutTimestamp(timestamp) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetPlayoutTimeStamp() failed to read playout"
" timestamp from the ACM");
return -1;
}
WebRtc_UWord16 delayMS(0);
if (_audioDeviceModulePtr->PlayoutDelay(&delayMS) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetPlayoutTimeStamp() failed to read playout"
" delay from the ADM");
return -1;
}
WebRtc_Word32 playoutFrequency = _audioCodingModule.PlayoutFrequency();
if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0)
{
if (STR_CASE_CMP("G722", currRecCodec.plname) == 0)
{
playoutFrequency = 8000;
}
}
timestamp -= (delayMS * (playoutFrequency/1000));
playoutTimestamp = timestamp;
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetPlayoutTimeStamp() => playoutTimestamp = %lu",
playoutTimestamp);
return 0;
}
void
Channel::ResetDeadOrAliveCounters()
{
_countDeadDetections = 0;
_countAliveDetections = 0;
}
void
Channel::UpdateDeadOrAliveCounters(bool alive)
{
if (alive)
_countAliveDetections++;
else
_countDeadDetections++;
}
int
Channel::GetDeadOrAliveCounters(int& countDead, int& countAlive) const
{
bool enabled;
WebRtc_UWord8 timeSec;
_rtpRtcpModule->PeriodicDeadOrAliveStatus(enabled, timeSec);
if (!enabled)
return (-1);
countDead = static_cast<int> (_countDeadDetections);
countAlive = static_cast<int> (_countAliveDetections);
return 0;
}
WebRtc_Word32
Channel::SendPacketRaw(const void *data, int len, bool RTCP)
{
if (_transportPtr == NULL)
{
return -1;
}
if (!RTCP)
{
return _transportPtr->SendPacket(_channelId, data, len);
}
else
{
return _transportPtr->SendRTCPPacket(_channelId, data, len);
}
}
WebRtc_Word32
Channel::UpdatePacketDelay(const WebRtc_UWord32 timestamp,
const WebRtc_UWord16 sequenceNumber)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
timestamp, sequenceNumber);
WebRtc_Word32 rtpReceiveFrequency(0);
// Get frequency of last received payload
rtpReceiveFrequency = _audioCodingModule.ReceiveFrequency();
CodecInst currRecCodec;
if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0)
{
if (STR_CASE_CMP("G722", currRecCodec.plname) == 0)
{
// Even though the actual sampling rate for G.722 audio is
// 16,000 Hz, the RTP clock rate for the G722 payload format is
// 8,000 Hz because that value was erroneously assigned in
// RFC 1890 and must remain unchanged for backward compatibility.
rtpReceiveFrequency = 8000;
}
}
const WebRtc_UWord32 timeStampDiff = timestamp - _playoutTimeStampRTP;
WebRtc_UWord32 timeStampDiffMs(0);
if (timeStampDiff > 0)
{
switch (rtpReceiveFrequency)
{
case 8000:
timeStampDiffMs = timeStampDiff >> 3;
break;
case 16000:
timeStampDiffMs = timeStampDiff >> 4;
break;
case 32000:
timeStampDiffMs = timeStampDiff >> 5;
break;
default:
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::UpdatePacketDelay() invalid sample "
"rate");
timeStampDiffMs = 0;
return -1;
}
if (timeStampDiffMs > 5000)
{
timeStampDiffMs = 0;
}
if (_averageDelayMs == 0)
{
_averageDelayMs = timeStampDiffMs;
}
else
{
// Filter average delay value using exponential filter (alpha is
// 7/8). We derive 10*_averageDelayMs here (reduces risk of
// rounding error) and compensate for it in GetDelayEstimate()
// later. Adding 4/8 results in correct rounding.
_averageDelayMs = ((_averageDelayMs*7 + 10*timeStampDiffMs + 4)>>3);
}
if (sequenceNumber - _previousSequenceNumber == 1)
{
WebRtc_UWord16 packetDelayMs = 0;
switch (rtpReceiveFrequency)
{
case 8000:
packetDelayMs = (WebRtc_UWord16)(
(timestamp - _previousTimestamp) >> 3);
break;
case 16000:
packetDelayMs = (WebRtc_UWord16)(
(timestamp - _previousTimestamp) >> 4);
break;
case 32000:
packetDelayMs = (WebRtc_UWord16)(
(timestamp - _previousTimestamp) >> 5);
break;
}
if (packetDelayMs >= 10 && packetDelayMs <= 60)
_recPacketDelayMs = packetDelayMs;
}
}
_previousSequenceNumber = sequenceNumber;
_previousTimestamp = timestamp;
return 0;
}
void
Channel::RegisterReceiveCodecsToRTPModule()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::RegisterReceiveCodecsToRTPModule()");
CodecInst codec;
const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
for (int idx = 0; idx < nSupportedCodecs; idx++)
{
// Open up the RTP/RTCP receiver for all supported codecs
if ((_audioCodingModule.Codec(idx, codec) == -1) ||
(_rtpRtcpModule->RegisterReceivePayload(codec) == -1))
{
WEBRTC_TRACE(
kTraceWarning,
kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule() unable"
" to register %s (%d/%d/%d/%d) to RTP/RTCP receiver",
codec.plname, codec.pltype, codec.plfreq,
codec.channels, codec.rate);
}
else
{
WEBRTC_TRACE(
kTraceInfo,
kTraceVoice,
VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule() %s "
"(%d/%d/%d/%d) has been added to the RTP/RTCP "
"receiver",
codec.plname, codec.pltype, codec.plfreq,
codec.channels, codec.rate);
}
}
}
int
Channel::ApmProcessRx(AudioFrame& audioFrame)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::ApmProcessRx()");
// Reset the APM frequency if the frequency has changed
if (_rxAudioProcessingModulePtr->sample_rate_hz() !=
audioFrame.sample_rate_hz_)
{
if (_rxAudioProcessingModulePtr->set_sample_rate_hz(
audioFrame.sample_rate_hz_) != 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
"AudioProcessingModule::set_sample_rate_hz("
"sample_rate_hz_=%u) => error",
_audioFrame.sample_rate_hz_);
}
}
if (_rxAudioProcessingModulePtr->ProcessStream(&audioFrame) != 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
"AudioProcessingModule::ProcessStream() => error");
}
return 0;
}
} // namespace voe
} // namespace webrtc