| /* |
| * 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 "rtp_receiver_audio.h" |
| |
| #include <cassert> //assert |
| #include <cstring> // memcpy() |
| #include <math.h> // pow() |
| |
| #include "critical_section_wrapper.h" |
| |
| namespace webrtc { |
| RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id): |
| _id(id), |
| _lastReceivedFrequency(8000), |
| _telephoneEvent(false), |
| _telephoneEventForwardToDecoder(false), |
| _telephoneEventDetectEndOfTone(false), |
| _telephoneEventPayloadType(-1), |
| _cngNBPayloadType(-1), |
| _cngWBPayloadType(-1), |
| _cngSWBPayloadType(-1), |
| _cngPayloadType(-1), |
| _G722PayloadType(-1), |
| _lastReceivedG722(false), |
| _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()), |
| _cbAudioFeedback(NULL) |
| { |
| } |
| |
| RTPReceiverAudio::~RTPReceiverAudio() |
| { |
| delete _criticalSectionFeedback; |
| } |
| |
| WebRtc_Word32 RTPReceiverAudio::Init() { |
| _lastReceivedFrequency = 8000; |
| _telephoneEvent = false; |
| _telephoneEventForwardToDecoder = false; |
| _telephoneEventDetectEndOfTone = false; |
| _telephoneEventPayloadType = -1; |
| |
| _telephoneEventReported.clear(); |
| |
| _cngNBPayloadType = -1; |
| _cngWBPayloadType = -1; |
| _cngSWBPayloadType = -1; |
| _cngPayloadType = -1; |
| _G722PayloadType = -1; |
| _lastReceivedG722 = false; |
| return 0; |
| } |
| |
| void |
| RTPReceiverAudio::ChangeUniqueId(const WebRtc_Word32 id) |
| { |
| _id = id; |
| } |
| |
| WebRtc_Word32 |
| RTPReceiverAudio::RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback) |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedback); |
| _cbAudioFeedback = incomingMessagesCallback; |
| return 0; |
| } |
| |
| WebRtc_UWord32 |
| RTPReceiverAudio::AudioFrequency() const |
| { |
| if(_lastReceivedG722) |
| { |
| return 8000; |
| } |
| return _lastReceivedFrequency; |
| } |
| |
| // Outband TelephoneEvent(DTMF) detection |
| WebRtc_Word32 |
| RTPReceiverAudio::SetTelephoneEventStatus(const bool enable, |
| const bool forwardToDecoder, |
| const bool detectEndOfTone) |
| { |
| _telephoneEvent= enable; |
| _telephoneEventDetectEndOfTone = detectEndOfTone; |
| _telephoneEventForwardToDecoder = forwardToDecoder; |
| return 0; |
| } |
| |
| // Is outband TelephoneEvent(DTMF) turned on/off? |
| bool |
| RTPReceiverAudio::TelephoneEvent() const |
| { |
| return _telephoneEvent; |
| } |
| |
| // Is forwarding of outband telephone events turned on/off? |
| bool |
| RTPReceiverAudio::TelephoneEventForwardToDecoder() const |
| { |
| return _telephoneEventForwardToDecoder; |
| } |
| |
| bool |
| RTPReceiverAudio::TelephoneEventPayloadType(const WebRtc_Word8 payloadType) const |
| { |
| return (_telephoneEventPayloadType == payloadType)?true:false; |
| } |
| |
| bool |
| RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payloadType, |
| WebRtc_UWord32& frequency) |
| { |
| // we can have three CNG on 8000Hz, 16000Hz and 32000Hz |
| if(_cngNBPayloadType == payloadType) |
| { |
| frequency = 8000; |
| if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngNBPayloadType)) |
| { |
| ResetStatistics(); |
| } |
| _cngPayloadType = _cngNBPayloadType; |
| return true; |
| } else if(_cngWBPayloadType == payloadType) |
| { |
| // if last received codec is G.722 we must use frequency 8000 |
| if(_lastReceivedG722) |
| { |
| frequency = 8000; |
| } else |
| { |
| frequency = 16000; |
| } |
| if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngWBPayloadType)) |
| { |
| ResetStatistics(); |
| } |
| _cngPayloadType = _cngWBPayloadType; |
| return true; |
| }else if(_cngSWBPayloadType == payloadType) |
| { |
| frequency = 32000; |
| if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngSWBPayloadType)) |
| { |
| ResetStatistics(); |
| } |
| _cngPayloadType = _cngSWBPayloadType; |
| return true; |
| }else |
| { |
| // not CNG |
| if(_G722PayloadType == payloadType) |
| { |
| _lastReceivedG722 = true; |
| }else |
| { |
| _lastReceivedG722 = false; |
| } |
| } |
| return false; |
| } |
| |
| /* |
| Sample based or frame based codecs based on RFC 3551 |
| |
| NOTE! There is one error in the RFC, stating G.722 uses 8 bits/samples. |
| The correct rate is 4 bits/sample. |
| |
| name of sampling default |
| encoding sample/frame bits/sample rate ms/frame ms/packet |
| |
| Sample based audio codecs |
| DVI4 sample 4 var. 20 |
| G722 sample 4 16,000 20 |
| G726-40 sample 5 8,000 20 |
| G726-32 sample 4 8,000 20 |
| G726-24 sample 3 8,000 20 |
| G726-16 sample 2 8,000 20 |
| L8 sample 8 var. 20 |
| L16 sample 16 var. 20 |
| PCMA sample 8 var. 20 |
| PCMU sample 8 var. 20 |
| |
| Frame based audio codecs |
| G723 frame N/A 8,000 30 30 |
| G728 frame N/A 8,000 2.5 20 |
| G729 frame N/A 8,000 10 20 |
| G729D frame N/A 8,000 10 20 |
| G729E frame N/A 8,000 10 20 |
| GSM frame N/A 8,000 20 20 |
| GSM-EFR frame N/A 8,000 20 20 |
| LPC frame N/A 8,000 20 20 |
| MPA frame N/A var. var. |
| |
| G7221 frame N/A |
| */ |
| |
| ModuleRTPUtility::Payload* RTPReceiverAudio::RegisterReceiveAudioPayload( |
| const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| const WebRtc_Word8 payloadType, |
| const WebRtc_UWord32 frequency, |
| const WebRtc_UWord8 channels, |
| const WebRtc_UWord32 rate) { |
| if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) { |
| _telephoneEventPayloadType = payloadType; |
| } |
| if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) { |
| // we can have three CNG on 8000Hz, 16000Hz and 32000Hz |
| if(frequency == 8000){ |
| _cngNBPayloadType = payloadType; |
| } else if(frequency == 16000) { |
| _cngWBPayloadType = payloadType; |
| } else if(frequency == 32000) { |
| _cngSWBPayloadType = payloadType; |
| } else { |
| assert(false); |
| return NULL; |
| } |
| } |
| WebRtc_UWord8 bitsPerSample = 0; // zero implies frame based |
| if (ModuleRTPUtility::StringCompare(payloadName, "DVI4", 4)) { |
| bitsPerSample = 4; |
| } else if(ModuleRTPUtility::StringCompare(payloadName, "G722", 4)) { |
| if(ModuleRTPUtility::StringCompare(payloadName, "G7221", 5)) { |
| // frame based |
| } else { |
| _G722PayloadType = payloadType; |
| bitsPerSample = 4; |
| } |
| } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-40",7)) { |
| bitsPerSample = 5; |
| } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-32",7)) { |
| bitsPerSample = 4; |
| } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-24",7)) { |
| bitsPerSample = 3; |
| } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-16",7)) { |
| bitsPerSample = 2; |
| } else if(ModuleRTPUtility::StringCompare(payloadName,"L8",2)) { |
| bitsPerSample = 8; |
| } else if(ModuleRTPUtility::StringCompare(payloadName,"L16",3)) { |
| bitsPerSample = 16; |
| } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMU",4)) { |
| bitsPerSample = 8; |
| } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMA",4)) { |
| bitsPerSample = 8; |
| } |
| ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload; |
| payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; |
| strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); |
| payload->typeSpecific.Audio.frequency = frequency; |
| payload->typeSpecific.Audio.channels = channels; |
| payload->typeSpecific.Audio.bitsPerSample = bitsPerSample; |
| payload->typeSpecific.Audio.rate = rate; |
| payload->audio = true; |
| return payload; |
| } |
| |
| // we are not allowed to have any critsects when calling CallbackOfReceivedPayloadData |
| WebRtc_Word32 |
| RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader, |
| const WebRtc_UWord8* payloadData, |
| const WebRtc_UWord16 payloadLength, |
| const ModuleRTPUtility::AudioPayload& audioSpecific, |
| const bool isRED) |
| { |
| WebRtc_UWord8 newEvents[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS]; |
| WebRtc_UWord8 removedEvents[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS]; |
| WebRtc_UWord8 numberOfNewEvents = 0; |
| WebRtc_UWord8 numberOfRemovedEvents = 0; |
| bool telephoneEventPacket = TelephoneEventPayloadType(rtpHeader->header.payloadType); |
| |
| if(payloadLength == 0) |
| { |
| return 0; |
| } |
| |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedback); |
| |
| if(telephoneEventPacket) |
| { |
| // RFC 4733 2.3 |
| /* |
| 0 1 2 3 |
| 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | event |E|R| volume | duration | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| */ |
| if(payloadLength % 4 != 0) |
| { |
| return -1; |
| } |
| WebRtc_UWord8 numberOfEvents = payloadLength / 4; |
| |
| // sanity |
| if(numberOfEvents >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS) |
| { |
| numberOfEvents = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS; |
| } |
| for (int n = 0; n < numberOfEvents; n++) |
| { |
| bool end = (payloadData[(4*n)+1] & 0x80)? true:false; |
| |
| std::set<WebRtc_UWord8>::iterator event = |
| _telephoneEventReported.find(payloadData[4*n]); |
| |
| if(event != _telephoneEventReported.end()) |
| { |
| // we have already seen this event |
| if(end) |
| { |
| removedEvents[numberOfRemovedEvents]= payloadData[4*n]; |
| numberOfRemovedEvents++; |
| _telephoneEventReported.erase(payloadData[4*n]); |
| } |
| }else |
| { |
| if(end) |
| { |
| // don't add if it's a end of a tone |
| }else |
| { |
| newEvents[numberOfNewEvents] = payloadData[4*n]; |
| numberOfNewEvents++; |
| _telephoneEventReported.insert(payloadData[4*n]); |
| } |
| } |
| } |
| |
| // RFC 4733 2.5.1.3 & 2.5.2.3 Long-Duration Events |
| // should not be a problem since we don't care about the duration |
| |
| // RFC 4733 See 2.5.1.5. & 2.5.2.4. Multiple Events in a Packet |
| } |
| |
| if(_telephoneEvent && _cbAudioFeedback) |
| { |
| for (int n = 0; n < numberOfNewEvents; n++) |
| { |
| _cbAudioFeedback->OnReceivedTelephoneEvent(_id, newEvents[n], false); |
| } |
| if(_telephoneEventDetectEndOfTone) |
| { |
| for (int n = 0; n < numberOfRemovedEvents; n++) |
| { |
| _cbAudioFeedback->OnReceivedTelephoneEvent(_id, removedEvents[n], true); |
| } |
| } |
| } |
| } |
| if(! telephoneEventPacket ) |
| { |
| _lastReceivedFrequency = audioSpecific.frequency; |
| } |
| |
| // Check if this is a CNG packet, receiver might want to know |
| WebRtc_UWord32 dummy; |
| if(CNGPayloadType(rtpHeader->header.payloadType, dummy)) |
| { |
| rtpHeader->type.Audio.isCNG=true; |
| rtpHeader->frameType = kAudioFrameCN; |
| }else |
| { |
| rtpHeader->frameType = kAudioFrameSpeech; |
| rtpHeader->type.Audio.isCNG=false; |
| } |
| |
| // check if it's a DTMF event, hence something we can playout |
| if(telephoneEventPacket) |
| { |
| if(!_telephoneEventForwardToDecoder) |
| { |
| // don't forward event to decoder |
| return 0; |
| } |
| std::set<WebRtc_UWord8>::iterator first = |
| _telephoneEventReported.begin(); |
| if(first != _telephoneEventReported.end() && *first > 15) |
| { |
| // don't forward non DTMF events |
| return 0; |
| } |
| } |
| if(isRED && !(payloadData[0] & 0x80)) |
| { |
| // we recive only one frame packed in a RED packet remove the RED wrapper |
| rtpHeader->header.payloadType = payloadData[0]; |
| |
| // only one frame in the RED strip the one byte to help NetEq |
| return CallbackOfReceivedPayloadData(payloadData+1, |
| payloadLength-1, |
| rtpHeader); |
| } |
| if(audioSpecific.channels > 1) |
| { |
| WebRtc_Word32 retVal = 0; |
| WebRtc_UWord16 channelLength = payloadLength/audioSpecific.channels; |
| |
| if(audioSpecific.bitsPerSample > 0) |
| { |
| // sanity |
| assert((payloadLength*8)%audioSpecific.bitsPerSample == 0); |
| |
| // sample based codec |
| |
| // build matrix |
| WebRtc_UWord8 matrix[IP_PACKET_SIZE]; |
| WebRtc_UWord32 offsetBytes = 0; |
| WebRtc_UWord32 offsetBytesInsert = 0; |
| // initialize matrix to 0 |
| memset(matrix, 0, audioSpecific.channels*channelLength); |
| |
| switch(audioSpecific.bitsPerSample) |
| { |
| case 1: |
| case 2: |
| case 3: |
| case 4: |
| case 5: |
| case 6: |
| case 7: |
| { |
| WebRtc_UWord32 offsetSamples = 0; |
| WebRtc_UWord32 offsetSamplesInsert = 0; |
| WebRtc_UWord16 bitMask = (WebRtc_UWord16)ModuleRTPUtility::pow2(audioSpecific.bitsPerSample)-1; |
| WebRtc_UWord16 samplesPerChannel =payloadLength*8/audioSpecific.bitsPerSample/audioSpecific.channels; |
| |
| for(WebRtc_UWord32 i = 0; i < samplesPerChannel; i++) |
| { |
| WebRtc_UWord8 insertShift = (WebRtc_UWord8)((offsetSamplesInsert+audioSpecific.bitsPerSample)%16); |
| insertShift = 16 - insertShift; // inverse the calculation |
| |
| for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++) |
| { |
| // get sample |
| WebRtc_UWord16 s = payloadData[offsetBytes] << 8; |
| |
| // check that we don't read outside the memory |
| if(offsetBytes < (WebRtc_UWord32)payloadLength -2) |
| { |
| s += payloadData[offsetBytes+1]; |
| } |
| |
| WebRtc_UWord8 readShift = (WebRtc_UWord8)((offsetSamples+audioSpecific.bitsPerSample)%16); |
| readShift = 16 - readShift; // inverse the calculation |
| s >>= readShift; |
| s &= bitMask; |
| |
| // prepare for reading next sample |
| offsetSamples += audioSpecific.bitsPerSample; |
| if(readShift <= audioSpecific.bitsPerSample) |
| { |
| // next does not fitt |
| // or fitt exactly |
| offsetSamples -= 8; |
| offsetBytes++; |
| } |
| |
| // insert sample into matrix |
| WebRtc_UWord32 columOffset = j*channelLength; |
| |
| WebRtc_UWord16 insert = s << insertShift; |
| #if defined(WEBRTC_LITTLE_ENDIAN) |
| matrix[columOffset+offsetBytesInsert] |= static_cast<WebRtc_UWord8>(insert>>8); |
| matrix[columOffset+offsetBytesInsert+1] |= static_cast<WebRtc_UWord8>(insert); |
| #else |
| WebRtc_UWord16* matrixU16 = (WebRtc_UWord16*)&(matrix[columOffset+offsetBytesInsert]); |
| matrixU16[0] |= (s << insertShift); |
| #endif |
| } |
| // prepare for writing next sample |
| offsetSamplesInsert += audioSpecific.bitsPerSample; |
| if(insertShift <= audioSpecific.bitsPerSample) |
| { |
| // next does not fitt |
| // or fitt exactly |
| offsetSamplesInsert -= 8; |
| offsetBytesInsert++; |
| } |
| } |
| } |
| break; |
| case 8: |
| { |
| WebRtc_UWord32 sample = 0; |
| for(WebRtc_UWord32 i = 0; i < channelLength; i++) |
| { |
| for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++) |
| { |
| WebRtc_UWord32 columOffset = j*channelLength; |
| matrix[columOffset + i] = payloadData[sample++]; |
| } |
| } |
| } |
| break; |
| case 16: |
| { |
| WebRtc_UWord32 sample = 0; |
| for(WebRtc_UWord32 i = 0; i < channelLength; i +=2) |
| { |
| for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++) |
| { |
| WebRtc_UWord32 columOffset = j*channelLength; |
| matrix[columOffset + i] = payloadData[sample++]; |
| matrix[columOffset + i + 1] = payloadData[sample++]; |
| } |
| } |
| } |
| break; |
| default: |
| assert(false); |
| return -1; |
| } |
| // we support 16 bits sample |
| // callback for all channels |
| for(int channel = 0; channel < audioSpecific.channels && retVal == 0; channel++) |
| { |
| // one callback per channel |
| rtpHeader->type.Audio.channel = channel+1; |
| |
| if(channel == 0) |
| { |
| // include the original packet only in the first callback |
| retVal = CallbackOfReceivedPayloadData(&matrix[channel*channelLength], |
| channelLength, |
| rtpHeader); |
| } else |
| { |
| retVal = CallbackOfReceivedPayloadData(&matrix[channel*channelLength], |
| channelLength, |
| rtpHeader); |
| } |
| } |
| } else |
| { |
| for(int channel = 1; channel <= audioSpecific.channels && retVal == 0; channel++) |
| { |
| // one callback per channel |
| rtpHeader->type.Audio.channel = channel; |
| |
| if(channel == 1) |
| { |
| // include the original packet only in the first callback |
| retVal = CallbackOfReceivedPayloadData(payloadData, |
| channelLength, |
| rtpHeader); |
| } else |
| { |
| retVal = CallbackOfReceivedPayloadData(payloadData, |
| channelLength, |
| rtpHeader); |
| } |
| payloadData += channelLength; |
| } |
| } |
| return retVal; |
| }else |
| { |
| rtpHeader->type.Audio.channel = 1; |
| return CallbackOfReceivedPayloadData(payloadData, |
| payloadLength, |
| rtpHeader); |
| } |
| } |
| } // namespace webrtc |
