| /********** |
| This library is free software; you can redistribute it and/or modify it under |
| the terms of the GNU Lesser General Public License as published by the |
| Free Software Foundation; either version 3 of the License, or (at your |
| option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.) |
| |
| This library is distributed in the hope that it will be useful, but WITHOUT |
| ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS |
| FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for |
| more details. |
| |
| You should have received a copy of the GNU Lesser General Public License |
| along with this library; if not, write to the Free Software Foundation, Inc., |
| 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| **********/ |
| // "liveMedia" |
| // Copyright (c) 1996-2020 Live Networks, Inc. All rights reserved. |
| // AC3 Audio RTP Sources |
| // Implementation |
| |
| #include "AC3AudioRTPSource.hh" |
| |
| AC3AudioRTPSource* |
| AC3AudioRTPSource::createNew(UsageEnvironment& env, |
| Groupsock* RTPgs, |
| unsigned char rtpPayloadFormat, |
| unsigned rtpTimestampFrequency) { |
| return new AC3AudioRTPSource(env, RTPgs, rtpPayloadFormat, |
| rtpTimestampFrequency); |
| } |
| |
| AC3AudioRTPSource::AC3AudioRTPSource(UsageEnvironment& env, |
| Groupsock* rtpGS, |
| unsigned char rtpPayloadFormat, |
| unsigned rtpTimestampFrequency) |
| : MultiFramedRTPSource(env, rtpGS, |
| rtpPayloadFormat, rtpTimestampFrequency) { |
| } |
| |
| AC3AudioRTPSource::~AC3AudioRTPSource() { |
| } |
| |
| Boolean AC3AudioRTPSource |
| ::processSpecialHeader(BufferedPacket* packet, |
| unsigned& resultSpecialHeaderSize) { |
| unsigned char* headerStart = packet->data(); |
| unsigned packetSize = packet->dataSize(); |
| |
| // There's a 2-byte payload header at the beginning: |
| if (packetSize < 2) return False; |
| resultSpecialHeaderSize = 2; |
| |
| unsigned char FT = headerStart[0]&0x03; |
| fCurrentPacketBeginsFrame = FT != 3; |
| |
| // The RTP "M" (marker) bit indicates the last fragment of a frame. |
| // In case the sender did not set the "M" bit correctly, we also test for FT == 0: |
| fCurrentPacketCompletesFrame = packet->rtpMarkerBit() || FT == 0; |
| |
| return True; |
| } |
| |
| char const* AC3AudioRTPSource::MIMEtype() const { |
| return "audio/AC3"; |
| } |
| |