| /********** |
| 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. |
| // RTP sink for AC3 audio |
| // Implementation |
| |
| #include "AC3AudioRTPSink.hh" |
| |
| AC3AudioRTPSink::AC3AudioRTPSink(UsageEnvironment& env, Groupsock* RTPgs, |
| u_int8_t rtpPayloadFormat, |
| u_int32_t rtpTimestampFrequency) |
| : AudioRTPSink(env, RTPgs, rtpPayloadFormat, |
| rtpTimestampFrequency, "AC3"), |
| fTotNumFragmentsUsed(0) { |
| } |
| |
| AC3AudioRTPSink::~AC3AudioRTPSink() { |
| } |
| |
| AC3AudioRTPSink* |
| AC3AudioRTPSink::createNew(UsageEnvironment& env, Groupsock* RTPgs, |
| u_int8_t rtpPayloadFormat, |
| u_int32_t rtpTimestampFrequency) { |
| return new AC3AudioRTPSink(env, RTPgs, |
| rtpPayloadFormat, rtpTimestampFrequency); |
| } |
| |
| Boolean AC3AudioRTPSink |
| ::frameCanAppearAfterPacketStart(unsigned char const* /*frameStart*/, |
| unsigned /*numBytesInFrame*/) const { |
| // (For now) allow at most 1 frame in a single packet: |
| return False; |
| } |
| |
| void AC3AudioRTPSink |
| ::doSpecialFrameHandling(unsigned fragmentationOffset, |
| unsigned char* frameStart, |
| unsigned numBytesInFrame, |
| struct timeval framePresentationTime, |
| unsigned numRemainingBytes) { |
| // Set the 2-byte "payload header", as defined in RFC 4184. |
| unsigned char headers[2]; |
| |
| Boolean isFragment = numRemainingBytes > 0 || fragmentationOffset > 0; |
| if (!isFragment) { |
| headers[0] = 0; // One or more complete frames |
| headers[1] = 1; // because we (for now) allow at most 1 frame per packet |
| } else { |
| if (fragmentationOffset > 0) { |
| headers[0] = 3; // Fragment of frame other than initial fragment |
| } else { |
| // An initial fragment of the frame |
| unsigned const totalFrameSize = fragmentationOffset + numBytesInFrame + numRemainingBytes; |
| unsigned const fiveEighthsPoint = totalFrameSize/2 + totalFrameSize/8; |
| headers[0] = numBytesInFrame >= fiveEighthsPoint ? 1 : 2; |
| |
| // Because this outgoing packet will be full (because it's an initial fragment), we can compute how many total |
| // fragments (and thus packets) will make up the complete AC-3 frame: |
| fTotNumFragmentsUsed = (totalFrameSize + (numBytesInFrame-1))/numBytesInFrame; |
| } |
| |
| headers[1] = fTotNumFragmentsUsed; |
| } |
| |
| setSpecialHeaderBytes(headers, sizeof headers); |
| |
| if (numRemainingBytes == 0) { |
| // This packet contains the last (or only) fragment of the frame. |
| // Set the RTP 'M' ('marker') bit: |
| setMarkerBit(); |
| } |
| |
| // Important: Also call our base class's doSpecialFrameHandling(), |
| // to set the packet's timestamp: |
| MultiFramedRTPSink::doSpecialFrameHandling(fragmentationOffset, |
| frameStart, numBytesInFrame, |
| framePresentationTime, |
| numRemainingBytes); |
| } |
| |
| unsigned AC3AudioRTPSink::specialHeaderSize() const { |
| return 2; |
| } |