/**********
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) 1996-2022 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;
}
