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Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData

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It appears unused everywhere. It will be deleted in a followup cl.

Bug: webrtc:6471
Change-Id: Ief992db6e52aee3cf1bc77ffd659ffbc072672ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134212
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27787}
This commit is contained in:
Niels Möller
2019-04-25 16:31:18 +02:00
committed by Commit Bot
parent e670fd9795
commit c35b6e675a
17 changed files with 54 additions and 121 deletions
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49
modules/audio_coding/test/Channel.cc
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@ -22,8 +22,7 @@ int32_t Channel::SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) {
size_t payloadSize) {
RTPHeader rtp_header;
int32_t status;
size_t payloadDataSize = payloadSize;
@ -46,46 +45,14 @@ int32_t Channel::SendData(AudioFrameType frameType,
return 0;
}
// Treat fragmentation separately
if (fragmentation != NULL) {
// If silence for too long, send only new data.
if ((fragmentation->fragmentationVectorSize == 2) &&
(fragmentation->fragmentationTimeDiff[1] <= 0x3fff)) {
// only 0x80 if we have multiple blocks
_payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
fragmentation->fragmentationLength[1];
_payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
_payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
_payloadData[3] = uint8_t(REDheader & 0x000000FF);
_payloadData[4] = fragmentation->fragmentationPlType[0];
// copy the RED data
memcpy(_payloadData + 5,
payloadData + fragmentation->fragmentationOffset[1],
fragmentation->fragmentationLength[1]);
// copy the normal data
memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
payloadData + fragmentation->fragmentationOffset[0],
fragmentation->fragmentationLength[0]);
payloadDataSize += 5;
memcpy(_payloadData, payloadData, payloadDataSize);
if (_isStereo) {
if (_leftChannel) {
_rtp_header = rtp_header;
_leftChannel = false;
} else {
// single block (newest one)
memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
fragmentation->fragmentationLength[0]);
payloadDataSize = fragmentation->fragmentationLength[0];
rtp_header.payloadType = fragmentation->fragmentationPlType[0];
}
} else {
memcpy(_payloadData, payloadData, payloadDataSize);
if (_isStereo) {
if (_leftChannel) {
_rtp_header = rtp_header;
_leftChannel = false;
} else {
rtp_header = _rtp_header;
_leftChannel = true;
}
rtp_header = _rtp_header;
_leftChannel = true;
}
}