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rtc::Buffer: Rename length to size, for conformance with the STL

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And add a constructor for creating an uninitialized Buffer of a
specified size.

(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48579004

Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2015-03-24 09:19:06 +00:00
parent e815290828
commit eebcab5ce9
33 changed files with 168 additions and 188 deletions
Download Patch File Download Diff File Expand all files Collapse all files

23
talk/media/base/rtpdataengine.cc
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@@ -216,7 +216,7 @@ bool RtpDataMediaChannel::RemoveRecvStream(uint32 ssrc) {
void RtpDataMediaChannel::OnPacketReceived(
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
RtpHeader header;
if (!GetRtpHeader(packet->data(), packet->length(), &header)) {
if (!GetRtpHeader(packet->data(), packet->size(), &header)) {
// Don't want to log for every corrupt packet.
// LOG(LS_WARNING) << "Could not read rtp header from packet of length "
// << packet->length() << ".";
@@ -224,7 +224,7 @@ void RtpDataMediaChannel::OnPacketReceived(
}
size_t header_length;
if (!GetRtpHeaderLen(packet->data(), packet->length(), &header_length)) {
if (!GetRtpHeaderLen(packet->data(), packet->size(), &header_length)) {
// Don't want to log for every corrupt packet.
// LOG(LS_WARNING) << "Could not read rtp header"
// << length from packet of length "
@@ -232,7 +232,7 @@ void RtpDataMediaChannel::OnPacketReceived(
return;
}
const char* data = packet->data() + header_length + sizeof(kReservedSpace);
size_t data_len = packet->length() - header_length - sizeof(kReservedSpace);
size_t data_len = packet->size() - header_length - sizeof(kReservedSpace);
if (!receiving_) {
LOG(LS_WARNING) << "Not receiving packet "
@@ -292,7 +292,7 @@ bool RtpDataMediaChannel::SendData(
}
if (!sending_) {
LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssrc
<< " len=" << payload.length() << " before SetSend(true).";
<< " len=" << payload.size() << " before SetSend(true).";
return false;
}
@@ -316,8 +316,8 @@ bool RtpDataMediaChannel::SendData(
return false;
}
size_t packet_len = (kMinRtpPacketLen + sizeof(kReservedSpace)
+ payload.length() + kMaxSrtpHmacOverhead);
size_t packet_len = (kMinRtpPacketLen + sizeof(kReservedSpace) +
payload.size() + kMaxSrtpHmacOverhead);
if (packet_len > kDataMaxRtpPacketLen) {
return false;
}
@@ -339,19 +339,18 @@ bool RtpDataMediaChannel::SendData(
rtc::Buffer packet;
packet.SetCapacity(packet_len);
packet.SetLength(kMinRtpPacketLen);
if (!SetRtpHeader(packet.data(), packet.length(), header)) {
packet.SetSize(kMinRtpPacketLen);
if (!SetRtpHeader(packet.data(), packet.size(), header)) {
return false;
}
packet.AppendData(&kReservedSpace, sizeof(kReservedSpace));
packet.AppendData(payload.data(), payload.length());
packet.AppendData(payload.data(), payload.size());
LOG(LS_VERBOSE) << "Sent RTP data packet: "
<< " stream=" << found_stream->id
<< " ssrc=" << header.ssrc
<< " stream=" << found_stream->id << " ssrc=" << header.ssrc
<< ", seqnum=" << header.seq_num
<< ", timestamp=" << header.timestamp
<< ", len=" << payload.length();
<< ", len=" << payload.size();
MediaChannel::SendPacket(&packet);
send_limiter_->Use(packet_len, now);