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Wired up max packet size and added simple test.

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BUG=2428
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2384004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4973 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
sprang@webrtc.org 2013-10-16 11:37:54 +00:00
parent 9401524211
commit 5d957e29f7
3 changed files with 52 additions and 1 deletions
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2
webrtc/video_engine/internal/video_send_stream.cc
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@ -166,6 +166,8 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
assert(network_ != NULL); assert(network_ != NULL);
network_->RegisterSendTransport(channel_, transport_adapter_); network_->RegisterSendTransport(channel_, transport_adapter_);
network_->SetMTU(channel_, config_.rtp.max_packet_size +
VideoSendStream::Config::kDefaultPacketOverheader);
if (config.encoder) { if (config.encoder) {
external_codec_ = ViEExternalCodec::GetInterface(video_engine); external_codec_ = ViEExternalCodec::GetInterface(video_engine);

5
webrtc/video_engine/new_include/video_send_stream.h
View File

@ -84,8 +84,11 @@ class VideoSendStream {
start_state(NULL) {} start_state(NULL) {}
VideoCodec codec; VideoCodec codec;
static const size_t kDefaultPacketOverheader = 20 + 20; // IPv4 + TCP
static const size_t kDefaultMaxPacketSize = 1500 - kDefaultPacketOverheader;
struct Rtp { struct Rtp {
Rtp() : mode(newapi::kRtcpReducedSize), max_packet_size(0) {} Rtp() : mode(newapi::kRtcpReducedSize),
max_packet_size(kDefaultMaxPacketSize) {}
newapi::RtcpMode mode; newapi::RtcpMode mode;
std::vector<uint32_t> ssrcs; std::vector<uint32_t> ssrcs;

46
webrtc/video_engine/test/send_stream_tests.cc
View File

@ -105,6 +105,7 @@ TEST_F(VideoSendStreamTest, SendsSetSsrc) {
scoped_ptr<Call> call(Call::Create(call_config)); scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = GetSendTestConfig(call.get()); VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.max_packet_size = 128;
RunSendTest(call.get(), send_config, &observer); RunSendTest(call.get(), send_config, &observer);
} }
@ -431,4 +432,49 @@ TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
TestNackRetransmission(kSendRtxSsrc); TestNackRetransmission(kSendRtxSsrc);
} }
TEST_F(VideoSendStreamTest, MaxPacketSize) {
class PacketSizeObserver : public SendTransportObserver {
public:
PacketSizeObserver(size_t max_length) : SendTransportObserver(30 * 1000),
max_length_(max_length), accumulated_size_(0) {}
virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
RTPHeader header;
EXPECT_TRUE(
rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));
EXPECT_TRUE(length <= max_length_);
accumulated_size_ += length;
// Marker bit set indicates last fragment of a packet
if (header.markerBit) {
if (accumulated_size_ + length > max_length_) {
// The packet was fragmented, total size was larger than max size,
// but size of individual fragments were within size limit => pass!
send_test_complete_->Set();
}
accumulated_size_ = 0; // Last fragment, reset packet size
}
return true;
}
private:
size_t max_length_;
size_t accumulated_size_;
};
static const uint32_t kMaxPacketSize = 128;
PacketSizeObserver observer(kMaxPacketSize);
Call::Config call_config(&observer);
scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.max_packet_size = kMaxPacketSize;
RunSendTest(call.get(), send_config, &observer);
}
} // namespace webrtc } // namespace webrtc