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Add callbacks for send channel rtp statistics

Browse Source 
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
sprang@webrtc.org
2013-12-05 14:29:02 +00:00
parent 5cea89f3e1
commit ebad765ee0
11 changed files with 253 additions and 47 deletions
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79
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -791,6 +791,85 @@ class RtpSenderAudioTest : public RtpSenderTest {
}
};
TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
class TestCallback : public StreamDataCountersCallback {
public:
TestCallback()
: StreamDataCountersCallback(), ssrc_(0), counters_() {}
virtual ~TestCallback() {}
virtual void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) {
ssrc_ = ssrc;
counters_ = counters;
}
uint32_t ssrc_;
StreamDataCounters counters_;
bool Matches(uint32_t ssrc, uint32_t bytes, uint32_t header_bytes,
uint32_t padding, uint32_t packets, uint32_t retransmits,
uint32_t fec) {
return ssrc_ == ssrc &&
counters_.bytes == bytes &&
counters_.header_bytes == header_bytes &&
counters_.padding_bytes == padding &&
counters_.packets == packets &&
counters_.retransmitted_packets == retransmits &&
counters_.fec_packets == fec;
}
} callback;
const uint8_t kRedPayloadType = 96;
const uint8_t kUlpfecPayloadType = 97;
const uint32_t kMaxPaddingSize = 224;
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
rtp_sender_->RegisterRtpStatisticsCallback(&callback);
// Send a frame.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
4321, payload, sizeof(payload),
NULL));
// {bytes = 6, header = 12, padding = 0, packets = 1, retrans = 0, fec = 0}
EXPECT_TRUE(callback.Matches(ssrc, 6, 12, 0, 1, 0, 0));
// Retransmit a frame.
uint16_t seqno = rtp_sender_->SequenceNumber() - 1;
rtp_sender_->ReSendPacket(seqno, 0);
// bytes = 6, header = 12, padding = 0, packets = 2, retrans = 1, fec = 0}
EXPECT_TRUE(callback.Matches(ssrc, 6, 12, 0, 2, 1, 0));
// Send padding.
rtp_sender_->TimeToSendPadding(kMaxPaddingSize);
// {bytes = 6, header = 24, padding = 224, packets = 3, retrans = 1, fec = 0}
EXPECT_TRUE(callback.Matches(ssrc, 6, 24, 224, 3, 1, 0));
// Send FEC.
rtp_sender_->SetGenericFECStatus(true, kRedPayloadType, kUlpfecPayloadType);
FecProtectionParams fec_params;
fec_params.fec_mask_type = kFecMaskRandom;
fec_params.fec_rate = 1;
fec_params.max_fec_frames = 1;
rtp_sender_->SetFecParameters(&fec_params, &fec_params);
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
1234, 4321, payload,
sizeof(payload), NULL));
// {bytes = 34, header = 48, padding = 224, packets = 5, retrans = 1, fec = 1}
EXPECT_TRUE(callback.Matches(ssrc, 34, 48, 224, 5, 1, 1));
rtp_sender_->RegisterRtpStatisticsCallback(NULL);
}
TEST_F(RtpSenderAudioTest, BuildRTPPacketWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(true,
kAudioLevelExtensionId));