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Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.

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R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andresp@webrtc.org
2014-07-10 09:39:23 +00:00
parent 5bde66e913
commit 8f1512140e
8 changed files with 34 additions and 56 deletions
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15
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -94,7 +94,7 @@ class RtpSenderTest : public ::testing::Test {
virtual void SetUp() {
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
&mock_paced_sender_, NULL));
&mock_paced_sender_, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
@ -672,7 +672,7 @@ TEST_F(RtpSenderTest, SendPadding) {
TEST_F(RtpSenderTest, SendRedundantPayloads) {
MockTransport transport;
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL,
&mock_paced_sender_, NULL));
&mock_paced_sender_, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
// Make all packets go through the pacer.
EXPECT_CALL(mock_paced_sender_,
@ -817,6 +817,9 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
uint32_t delta_frames_;
} callback;
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
&mock_paced_sender_, NULL, &callback));
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,
@ -825,8 +828,6 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
rtp_sender_->RegisterFrameCountObserver(&callback);
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
4321, payload, sizeof(payload),
NULL));
@ -845,7 +846,7 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
EXPECT_EQ(1U, callback.key_frames_);
EXPECT_EQ(1U, callback.delta_frames_);
rtp_sender_->RegisterFrameCountObserver(NULL);
rtp_sender_.reset();
}
TEST_F(RtpSenderTest, BitrateCallbacks) {
@ -866,7 +867,7 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
BitrateStatistics bitrate_;
} callback;
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
&mock_paced_sender_, &callback));
&mock_paced_sender_, &callback, NULL));
// Simulate kNumPackets sent with kPacketInterval ms intervals.
const uint32_t kNumPackets = 15;
@ -922,7 +923,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
virtual void SetUp() {
payload_ = kAudioPayload;
rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, NULL,
&mock_paced_sender_, NULL));
&mock_paced_sender_, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
};