Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(215)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Issue 2089773002: Add EncodedImageCallback::OnEncodedImage(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: update SendOutgoingData() Created 4 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('K') | « webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('k') | webrtc/modules/video_coding/codecs/test/videoprocessor.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 99cef009e733a38b9727ddeeb0305951005751dd..73bd2f26234c701e4ca4c84dfa356f77a83f8e2d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -206,7 +206,7 @@ class RtpSenderTest : public ::testing::Test {
EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
- kPayload, sizeof(kPayload), nullptr));
+ kPayload, sizeof(kPayload), nullptr, nullptr, nullptr));
}
};
@@ -1109,9 +1109,9 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
uint8_t payload[] = {47, 11, 32, 93, 89};
// Send keyframe
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1135,9 +1135,9 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
payload[1] = 42;
payload[4] = 13;
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
- 1234, 4321, payload,
- sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1188,18 +1188,18 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
.Times(::testing::AtLeast(2));
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr, nullptr));
EXPECT_EQ(1U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
EXPECT_EQ(1, callback.frame_counts_.key_frames);
EXPECT_EQ(0, callback.frame_counts_.delta_frames);
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
- 1234, 4321, payload,
- sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr, nullptr));
EXPECT_EQ(2U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
@@ -1261,9 +1261,9 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
// Send a few frames.
for (uint32_t i = 0; i < kNumPackets; ++i) {
- ASSERT_EQ(0,
- rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
- 4321, payload, sizeof(payload), 0));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameKey, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr, nullptr));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
@@ -1342,9 +1342,9 @@ TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
rtp_sender_->RegisterRtpStatisticsCallback(&callback);
// Send a frame.
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr, nullptr));
StreamDataCounters expected;
expected.transmitted.payload_bytes = 6;
expected.transmitted.header_bytes = 12;
@@ -1384,9 +1384,9 @@ TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
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), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr, nullptr));
expected.transmitted.payload_bytes = 40;
expected.transmitted.header_bytes = 60;
expected.transmitted.packets = 5;
@@ -1403,9 +1403,9 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1432,9 +1432,9 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1485,13 +1485,13 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
// timestamp. So for first call it will skip since the duration is zero.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms, 0, nullptr, 0,
- nullptr));
+ nullptr, nullptr, nullptr));
// DTMF Sample Length is (Frequency/1000) * Duration.
// So in this case, it is (8000/1000) * 500 = 4000.
// Sending it as two packets.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms + 2000, 0, nullptr,
- 0, nullptr));
+ 0, nullptr, nullptr, nullptr));
std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
webrtc::RtpHeaderParser::Create());
ASSERT_TRUE(rtp_parser.get() != nullptr);
@@ -1503,7 +1503,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms + 4000, 0, nullptr,
- 0, nullptr));
+ 0, nullptr, nullptr, nullptr));
ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
transport_.last_sent_packet_len_, &rtp_header));
// Marker Bit should be set to 0 for rest of the packets.
@@ -1522,9 +1522,9 @@ TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234, 4321,
- payload, sizeof(payload), 0));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr, nullptr));
// Will send 2 full-size padding packets.
rtp_sender_->TimeToSendPadding(1, PacketInfo::kNotAProbe);
« webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('K') | « webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('k') | webrtc/modules/video_coding/codecs/test/videoprocessor.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698