Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(44)

Issues Search
    My Issues | Starred     Open | Closed | All

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

Issue 2203233002: Revert of Add EncodedImageCallback::OnEncodedImage(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 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
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.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 ce032eca17a24219e1bf1b3141f62ef47e7bbe20..fed767b687c00e864ba8fd04ac9c243fdab5d2a3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -190,9 +190,9 @@
ASSERT_GE(rtp_length, 0);
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(
- packet_, payload_length, rtp_length, capture_time_ms,
- kAllowRetransmission, RtpPacketSender::kNormalPriority));
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(
+ packet_, payload_length, rtp_length, capture_time_ms,
+ kAllowRetransmission, RtpPacketSender::kNormalPriority));
}
void SendGenericPayload() {
@@ -204,9 +204,9 @@
EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
0, 1500));
- EXPECT_TRUE(rtp_sender_->SendOutgoingData(
- kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs, kPayload,
- sizeof(kPayload), nullptr, nullptr, nullptr));
+ EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
+ kPayload, sizeof(kPayload), nullptr, nullptr));
}
};
@@ -753,9 +753,9 @@
size_t rtp_length = static_cast<size_t>(rtp_length_int);
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- capture_time_ms, kAllowRetransmission,
- RtpPacketSender::kNormalPriority));
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+ capture_time_ms, kAllowRetransmission,
+ RtpPacketSender::kNormalPriority));
EXPECT_EQ(0, transport_.packets_sent_);
@@ -806,9 +806,9 @@
size_t rtp_length = static_cast<size_t>(rtp_length_int);
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- capture_time_ms, kAllowRetransmission,
- RtpPacketSender::kNormalPriority));
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+ capture_time_ms, kAllowRetransmission,
+ RtpPacketSender::kNormalPriority));
EXPECT_EQ(0, transport_.packets_sent_);
@@ -888,9 +888,9 @@
size_t rtp_length = static_cast<size_t>(rtp_length_int);
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- capture_time_ms, kAllowRetransmission,
- RtpPacketSender::kNormalPriority));
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+ capture_time_ms, kAllowRetransmission,
+ RtpPacketSender::kNormalPriority));
int total_packets_sent = 0;
EXPECT_EQ(total_packets_sent, transport_.packets_sent_);
@@ -948,9 +948,9 @@
InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _));
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- capture_time_ms, kAllowRetransmission,
- RtpPacketSender::kNormalPriority));
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+ capture_time_ms, kAllowRetransmission,
+ RtpPacketSender::kNormalPriority));
rtp_sender_->TimeToSendPacket(seq_num, capture_time_ms, false,
PacketInfo::kNotAProbe);
@@ -1115,9 +1115,9 @@
uint8_t payload[] = {47, 11, 32, 93, 89};
// Send keyframe
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
- 4321, payload, sizeof(payload),
- nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1141,9 +1141,9 @@
payload[1] = 42;
payload[4] = 13;
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
- nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1195,18 +1195,18 @@
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
.Times(::testing::AtLeast(2));
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
- 4321, payload, sizeof(payload),
- nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ 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_TRUE(rtp_sender_->SendOutgoingData(
- kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
- nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr));
EXPECT_EQ(2U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
@@ -1268,9 +1268,9 @@
// Send a few frames.
for (uint32_t i = 0; i < kNumPackets; ++i) {
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
- nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameKey, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
@@ -1349,9 +1349,9 @@
rtp_sender_->RegisterRtpStatisticsCallback(&callback);
// Send a frame.
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kVideoFrameKey, payload_type, 1234, 4321, payload,
- sizeof(payload), nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
StreamDataCounters expected;
expected.transmitted.payload_bytes = 6;
expected.transmitted.header_bytes = 12;
@@ -1391,9 +1391,9 @@
fec_params.fec_rate = 1;
fec_params.max_fec_frames = 1;
rtp_sender_->SetFecParameters(&fec_params, &fec_params);
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kVideoFrameDelta, payload_type, 1234, 4321, payload,
- sizeof(payload), nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr));
expected.transmitted.payload_bytes = 40;
expected.transmitted.header_bytes = 60;
expected.transmitted.packets = 5;
@@ -1410,9 +1410,9 @@
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kAudioFrameCN, payload_type, 1234, 4321, payload,
- sizeof(payload), nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1439,9 +1439,9 @@
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kAudioFrameCN, payload_type, 1234, 4321, payload,
- sizeof(payload), nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1490,15 +1490,15 @@
// During start, it takes the starting timestamp as last sent timestamp.
// The duration is calculated as the difference of current and last sent
// timestamp. So for first call it will skip since the duration is zero.
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
- capture_time_ms, 0, nullptr, 0,
- nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
+ capture_time_ms, 0, nullptr, 0,
+ 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_TRUE(rtp_sender_->SendOutgoingData(
- kEmptyFrame, payload_type, capture_time_ms + 2000, 0,
- nullptr, 0, nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
+ capture_time_ms + 2000, 0, nullptr,
+ 0, nullptr, nullptr));
std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
webrtc::RtpHeaderParser::Create());
ASSERT_TRUE(rtp_parser.get() != nullptr);
@@ -1508,9 +1508,9 @@
// Marker Bit should be set to 1 for first packet.
EXPECT_TRUE(rtp_header.markerBit);
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kEmptyFrame, payload_type, capture_time_ms + 4000, 0,
- nullptr, 0, nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
+ capture_time_ms + 4000, 0, nullptr,
+ 0, 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.
@@ -1529,9 +1529,9 @@
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kVideoFrameKey, kPayloadType, 1234, 4321, payload,
- sizeof(payload), nullptr, nullptr, nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
// Will send 2 full-size padding packets.
rtp_sender_->TimeToSendPadding(1, PacketInfo::kNotAProbe);
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698