zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Add EncodedImageCallback::OnEncodedImage().

Browse Source 
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Original-Commit-Position: refs/heads/master@{#13615}
Cr-Commit-Position: refs/heads/master@{#13617}
This commit is contained in:
Sergey Ulanov
2016-08-02 17:46:41 -07:00
parent 51db4dd1bd
commit 525df3ffd1
43 changed files with 477 additions and 382 deletions
Download Patch File Download Diff File Expand all files Collapse all files

114
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -190,9 +190,9 @@ class RtpSenderTest : public ::testing::Test {
ASSERT_GE(rtp_length, 0);
// Packet should be stored in a send bucket.
EXPECT_EQ(0, rtp_sender_->SendToNetwork(
packet_, payload_length, rtp_length, capture_time_ms,
kAllowRetransmission, RtpPacketSender::kNormalPriority));
EXPECT_TRUE(rtp_sender_->SendToNetwork(
packet_, payload_length, rtp_length, capture_time_ms,
kAllowRetransmission, RtpPacketSender::kNormalPriority));
}
void SendGenericPayload() {
@ -204,9 +204,9 @@ class RtpSenderTest : public ::testing::Test {
EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
0, 1500));
EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
kPayload, sizeof(kPayload), nullptr, nullptr));
EXPECT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs, kPayload,
sizeof(kPayload), nullptr, nullptr, nullptr));
}
};
@ -753,9 +753,9 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
size_t rtp_length = static_cast<size_t>(rtp_length_int);
// Packet should be stored in a send bucket.
EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
capture_time_ms, kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
capture_time_ms, kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_EQ(0, transport_.packets_sent_);
@ -806,9 +806,9 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
size_t rtp_length = static_cast<size_t>(rtp_length_int);
// Packet should be stored in a send bucket.
EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
capture_time_ms, kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
capture_time_ms, kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_EQ(0, transport_.packets_sent_);
@ -888,9 +888,9 @@ TEST_F(RtpSenderTest, SendPadding) {
size_t rtp_length = static_cast<size_t>(rtp_length_int);
// Packet should be stored in a send bucket.
EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
capture_time_ms, kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_TRUE(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 @@ TEST_F(RtpSenderTest, SendPadding) {
InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _));
// Packet should be stored in a send bucket.
EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
capture_time_ms, kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_TRUE(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 @@ 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, nullptr));
ASSERT_TRUE(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_);
@ -1141,9 +1141,9 @@ 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, nullptr));
ASSERT_TRUE(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_);
@ -1195,18 +1195,18 @@ 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, nullptr));
ASSERT_TRUE(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, nullptr));
ASSERT_TRUE(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_);
@ -1268,9 +1268,9 @@ 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), nullptr, nullptr));
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
@ -1349,9 +1349,9 @@ 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, nullptr));
ASSERT_TRUE(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;
@ -1391,9 +1391,9 @@ 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, nullptr));
ASSERT_TRUE(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;
@ -1410,9 +1410,9 @@ 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, nullptr));
ASSERT_TRUE(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_);
@ -1439,9 +1439,9 @@ 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, nullptr));
ASSERT_TRUE(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_);
@ -1490,15 +1490,15 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
// 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_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms, 0, nullptr, 0,
nullptr, nullptr));
ASSERT_TRUE(rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms, 0, nullptr, 0,
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, nullptr));
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kEmptyFrame, payload_type, capture_time_ms + 2000, 0,
nullptr, 0, nullptr, nullptr, nullptr));
std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
webrtc::RtpHeaderParser::Create());
ASSERT_TRUE(rtp_parser.get() != nullptr);
@ -1508,9 +1508,9 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
// Marker Bit should be set to 1 for first packet.
EXPECT_TRUE(rtp_header.markerBit);
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms + 4000, 0, nullptr,
0, nullptr, nullptr));
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kEmptyFrame, payload_type, capture_time_ms + 4000, 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.
@ -1529,9 +1529,9 @@ 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),
nullptr, nullptr));
ASSERT_TRUE(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);