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Revert 8028 "Support associated payload type when registering Rt..."

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Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.

> Support associated payload type when registering Rtx payload type.
> 
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
> 
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26259004
> 
> Patch from Changbin Shao <changbin.shao@intel.com>.

TBR=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2015-01-09 20:22:46 +00:00
parent 80452d70cb
commit 8f27fcce79
39 changed files with 206 additions and 400 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
talk/libjingle_tests.gyp
View File

@ -88,16 +88,10 @@
'target_name': 'libjingle_media_unittest',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle_media',
'libjingle_unittest_main',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/testing/gmock/include',
],
},
'sources': [
# TODO(ronghuawu): Reenable this test.
# 'media/base/capturemanager_unittest.cc',

58
talk/media/webrtc/fakewebrtcvideoengine.h
View File

@ -266,6 +266,8 @@ class FakeWebRtcVideoEngine
receive_(false),
can_transmit_(true),
remote_rtx_ssrc_(-1),
rtx_send_payload_type(-1),
rtx_recv_payload_type(-1),
rtcp_status_(webrtc::kRtcpNone),
key_frame_request_method_(webrtc::kViEKeyFrameRequestNone),
tmmbr_(false),
@ -303,8 +305,8 @@ class FakeWebRtcVideoEngine
std::map<int, int> ssrcs_;
std::map<int, int> rtx_ssrcs_;
int remote_rtx_ssrc_;
std::map<int, int> rtx_send_payload_types;
std::map<int, int> rtx_recv_payload_types;
int rtx_send_payload_type;
int rtx_recv_payload_type;
std::string cname_;
webrtc::ViERTCPMode rtcp_status_;
webrtc::ViEKeyFrameRequestMethod key_frame_request_method_;
@ -612,30 +614,14 @@ class FakeWebRtcVideoEngine
WEBRTC_ASSERT_CHANNEL(channel);
channels_[GetOriginalChannelId(channel)]->receive_bandwidth_ =
receive_bandwidth;
};
int GetRtxSendPayloadType(int channel) {
WEBRTC_CHECK_CHANNEL(channel);
return channels_[channel]->rtx_send_payload_type;
}
std::set<int> GetRtxSendPayloadType(int channel) {
std::set<int> rtx_payload_types;
if (channels_.find(channel) == channels_.end())
return rtx_payload_types;
std::map<int, int>::const_iterator it;
for (it = channels_[channel]->rtx_send_payload_types.begin();
it != channels_[channel]->rtx_send_payload_types.end(); ++it) {
rtx_payload_types.insert(it->first);
}
return rtx_payload_types;
}
std::set<int> GetRtxRecvPayloadType(int channel) {
std::set<int> rtx_payload_types;
if (channels_.find(channel) == channels_.end())
return rtx_payload_types;
std::map<int, int>::const_iterator it;
for (it = channels_[channel]->rtx_recv_payload_types.begin();
it != channels_[channel]->rtx_recv_payload_types.end(); ++it) {
rtx_payload_types.insert(it->first);
}
return rtx_payload_types;
int GetRtxRecvPayloadType(int channel) {
WEBRTC_CHECK_CHANNEL(channel);
return channels_[channel]->rtx_recv_payload_type;
}
int GetRemoteRtxSsrc(int channel) {
WEBRTC_CHECK_CHANNEL(channel);
@ -1020,27 +1006,17 @@ class FakeWebRtcVideoEngine
WEBRTC_STUB_CONST(GetRemoteSSRC, (const int, unsigned int&));
WEBRTC_STUB_CONST(GetRemoteCSRCs, (const int, unsigned int*));
WEBRTC_FUNC(SetRtxSendPayloadType,
(const int channel,
const uint8 payload_type,
const uint8 associated_payload_type)) {
WEBRTC_FUNC(SetRtxSendPayloadType, (const int channel,
const uint8 payload_type)) {
WEBRTC_CHECK_CHANNEL(channel);
assert(payload_type >= 0);
assert(associated_payload_type >= 0);
channels_[channel]->rtx_send_payload_types[payload_type] =
associated_payload_type;
channels_[channel]->rtx_send_payload_type = payload_type;
return 0;
}
WEBRTC_FUNC(SetRtxReceivePayloadType,
(const int channel,
const uint8 payload_type,
const uint8 associated_payload_type)) {
WEBRTC_FUNC(SetRtxReceivePayloadType, (const int channel,
const uint8 payload_type)) {
WEBRTC_CHECK_CHANNEL(channel);
assert(payload_type >= 0);
assert(associated_payload_type >= 0);
channels_[channel]->rtx_recv_payload_types[payload_type] =
associated_payload_type;
channels_[channel]->rtx_recv_payload_type = payload_type;
return 0;
}

62
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -180,13 +180,6 @@ const int kVideoRtpBufferSize = 65536;
const char kVp8CodecName[] = "VP8";
const char kVp9CodecName[] = "VP9";
const int kDefaultVp8PlType = 100;
const int kDefaultVp9PlType = 101;
const int kDefaultRedPlType = 116;
const int kDefaultUlpfecType = 117;
const int kDefaultRtxVp8PlType = 96;
const int kDefaultRtxRedPlType = 97;
// TODO(ronghuawu): Change to 640x360.
const int kDefaultVideoMaxWidth = 640;
const int kDefaultVideoMaxHeight = 400;
@ -312,16 +305,14 @@ bool CodecIsInternallySupported(const std::string& codec_name) {
std::vector<VideoCodec> DefaultVideoCodecList() {
std::vector<VideoCodec> codecs;
if (CodecIsInternallySupported(kVp9CodecName)) {
codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp9PlType,
kVp9CodecName));
codecs.push_back(
MakeVideoCodecWithDefaultFeedbackParams(101, kVp9CodecName));
// TODO(andresp): Add rtx codec for vp9 and verify it works.
}
codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType,
kVp8CodecName));
codecs.push_back(MakeRtxCodec(kDefaultRtxVp8PlType, kDefaultVp8PlType));
codecs.push_back(MakeVideoCodec(kDefaultRedPlType, kRedCodecName));
codecs.push_back(MakeRtxCodec(kDefaultRtxRedPlType, kDefaultRedPlType));
codecs.push_back(MakeVideoCodec(kDefaultUlpfecType, kUlpfecCodecName));
codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(100, kVp8CodecName));
codecs.push_back(MakeRtxCodec(96, 100));
codecs.push_back(MakeVideoCodec(116, kRedCodecName));
codecs.push_back(MakeVideoCodec(117, kUlpfecCodecName));
return codecs;
}
@ -1738,6 +1729,7 @@ WebRtcVideoMediaChannel::WebRtcVideoMediaChannel(
first_receive_ssrc_(kSsrcUnset),
receiver_report_ssrc_(kSsrcUnset),
num_unsignalled_recv_channels_(0),
send_rtx_type_(-1),
send_red_type_(-1),
send_fec_type_(-1),
sending_(false),
@ -1827,6 +1819,7 @@ bool WebRtcVideoMediaChannel::SetSendCodecs(
std::vector<webrtc::VideoCodec> send_codecs;
VideoCodec checked_codec;
VideoCodec dummy_current; // Will be ignored by CanSendCodec.
std::map<int, int> primary_rtx_pt_mapping;
bool nack_enabled = nack_enabled_;
bool remb_enabled = remb_enabled_;
for (std::vector<VideoCodec>::const_iterator iter = codecs.begin();
@ -1839,7 +1832,7 @@ bool WebRtcVideoMediaChannel::SetSendCodecs(
int rtx_type = iter->id;
int rtx_primary_type = -1;
if (iter->GetParam(kCodecParamAssociatedPayloadType, &rtx_primary_type)) {
send_rtx_apt_types_[rtx_type] = rtx_primary_type;
primary_rtx_pt_mapping[rtx_primary_type] = rtx_type;
}
} else if (engine()->CanSendCodec(*iter, dummy_current, &checked_codec)) {
webrtc::VideoCodec wcodec;
@ -1904,6 +1897,14 @@ bool WebRtcVideoMediaChannel::SetSendCodecs(
// Select the first matched codec.
webrtc::VideoCodec& codec(send_codecs[0]);
// Set RTX payload type if primary now active. This value will be used in
// SetSendCodec.
std::map<int, int>::const_iterator rtx_it =
primary_rtx_pt_mapping.find(static_cast<int>(codec.plType));
if (rtx_it != primary_rtx_pt_mapping.end()) {
send_rtx_type_ = rtx_it->second;
}
if (BitrateIsSet(codec.minBitrate) && BitrateIsSet(codec.maxBitrate) &&
codec.minBitrate > codec.maxBitrate) {
// TODO(pthatcher): This behavior contradicts other behavior in
@ -3830,11 +3831,8 @@ void WebRtcVideoMediaChannel::LogSendCodecChange(const std::string& reason) {
<< vie_codec.codecSpecific.VP8.keyFrameInterval;
}
std::map<int, int>::const_iterator it;
for (it = send_rtx_apt_types_.begin(); it != send_rtx_apt_types_.end();
++it) {
LOG(LS_INFO) << "RTX payload type: " << it->first
<< ", associated payload type:" << it->second;
if (send_rtx_type_ != -1) {
LOG(LS_INFO) << "RTX payload type: " << send_rtx_type_;
}
LogSimulcastSubstreams(vie_codec);
@ -3852,6 +3850,7 @@ bool WebRtcVideoMediaChannel::SetReceiveCodecs(
it != receive_codecs_.end(); ++it) {
pt_to_codec[it->plType] = &(*it);
}
bool rtx_registered = false;
for (std::vector<webrtc::VideoCodec>::iterator it = receive_codecs_.begin();
it != receive_codecs_.end(); ++it) {
if (it->codecType == webrtc::kVideoCodecRED) {
@ -3862,6 +3861,11 @@ bool WebRtcVideoMediaChannel::SetReceiveCodecs(
// If this is an RTX codec we have to verify that it is associated with
// a valid video codec which we have RTX support for.
if (_stricmp(it->plName, kRtxCodecName) == 0) {
// WebRTC only supports one RTX codec at a time.
if (rtx_registered) {
LOG(LS_ERROR) << "Only one RTX codec at a time is supported.";
return false;
}
std::map<int, int>::iterator apt_it = associated_payload_types_.find(
it->plType);
bool valid_apt = false;
@ -3876,10 +3880,11 @@ bool WebRtcVideoMediaChannel::SetReceiveCodecs(
return false;
}
if (engine()->vie()->rtp()->SetRtxReceivePayloadType(
channel_id, it->plType, apt_it->second) != 0) {
channel_id, it->plType) != 0) {
LOG_RTCERR2(SetRtxReceivePayloadType, channel_id, it->plType);
return false;
}
rtx_registered = true;
continue;
}
if (engine()->vie()->codec()->SetReceiveCodec(channel_id, *it) != 0) {
@ -4015,14 +4020,11 @@ bool WebRtcVideoMediaChannel::SetSendParams(
// NOTE: SetRtxSendPayloadType must be called after all SSRCs are
// configured. Otherwise ssrc's configured after this point will use
// the primary PT for RTX.
std::map<int, int>::const_iterator it;
for (it = send_rtx_apt_types_.begin(); it != send_rtx_apt_types_.end();
++it) {
if (engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id, it->first,
it->second) != 0) {
LOG_RTCERR3(SetRtxSendPayloadType, channel_id, it->first, it->second);
return false;
}
if (send_rtx_type_ != -1 &&
engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id,
send_rtx_type_) != 0) {
LOG_RTCERR2(SetRtxSendPayloadType, channel_id, send_rtx_type_);
return false;
}
send_channel->set_send_params(send_params);

3
talk/media/webrtc/webrtcvideoengine.h
View File

@ -511,8 +511,7 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
// Global send side state.
SendChannelMap send_channels_;
rtc::scoped_ptr<webrtc::VideoCodec> send_codec_;
// A map from RTX payload types to their associated payload types.
std::map<int, int> send_rtx_apt_types_;
int send_rtx_type_;
int send_red_type_;
int send_fec_type_;
bool sending_;

23
talk/media/webrtc/webrtcvideoengine2.cc
View File

@ -121,15 +121,6 @@ static void MergeFecConfig(const webrtc::FecConfig& other,
}
output->red_payload_type = other.red_payload_type;
}
if (other.rtx_payload_type != -1) {
if (output->rtx_payload_type != -1 &&
output->rtx_payload_type != other.rtx_payload_type) {
LOG(LS_WARNING) << "Conflict merging rtx_payload_type configs: "
<< output->rtx_payload_type << " and "
<< other.rtx_payload_type;
}
output->rtx_payload_type = other.rtx_payload_type;
}
}
} // namespace
@ -2079,7 +2070,6 @@ bool WebRtcVideoChannel2::VideoCodecSettings::operator==(
return codec == other.codec &&
fec.ulpfec_payload_type == other.fec.ulpfec_payload_type &&
fec.red_payload_type == other.fec.red_payload_type &&
fec.rtx_payload_type == other.fec.rtx_payload_type &&
rtx_payload_type == other.rtx_payload_type;
}
@ -2152,24 +2142,17 @@ WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) {
LOG(LS_ERROR) << "RTX mapped to payload not in codec list.";
return std::vector<VideoCodecSettings>();
}
if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO &&
payload_codec_type[it->first] != VideoCodec::CODEC_RED) {
LOG(LS_ERROR) << "RTX not mapped to regular video codec or RED codec.";
if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO) {
LOG(LS_ERROR) << "RTX not mapped to regular video codec.";
return std::vector<VideoCodecSettings>();
}
if (it->first == fec_settings.red_payload_type) {
fec_settings.rtx_payload_type = it->second;
}
}
// TODO(pbos): Write tests that figure out that I have not verified that RTX
// codecs aren't mapped to bogus payloads.
for (size_t i = 0; i < video_codecs.size(); ++i) {
video_codecs[i].fec = fec_settings;
if (rtx_mapping[video_codecs[i].codec.id] != 0 &&
rtx_mapping[video_codecs[i].codec.id] !=
fec_settings.red_payload_type) {
if (rtx_mapping[video_codecs[i].codec.id] != 0) {
video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
}
}

1
talk/media/webrtc/webrtcvideoengine2.h
View File

@ -285,6 +285,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
struct VideoCodecSettings {
VideoCodecSettings();
bool operator ==(const VideoCodecSettings& other) const;
VideoCodec codec;

27
talk/media/webrtc/webrtcvideoengine2_unittest.cc
View File

@ -34,7 +34,6 @@
#include "talk/media/webrtc/fakewebrtcvideoengine.h"
#include "talk/media/webrtc/simulcast.h"
#include "talk/media/webrtc/webrtcvideochannelfactory.h"
#include "talk/media/webrtc/webrtcvideoengine.h"
#include "talk/media/webrtc/webrtcvideoengine2.h"
#include "talk/media/webrtc/webrtcvideoengine2_unittest.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
@ -74,19 +73,6 @@ void VerifyCodecHasDefaultFeedbackParams(const cricket::VideoCodec& codec) {
cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir)));
}
void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config,
const std::map<int, int>& rtx_types) {
std::map<int, int>::const_iterator it;
it = rtx_types.find(config.encoder_settings.payload_type);
EXPECT_TRUE(it != rtx_types.end() &&
it->second == config.rtp.rtx.payload_type);
if (config.rtp.fec.rtx_payload_type != -1) {
it = rtx_types.find(config.rtp.fec.red_payload_type);
EXPECT_TRUE(it != rtx_types.end() &&
it->second == config.rtp.fec.rtx_payload_type);
}
}
} // namespace
namespace cricket {
@ -363,11 +349,7 @@ class WebRtcVideoEngine2Test : public ::testing::Test {
} else if (engine_codecs[i].name == "ulpfec") {
default_ulpfec_codec_ = engine_codecs[i];
} else if (engine_codecs[i].name == "rtx") {
int associated_payload_type;
if (engine_codecs[i].GetParam(kCodecParamAssociatedPayloadType,
&associated_payload_type)) {
default_apt_rtx_types_[associated_payload_type] = engine_codecs[i].id;
}
default_rtx_codec_ = engine_codecs[i];
} else if (!codec_set) {
default_codec_ = engine_codecs[i];
codec_set = true;
@ -406,7 +388,7 @@ class WebRtcVideoEngine2Test : public ::testing::Test {
VideoCodec default_codec_;
VideoCodec default_red_codec_;
VideoCodec default_ulpfec_codec_;
std::map<int, int> default_apt_rtx_types_;
VideoCodec default_rtx_codec_;
};
TEST_F(WebRtcVideoEngine2Test, ConfiguresAvSyncForFirstReceiveChannel) {
@ -449,7 +431,7 @@ TEST_F(WebRtcVideoEngine2Test, ConfiguresAvSyncForFirstReceiveChannel) {
TEST_F(WebRtcVideoEngine2Test, FindCodec) {
const std::vector<cricket::VideoCodec>& c = engine_.codecs();
EXPECT_EQ(cricket::DefaultVideoCodecList().size(), c.size());
EXPECT_EQ(4U, c.size());
cricket::VideoCodec vp8(104, "VP8", 320, 200, 30, 0);
EXPECT_TRUE(engine_.FindCodec(vp8));
@ -1601,7 +1583,8 @@ TEST_F(WebRtcVideoChannel2Test, SetDefaultSendCodecs) {
EXPECT_EQ(1u, config.rtp.rtx.ssrcs.size());
EXPECT_EQ(kRtxSsrcs1[0], config.rtp.rtx.ssrcs[0]);
VerifySendStreamHasRtxTypes(config, default_apt_rtx_types_);
EXPECT_EQ(static_cast<int>(default_rtx_codec_.id),
config.rtp.rtx.payload_type);
// TODO(juberti): Check RTCP, PLI, TMMBR.
}

37
talk/media/webrtc/webrtcvideoengine_unittest.cc
View File

@ -38,8 +38,6 @@
#include "talk/media/webrtc/webrtcvideoframe.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
#include "talk/session/media/mediasession.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/fakecpumonitor.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
@ -756,8 +754,7 @@ TEST_F(WebRtcVideoEngineTestFake, SetRecvCodecsWithRtx) {
EXPECT_FALSE(vie_.ReceiveCodecRegistered(channel_num, wcodec));
// The RTX payload type should have been set.
EXPECT_THAT(vie_.GetRtxRecvPayloadType(channel_num),
::testing::ElementsAre(rtx_codec.id));
EXPECT_EQ(rtx_codec.id, vie_.GetRtxRecvPayloadType(channel_num));
}
// Test that RTX packets are routed to the default video channel if
@ -2082,8 +2079,7 @@ TEST_F(WebRtcVideoEngineTestFake, SetSendCodecsWithExternalH264) {
codecs.push_back(rtx_codec);
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_THAT(vie_.GetRtxSendPayloadType(channel_num),
::testing::ElementsAre(96));
EXPECT_EQ(96, vie_.GetRtxSendPayloadType(channel_num));
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
@ -2121,8 +2117,7 @@ TEST_F(WebRtcVideoEngineTestFake, SetSendCodecsWithVP8AndExternalH264) {
// The first matched codec should be set, i.e., H.264.
EXPECT_THAT(vie_.GetRtxSendPayloadType(channel_num),
::testing::ElementsAre(96, 97));
EXPECT_EQ(96, vie_.GetRtxSendPayloadType(channel_num));
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
@ -2159,8 +2154,7 @@ TEST_F(WebRtcVideoEngineTestFake, SetRecvCodecsWithExternalH264) {
codecs.push_back(rtx_codec);
EXPECT_TRUE(channel_->SetRecvCodecs(codecs));
EXPECT_THAT(vie_.GetRtxRecvPayloadType(channel_num),
::testing::ElementsAre(96));
EXPECT_EQ(96, vie_.GetRtxRecvPayloadType(channel_num));
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
@ -2195,21 +2189,17 @@ TEST_F(WebRtcVideoEngineTestFake, SetRecvCodecsWithVP8AndExternalH264) {
cricket::VideoCodec rtx_codec2(96, "rtx", 0, 0, 0, 0);
rtx_codec2.SetParam("apt", kVP8Codec.id);
codecs.push_back(kVP8Codec);
codecs.push_back(rtx_codec2);
// Now WebRTC supports setting multiple RTX codecs at a time.
EXPECT_TRUE(channel_->SetRecvCodecs(codecs));
EXPECT_THAT(vie_.GetRtxRecvPayloadType(channel_num),
::testing::ElementsAre(96, 97));
codecs.push_back(rtx_codec);
// Should fail since WebRTC only supports one RTX codec at a time.
EXPECT_FALSE(channel_->SetRecvCodecs(codecs));
codecs.pop_back();
// One RTX codec should be fine.
EXPECT_TRUE(channel_->SetRecvCodecs(codecs));
// TODO(changbin): The Rtx key can still be found from the Rtx-Apt map
// if the new codec list doesn't assign it with a new value.
// Should pass a map to SetRtxRecvPayloadType in future.
EXPECT_THAT(vie_.GetRtxRecvPayloadType(channel_num),
::testing::ElementsAre(96, 97));
// The RTX payload type should have been set.
EXPECT_EQ(rtx_codec.id, vie_.GetRtxRecvPayloadType(channel_num));
}
#endif
@ -2259,7 +2249,7 @@ TEST_F(WebRtcVideoEngineTestFake, CaptureFrameTimestampToNtpTimestamp) {
TEST_F(WebRtcVideoEngineTest, FindCodec) {
// We should not need to init engine in order to get codecs.
const std::vector<cricket::VideoCodec>& c = engine_.codecs();
EXPECT_EQ(cricket::DefaultVideoCodecList().size(), c.size());
EXPECT_EQ(4U, c.size());
cricket::VideoCodec vp8(104, "VP8", 320, 200, 30, 0);
EXPECT_TRUE(engine_.FindCodec(vp8));
@ -2306,7 +2296,7 @@ TEST_F(WebRtcVideoEngineTest, RtxCodecHasAptSet) {
std::vector<cricket::VideoCodec>::const_iterator it;
bool apt_checked = false;
for (it = engine_.codecs().begin(); it != engine_.codecs().end(); ++it) {
if (_stricmp(cricket::kRtxCodecName, it->name.c_str()) || it->id != 96) {
if (_stricmp(cricket::kRtxCodecName, it->name.c_str()) && it->id != 96) {
continue;
}
int apt;
@ -3212,8 +3202,7 @@ TEST_F(WebRtcVideoEngineSimulcastTestFake, TestStreamWithRtx) {
EXPECT_TRUE(channel_->SetSendCodecs(codec_list));
// RTX payload type should now be set.
EXPECT_THAT(vie_.GetRtxSendPayloadType(channel_num),
::testing::ElementsAre(96));
EXPECT_EQ(96, vie_.GetRtxSendPayloadType(channel_num));
// Verify all SSRCs are set after SetSendCodecs.
EXPECT_EQ(3, vie_.GetNumSsrcs(channel_num));

1
webrtc/config.cc
View File

@ -17,7 +17,6 @@ std::string FecConfig::ToString() const {
std::stringstream ss;
ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
ss << ", red_payload_type: " << red_payload_type;
ss << ", rtx_payload_type: " << rtx_payload_type;
ss << '}';
return ss.str();
}

6
webrtc/config.h
View File

@ -54,17 +54,13 @@ struct NackConfig {
// Settings for forward error correction, see RFC 5109 for details. Set the
// payload types to '-1' to disable.
struct FecConfig {
FecConfig()
: ulpfec_payload_type(-1), red_payload_type(-1), rtx_payload_type(-1) {}
FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {}
std::string ToString() const;
// Payload type used for ULPFEC packets.
int ulpfec_payload_type;
// Payload type used for RED packets.
int red_payload_type;
// Rtx payload type for RED payload.
int rtx_payload_type;
};
// RTP header extension to use for the video stream, see RFC 5285.

5
webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
View File

@ -79,7 +79,7 @@ class RTPPayloadRegistry {
bool GetRtxSsrc(uint32_t* ssrc) const;
void SetRtxPayloadType(int payload_type, int associated_payload_type);
void SetRtxPayloadType(int payload_type);
bool IsRtx(const RTPHeader& header) const;
@ -158,8 +158,7 @@ class RTPPayloadRegistry {
int8_t last_received_payload_type_;
int8_t last_received_media_payload_type_;
bool rtx_;
// Mapping rtx_apt_types_[rtx] = apt.
std::map<int, int> rtx_apt_types_;
int8_t payload_type_rtx_;
uint32_t ssrc_rtx_;
};

3
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
View File

@ -228,8 +228,7 @@ class RtpRtcp : public Module {
// Sets the payload type to use when sending RTX packets. Note that this
// doesn't enable RTX, only the payload type is set.
virtual void SetRtxSendPayloadType(int payload_type,
int associated_payload_type) = 0;
virtual void SetRtxSendPayloadType(int payload_type) = 0;
/*
* Get status of sending RTX (RFC 4588) on a specific SSRC.

3
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -101,7 +101,8 @@ class MockRtpRtcp : public RtpRtcp {
void(int* modes, uint32_t* ssrc, int* payload_type));
MOCK_METHOD1(SetRtxSsrc,
void(uint32_t));
MOCK_METHOD2(SetRtxSendPayloadType, void(int, int));
MOCK_METHOD1(SetRtxSendPayloadType,
void(int));
MOCK_METHOD1(SetSendingStatus,
int32_t(const bool sending));
MOCK_CONST_METHOD0(Sending,

24
webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
View File

@ -32,8 +32,6 @@ const uint16_t kTestSequenceNumber = 2345;
const uint32_t kTestNumberOfPackets = 1350;
const int kTestNumberOfRtxPackets = 149;
const int kNumFrames = 30;
const int kPayloadType = 123;
const int kRtxPayloadType = 98;
class VerifyingRtxReceiver : public NullRtpData
{
@ -208,17 +206,15 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
VideoCodec video_codec;
memset(&video_codec, 0, sizeof(video_codec));
video_codec.plType = kPayloadType;
video_codec.plType = 123;
memcpy(video_codec.plName, "I420", 5);
EXPECT_EQ(0, rtp_rtcp_module_->RegisterSendPayload(video_codec));
rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(video_codec.plName,
video_codec.plType,
90000,
0,
video_codec.maxBitrate));
rtp_payload_registry_.SetRtxPayloadType(kRtxPayloadType, kPayloadType);
for (size_t n = 0; n < payload_data_length; n++) {
payload_data[n] = n % 10;
@ -269,9 +265,12 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
uint32_t timestamp = 3000;
uint16_t nack_list[kVideoNackListSize];
for (int frame = 0; frame < kNumFrames; ++frame) {
EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(
webrtc::kVideoFrameDelta, kPayloadType, timestamp,
timestamp / 90, payload_data, payload_data_length));
EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
123,
timestamp,
timestamp / 90,
payload_data,
payload_data_length));
int length = BuildNackList(nack_list);
if (length > 0)
rtp_rtcp_module_->SendNACK(nack_list, length);
@ -314,9 +313,12 @@ TEST_F(RtpRtcpRtxNackTest, LongNackList) {
// Send 30 frames which at the default size is roughly what we need to get
// enough packets.
for (int frame = 0; frame < kNumFrames; ++frame) {
EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(
webrtc::kVideoFrameDelta, kPayloadType, timestamp,
timestamp / 90, payload_data, payload_data_length));
EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
123,
timestamp,
timestamp / 90,
payload_data,
payload_data_length));
// Prepare next frame.
timestamp += 3000;
fake_clock.AdvanceTimeMilliseconds(33);

33
webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
View File

@ -24,6 +24,7 @@ RTPPayloadRegistry::RTPPayloadRegistry(
last_received_payload_type_(-1),
last_received_media_payload_type_(-1),
rtx_(false),
payload_type_rtx_(-1),
ssrc_rtx_(0) {}
RTPPayloadRegistry::~RTPPayloadRegistry() {
@ -263,25 +264,19 @@ bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t** restored_packet,
RtpUtility::AssignUWord32ToBuffer(*restored_packet + 8, original_ssrc);
CriticalSectionScoped cs(crit_sect_.get());
if (!rtx_)
return true;
if (rtx_apt_types_.empty()) {
LOG(LS_WARNING) << "No RTX is set, dropping packet.";
return false;
}
std::map<int, int>::const_iterator it =
rtx_apt_types_.find(header.payloadType);
if (it != rtx_apt_types_.end()) {
(*restored_packet)[1] = static_cast<uint8_t>(it->second);
if (header.markerBit) {
(*restored_packet)[1] |= kRtpMarkerBitMask; // Marker bit is set.
if (payload_type_rtx_ != -1) {
if (header.payloadType == payload_type_rtx_ &&
incoming_payload_type_ != -1) {
(*restored_packet)[1] = static_cast<uint8_t>(incoming_payload_type_);
if (header.markerBit) {
(*restored_packet)[1] |= kRtpMarkerBitMask; // Marker bit is set.
}
} else {
LOG(LS_WARNING) << "Incorrect RTX configuration, dropping packet.";
return false;
}
} else {
LOG(LS_WARNING) << "Incorrect RTX configuration, dropping packet.";
return false;
}
return true;
}
@ -297,12 +292,10 @@ bool RTPPayloadRegistry::GetRtxSsrc(uint32_t* ssrc) const {
return rtx_;
}
void RTPPayloadRegistry::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
void RTPPayloadRegistry::SetRtxPayloadType(int payload_type) {
CriticalSectionScoped cs(crit_sect_.get());
assert(payload_type >= 0);
assert(associated_payload_type >= 0);
rtx_apt_types_[payload_type] = associated_payload_type;
payload_type_rtx_ = payload_type;
rtx_ = true;
}

5
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -257,9 +257,8 @@ void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
rtp_sender_.SetRtxSsrc(ssrc);
}
void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
int associated_payload_type) {
rtp_sender_.SetRtxPayloadType(payload_type, associated_payload_type);
void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type) {
rtp_sender_.SetRtxPayloadType(payload_type);
}
int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(

3
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -94,8 +94,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
virtual void SetRtxSsrc(uint32_t ssrc) OVERRIDE;
virtual void SetRtxSendPayloadType(int payload_type,
int associated_payload_type) OVERRIDE;
virtual void SetRtxSendPayloadType(int payload_type) OVERRIDE;
// Sends kRtcpByeCode when going from true to false.
virtual int32_t SetSendingStatus(bool sending) OVERRIDE;

4
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
View File

@ -691,7 +691,7 @@ TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtx) {
// as otherwise the timestmap used for BWE will be broken.
senders_[i]->RegisterSendRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
1);
senders_[i]->SetRtxSendPayloadType(96, 100);
senders_[i]->SetRtxSendPayloadType(96);
senders_[i]->SetRtxSsrc(kSenderRtxSsrc + i);
senders_[i]->SetRTXSendStatus(kRtxRetransmitted);
}
@ -716,7 +716,7 @@ TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtx) {
TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtxRedundantPayloads) {
for (int i = 1; i < codec_.numberOfSimulcastStreams + 1; ++i) {
senders_[i]->SetRtxSendPayloadType(96, 100);
senders_[i]->SetRtxSendPayloadType(96);
senders_[i]->SetRtxSsrc(kSenderRtxSsrc + i);
senders_[i]->SetRTXSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
senders_[i]->SetStorePacketsStatus(true, 100);

29
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -147,6 +147,7 @@ RTPSender::RTPSender(int32_t id,
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
payload_type_rtx_(-1),
target_bitrate_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
target_bitrate_(0) {
memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
@ -402,17 +403,12 @@ void RTPSender::RTXStatus(int* mode, uint32_t* ssrc,
CriticalSectionScoped cs(send_critsect_);
*mode = rtx_;
*ssrc = ssrc_rtx_;
std::map<int, int>::const_iterator it = apt_rtx_types_.find(payload_type_);
*payload_type = (it == apt_rtx_types_.end()) ? -1 : it->second;
*payload_type = payload_type_rtx_;
}
void RTPSender::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
void RTPSender::SetRtxPayloadType(int payload_type) {
CriticalSectionScoped cs(send_critsect_);
assert(payload_type >= 0);
assert(associated_payload_type >= 0);
apt_rtx_types_[associated_payload_type] = payload_type;
payload_type_rtx_ = payload_type;
}
int32_t RTPSender::CheckPayloadType(int8_t payload_type,
@ -614,14 +610,8 @@ size_t RTPSender::SendPadData(uint32_t timestamp,
ssrc = ssrc_rtx_;
sequence_number = sequence_number_rtx_;
++sequence_number_rtx_;
if (apt_rtx_types_.empty())
return 0;
std::map<int, int>::const_iterator it =
apt_rtx_types_.find(payload_type_);
payload_type = it != apt_rtx_types_.end()
? it->second
: apt_rtx_types_.begin()->second;
payload_type = ((rtx_ & kRtxRedundantPayloads) > 0) ? payload_type_rtx_
: payload_type_;
over_rtx = true;
}
}
@ -1670,10 +1660,9 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
// Add original RTP header.
memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
std::map<int, int>::const_iterator it =
apt_rtx_types_.find(rtp_header.payloadType);
if (it != apt_rtx_types_.end()) {
data_buffer_rtx[1] = static_cast<uint8_t>(it->second);
// Replace payload type, if a specific type is set for RTX.
if (payload_type_rtx_ != -1) {
data_buffer_rtx[1] = static_cast<uint8_t>(payload_type_rtx_);
if (rtp_header.markerBit)
data_buffer_rtx[1] |= kRtpMarkerBitMask;
}

5
webrtc/modules/rtp_rtcp/source/rtp_sender.h
View File

@ -191,7 +191,7 @@ class RTPSender : public RTPSenderInterface {
uint32_t RtxSsrc() const;
void SetRtxSsrc(uint32_t ssrc);
void SetRtxPayloadType(int payloadType, int associated_payload_type);
void SetRtxPayloadType(int payloadType);
// Functions wrapping RTPSenderInterface.
virtual int32_t BuildRTPheader(
@ -391,8 +391,7 @@ class RTPSender : public RTPSenderInterface {
std::vector<uint32_t> csrcs_ GUARDED_BY(send_critsect_);
int rtx_ GUARDED_BY(send_critsect_);
uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
// Mapping apt_rtx_types_[apt] = rtx.
std::map<int, int> apt_rtx_types_ GUARDED_BY(send_critsect_);
int payload_type_rtx_ GUARDED_BY(send_critsect_);
// Note: Don't access this variable directly, always go through
// SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember

4
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -30,7 +30,6 @@ namespace {
const int kTransmissionTimeOffsetExtensionId = 1;
const int kAbsoluteSendTimeExtensionId = 14;
const int kPayload = 100;
const int kRtxPayload = 98;
const uint32_t kTimestamp = 10;
const uint16_t kSeqNum = 33;
const int kTimeOffset = 22222;
@ -655,7 +654,6 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL,
&mock_paced_sender_, NULL, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
// Make all packets go through the pacer.
EXPECT_CALL(mock_paced_sender_,
SendPacket(PacedSender::kNormalPriority, _, _, _, _, _)).
@ -1125,7 +1123,7 @@ TEST_F(RtpSenderTest, BytesReportedCorrectly) {
const uint8_t kPayloadType = 127;
rtp_sender_->SetSSRC(1234);
rtp_sender_->SetRtxSsrc(4321);
rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
rtp_sender_->SetRtxPayloadType(kPayloadType - 1);
rtp_sender_->SetRTXStatus(kRtxRetransmitted | kRtxRedundantPayloads);
ASSERT_EQ(

14
webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
View File

@ -106,31 +106,33 @@ TEST_F(RtpRtcpAPITest, RtxSender) {
unsigned int ssrc = 0;
int rtx_mode = kRtxOff;
const int kRtxPayloadType = 119;
const int kPayloadType = 100;
int payload_type = -1;
module->SetRTXSendStatus(kRtxRetransmitted);
module->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
module->SetRtxSendPayloadType(kRtxPayloadType);
module->SetRtxSsrc(1);
int payload_type;
module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
EXPECT_EQ(kRtxRetransmitted, rtx_mode);
EXPECT_EQ(1u, ssrc);
EXPECT_EQ(kRtxPayloadType, payload_type);
rtx_mode = kRtxOff;
module->SetRTXSendStatus(kRtxOff);
module->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
payload_type = -1;
module->SetRtxSendPayloadType(kRtxPayloadType);
module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
EXPECT_EQ(kRtxOff, rtx_mode);
EXPECT_EQ(kRtxPayloadType, payload_type);
module->SetRTXSendStatus(kRtxRetransmitted);
module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
EXPECT_EQ(kRtxRetransmitted, rtx_mode);
EXPECT_EQ(kRtxPayloadType, payload_type);
}
TEST_F(RtpRtcpAPITest, RtxReceiver) {
const uint32_t kRtxSsrc = 1;
const int kRtxPayloadType = 119;
const int kPayloadType = 100;
EXPECT_FALSE(rtp_payload_registry_->RtxEnabled());
rtp_payload_registry_->SetRtxSsrc(kRtxSsrc);
rtp_payload_registry_->SetRtxPayloadType(kRtxPayloadType, kPayloadType);
rtp_payload_registry_->SetRtxPayloadType(kRtxPayloadType);
EXPECT_TRUE(rtp_payload_registry_->RtxEnabled());
RTPHeader rtx_header;
rtx_header.ssrc = kRtxSsrc;

1
webrtc/test/call_test.cc
View File

@ -151,7 +151,6 @@ const uint8_t CallTest::kSendPayloadType = 100;
const uint8_t CallTest::kFakeSendPayloadType = 125;
const uint8_t CallTest::kSendRtxPayloadType = 98;
const uint8_t CallTest::kRedPayloadType = 118;
const uint8_t CallTest::kRtxRedPayloadType = 99;
const uint8_t CallTest::kUlpfecPayloadType = 119;
const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
0xBADCAFF};

1
webrtc/test/call_test.h
View File

@ -37,7 +37,6 @@ class CallTest : public ::testing::Test {
static const uint8_t kSendRtxPayloadType;
static const uint8_t kFakeSendPayloadType;
static const uint8_t kRedPayloadType;
static const uint8_t kRtxRedPayloadType;
static const uint8_t kUlpfecPayloadType;
static const uint32_t kSendRtxSsrcs[kNumSsrcs];
static const uint32_t kSendSsrcs[kNumSsrcs];

51
webrtc/video/end_to_end_tests.cc
View File

@ -69,7 +69,7 @@ class EndToEndTest : public test::CallTest {
}
};
void DecodesRetransmittedFrame(bool use_rtx, bool use_red);
void DecodesRetransmittedFrame(bool retransmit_over_rtx);
void ReceivesPliAndRecovers(int rtp_history_ms);
void RespectsRtcpMode(newapi::RtcpMode rtcp_mode);
void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
@ -538,16 +538,16 @@ TEST_F(EndToEndTest, CanReceiveFec) {
// This test drops second RTP packet with a marker bit set, makes sure it's
// retransmitted and renders. Retransmission SSRCs are also checked.
void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {
void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
static const int kDroppedFrameNumber = 2;
class RetransmissionObserver : public test::EndToEndTest,
public I420FrameCallback {
public:
explicit RetransmissionObserver(bool use_rtx, bool use_red)
explicit RetransmissionObserver(bool expect_rtx)
: EndToEndTest(kDefaultTimeoutMs),
payload_type_(GetPayloadType(false, use_red)),
retransmission_ssrc_(use_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]),
retransmission_payload_type_(GetPayloadType(use_rtx, use_red)),
retransmission_ssrc_(expect_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]),
retransmission_payload_type_(expect_rtx ? kSendRtxPayloadType
: kFakeSendPayloadType),
marker_bits_observed_(0),
retransmitted_timestamp_(0),
frame_retransmitted_(false) {}
@ -565,7 +565,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {
}
EXPECT_EQ(kSendSsrcs[0], header.ssrc);
EXPECT_EQ(payload_type_, header.payloadType);
EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
// Found the second frame's final packet, drop this and expect a
// retransmission.
@ -592,25 +592,13 @@ void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].pre_render_callback = this;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (payload_type_ == kRedPayloadType) {
send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
send_config->rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
}
if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
(*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].ssrc =
(*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].ssrc =
kSendRtxSsrcs[0];
(*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].payload_type =
(*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].payload_type =
kSendRtxPayloadType;
if (payload_type_ == kRedPayloadType) {
send_config->rtp.fec.rtx_payload_type = kRtxRedPayloadType;
(*receive_configs)[0].rtp.fec.rtx_payload_type = kRtxRedPayloadType;
}
}
}
@ -619,36 +607,22 @@ void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {
<< "Timed out while waiting for retransmission to render.";
}
int GetPayloadType(bool use_rtx, bool use_red) {
return use_red ? (use_rtx ? kRtxRedPayloadType : kRedPayloadType)
: (use_rtx ? kSendRtxPayloadType : kFakeSendPayloadType);
}
const int payload_type_;
const uint32_t retransmission_ssrc_;
const int retransmission_payload_type_;
int marker_bits_observed_;
uint32_t retransmitted_timestamp_;
bool frame_retransmitted_;
} test(use_rtx, use_red);
} test(retransmit_over_rtx);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrame) {
DecodesRetransmittedFrame(false, false);
DecodesRetransmittedFrame(false);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) {
DecodesRetransmittedFrame(true, false);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrameByRed) {
DecodesRetransmittedFrame(false, true);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
DecodesRetransmittedFrame(true, true);
DecodesRetransmittedFrame(true);
}
TEST_F(EndToEndTest, UsesFrameCallbacks) {
@ -2294,4 +2268,5 @@ TEST_F(EndToEndTest, CanCreateAndDestroyManyVideoStreams) {
call->DestroyVideoReceiveStream(receive_stream);
}
}
} // namespace webrtc

11
webrtc/video/loopback.cc
View File

@ -103,8 +103,7 @@ static const uint32_t kSendSsrc = 0x654321;
static const uint32_t kSendRtxSsrc = 0x654322;
static const uint32_t kReceiverLocalSsrc = 0x123456;
static const uint8_t kRtxVideoPayloadType = 96;
static const uint8_t kVideoPayloadType = 124;
static const uint8_t kRtxPayloadType = 96;
void Loopback() {
scoped_ptr<test::TraceToStderr> trace_to_stderr_;
@ -138,7 +137,7 @@ void Loopback() {
VideoSendStream::Config send_config;
send_config.rtp.ssrcs.push_back(kSendSsrc);
send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc);
send_config.rtp.rtx.payload_type = kRtxVideoPayloadType;
send_config.rtp.rtx.payload_type = kRtxPayloadType;
send_config.rtp.nack.rtp_history_ms = 1000;
send_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
@ -156,7 +155,7 @@ void Loopback() {
}
send_config.encoder_settings.encoder = encoder.get();
send_config.encoder_settings.payload_name = flags::Codec();
send_config.encoder_settings.payload_type = kVideoPayloadType;
send_config.encoder_settings.payload_type = 124;
VideoEncoderConfig encoder_config;
encoder_config.streams = test::CreateVideoStreams(1);
VideoStream* stream = &encoder_config.streams[0];
@ -184,8 +183,8 @@ void Loopback() {
receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
receive_config.rtp.nack.rtp_history_ms = 1000;
receive_config.rtp.rtx[kVideoPayloadType].ssrc = kSendRtxSsrc;
receive_config.rtp.rtx[kVideoPayloadType].payload_type = kRtxVideoPayloadType;
receive_config.rtp.rtx[kRtxPayloadType].ssrc = kSendRtxSsrc;
receive_config.rtp.rtx[kRtxPayloadType].payload_type = kRtxPayloadType;
receive_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
receive_config.renderer = loopback_video.get();

85
webrtc/video/rampup_tests.cc
View File

@ -76,10 +76,6 @@ StreamObserver::StreamObserver(const SsrcMap& rtx_media_ssrcs,
remote_bitrate_estimator_.reset(
rbe_factory->Create(this, clock, control_type,
kRemoteBitrateEstimatorMinBitrateBps));
payload_registry_->SetRtxPayloadType(RampUpTest::kSendRtxPayloadType,
RampUpTest::kFakeSendPayloadType);
payload_registry_->SetRtxPayloadType(RampUpTest::kRtxRedPayloadType,
RampUpTest::kRedPayloadType);
}
void StreamObserver::set_expected_bitrate_bps(
@ -122,6 +118,7 @@ bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
receive_stats_->IncomingPacket(header, length, false);
payload_registry_->SetIncomingPayloadType(header);
remote_bitrate_estimator_->IncomingPacket(
clock_->TimeInMilliseconds(), length - 12, header);
if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
@ -139,9 +136,11 @@ bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
uint8_t restored_packet[kMaxPacketSize];
uint8_t* restored_packet_ptr = restored_packet;
size_t restored_length = length;
EXPECT_TRUE(payload_registry_->RestoreOriginalPacket(
&restored_packet_ptr, packet, &restored_length,
rtx_media_ssrcs_[header.ssrc], header));
payload_registry_->RestoreOriginalPacket(&restored_packet_ptr,
packet,
&restored_length,
rtx_media_ssrcs_[header.ssrc],
header);
length = restored_length;
EXPECT_TRUE(rtp_parser_->Parse(
restored_packet, static_cast<int>(length), &header));
@ -367,11 +366,10 @@ EventTypeWrapper LowRateStreamObserver::Wait() {
return test_done_->Wait(test::CallTest::kLongTimeoutMs);
}
void RampUpTest::RunRampUpTest(size_t num_streams,
void RampUpTest::RunRampUpTest(bool rtx,
size_t num_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,
bool red) {
const std::string& extension_type) {
std::vector<uint32_t> ssrcs(GenerateSsrcs(num_streams, 100));
std::vector<uint32_t> rtx_ssrcs(GenerateSsrcs(num_streams, 200));
StreamObserver::SsrcMap rtx_ssrc_map;
@ -424,11 +422,6 @@ void RampUpTest::RunRampUpTest(size_t num_streams,
send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
send_config_.rtp.rtx.ssrcs = rtx_ssrcs;
}
if (red) {
send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
send_config_.rtp.fec.red_payload_type = kRedPayloadType;
send_config_.rtp.fec.rtx_payload_type = kRtxRedPayloadType;
}
if (num_streams == 1) {
// For single stream rampup until 1mbps
@ -455,9 +448,7 @@ void RampUpTest::RunRampUpTest(size_t num_streams,
DestroyStreams();
}
void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams,
bool rtx,
bool red) {
void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams, bool rtx) {
test::DirectTransport receiver_transport;
LowRateStreamObserver stream_observer(
&receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx);
@ -476,11 +467,6 @@ void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams,
send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
send_config_.rtp.rtx.ssrcs = GenerateSsrcs(number_of_streams, 200);
}
if (red) {
send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
send_config_.rtp.fec.red_payload_type = kRedPayloadType;
send_config_.rtp.fec.rtx_payload_type = kRtxRedPayloadType;
}
CreateStreams();
stream_observer.SetSendStream(send_stream_);
@ -496,68 +482,43 @@ void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams,
}
TEST_F(RampUpTest, SingleStream) {
RunRampUpTest(1, 0, RtpExtension::kTOffset, false, false);
RunRampUpTest(false, 1, 0, RtpExtension::kTOffset);
}
TEST_F(RampUpTest, Simulcast) {
RunRampUpTest(3, 0, RtpExtension::kTOffset, false, false);
RunRampUpTest(false, 3, 0, RtpExtension::kTOffset);
}
TEST_F(RampUpTest, SimulcastWithRtx) {
RunRampUpTest(3, 0, RtpExtension::kTOffset, true, false);
}
TEST_F(RampUpTest, SimulcastByRedWithRtx) {
RunRampUpTest(3, 0, RtpExtension::kTOffset, true, true);
RunRampUpTest(true, 3, 0, RtpExtension::kTOffset);
}
TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
RunRampUpTest(1, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset, false,
false);
RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset);
}
TEST_F(RampUpTest, UpDownUpOneStream) {
RunRampUpDownUpTest(1, false, false);
}
TEST_F(RampUpTest, UpDownUpOneStream) { RunRampUpDownUpTest(1, false); }
TEST_F(RampUpTest, UpDownUpThreeStreams) {
RunRampUpDownUpTest(3, false, false);
}
TEST_F(RampUpTest, UpDownUpThreeStreams) { RunRampUpDownUpTest(3, false); }
TEST_F(RampUpTest, UpDownUpOneStreamRtx) {
RunRampUpDownUpTest(1, true, false);
}
TEST_F(RampUpTest, UpDownUpOneStreamRtx) { RunRampUpDownUpTest(1, true); }
TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) {
RunRampUpDownUpTest(3, true, false);
}
TEST_F(RampUpTest, UpDownUpOneStreamByRedRtx) {
RunRampUpDownUpTest(1, true, true);
}
TEST_F(RampUpTest, UpDownUpThreeStreamsByRedRtx) {
RunRampUpDownUpTest(3, true, true);
}
TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) { RunRampUpDownUpTest(3, true); }
TEST_F(RampUpTest, AbsSendTimeSingleStream) {
RunRampUpTest(1, 0, RtpExtension::kAbsSendTime, false, false);
RunRampUpTest(false, 1, 0, RtpExtension::kAbsSendTime);
}
TEST_F(RampUpTest, AbsSendTimeSimulcast) {
RunRampUpTest(3, 0, RtpExtension::kAbsSendTime, false, false);
RunRampUpTest(false, 3, 0, RtpExtension::kAbsSendTime);
}
TEST_F(RampUpTest, AbsSendTimeSimulcastWithRtx) {
RunRampUpTest(3, 0, RtpExtension::kAbsSendTime, true, false);
}
TEST_F(RampUpTest, AbsSendTimeSimulcastByRedWithRtx) {
RunRampUpTest(3, 0, RtpExtension::kAbsSendTime, true, true);
RunRampUpTest(true, 3, 0, RtpExtension::kAbsSendTime);
}
TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
RunRampUpTest(1, 0.9 * kSingleStreamTargetBps, RtpExtension::kAbsSendTime,
false, false);
RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps,
RtpExtension::kAbsSendTime);
}
} // namespace webrtc

9
webrtc/video/rampup_tests.h
View File

@ -148,13 +148,12 @@ class LowRateStreamObserver : public test::DirectTransport,
class RampUpTest : public test::CallTest {
protected:
void RunRampUpTest(size_t num_streams,
void RunRampUpTest(bool rtx,
size_t num_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,
bool red);
const std::string& extension_type);
void RunRampUpDownUpTest(size_t number_of_streams, bool rtx, bool red);
void RunRampUpDownUpTest(size_t number_of_streams, bool rtx);
};
} // namespace webrtc
#endif // WEBRTC_VIDEO_RAMPUP_TESTS_H_

10
webrtc/video/video_receive_stream.cc
View File

@ -92,13 +92,12 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.local_ssrc);
// TODO(pbos): Support multiple RTX, per video payload.
Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin();
for (; it != config_.rtp.rtx.end(); ++it) {
if (it != config_.rtp.rtx.end()) {
assert(it->second.ssrc != 0);
assert(it->second.payload_type != 0);
rtp_rtcp_->SetRemoteSSRCType(channel_, kViEStreamTypeRtx, it->second.ssrc);
rtp_rtcp_->SetRtxReceivePayloadType(channel_, it->second.payload_type,
it->first);
rtp_rtcp_->SetRtxReceivePayloadType(channel_, it->second.payload_type);
}
rtp_rtcp_->SetRembStatus(channel_, false, config_.rtp.remb);
@ -147,11 +146,6 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
LOG(LS_ERROR) << "Could not set RED codec. This shouldn't happen.";
abort();
}
if (config_.rtp.fec.rtx_payload_type != -1) {
rtp_rtcp_->SetRtxReceivePayloadType(channel_,
config_.rtp.fec.rtx_payload_type,
config_.rtp.fec.red_payload_type);
}
}
stats_proxy_.reset(

9
webrtc/video/video_send_stream.cc
View File

@ -49,6 +49,7 @@ std::string VideoSendStream::Config::Rtp::Rtx::ToString()
ss << "}, {";
}
ss << '}';
ss << ", payload_type: " << payload_type;
ss << '}';
return ss.str();
@ -172,11 +173,6 @@ VideoSendStream::VideoSendStream(
static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
}
if (config_.rtp.fec.rtx_payload_type != -1) {
rtp_rtcp_->SetRtxSendPayloadType(channel_,
config_.rtp.fec.rtx_payload_type,
config_.rtp.fec.red_payload_type);
}
} else {
rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
}
@ -479,8 +475,7 @@ void VideoSendStream::ConfigureSsrcs() {
}
assert(config_.rtp.rtx.payload_type >= 0);
rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type,
config_.encoder_settings.payload_type);
rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type);
}
std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {

9
webrtc/video_engine/include/vie_rtp_rtcp.h
View File

@ -114,13 +114,10 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
// This sets a specific payload type for the RTX stream. Note that this
// doesn't enable RTX, SetLocalSSRC must still be called to enable RTX.
virtual int SetRtxSendPayloadType(const int video_channel,
const uint8_t payload_type,
const uint8_t associated_payload_type) = 0;
const uint8_t payload_type) = 0;
virtual int SetRtxReceivePayloadType(
const int video_channel,
const uint8_t payload_type,
const uint8_t associated_payload_type) = 0;
virtual int SetRtxReceivePayloadType(const int video_channel,
const uint8_t payload_type) = 0;
// This function enables manual initialization of the sequence number. The
// start sequence number is normally a random number.

8
webrtc/video_engine/test/auto_test/source/vie_autotest_loopback.cc
View File

@ -40,7 +40,6 @@
const uint32_t kSsrc = 0x01234567;
const uint32_t kRtxSsrc = 0x01234568;
const int kRtxPayloadType = 98;
const int kPayloadType = 100;
#define VCM_RED_PAYLOAD_TYPE 96
#define VCM_ULPFEC_PAYLOAD_TYPE 97
@ -245,15 +244,14 @@ int VideoEngineSampleCode(void* window1, void* window2)
return -1;
}
error = ptrViERtpRtcp->SetRtxSendPayloadType(videoChannel, kRtxPayloadType,
kPayloadType);
error = ptrViERtpRtcp->SetRtxSendPayloadType(videoChannel, kRtxPayloadType);
if (error == -1) {
printf("ERROR in ViERTP_RTCP::SetRtxSendPayloadType\n");
return -1;
}
error = ptrViERtpRtcp->SetRtxReceivePayloadType(
videoChannel, kRtxPayloadType, kPayloadType);
error = ptrViERtpRtcp->SetRtxReceivePayloadType(videoChannel,
kRtxPayloadType);
if (error == -1) {
printf("ERROR in ViERTP_RTCP::SetRtxReceivePayloadType\n");
return -1;

9
webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
View File

@ -172,15 +172,14 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
myTransport.ClearStats();
const uint8_t kRtxPayloadType = 96;
const uint8_t kPayloadType = 100;
// Temporarily disable pacing.
EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
tbChannel.videoChannel, false));
EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
EXPECT_EQ(0, ViE.rtp_rtcp->SetRtxSendPayloadType(
tbChannel.videoChannel, kRtxPayloadType, kPayloadType));
EXPECT_EQ(0, ViE.rtp_rtcp->SetRtxReceivePayloadType(
tbChannel.videoChannel, kRtxPayloadType, kPayloadType));
EXPECT_EQ(0, ViE.rtp_rtcp->SetRtxSendPayloadType(tbChannel.videoChannel,
kRtxPayloadType));
EXPECT_EQ(0, ViE.rtp_rtcp->SetRtxReceivePayloadType(tbChannel.videoChannel,
kRtxPayloadType));
EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(tbChannel.videoChannel, 1234,
webrtc::kViEStreamTypeRtx, 0));
EXPECT_EQ(0, ViE.rtp_rtcp->SetRemoteSSRCType(tbChannel.videoChannel,

12
webrtc/video_engine/vie_channel.cc
View File

@ -899,12 +899,11 @@ int32_t ViEChannel::GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]) {
return 0;
}
int ViEChannel::SetRtxSendPayloadType(int payload_type,
int associated_payload_type) {
rtp_rtcp_->SetRtxSendPayloadType(payload_type, associated_payload_type);
int ViEChannel::SetRtxSendPayloadType(int payload_type) {
rtp_rtcp_->SetRtxSendPayloadType(payload_type);
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end(); it++) {
(*it)->SetRtxSendPayloadType(payload_type, associated_payload_type);
(*it)->SetRtxSendPayloadType(payload_type);
}
SetRtxSendStatus(true);
return 0;
@ -921,9 +920,8 @@ void ViEChannel::SetRtxSendStatus(bool enable) {
}
}
void ViEChannel::SetRtxReceivePayloadType(int payload_type,
int associated_payload_type) {
vie_receiver_.SetRtxPayloadType(payload_type, associated_payload_type);
void ViEChannel::SetRtxReceivePayloadType(int payload_type) {
vie_receiver_.SetRtxPayloadType(payload_type);
}
int32_t ViEChannel::SetStartSequenceNumber(uint16_t sequence_number) {

4
webrtc/video_engine/vie_channel.h
View File

@ -145,8 +145,8 @@ class ViEChannel
// Gets the CSRC for the incoming stream.
int32_t GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]);
int SetRtxSendPayloadType(int payload_type, int associated_payload_type);
void SetRtxReceivePayloadType(int payload_type, int associated_payload_type);
int SetRtxSendPayloadType(int payload_type);
void SetRtxReceivePayloadType(int payload_type);
// Sets the starting sequence number, must be called before StartSend.
int32_t SetStartSequenceNumber(uint16_t sequence_number);

6
webrtc/video_engine/vie_receiver.cc
View File

@ -102,10 +102,8 @@ void ViEReceiver::SetNackStatus(bool enable,
rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
}
void ViEReceiver::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
rtp_payload_registry_->SetRtxPayloadType(payload_type,
associated_payload_type);
void ViEReceiver::SetRtxPayloadType(int payload_type) {
rtp_payload_registry_->SetRtxPayloadType(payload_type);
}
void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {

2
webrtc/video_engine/vie_receiver.h
View File

@ -47,7 +47,7 @@ class ViEReceiver : public RtpData {
bool RegisterPayload(const VideoCodec& video_codec);
void SetNackStatus(bool enable, int max_nack_reordering_threshold);
void SetRtxPayloadType(int payload_type, int associated_payload_type);
void SetRtxPayloadType(int payload_type);
void SetRtxSsrc(uint32_t ssrc);
bool GetRtxSsrc(uint32_t* ssrc) const;

17
webrtc/video_engine/vie_rtp_rtcp_impl.cc
View File

@ -191,10 +191,8 @@ int ViERTP_RTCPImpl::GetRemoteCSRCs(const int video_channel,
return 0;
}
int ViERTP_RTCPImpl::SetRtxSendPayloadType(
const int video_channel,
const uint8_t payload_type,
const uint8_t associated_payload_type) {
int ViERTP_RTCPImpl::SetRtxSendPayloadType(const int video_channel,
const uint8_t payload_type) {
LOG_F(LS_INFO) << "channel: " << video_channel
<< " payload_type: " << static_cast<int>(payload_type);
ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
@ -203,17 +201,14 @@ int ViERTP_RTCPImpl::SetRtxSendPayloadType(
shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
return -1;
}
if (vie_channel->SetRtxSendPayloadType(payload_type,
associated_payload_type) != 0) {
if (vie_channel->SetRtxSendPayloadType(payload_type) != 0) {
return -1;
}
return 0;
}
int ViERTP_RTCPImpl::SetRtxReceivePayloadType(
const int video_channel,
const uint8_t payload_type,
const uint8_t associated_payload_type) {
int ViERTP_RTCPImpl::SetRtxReceivePayloadType(const int video_channel,
const uint8_t payload_type) {
LOG_F(LS_INFO) << "channel: " << video_channel
<< " payload_type: " << static_cast<int>(payload_type);
ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
@ -222,7 +217,7 @@ int ViERTP_RTCPImpl::SetRtxReceivePayloadType(
shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
return -1;
}
vie_channel->SetRtxReceivePayloadType(payload_type, associated_payload_type);
vie_channel->SetRtxReceivePayloadType(payload_type);
return 0;
}

6
webrtc/video_engine/vie_rtp_rtcp_impl.h
View File

@ -40,11 +40,9 @@ class ViERTP_RTCPImpl
virtual int GetRemoteCSRCs(const int video_channel,
unsigned int CSRCs[kRtpCsrcSize]) const;
virtual int SetRtxSendPayloadType(const int video_channel,
const uint8_t payload_type,
const uint8_t associated_payload_type);
const uint8_t payload_type);
virtual int SetRtxReceivePayloadType(const int video_channel,
const uint8_t payload_type,
const uint8_t associated_payload_type);
const uint8_t payload_type);
virtual int SetStartSequenceNumber(const int video_channel,
uint16_t sequence_number);
virtual void SetRtpStateForSsrc(int video_channel,