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(Auto)update libjingle 66236292-> 66294299

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-05-06 15:54:38 +00:00
parent cfb18dd7a3
commit 150835ea34
7 changed files with 304 additions and 222 deletions
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43
talk/media/webrtc/fakewebrtcvideoengine.h
View File

@ -276,10 +276,10 @@ class FakeWebRtcVideoEngine
tmmbr_(false),
remb_contribute_(false),
remb_bw_partition_(false),
rtp_offset_send_id_(0),
rtp_offset_receive_id_(0),
rtp_absolute_send_time_send_id_(0),
rtp_absolute_send_time_receive_id_(0),
rtp_offset_send_id_(-1),
rtp_offset_receive_id_(-1),
rtp_absolute_send_time_send_id_(-1),
rtp_absolute_send_time_receive_id_(-1),
sender_target_delay_(0),
receiver_target_delay_(0),
transmission_smoothing_(false),
@ -483,21 +483,24 @@ class FakeWebRtcVideoEngine
WEBRTC_ASSERT_CHANNEL(channel);
return channels_.find(channel)->second->remb_contribute_;
}
int GetSendRtpTimestampOffsetExtensionId(int channel) {
int GetSendRtpExtensionId(int channel, const std::string& extension) {
WEBRTC_ASSERT_CHANNEL(channel);
return channels_.find(channel)->second->rtp_offset_send_id_;
if (extension == kRtpTimestampOffsetHeaderExtension) {
return channels_.find(channel)->second->rtp_offset_send_id_;
} else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) {
return channels_.find(channel)->second->rtp_absolute_send_time_send_id_;
}
return -1;
}
int GetReceiveRtpTimestampOffsetExtensionId(int channel) {
int GetReceiveRtpExtensionId(int channel, const std::string& extension) {
WEBRTC_ASSERT_CHANNEL(channel);
return channels_.find(channel)->second->rtp_offset_receive_id_;
}
int GetSendAbsoluteSendTimeExtensionId(int channel) {
WEBRTC_ASSERT_CHANNEL(channel);
return channels_.find(channel)->second->rtp_absolute_send_time_send_id_;
}
int GetReceiveAbsoluteSendTimeExtensionId(int channel) {
WEBRTC_ASSERT_CHANNEL(channel);
return channels_.find(channel)->second->rtp_absolute_send_time_receive_id_;
if (extension == kRtpTimestampOffsetHeaderExtension) {
return channels_.find(channel)->second->rtp_offset_receive_id_;
} else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) {
return
channels_.find(channel)->second->rtp_absolute_send_time_receive_id_;
}
return -1;
}
bool GetTransmissionSmoothingStatus(int channel) {
WEBRTC_ASSERT_CHANNEL(channel);
@ -1046,25 +1049,25 @@ class FakeWebRtcVideoEngine
WEBRTC_FUNC(SetSendTimestampOffsetStatus, (int channel, bool enable,
int id)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->rtp_offset_send_id_ = (enable) ? id : 0;
channels_[channel]->rtp_offset_send_id_ = (enable) ? id : -1;
return 0;
}
WEBRTC_FUNC(SetReceiveTimestampOffsetStatus, (int channel, bool enable,
int id)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->rtp_offset_receive_id_ = (enable) ? id : 0;
channels_[channel]->rtp_offset_receive_id_ = (enable) ? id : -1;
return 0;
}
WEBRTC_FUNC(SetSendAbsoluteSendTimeStatus, (int channel, bool enable,
int id)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->rtp_absolute_send_time_send_id_ = (enable) ? id : 0;
channels_[channel]->rtp_absolute_send_time_send_id_ = (enable) ? id : -1;
return 0;
}
WEBRTC_FUNC(SetReceiveAbsoluteSendTimeStatus, (int channel, bool enable,
int id)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->rtp_absolute_send_time_receive_id_ = (enable) ? id : 0;
channels_[channel]->rtp_absolute_send_time_receive_id_ = (enable) ? id : -1;
return 0;
}
WEBRTC_STUB(SetRtcpXrRrtrStatus, (int, bool));

34
talk/media/webrtc/fakewebrtcvoiceengine.h
View File

@ -110,6 +110,7 @@ class FakeWebRtcVoiceEngine
video_channel(-1),
send_ssrc(0),
send_audio_level_ext_(-1),
receive_audio_level_ext_(-1),
send_absolute_sender_time_ext_(-1),
receive_absolute_sender_time_ext_(-1) {
memset(&send_codec, 0, sizeof(send_codec));
@ -138,6 +139,7 @@ class FakeWebRtcVoiceEngine
int video_channel;
uint32 send_ssrc;
int send_audio_level_ext_;
int receive_audio_level_ext_;
int send_absolute_sender_time_ext_;
int receive_absolute_sender_time_ext_;
DtmfInfo dtmf_info;
@ -299,14 +301,23 @@ class FakeWebRtcVoiceEngine
channels_[++last_channel_] = ch;
return last_channel_;
}
int GetSendAudioLevelId(int channel) {
return channels_[channel]->send_audio_level_ext_;
int GetSendRtpExtensionId(int channel, const std::string& extension) {
WEBRTC_ASSERT_CHANNEL(channel);
if (extension == kRtpAudioLevelHeaderExtension) {
return channels_[channel]->send_audio_level_ext_;
} else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) {
return channels_[channel]->send_absolute_sender_time_ext_;
}
return -1;
}
int GetSendAbsoluteSenderTimeId(int channel) {
return channels_[channel]->send_absolute_sender_time_ext_;
}
int GetReceiveAbsoluteSenderTimeId(int channel) {
return channels_[channel]->receive_absolute_sender_time_ext_;
int GetReceiveRtpExtensionId(int channel, const std::string& extension) {
WEBRTC_ASSERT_CHANNEL(channel);
if (extension == kRtpAudioLevelHeaderExtension) {
return channels_[channel]->receive_audio_level_ext_;
} else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) {
return channels_[channel]->receive_absolute_sender_time_ext_;
}
return -1;
}
int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
@ -736,6 +747,15 @@ class FakeWebRtcVoiceEngine
channels_[channel]->send_audio_level_ext_ = (enable) ? id : -1;
return 0;
}
#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
unsigned char id)) {
WEBRTC_CHECK_CHANNEL(channel);
WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
channels_[channel]->receive_audio_level_ext_ = (enable) ? id : -1;
return 0;
}
#endif // USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
unsigned char id)) {
WEBRTC_CHECK_CHANNEL(channel);

9
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -2728,7 +2728,6 @@ bool WebRtcVideoMediaChannel::SetRecvRtpHeaderExtensions(
if (receive_extensions_ == extensions) {
return true;
}
receive_extensions_ = extensions;
const RtpHeaderExtension* offset_extension =
FindHeaderExtension(extensions, kRtpTimestampOffsetHeaderExtension);
@ -2750,12 +2749,16 @@ bool WebRtcVideoMediaChannel::SetRecvRtpHeaderExtensions(
return false;
}
}
receive_extensions_ = extensions;
return true;
}
bool WebRtcVideoMediaChannel::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
send_extensions_ = extensions;
if (send_extensions_ == extensions) {
return true;
}
const RtpHeaderExtension* offset_extension =
FindHeaderExtension(extensions, kRtpTimestampOffsetHeaderExtension);
@ -2786,6 +2789,8 @@ bool WebRtcVideoMediaChannel::SetSendRtpHeaderExtensions(
talk_base::Socket::OPT_RTP_SENDTIME_EXTN_ID,
send_time_extension->id);
}
send_extensions_ = extensions;
return true;
}

163
talk/media/webrtc/webrtcvideoengine_unittest.cc
View File

@ -50,6 +50,9 @@
// Tests for the WebRtcVideoEngine/VideoChannel code.
using cricket::kRtpTimestampOffsetHeaderExtension;
using cricket::kRtpAbsoluteSenderTimeHeaderExtension;
static const cricket::VideoCodec kVP8Codec720p(100, "VP8", 1280, 720, 30, 0);
static const cricket::VideoCodec kVP8Codec360p(100, "VP8", 640, 360, 30, 0);
static const cricket::VideoCodec kVP8Codec270p(100, "VP8", 480, 270, 30, 0);
@ -148,6 +151,76 @@ class WebRtcVideoEngineTestFake : public testing::Test,
channel_->SendFrame(&capturer, &frame);
return true;
}
void TestSetSendRtpHeaderExtensions(const std::string& ext) {
EXPECT_TRUE(SetupEngine());
int channel_num = vie_.GetLastChannel();
// Verify extensions are off by default.
EXPECT_EQ(-1, vie_.GetSendRtpExtensionId(channel_num, ext));
// Enable extension.
const int id = 1;
std::vector<cricket::RtpHeaderExtension> extensions;
extensions.push_back(cricket::RtpHeaderExtension(ext, id));
// Verify the send extension id.
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(id, vie_.GetSendRtpExtensionId(channel_num, ext));
// Verify call with same set of extensions returns true.
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(id, vie_.GetSendRtpExtensionId(channel_num, ext));
// Add a new send stream and verify the extension is set.
// The first send stream to occupy the default channel.
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(123)));
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(234)));
int new_send_channel_num = vie_.GetLastChannel();
EXPECT_NE(channel_num, new_send_channel_num);
EXPECT_EQ(id, vie_.GetSendRtpExtensionId(new_send_channel_num, ext));
// Remove the extension id.
std::vector<cricket::RtpHeaderExtension> empty_extensions;
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(empty_extensions));
EXPECT_EQ(-1, vie_.GetSendRtpExtensionId(channel_num, ext));
EXPECT_EQ(-1, vie_.GetSendRtpExtensionId(new_send_channel_num, ext));
}
void TestSetRecvRtpHeaderExtensions(const std::string& ext) {
EXPECT_TRUE(SetupEngine());
int channel_num = vie_.GetLastChannel();
// Verify extensions are off by default.
EXPECT_EQ(-1, vie_.GetReceiveRtpExtensionId(channel_num, ext));
// Enable extension.
const int id = 2;
std::vector<cricket::RtpHeaderExtension> extensions;
extensions.push_back(cricket::RtpHeaderExtension(ext, id));
// Verify receive extension id.
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
EXPECT_EQ(id, vie_.GetReceiveRtpExtensionId(channel_num, ext));
// Verify call with same set of extensions returns true.
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
EXPECT_EQ(id, vie_.GetReceiveRtpExtensionId(channel_num, ext));
// Add a new receive stream and verify the extension is set.
// The first send stream to occupy the default channel.
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(345)));
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(456)));
int new_recv_channel_num = vie_.GetLastChannel();
EXPECT_NE(channel_num, new_recv_channel_num);
EXPECT_EQ(id, vie_.GetReceiveRtpExtensionId(new_recv_channel_num, ext));
// Remove the extension id.
std::vector<cricket::RtpHeaderExtension> empty_extensions;
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(empty_extensions));
EXPECT_EQ(-1, vie_.GetReceiveRtpExtensionId(channel_num, ext));
EXPECT_EQ(-1, vie_.GetReceiveRtpExtensionId(new_recv_channel_num, ext));
}
void VerifyCodecFeedbackParams(const cricket::VideoCodec& codec) {
EXPECT_TRUE(codec.HasFeedbackParam(
cricket::FeedbackParam(cricket::kRtcpFbParamNack,
@ -730,89 +803,19 @@ TEST_F(WebRtcVideoEngineTestFake, RecvStreamNoRtx) {
}
// Test support for RTP timestamp offset header extension.
TEST_F(WebRtcVideoEngineTestFake, RtpTimestampOffsetHeaderExtensions) {
EXPECT_TRUE(SetupEngine());
int channel_num = vie_.GetLastChannel();
cricket::VideoOptions options;
options.conference_mode.Set(true);
EXPECT_TRUE(channel_->SetOptions(options));
// Verify extensions are off by default.
EXPECT_EQ(0, vie_.GetSendRtpTimestampOffsetExtensionId(channel_num));
EXPECT_EQ(0, vie_.GetReceiveRtpTimestampOffsetExtensionId(channel_num));
// Enable RTP timestamp extension.
const int id = 14;
std::vector<cricket::RtpHeaderExtension> extensions;
extensions.push_back(cricket::RtpHeaderExtension(
"urn:ietf:params:rtp-hdrext:toffset", id));
// Verify the send extension id.
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(id, vie_.GetSendRtpTimestampOffsetExtensionId(channel_num));
// Remove the extension id.
std::vector<cricket::RtpHeaderExtension> empty_extensions;
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(empty_extensions));
EXPECT_EQ(0, vie_.GetSendRtpTimestampOffsetExtensionId(channel_num));
// Verify receive extension id.
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
EXPECT_EQ(id, vie_.GetReceiveRtpTimestampOffsetExtensionId(channel_num));
// Add a new receive stream and verify the extension is set.
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
int new_channel_num = vie_.GetLastChannel();
EXPECT_NE(channel_num, new_channel_num);
EXPECT_EQ(id, vie_.GetReceiveRtpTimestampOffsetExtensionId(new_channel_num));
// Remove the extension id.
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(empty_extensions));
EXPECT_EQ(0, vie_.GetReceiveRtpTimestampOffsetExtensionId(channel_num));
EXPECT_EQ(0, vie_.GetReceiveRtpTimestampOffsetExtensionId(new_channel_num));
TEST_F(WebRtcVideoEngineTestFake, SendRtpTimestampOffsetHeaderExtensions) {
TestSetSendRtpHeaderExtensions(kRtpTimestampOffsetHeaderExtension);
}
TEST_F(WebRtcVideoEngineTestFake, RecvRtpTimestampOffsetHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(kRtpTimestampOffsetHeaderExtension);
}
// Test support for absolute send time header extension.
TEST_F(WebRtcVideoEngineTestFake, AbsoluteSendTimeHeaderExtensions) {
EXPECT_TRUE(SetupEngine());
int channel_num = vie_.GetLastChannel();
cricket::VideoOptions options;
options.conference_mode.Set(true);
EXPECT_TRUE(channel_->SetOptions(options));
// Verify extensions are off by default.
EXPECT_EQ(0, vie_.GetSendAbsoluteSendTimeExtensionId(channel_num));
EXPECT_EQ(0, vie_.GetReceiveAbsoluteSendTimeExtensionId(channel_num));
// Enable RTP timestamp extension.
const int id = 12;
std::vector<cricket::RtpHeaderExtension> extensions;
extensions.push_back(cricket::RtpHeaderExtension(
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", id));
// Verify the send extension id.
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(id, vie_.GetSendAbsoluteSendTimeExtensionId(channel_num));
// Remove the extension id.
std::vector<cricket::RtpHeaderExtension> empty_extensions;
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(empty_extensions));
EXPECT_EQ(0, vie_.GetSendAbsoluteSendTimeExtensionId(channel_num));
// Verify receive extension id.
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
EXPECT_EQ(id, vie_.GetReceiveAbsoluteSendTimeExtensionId(channel_num));
// Add a new receive stream and verify the extension is set.
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
int new_channel_num = vie_.GetLastChannel();
EXPECT_NE(channel_num, new_channel_num);
EXPECT_EQ(id, vie_.GetReceiveAbsoluteSendTimeExtensionId(new_channel_num));
// Remove the extension id.
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(empty_extensions));
EXPECT_EQ(0, vie_.GetReceiveAbsoluteSendTimeExtensionId(channel_num));
EXPECT_EQ(0, vie_.GetReceiveAbsoluteSendTimeExtensionId(new_channel_num));
TEST_F(WebRtcVideoEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) {
TestSetSendRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension);
}
TEST_F(WebRtcVideoEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension);
}
TEST_F(WebRtcVideoEngineTestFake, LeakyBucketTest) {

106
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -2202,53 +2202,96 @@ bool WebRtcVoiceMediaChannel::SetSendCodec(
bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
if (receive_extensions_ == extensions) {
return true;
}
// The default channel may or may not be in |receive_channels_|. Set the rtp
// header extensions for default channel regardless.
if (!SetChannelRecvRtpHeaderExtensions(voe_channel(), extensions)) {
return false;
}
// Loop through all receive channels and enable/disable the extensions.
for (ChannelMap::const_iterator channel_it = receive_channels_.begin();
channel_it != receive_channels_.end(); ++channel_it) {
int channel_id = channel_it->second->channel();
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id,
send_time_extension)) {
if (!SetChannelRecvRtpHeaderExtensions(channel_it->second->channel(),
extensions)) {
return false;
}
}
receive_extensions_ = extensions;
return true;
}
bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
#ifdef USE_WEBRTC_DEV_BRANCH
const RtpHeaderExtension* audio_level_extension =
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus, channel_id,
audio_level_extension)) {
return false;
}
#endif // USE_WEBRTC_DEV_BRANCH
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id,
send_time_extension)) {
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
if (send_extensions_ == extensions) {
return true;
}
// The default channel may or may not be in |send_channels_|. Set the rtp
// header extensions for default channel regardless.
if (!SetChannelSendRtpHeaderExtensions(voe_channel(), extensions)) {
return false;
}
// Loop through all send channels and enable/disable the extensions.
for (ChannelMap::const_iterator channel_it = send_channels_.begin();
channel_it != send_channels_.end(); ++channel_it) {
if (!SetChannelSendRtpHeaderExtensions(channel_it->second->channel(),
extensions)) {
return false;
}
}
send_extensions_ = extensions;
return true;
}
bool WebRtcVoiceMediaChannel::SetChannelSendRtpHeaderExtensions(
int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
const RtpHeaderExtension* audio_level_extension =
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, voe_channel(),
&webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id,
audio_level_extension)) {
return false;
}
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, voe_channel(),
&webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id,
send_time_extension)) {
return false;
}
for (ChannelMap::const_iterator channel_it = send_channels_.begin();
channel_it != send_channels_.end(); ++channel_it) {
int channel_id = channel_it->second->channel();
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id,
audio_level_extension)) {
return false;
}
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id,
send_time_extension)) {
return false;
}
}
return true;
}
@ -2352,6 +2395,7 @@ bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
return true;
}
// TODO(ronghuawu): Change this method to return bool.
void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) {
if (engine()->voe()->network()->RegisterExternalTransport(
channel, *this) == -1) {
@ -2363,6 +2407,9 @@ void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) {
// Reset all recv codecs; they will be enabled via SetRecvCodecs.
ResetRecvCodecs(channel);
// Set RTP header extension for the new channel.
SetChannelSendRtpHeaderExtensions(channel, send_extensions_);
}
bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) {
@ -2599,6 +2646,11 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
}
SetNack(channel, nack_enabled_);
// Set RTP header extension for the new channel.
if (!SetChannelRecvRtpHeaderExtensions(channel, receive_extensions_)) {
return false;
}
return SetPlayout(channel, playout_);
}
@ -3474,13 +3526,15 @@ VoiceMediaChannel::Error
bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
int channel_id, const RtpHeaderExtension* extension) {
bool enable = false;
unsigned char id = 0;
int id = 0;
std::string uri;
if (extension) {
enable = true;
id = extension->id;
uri = extension->uri;
}
if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) {
LOG_RTCERR4(*setter, extension->uri, channel_id, enable, id);
LOG_RTCERR4(*setter, uri, channel_id, enable, id);
return false;
}
return true;

9
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -416,6 +416,13 @@ class WebRtcVoiceMediaChannel
bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
const RtpHeaderExtension* extension);
bool SetChannelRecvRtpHeaderExtensions(
int channel_id,
const std::vector<RtpHeaderExtension>& extensions);
bool SetChannelSendRtpHeaderExtensions(
int channel_id,
const std::vector<RtpHeaderExtension>& extensions);
talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
std::set<int> ringback_channels_; // channels playing ringback
std::vector<AudioCodec> recv_codecs_;
@ -436,6 +443,7 @@ class WebRtcVoiceMediaChannel
// When the default channel (voe_channel) is used for sending, it is
// contained in send_channels_, otherwise not.
ChannelMap send_channels_;
std::vector<RtpHeaderExtension> send_extensions_;
uint32 default_receive_ssrc_;
// Note the default channel (voe_channel()) can reside in both
// receive_channels_ and send_channels_ in non-conference mode and in that
@ -445,6 +453,7 @@ class WebRtcVoiceMediaChannel
// the WebRtc thread must be synchronized with edits on the worker thread.
// Reads on the worker thread are ok.
//
std::vector<RtpHeaderExtension> receive_extensions_;
// Do not lock this on the VoE media processor thread; potential for deadlock
// exists.
mutable talk_base::CriticalSection receive_channels_cs_;

162
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -44,6 +44,9 @@
// Tests for the WebRtcVoiceEngine/VoiceChannel code.
using cricket::kRtpAudioLevelHeaderExtension;
using cricket::kRtpAbsoluteSenderTimeHeaderExtension;
static const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0);
static const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0);
static const cricket::AudioCodec kCeltCodec(110, "CELT", 32000, 64000, 2, 0);
@ -136,17 +139,19 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
options_conference_.conference_mode.Set(true);
options_adjust_agc_.adjust_agc_delta.Set(-10);
}
bool SetupEngineWithoutStream() {
if (!engine_.Init(talk_base::Thread::Current())) {
return false;
}
channel_ = engine_.CreateChannel();
return (channel_ != NULL);
}
bool SetupEngine() {
bool result = engine_.Init(talk_base::Thread::Current());
if (result) {
channel_ = engine_.CreateChannel();
result = (channel_ != NULL);
if (!SetupEngineWithoutStream()) {
return false;
}
if (result) {
result = channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrc1));
}
return result;
return channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrc1));
}
void SetupForMultiSendStream() {
EXPECT_TRUE(SetupEngine());
@ -246,75 +251,88 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
EXPECT_EQ(expected_bitrate, temp_codec.rate);
}
void TestSetSendRtpHeaderExtensions(int channel_id) {
std::vector<cricket::RtpHeaderExtension> extensions;
void TestSetSendRtpHeaderExtensions(const std::string& ext) {
EXPECT_TRUE(SetupEngineWithoutStream());
int channel_num = voe_.GetLastChannel();
// Ensure extensions are off by default.
EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
std::vector<cricket::RtpHeaderExtension> extensions;
// Ensure unknown extensions won't cause an error.
extensions.push_back(cricket::RtpHeaderExtension(
"urn:ietf:params:unknownextention", 1));
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
// Ensure extensions stay off with an empty list of headers.
extensions.clear();
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
// Ensure audio levels are enabled if the audio-level header is specified
// (but AST is still off).
extensions.push_back(cricket::RtpHeaderExtension(
"urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8));
// Ensure extension is set properly.
const int id = 1;
extensions.push_back(cricket::RtpHeaderExtension(ext, id));
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(8, voe_.GetSendAudioLevelId(channel_id));
EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(id, voe_.GetSendRtpExtensionId(channel_num, ext));
// Ensure audio level and AST are enabled if the extensions are specified.
extensions.push_back(cricket::RtpHeaderExtension(
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 12));
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(8, voe_.GetSendAudioLevelId(channel_id));
EXPECT_EQ(12, voe_.GetSendAbsoluteSenderTimeId(channel_id));
// Ensure extension is set properly on new channel.
// The first stream to occupy the default channel.
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(123)));
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(234)));
int new_channel_num = voe_.GetLastChannel();
EXPECT_NE(channel_num, new_channel_num);
EXPECT_EQ(id, voe_.GetSendRtpExtensionId(new_channel_num, ext));
// Ensure all extensions go back off with an empty list.
extensions.clear();
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(new_channel_num, ext));
}
void TestSetRecvRtpHeaderExtensions(int channel_id) {
std::vector<cricket::RtpHeaderExtension> extensions;
void TestSetRecvRtpHeaderExtensions(const std::string& ext) {
EXPECT_TRUE(SetupEngineWithoutStream());
int channel_num = voe_.GetLastChannel();
// Ensure extensions are off by default.
EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
std::vector<cricket::RtpHeaderExtension> extensions;
// Ensure unknown extensions won't cause an error.
extensions.push_back(cricket::RtpHeaderExtension(
"urn:ietf:params:unknownextention", 1));
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
// An empty list shouldn't cause any headers to be enabled.
// Ensure extensions stay off with an empty list of headers.
extensions.clear();
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
// Nor should indicating we can receive the absolute sender time header.
extensions.push_back(cricket::RtpHeaderExtension(
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 11));
// Ensure extension is set properly.
const int id = 2;
extensions.push_back(cricket::RtpHeaderExtension(ext, id));
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
EXPECT_EQ(11, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(id, voe_.GetReceiveRtpExtensionId(channel_num, ext));
// Resetting to an empty list shouldn't cause any headers to be enabled.
// Ensure extension is set properly on new channel.
// The first stream to occupy the default channel.
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(345)));
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(456)));
int new_channel_num = voe_.GetLastChannel();
EXPECT_NE(channel_num, new_channel_num);
EXPECT_EQ(id, voe_.GetReceiveRtpExtensionId(new_channel_num, ext));
// Ensure all extensions go back off with an empty list.
extensions.clear();
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(new_channel_num, ext));
}
protected:
@ -1559,17 +1577,22 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED5) {
EXPECT_FALSE(voe_.GetFEC(channel_num));
}
// Test that we support setting certain send header extensions.
TEST_F(WebRtcVoiceEngineTestFake, SetSendRtpHeaderExtensions) {
EXPECT_TRUE(SetupEngine());
TestSetSendRtpHeaderExtensions(voe_.GetLastChannel());
// Test support for audio level header extension.
TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) {
TestSetSendRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);
}
#ifdef USE_WEBRTC_DEV_BRANCH
TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);
}
#endif // USE_WEBRTC_DEV_BRANCH
// Test that we support setting recv header extensions.
TEST_F(WebRtcVoiceEngineTestFake, SetRecvRtpHeaderExtensions) {
EXPECT_TRUE(SetupEngine());
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
TestSetRecvRtpHeaderExtensions(voe_.GetLastChannel());
// Test support for absolute send time header extension.
TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) {
TestSetSendRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension);
}
TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension);
}
// Test that we can create a channel and start sending/playing out on it.
@ -1729,41 +1752,6 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
EXPECT_EQ(1u, info.receivers.size());
}
// Test that we support setting header extensions on multiple send streams.
TEST_F(WebRtcVoiceEngineTestFake,
SetSendRtpHeaderExtensionsWithMultipleSendStreams) {
SetupForMultiSendStream();
static const uint32 kSsrcs[] = {1, 2, 3, 4};
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrcs[i])));
}
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs[i]);
TestSetSendRtpHeaderExtensions(channel_num);
}
}
// Test that we support setting header extensions on multiple receive streams.
TEST_F(WebRtcVoiceEngineTestFake,
SetRecvRtpHeaderExtensionsWithMultipleRecvStreams) {
EXPECT_TRUE(SetupEngine());
static const uint32 kSsrcs[] = {1, 2, 3, 4};
int channel_ids[ARRAY_SIZE(kSsrcs)] = {0};
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrcs[i])));
channel_ids[i] = voe_.GetLastChannel();
}
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
TestSetRecvRtpHeaderExtensions(channel_ids[i]);
}
}
// Test that we can add and remove receive streams, and do proper send/playout.
// We can receive on multiple streams while sending one stream.
TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {