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Return a copy of the supported RTP header extensions instead of a reference.

Browse Source 
This also renames the method to better reflect what it does.

BUG=webrtc:5187
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1486123002 .

Cr-Commit-Position: refs/heads/master@{#10910}
This commit is contained in:
Stefan Holmer 2015-12-07 10:45:43 +01:00
parent b86d4e4a8d
commit 9d69c3f4d9
9 changed files with 71 additions and 76 deletions
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8
talk/media/base/fakemediaengine.h
View File

@ -694,12 +694,10 @@ class FakeBaseEngine {
fail_create_channel_(false) {}
void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
return rtp_header_extensions_;
}
RtpCapabilities GetCapabilities() const { return capabilities_; }
void set_rtp_header_extensions(
const std::vector<RtpHeaderExtension>& extensions) {
rtp_header_extensions_ = extensions;
capabilities_.header_extensions = extensions;
}
protected:
@ -708,7 +706,7 @@ class FakeBaseEngine {
// TODO(thaloun): Replace with explicit checks of before & after values.
bool options_changed_;
bool fail_create_channel_;
std::vector<RtpHeaderExtension> rtp_header_extensions_;
RtpCapabilities capabilities_;
};
class FakeVoiceEngine : public FakeBaseEngine {

18
talk/media/base/mediaengine.h
View File

@ -57,6 +57,10 @@ namespace cricket {
class VideoCapturer;
struct RtpCapabilities {
std::vector<RtpHeaderExtension> header_extensions;
};
// MediaEngineInterface is an abstraction of a media engine which can be
// subclassed to support different media componentry backends.
// It supports voice and video operations in the same class to facilitate
@ -108,11 +112,9 @@ class MediaEngineInterface {
virtual int GetInputLevel() = 0;
virtual const std::vector<AudioCodec>& audio_codecs() = 0;
virtual const std::vector<RtpHeaderExtension>&
audio_rtp_header_extensions() = 0;
virtual RtpCapabilities GetAudioCapabilities() = 0;
virtual const std::vector<VideoCodec>& video_codecs() = 0;
virtual const std::vector<RtpHeaderExtension>&
video_rtp_header_extensions() = 0;
virtual RtpCapabilities GetVideoCapabilities() = 0;
// Starts AEC dump using existing file.
virtual bool StartAecDump(rtc::PlatformFile file) = 0;
@ -202,14 +204,14 @@ class CompositeMediaEngine : public MediaEngineInterface {
virtual const std::vector<AudioCodec>& audio_codecs() {
return voice_.codecs();
}
virtual const std::vector<RtpHeaderExtension>& audio_rtp_header_extensions() {
return voice_.rtp_header_extensions();
virtual RtpCapabilities GetAudioCapabilities() {
return voice_.GetCapabilities();
}
virtual const std::vector<VideoCodec>& video_codecs() {
return video_.codecs();
}
virtual const std::vector<RtpHeaderExtension>& video_rtp_header_extensions() {
return video_.rtp_header_extensions();
virtual RtpCapabilities GetVideoCapabilities() {
return video_.GetCapabilities();
}
virtual bool StartAecDump(rtc::PlatformFile file) {

34
talk/media/webrtc/webrtcvideoengine2.cc
View File

@ -488,20 +488,6 @@ WebRtcVideoEngine2::WebRtcVideoEngine2()
external_encoder_factory_(NULL) {
LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
video_codecs_ = GetSupportedCodecs();
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
kRtpTimestampOffsetHeaderExtensionDefaultId));
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
kRtpVideoRotationHeaderExtensionDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
rtp_header_extensions_.push_back(RtpHeaderExtension(
kRtpTransportSequenceNumberHeaderExtension,
kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
}
WebRtcVideoEngine2::~WebRtcVideoEngine2() {
@ -549,9 +535,23 @@ const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const {
return video_codecs_;
}
const std::vector<RtpHeaderExtension>&
WebRtcVideoEngine2::rtp_header_extensions() const {
return rtp_header_extensions_;
RtpCapabilities WebRtcVideoEngine2::GetCapabilities() const {
RtpCapabilities capabilities;
capabilities.header_extensions.push_back(
RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
kRtpTimestampOffsetHeaderExtensionDefaultId));
capabilities.header_extensions.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
capabilities.header_extensions.push_back(
RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
kRtpVideoRotationHeaderExtensionDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
capabilities.header_extensions.push_back(RtpHeaderExtension(
kRtpTransportSequenceNumberHeaderExtension,
kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
return capabilities;
}
void WebRtcVideoEngine2::SetExternalDecoderFactory(

3
talk/media/webrtc/webrtcvideoengine2.h
View File

@ -118,7 +118,7 @@ class WebRtcVideoEngine2 {
const VideoOptions& options);
const std::vector<VideoCodec>& codecs() const;
const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
RtpCapabilities GetCapabilities() const;
// Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
// not take the ownership of |decoder_factory|. The caller needs to make sure
@ -143,7 +143,6 @@ class WebRtcVideoEngine2 {
std::vector<VideoCodec> GetSupportedCodecs() const;
std::vector<VideoCodec> video_codecs_;
std::vector<RtpHeaderExtension> rtp_header_extensions_;
bool initialized_;

41
talk/media/webrtc/webrtcvideoengine2_unittest.cc
View File

@ -242,11 +242,11 @@ TEST_F(WebRtcVideoEngine2Test, DefaultRtxCodecHasAssociatedPayloadTypeSet) {
}
TEST_F(WebRtcVideoEngine2Test, SupportsTimestampOffsetHeaderExtension) {
std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
ASSERT_FALSE(extensions.empty());
for (size_t i = 0; i < extensions.size(); ++i) {
if (extensions[i].uri == kRtpTimestampOffsetHeaderExtension) {
EXPECT_EQ(kRtpTimestampOffsetHeaderExtensionDefaultId, extensions[i].id);
RtpCapabilities capabilities = engine_.GetCapabilities();
ASSERT_FALSE(capabilities.header_extensions.empty());
for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
if (extension.uri == kRtpTimestampOffsetHeaderExtension) {
EXPECT_EQ(kRtpTimestampOffsetHeaderExtensionDefaultId, extension.id);
return;
}
}
@ -254,12 +254,11 @@ TEST_F(WebRtcVideoEngine2Test, SupportsTimestampOffsetHeaderExtension) {
}
TEST_F(WebRtcVideoEngine2Test, SupportsAbsoluteSenderTimeHeaderExtension) {
std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
ASSERT_FALSE(extensions.empty());
for (size_t i = 0; i < extensions.size(); ++i) {
if (extensions[i].uri == kRtpAbsoluteSenderTimeHeaderExtension) {
EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId,
extensions[i].id);
RtpCapabilities capabilities = engine_.GetCapabilities();
ASSERT_FALSE(capabilities.header_extensions.empty());
for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
if (extension.uri == kRtpAbsoluteSenderTimeHeaderExtension) {
EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId, extension.id);
return;
}
}
@ -274,12 +273,12 @@ class WebRtcVideoEngine2WithSendSideBweTest : public WebRtcVideoEngine2Test {
TEST_F(WebRtcVideoEngine2WithSendSideBweTest,
SupportsTransportSequenceNumberHeaderExtension) {
std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
ASSERT_FALSE(extensions.empty());
for (size_t i = 0; i < extensions.size(); ++i) {
if (extensions[i].uri == kRtpTransportSequenceNumberHeaderExtension) {
RtpCapabilities capabilities = engine_.GetCapabilities();
ASSERT_FALSE(capabilities.header_extensions.empty());
for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
if (extension.uri == kRtpTransportSequenceNumberHeaderExtension) {
EXPECT_EQ(kRtpTransportSequenceNumberHeaderExtensionDefaultId,
extensions[i].id);
extension.id);
return;
}
}
@ -287,11 +286,11 @@ TEST_F(WebRtcVideoEngine2WithSendSideBweTest,
}
TEST_F(WebRtcVideoEngine2Test, SupportsVideoRotationHeaderExtension) {
std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
ASSERT_FALSE(extensions.empty());
for (size_t i = 0; i < extensions.size(); ++i) {
if (extensions[i].uri == kRtpVideoRotationHeaderExtension) {
EXPECT_EQ(kRtpVideoRotationHeaderExtensionDefaultId, extensions[i].id);
RtpCapabilities capabilities = engine_.GetCapabilities();
ASSERT_FALSE(capabilities.header_extensions.empty());
for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
if (extension.uri == kRtpVideoRotationHeaderExtension) {
EXPECT_EQ(kRtpVideoRotationHeaderExtensionDefaultId, extension.id);
return;
}
}

28
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -517,18 +517,6 @@ void WebRtcVoiceEngine::Construct() {
// Load our audio codec list.
codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
// Load our RTP Header extensions.
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
kRtpAudioLevelHeaderExtensionDefaultId));
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
rtp_header_extensions_.push_back(RtpHeaderExtension(
kRtpTransportSequenceNumberHeaderExtension,
kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
options_ = GetDefaultEngineOptions();
voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
}
@ -1075,10 +1063,20 @@ const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
return codecs_;
}
const std::vector<RtpHeaderExtension>&
WebRtcVoiceEngine::rtp_header_extensions() const {
RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
return rtp_header_extensions_;
RtpCapabilities capabilities;
capabilities.header_extensions.push_back(RtpHeaderExtension(
kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId));
capabilities.header_extensions.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
capabilities.header_extensions.push_back(RtpHeaderExtension(
kRtpTransportSequenceNumberHeaderExtension,
kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
return capabilities;
}
int WebRtcVoiceEngine::GetLastEngineError() {

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

@ -80,7 +80,7 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
int GetInputLevel();
const std::vector<AudioCodec>& codecs();
const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
RtpCapabilities GetCapabilities() const;
// For tracking WebRtc channels. Needed because we have to pause them
// all when switching devices.
@ -140,7 +140,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
webrtc::AudioDeviceModule* adm_ = nullptr;
bool is_dumping_aec_ = false;
std::vector<AudioCodec> codecs_;
std::vector<RtpHeaderExtension> rtp_header_extensions_;
std::vector<WebRtcVoiceMediaChannel*> channels_;
webrtc::AgcConfig default_agc_config_;

8
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -3109,17 +3109,17 @@ TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
}
// Set up receive extensions.
const auto& e_exts = engine_.rtp_header_extensions();
cricket::RtpCapabilities capabilities = engine_.GetCapabilities();
cricket::AudioRecvParameters recv_parameters;
recv_parameters.extensions = e_exts;
recv_parameters.extensions = capabilities.header_extensions;
channel_->SetRecvParameters(recv_parameters);
EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
for (uint32_t ssrc : ssrcs) {
const auto* s = call_.GetAudioReceiveStream(ssrc);
EXPECT_NE(nullptr, s);
const auto& s_exts = s->GetConfig().rtp.extensions;
EXPECT_EQ(e_exts.size(), s_exts.size());
for (const auto& e_ext : e_exts) {
EXPECT_EQ(capabilities.header_extensions.size(), s_exts.size());
for (const auto& e_ext : capabilities.header_extensions) {
for (const auto& s_ext : s_exts) {
if (e_ext.id == s_ext.id) {
EXPECT_EQ(e_ext.uri, s_ext.name);

4
talk/session/media/channelmanager.cc
View File

@ -156,7 +156,7 @@ void ChannelManager::GetSupportedAudioCodecs(
void ChannelManager::GetSupportedAudioRtpHeaderExtensions(
RtpHeaderExtensions* ext) const {
*ext = media_engine_->audio_rtp_header_extensions();
*ext = media_engine_->GetAudioCapabilities().header_extensions;
}
void ChannelManager::GetSupportedVideoCodecs(
@ -175,7 +175,7 @@ void ChannelManager::GetSupportedVideoCodecs(
void ChannelManager::GetSupportedVideoRtpHeaderExtensions(
RtpHeaderExtensions* ext) const {
*ext = media_engine_->video_rtp_header_extensions();
*ext = media_engine_->GetVideoCapabilities().header_extensions;
}
void ChannelManager::GetSupportedDataCodecs(