Rename VideoCall to Call.
Call should encompass more than video, there's no point in calling it
VideoCall.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2191005
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4704 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/internal/video_call.cc b/video_engine/internal/call.cc
similarity index 81%
rename from video_engine/internal/video_call.cc
rename to video_engine/internal/call.cc
index 6e50395..acb65bf 100644
--- a/video_engine/internal/video_call.cc
+++ b/video_engine/internal/call.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/video_engine/internal/video_call.h"
+#include "webrtc/video_engine/internal/call.h"
#include <assert.h>
#include <string.h>
@@ -24,17 +24,16 @@
namespace webrtc {
-VideoCall* VideoCall::Create(const VideoCall::Config& config) {
- webrtc::VideoEngine* video_engine = webrtc::VideoEngine::Create();
+Call* Call::Create(const Call::Config& config) {
+ VideoEngine* video_engine = VideoEngine::Create();
assert(video_engine != NULL);
- return new internal::VideoCall(video_engine, config);
+ return new internal::Call(video_engine, config);
}
namespace internal {
-VideoCall::VideoCall(webrtc::VideoEngine* video_engine,
- const VideoCall::Config& config)
+Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config)
: config_(config),
receive_lock_(RWLockWrapper::CreateRWLock()),
send_lock_(RWLockWrapper::CreateRWLock()),
@@ -50,15 +49,15 @@
assert(codec_ != NULL);
}
-VideoCall::~VideoCall() {
+Call::~Call() {
codec_->Release();
rtp_rtcp_->Release();
webrtc::VideoEngine::Delete(video_engine_);
}
-PacketReceiver* VideoCall::Receiver() { return this; }
+PacketReceiver* Call::Receiver() { return this; }
-std::vector<VideoCodec> VideoCall::GetVideoCodecs() {
+std::vector<VideoCodec> Call::GetVideoCodecs() {
std::vector<VideoCodec> codecs;
VideoCodec codec;
@@ -70,14 +69,13 @@
return codecs;
}
-VideoSendStream::Config VideoCall::GetDefaultSendConfig() {
+VideoSendStream::Config Call::GetDefaultSendConfig() {
VideoSendStream::Config config;
codec_->GetCodec(0, config.codec);
return config;
}
-VideoSendStream* VideoCall::CreateSendStream(
- const VideoSendStream::Config& config) {
+VideoSendStream* Call::CreateSendStream(const VideoSendStream::Config& config) {
assert(config.rtp.ssrcs.size() > 0);
assert(config.codec.numberOfSimulcastStreams == 0 ||
config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size());
@@ -93,8 +91,7 @@
return send_stream;
}
-SendStreamState* VideoCall::DestroySendStream(
- webrtc::VideoSendStream* send_stream) {
+SendStreamState* Call::DestroySendStream(webrtc::VideoSendStream* send_stream) {
assert(send_stream != NULL);
VideoSendStream* send_stream_impl = NULL;
@@ -119,11 +116,11 @@
return NULL;
}
-VideoReceiveStream::Config VideoCall::GetDefaultReceiveConfig() {
+VideoReceiveStream::Config Call::GetDefaultReceiveConfig() {
return VideoReceiveStream::Config();
}
-VideoReceiveStream* VideoCall::CreateReceiveStream(
+VideoReceiveStream* Call::CreateReceiveStream(
const VideoReceiveStream::Config& config) {
VideoReceiveStream* receive_stream =
new VideoReceiveStream(video_engine_, config, config_.send_transport);
@@ -134,8 +131,7 @@
return receive_stream;
}
-void VideoCall::DestroyReceiveStream(
- webrtc::VideoReceiveStream* receive_stream) {
+void Call::DestroyReceiveStream(webrtc::VideoReceiveStream* receive_stream) {
assert(receive_stream != NULL);
VideoReceiveStream* receive_stream_impl = NULL;
@@ -157,17 +153,17 @@
delete receive_stream_impl;
}
-uint32_t VideoCall::SendBitrateEstimate() {
+uint32_t Call::SendBitrateEstimate() {
// TODO(pbos): Return send-bitrate estimate
return 0;
}
-uint32_t VideoCall::ReceiveBitrateEstimate() {
+uint32_t Call::ReceiveBitrateEstimate() {
// TODO(pbos): Return receive-bitrate estimate
return 0;
}
-bool VideoCall::DeliverRtcp(const uint8_t* packet, size_t length) {
+bool Call::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(pbos): Figure out what channel needs it actually.
// Do NOT broadcast! Also make sure it's a valid packet.
bool rtcp_delivered = false;
@@ -199,9 +195,9 @@
return rtcp_delivered;
}
-bool VideoCall::DeliverRtp(const RTPHeader& header,
- const uint8_t* packet,
- size_t length) {
+bool Call::DeliverRtp(const RTPHeader& header,
+ const uint8_t* packet,
+ size_t length) {
VideoReceiveStream* receiver;
{
ReadLockScoped read_lock(*receive_lock_);
@@ -217,7 +213,7 @@
return receiver->DeliverRtp(static_cast<const uint8_t*>(packet), length);
}
-bool VideoCall::DeliverPacket(const uint8_t* packet, size_t length) {
+bool Call::DeliverPacket(const uint8_t* packet, size_t length) {
// TODO(pbos): ExtensionMap if there are extensions.
if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)))
return DeliverRtcp(packet, length);
diff --git a/video_engine/internal/video_call.h b/video_engine/internal/call.h
similarity index 80%
rename from video_engine/internal/video_call.h
rename to video_engine/internal/call.h
index a271d48..13cdd89 100644
--- a/video_engine/internal/video_call.h
+++ b/video_engine/internal/call.h
@@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_VIDEO_ENGINE_VIDEO_CALL_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIDEO_CALL_IMPL_H_
+#ifndef WEBRTC_VIDEO_ENGINE_CALL_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_CALL_IMPL_H_
#include <map>
#include <vector>
@@ -18,7 +18,7 @@
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/video_engine/internal/video_receive_stream.h"
#include "webrtc/video_engine/internal/video_send_stream.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
namespace webrtc {
@@ -30,11 +30,10 @@
// TODO(pbos): Split out the packet receiver, should be sharable between
// VideoEngine and VoiceEngine.
-class VideoCall : public webrtc::VideoCall, public PacketReceiver {
+class Call : public webrtc::Call, public PacketReceiver {
public:
- VideoCall(webrtc::VideoEngine* video_engine,
- const VideoCall::Config& config);
- virtual ~VideoCall();
+ Call(webrtc::VideoEngine* video_engine, const Call::Config& config);
+ virtual ~Call();
virtual PacketReceiver* Receiver() OVERRIDE;
virtual std::vector<VideoCodec> GetVideoCodecs() OVERRIDE;
@@ -52,8 +51,8 @@
virtual VideoReceiveStream* CreateReceiveStream(
const VideoReceiveStream::Config& config) OVERRIDE;
- virtual void DestroyReceiveStream(webrtc::VideoReceiveStream* receive_stream)
- OVERRIDE;
+ virtual void DestroyReceiveStream(
+ webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
virtual uint32_t SendBitrateEstimate() OVERRIDE;
virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
@@ -66,7 +65,7 @@
const uint8_t* packet,
size_t length);
- VideoCall::Config config_;
+ Call::Config config_;
std::map<uint32_t, VideoReceiveStream*> receive_ssrcs_;
scoped_ptr<RWLockWrapper> receive_lock_;
@@ -80,9 +79,9 @@
ViERTP_RTCP* rtp_rtcp_;
ViECodec* codec_;
- DISALLOW_COPY_AND_ASSIGN(VideoCall);
+ DISALLOW_COPY_AND_ASSIGN(Call);
};
} // namespace internal
} // namespace webrtc
-#endif // WEBRTC_VIDEO_ENGINE_INTERNAL_VIDEO_CALL_H_
+#endif // WEBRTC_VIDEO_ENGINE_INTERNAL_CALL_H_
diff --git a/video_engine/new_include/video_call.h b/video_engine/new_include/call.h
similarity index 84%
rename from video_engine/new_include/video_call.h
rename to video_engine/new_include/call.h
index 96a93e5..6650216 100644
--- a/video_engine/new_include/video_call.h
+++ b/video_engine/new_include/call.h
@@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
-#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
+#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
+#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
#include <string>
#include <vector>
@@ -31,10 +31,10 @@
virtual ~PacketReceiver() {}
};
-// A VideoCall instance can contain several send and/or receive streams. All
-// streams are assumed to have the same remote endpoint and will share bitrate
-// estimates etc.
-class VideoCall {
+// A Call instance can contain several send and/or receive streams. All streams
+// are assumed to have the same remote endpoint and will share bitrate estimates
+// etc.
+class Call {
public:
struct Config {
explicit Config(newapi::Transport* send_transport)
@@ -47,14 +47,14 @@
newapi::Transport* send_transport;
bool overuse_detection;
- // VoiceEngine used for audio/video synchronization for this VideoCall.
+ // VoiceEngine used for audio/video synchronization for this Call.
VoiceEngine* voice_engine;
TraceCallback* trace_callback;
uint32_t trace_filter;
};
- static VideoCall* Create(const VideoCall::Config& config);
+ static Call* Create(const Call::Config& config);
virtual std::vector<VideoCodec> GetVideoCodecs() = 0;
@@ -76,7 +76,7 @@
// All received RTP and RTCP packets for the call should be inserted to this
// PacketReceiver. The PacketReceiver pointer is valid as long as the
- // VideoCall instance exists.
+ // Call instance exists.
virtual PacketReceiver* Receiver() = 0;
// Returns the estimated total send bandwidth. Note: this can differ from the
@@ -87,8 +87,8 @@
// differ from the actual receive bitrate.
virtual uint32_t ReceiveBitrateEstimate() = 0;
- virtual ~VideoCall() {}
+ virtual ~Call() {}
};
} // namespace webrtc
-#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
+#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
diff --git a/video_engine/test/common/direct_transport.cc b/video_engine/test/common/direct_transport.cc
index 7324401..f4f364b 100644
--- a/video_engine/test/common/direct_transport.cc
+++ b/video_engine/test/common/direct_transport.cc
@@ -10,7 +10,7 @@
#include "webrtc/video_engine/test/common/direct_transport.h"
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
namespace webrtc {
namespace test {
diff --git a/video_engine/test/engine_tests.cc b/video_engine/test/engine_tests.cc
index 3978c52..4ba659a 100644
--- a/video_engine/test/engine_tests.cc
+++ b/video_engine/test/engine_tests.cc
@@ -21,7 +21,7 @@
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
#include "webrtc/video_engine/test/common/direct_transport.h"
#include "webrtc/video_engine/test/common/fake_decoder.h"
#include "webrtc/video_engine/test/common/fake_encoder.h"
@@ -35,7 +35,8 @@
class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
public:
typedef std::map<uint32_t, int> BytesSentMap;
- StreamObserver(int num_expected_ssrcs, newapi::Transport* feedback_transport,
+ StreamObserver(int num_expected_ssrcs,
+ newapi::Transport* feedback_transport,
Clock* clock)
: critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
all_ssrcs_sent_(EventWrapper::Create()),
@@ -65,21 +66,19 @@
CriticalSectionScoped lock(critical_section_.get());
if (ssrcs.size() == num_expected_ssrcs_ && bitrate >= kExpectedBitrateBps)
all_ssrcs_sent_->Set();
- rtp_rtcp_->SetREMBData(bitrate, static_cast<uint8_t>(ssrcs.size()),
- &ssrcs[0]);
+ rtp_rtcp_->SetREMBData(
+ bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]);
rtp_rtcp_->Process();
}
virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
CriticalSectionScoped lock(critical_section_.get());
RTPHeader header;
- EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length),
- &header));
+ EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
receive_stats_->IncomingPacket(header, length, false);
rtp_rtcp_->SetRemoteSSRC(header.ssrc);
- remote_bitrate_estimator_->IncomingPacket(clock_->TimeInMilliseconds(),
- static_cast<int>(length - 12),
- header);
+ remote_bitrate_estimator_->IncomingPacket(
+ clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
remote_bitrate_estimator_->Process();
}
@@ -90,9 +89,7 @@
return true;
}
- EventTypeWrapper Wait() {
- return all_ssrcs_sent_->Wait(120 * 1000);
- }
+ EventTypeWrapper Wait() { return all_ssrcs_sent_->Wait(120 * 1000); }
private:
class TransportWrapper : public webrtc::Transport {
@@ -100,15 +97,18 @@
explicit TransportWrapper(newapi::Transport* new_transport)
: new_transport_(new_transport) {}
- virtual int SendPacket(int channel, const void *data, int len) OVERRIDE {
- return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len) ?
- len : -1;
+ virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
+ return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len)
+ ? len
+ : -1;
}
- virtual int SendRTCPPacket(int channel, const void *data,
+ virtual int SendRTCPPacket(int channel,
+ const void* data,
int len) OVERRIDE {
- return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len) ?
- len : -1;
+ return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len)
+ ? len
+ : -1;
}
private:
@@ -130,9 +130,7 @@
class RampUpTest : public ::testing::TestWithParam<bool> {
public:
- virtual void SetUp() {
- reserved_ssrcs_.clear();
- }
+ virtual void SetUp() { reserved_ssrcs_.clear(); }
protected:
std::map<uint32_t, bool> reserved_ssrcs_;
@@ -140,12 +138,11 @@
TEST_P(RampUpTest, RampUpWithPadding) {
test::DirectTransport receiver_transport;
- StreamObserver stream_observer(3, &receiver_transport,
- Clock::GetRealTimeClock());
- VideoCall::Config call_config(&stream_observer);
- scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
- VideoSendStream::Config send_config =
- call->GetDefaultSendConfig();
+ StreamObserver stream_observer(
+ 3, &receiver_transport, Clock::GetRealTimeClock());
+ Call::Config call_config(&stream_observer);
+ scoped_ptr<Call> call(Call::Create(call_config));
+ VideoSendStream::Config send_config = call->GetDefaultSendConfig();
receiver_transport.SetReceiver(call->Receiver());
@@ -157,22 +154,20 @@
test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
- VideoSendStream* send_stream =
- call->CreateSendStream(send_config);
+ VideoSendStream* send_stream = call->CreateSendStream(send_config);
VideoReceiveStream::Config receive_config;
receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
- receive_config.rtp.nack.rtp_history_ms =
- send_config.rtp.nack.rtp_history_ms;
- VideoReceiveStream* receive_stream = call->CreateReceiveStream(
- receive_config);
+ receive_config.rtp.nack.rtp_history_ms = send_config.rtp.nack.rtp_history_ms;
+ VideoReceiveStream* receive_stream =
+ call->CreateReceiveStream(receive_config);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream->Input(),
- test::FrameGenerator::Create(
- send_config.codec.width, send_config.codec.height,
- Clock::GetRealTimeClock()),
+ test::FrameGenerator::Create(send_config.codec.width,
+ send_config.codec.height,
+ Clock::GetRealTimeClock()),
30));
receive_stream->StartReceive();
@@ -213,10 +208,10 @@
protected:
void CreateCalls(newapi::Transport* sender_transport,
newapi::Transport* receiver_transport) {
- VideoCall::Config sender_config(sender_transport);
- VideoCall::Config receiver_config(receiver_transport);
- sender_call_.reset(VideoCall::Create(sender_config));
- receiver_call_.reset(VideoCall::Create(receiver_config));
+ Call::Config sender_config(sender_transport);
+ Call::Config receiver_config(receiver_transport);
+ sender_call_.reset(Call::Create(sender_config));
+ receiver_call_.reset(Call::Create(receiver_config));
}
void CreateTestConfigs() {
@@ -273,14 +268,14 @@
sender_call_->DestroySendStream(send_stream_);
if (receive_stream_ != NULL)
receiver_call_->DestroyReceiveStream(receive_stream_);
- send_stream_= NULL;
+ send_stream_ = NULL;
receive_stream_ = NULL;
}
void ReceivesPliAndRecovers(int rtp_history_ms);
- scoped_ptr<VideoCall> sender_call_;
- scoped_ptr<VideoCall> receiver_call_;
+ scoped_ptr<Call> sender_call_;
+ scoped_ptr<Call> receiver_call_;
VideoSendStream::Config send_config_;
VideoReceiveStream::Config receive_config_;
@@ -310,6 +305,7 @@
static const int kNumberOfNacksToObserve = 4;
static const int kInverseProbabilityToStartLossBurst = 20;
static const int kMaxLossBurst = 10;
+
public:
NackObserver()
: received_all_retransmissions_(EventWrapper::Create()),
@@ -444,9 +440,10 @@
class PliObserver : public test::RtpRtcpObserver {
static const int kInverseDropProbability = 16;
+
public:
- PliObserver(bool nack_enabled) :
- renderer_(this),
+ PliObserver(bool nack_enabled)
+ : renderer_(this),
rtp_header_parser_(RtpHeaderParser::Create()),
nack_enabled_(nack_enabled),
first_retransmitted_timestamp_(0),
@@ -572,9 +569,7 @@
explicit PacketInputObserver(PacketReceiver* receiver)
: receiver_(receiver), delivered_packet_(EventWrapper::Create()) {}
- EventTypeWrapper Wait() {
- return delivered_packet_->Wait(30 * 1000);
- }
+ EventTypeWrapper Wait() { return delivered_packet_->Wait(30 * 1000); }
private:
virtual bool DeliverPacket(const uint8_t* packet, size_t length) {
diff --git a/video_engine/test/full_stack.cc b/video_engine/test/full_stack.cc
index 8c9e755..365423f 100644
--- a/video_engine/test/full_stack.cc
+++ b/video_engine/test/full_stack.cc
@@ -23,7 +23,7 @@
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
#include "webrtc/video_engine/test/common/direct_transport.h"
#include "webrtc/video_engine/test/common/file_capturer.h"
#include "webrtc/video_engine/test/common/frame_generator_capturer.h"
@@ -47,14 +47,16 @@
double avg_ssim_threshold;
};
-FullStackTestParams paris_qcif = {"net_delay_0_0_plr_0",
- {"paris_qcif", 176, 144, 30}, 300, 36.0,
- 0.96};
+FullStackTestParams paris_qcif = {
+ "net_delay_0_0_plr_0", {"paris_qcif", 176, 144, 30}, 300, 36.0, 0.96};
// TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.
-FullStackTestParams foreman_cif = {"foreman_cif_net_delay_0_0_plr_0",
- {"foreman_cif", 352, 288, 30}, 700, 0.0,
- 0.0};
+FullStackTestParams foreman_cif = {
+ "foreman_cif_net_delay_0_0_plr_0",
+ {"foreman_cif", 352, 288, 30},
+ 700,
+ 0.0,
+ 0.0};
class FullStackTest : public ::testing::TestWithParam<FullStackTestParams> {
protected:
@@ -280,9 +282,9 @@
params.avg_ssim_threshold,
static_cast<uint64_t>(FLAGS_seconds * params.clip.fps));
- VideoCall::Config call_config(&analyzer);
+ Call::Config call_config(&analyzer);
- scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
+ scoped_ptr<Call> call(Call::Create(call_config));
analyzer.receiver_ = call->Receiver();
transport.SetReceiver(&analyzer);
@@ -314,8 +316,7 @@
test_clock),
params.clip.fps));
- VideoReceiveStream::Config receive_config =
- call->GetDefaultReceiveConfig();
+ VideoReceiveStream::Config receive_config = call->GetDefaultReceiveConfig();
receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
receive_config.renderer = &analyzer;
diff --git a/video_engine/test/loopback.cc b/video_engine/test/loopback.cc
index 19d99c1..93f2d83 100644
--- a/video_engine/test/loopback.cc
+++ b/video_engine/test/loopback.cc
@@ -17,7 +17,7 @@
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
#include "webrtc/video_engine/test/common/direct_transport.h"
#include "webrtc/video_engine/test/common/flags.h"
#include "webrtc/video_engine/test/common/generate_ssrcs.h"
@@ -40,9 +40,9 @@
"Loopback Video", test::flags::Width(), test::flags::Height()));
test::DirectTransport transport;
- VideoCall::Config call_config(&transport);
+ Call::Config call_config(&transport);
call_config.overuse_detection = true;
- scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
+ scoped_ptr<Call> call(Call::Create(call_config));
// Loopback, call sends to itself.
transport.SetReceiver(call->Receiver());
@@ -74,8 +74,7 @@
test::flags::Fps(),
test_clock));
- VideoReceiveStream::Config receive_config =
- call->GetDefaultReceiveConfig();
+ VideoReceiveStream::Config receive_config = call->GetDefaultReceiveConfig();
receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
receive_config.renderer = loopback_video.get();
diff --git a/video_engine/test/send_stream_tests.cc b/video_engine/test/send_stream_tests.cc
index 1e1ac27..dcf2ec1 100644
--- a/video_engine/test/send_stream_tests.cc
+++ b/video_engine/test/send_stream_tests.cc
@@ -16,7 +16,7 @@
#include "webrtc/video_engine/test/common/frame_generator.h"
#include "webrtc/video_engine/test/common/frame_generator_capturer.h"
#include "webrtc/video_engine/test/common/null_transport.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
#include "webrtc/video_engine/new_include/video_send_stream.h"
namespace webrtc {
@@ -28,9 +28,7 @@
send_test_complete_(EventWrapper::Create()),
timeout_ms_(timeout_ms) {}
- EventTypeWrapper Wait() {
- return send_test_complete_->Wait(timeout_ms_);
- }
+ EventTypeWrapper Wait() { return send_test_complete_->Wait(timeout_ms_); }
protected:
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
@@ -43,9 +41,10 @@
class VideoSendStreamTest : public ::testing::Test {
public:
VideoSendStreamTest() : fake_encoder_(Clock::GetRealTimeClock()) {}
+
protected:
static const uint32_t kSendSsrc;
- void RunSendTest(VideoCall* call,
+ void RunSendTest(Call* call,
const VideoSendStream::Config& config,
SendTransportObserver* observer) {
VideoSendStream* send_stream = call->CreateSendStream(config);
@@ -64,7 +63,7 @@
call->DestroySendStream(send_stream);
}
- VideoSendStream::Config GetSendTestConfig(VideoCall* call) {
+ VideoSendStream::Config GetSendTestConfig(Call* call) {
VideoSendStream::Config config = call->GetDefaultSendConfig();
config.encoder = &fake_encoder_;
config.internal_source = false;
@@ -94,8 +93,8 @@
}
} observer;
- VideoCall::Config call_config(&observer);
- scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
+ Call::Config call_config(&observer);
+ scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.ssrcs.push_back(kSendSsrc);
@@ -127,8 +126,8 @@
}
} observer;
- VideoCall::Config call_config(&observer);
- scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
+ Call::Config call_config(&observer);
+ scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.ssrcs.push_back(kSendSsrc);
diff --git a/video_engine/video_engine_core.gypi b/video_engine/video_engine_core.gypi
index 29849fa..5c7876a 100644
--- a/video_engine/video_engine_core.gypi
+++ b/video_engine/video_engine_core.gypi
@@ -115,16 +115,16 @@
'vie_sync_module.cc',
# New VideoEngine API
- 'internal/video_call.cc',
- 'internal/video_call.h',
+ 'internal/call.cc',
+ 'internal/call.h',
'internal/video_receive_stream.cc',
'internal/video_receive_stream.h',
'internal/video_send_stream.cc',
'internal/video_send_stream.h',
+ 'new_include/call.h',
'new_include/config.h',
'new_include/frame_callback.h',
'new_include/transport.h',
- 'new_include/video_call.h',
'new_include/video_receive_stream.h',
'new_include/video_renderer.h',
'new_include/video_send_stream.h',