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Rename Call::Create{Receive,Send}Stream().

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Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-11-20 10:40:25 +00:00
parent 0b72f5863b
commit 5a63655ab0
7 changed files with 19 additions and 17 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

9
webrtc/call.cc
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@ -38,14 +38,14 @@ class Call : public webrtc::Call, public PacketReceiver {
virtual VideoSendStream::Config GetDefaultSendConfig() OVERRIDE;
virtual VideoSendStream* CreateSendStream(
virtual VideoSendStream* CreateVideoSendStream(
const VideoSendStream::Config& config) OVERRIDE;
virtual void DestroySendStream(webrtc::VideoSendStream* send_stream) OVERRIDE;
virtual VideoReceiveStream::Config GetDefaultReceiveConfig() OVERRIDE;
virtual VideoReceiveStream* CreateReceiveStream(
virtual VideoReceiveStream* CreateVideoReceiveStream(
const VideoReceiveStream::Config& config) OVERRIDE;
virtual void DestroyReceiveStream(
@ -211,7 +211,8 @@ VideoSendStream::Config Call::GetDefaultSendConfig() {
return config;
}
VideoSendStream* Call::CreateSendStream(const VideoSendStream::Config& config) {
VideoSendStream* Call::CreateVideoSendStream(
const VideoSendStream::Config& config) {
assert(config.rtp.ssrcs.size() > 0);
assert(config.codec.numberOfSimulcastStreams == 0 ||
config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size());
@ -253,7 +254,7 @@ VideoReceiveStream::Config Call::GetDefaultReceiveConfig() {
return VideoReceiveStream::Config();
}
VideoReceiveStream* Call::CreateReceiveStream(
VideoReceiveStream* Call::CreateVideoReceiveStream(
const VideoReceiveStream::Config& config) {
VideoReceiveStream* receive_stream = new VideoReceiveStream(
video_engine_, config, config_.send_transport, config_.voice_engine);

4
webrtc/call.h
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@ -60,7 +60,7 @@ class Call {
virtual VideoSendStream::Config GetDefaultSendConfig() = 0;
virtual VideoSendStream* CreateSendStream(
virtual VideoSendStream* CreateVideoSendStream(
const VideoSendStream::Config& config) = 0;
// Returns the internal state of the send stream, for resume sending with a
@ -70,7 +70,7 @@ class Call {
virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;
virtual VideoReceiveStream* CreateReceiveStream(
virtual VideoReceiveStream* CreateVideoReceiveStream(
const VideoReceiveStream::Config& config) = 0;
virtual void DestroyReceiveStream(VideoReceiveStream* receive_stream) = 0;

9
webrtc/call_tests.cc
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@ -88,8 +88,8 @@ class CallTest : public ::testing::Test {
assert(send_stream_ == NULL);
assert(receive_stream_ == NULL);
send_stream_ = sender_call_->CreateSendStream(send_config_);
receive_stream_ = receiver_call_->CreateReceiveStream(receive_config_);
send_stream_ = sender_call_->CreateVideoSendStream(send_config_);
receive_stream_ = receiver_call_->CreateVideoReceiveStream(receive_config_);
}
void CreateFrameGenerator() {
@ -787,14 +787,15 @@ TEST_F(CallTest, SendsAndReceivesMultipleStreams) {
receiver_call->GetDefaultReceiveConfig();
receive_config.renderer = observers[i];
receive_config.rtp.ssrc = ssrc;
receive_streams[i] = receiver_call->CreateReceiveStream(receive_config);
receive_streams[i] =
receiver_call->CreateVideoReceiveStream(receive_config);
receive_streams[i]->StartReceive();
VideoSendStream::Config send_config = sender_call->GetDefaultSendConfig();
send_config.rtp.ssrcs.push_back(ssrc);
send_config.codec.width = width;
send_config.codec.height = height;
send_streams[i] = sender_call->CreateSendStream(send_config);
send_streams[i] = sender_call->CreateVideoSendStream(send_config);
send_streams[i]->StartSend();
frame_generators[i] = test::FrameGeneratorCapturer::Create(

4
webrtc/loopback.cc
View File

@ -63,7 +63,7 @@ TEST_F(LoopbackTest, Test) {
send_config.codec.maxBitrate =
static_cast<unsigned int>(test::flags::MaxBitrate());
VideoSendStream* send_stream = call->CreateSendStream(send_config);
VideoSendStream* send_stream = call->CreateVideoSendStream(send_config);
Clock* test_clock = Clock::GetRealTimeClock();
@ -79,7 +79,7 @@ TEST_F(LoopbackTest, Test) {
receive_config.renderer = loopback_video.get();
VideoReceiveStream* receive_stream =
call->CreateReceiveStream(receive_config);
call->CreateVideoReceiveStream(receive_config);
receive_stream->StartReceive();
send_stream->StartSend();

4
webrtc/video/full_stack.cc
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@ -402,7 +402,7 @@ TEST_P(FullStackTest, NoPacketLoss) {
send_config.codec.startBitrate = params.bitrate;
send_config.codec.maxBitrate = params.bitrate;
VideoSendStream* send_stream = call->CreateSendStream(send_config);
VideoSendStream* send_stream = call->CreateVideoSendStream(send_config);
analyzer.input_ = send_stream->Input();
scoped_ptr<test::FrameGeneratorCapturer> file_capturer(
@ -422,7 +422,7 @@ TEST_P(FullStackTest, NoPacketLoss) {
receive_config.renderer = &analyzer;
VideoReceiveStream* receive_stream =
call->CreateReceiveStream(receive_config);
call->CreateVideoReceiveStream(receive_config);
receive_stream->StartReceive();
send_stream->StartSend();

4
webrtc/video/rampup_tests.cc
View File

@ -159,13 +159,13 @@ TEST_P(RampUpTest, RampUpWithPadding) {
test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
VideoSendStream* send_stream = call->CreateSendStream(send_config);
VideoSendStream* send_stream = call->CreateVideoSendStream(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);
call->CreateVideoReceiveStream(receive_config);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(send_stream->Input(),

2
webrtc/video/video_send_stream_tests.cc
View File

@ -40,7 +40,7 @@ class VideoSendStreamTest : public ::testing::Test {
void RunSendTest(Call* call,
const VideoSendStream::Config& config,
test::RtpRtcpObserver* observer) {
send_stream_ = call->CreateSendStream(config);
send_stream_ = call->CreateVideoSendStream(config);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));