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@4704 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
parent
86136a0e8f
commit
841c8a44bb
@ -8,7 +8,7 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/internal/video_call.h"
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#include "webrtc/video_engine/internal/call.h"
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#include <assert.h>
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#include <string.h>
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@ -24,17 +24,16 @@
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namespace webrtc {
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VideoCall* VideoCall::Create(const VideoCall::Config& config) {
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webrtc::VideoEngine* video_engine = webrtc::VideoEngine::Create();
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Call* Call::Create(const Call::Config& config) {
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VideoEngine* video_engine = VideoEngine::Create();
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assert(video_engine != NULL);
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return new internal::VideoCall(video_engine, config);
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return new internal::Call(video_engine, config);
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}
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namespace internal {
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VideoCall::VideoCall(webrtc::VideoEngine* video_engine,
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const VideoCall::Config& config)
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Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config)
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: config_(config),
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receive_lock_(RWLockWrapper::CreateRWLock()),
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send_lock_(RWLockWrapper::CreateRWLock()),
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@ -50,15 +49,15 @@ VideoCall::VideoCall(webrtc::VideoEngine* video_engine,
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assert(codec_ != NULL);
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}
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VideoCall::~VideoCall() {
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Call::~Call() {
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codec_->Release();
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rtp_rtcp_->Release();
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webrtc::VideoEngine::Delete(video_engine_);
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}
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PacketReceiver* VideoCall::Receiver() { return this; }
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PacketReceiver* Call::Receiver() { return this; }
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std::vector<VideoCodec> VideoCall::GetVideoCodecs() {
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std::vector<VideoCodec> Call::GetVideoCodecs() {
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std::vector<VideoCodec> codecs;
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VideoCodec codec;
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@ -70,14 +69,13 @@ std::vector<VideoCodec> VideoCall::GetVideoCodecs() {
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return codecs;
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}
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VideoSendStream::Config VideoCall::GetDefaultSendConfig() {
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VideoSendStream::Config Call::GetDefaultSendConfig() {
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VideoSendStream::Config config;
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codec_->GetCodec(0, config.codec);
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return config;
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}
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VideoSendStream* VideoCall::CreateSendStream(
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const VideoSendStream::Config& config) {
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VideoSendStream* Call::CreateSendStream(const VideoSendStream::Config& config) {
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assert(config.rtp.ssrcs.size() > 0);
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assert(config.codec.numberOfSimulcastStreams == 0 ||
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config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size());
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@ -93,8 +91,7 @@ VideoSendStream* VideoCall::CreateSendStream(
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return send_stream;
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}
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SendStreamState* VideoCall::DestroySendStream(
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webrtc::VideoSendStream* send_stream) {
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SendStreamState* Call::DestroySendStream(webrtc::VideoSendStream* send_stream) {
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assert(send_stream != NULL);
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VideoSendStream* send_stream_impl = NULL;
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@ -119,11 +116,11 @@ SendStreamState* VideoCall::DestroySendStream(
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return NULL;
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}
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VideoReceiveStream::Config VideoCall::GetDefaultReceiveConfig() {
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VideoReceiveStream::Config Call::GetDefaultReceiveConfig() {
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return VideoReceiveStream::Config();
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}
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VideoReceiveStream* VideoCall::CreateReceiveStream(
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VideoReceiveStream* Call::CreateReceiveStream(
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const VideoReceiveStream::Config& config) {
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VideoReceiveStream* receive_stream =
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new VideoReceiveStream(video_engine_, config, config_.send_transport);
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@ -134,8 +131,7 @@ VideoReceiveStream* VideoCall::CreateReceiveStream(
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return receive_stream;
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}
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void VideoCall::DestroyReceiveStream(
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webrtc::VideoReceiveStream* receive_stream) {
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void Call::DestroyReceiveStream(webrtc::VideoReceiveStream* receive_stream) {
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assert(receive_stream != NULL);
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VideoReceiveStream* receive_stream_impl = NULL;
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@ -157,17 +153,17 @@ void VideoCall::DestroyReceiveStream(
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delete receive_stream_impl;
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}
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uint32_t VideoCall::SendBitrateEstimate() {
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uint32_t Call::SendBitrateEstimate() {
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// TODO(pbos): Return send-bitrate estimate
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return 0;
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}
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uint32_t VideoCall::ReceiveBitrateEstimate() {
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uint32_t Call::ReceiveBitrateEstimate() {
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// TODO(pbos): Return receive-bitrate estimate
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return 0;
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}
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bool VideoCall::DeliverRtcp(const uint8_t* packet, size_t length) {
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bool Call::DeliverRtcp(const uint8_t* packet, size_t length) {
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// TODO(pbos): Figure out what channel needs it actually.
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// Do NOT broadcast! Also make sure it's a valid packet.
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bool rtcp_delivered = false;
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@ -199,7 +195,7 @@ bool VideoCall::DeliverRtcp(const uint8_t* packet, size_t length) {
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return rtcp_delivered;
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}
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bool VideoCall::DeliverRtp(const RTPHeader& header,
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bool Call::DeliverRtp(const RTPHeader& header,
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const uint8_t* packet,
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size_t length) {
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VideoReceiveStream* receiver;
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@ -217,7 +213,7 @@ bool VideoCall::DeliverRtp(const RTPHeader& header,
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return receiver->DeliverRtp(static_cast<const uint8_t*>(packet), length);
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}
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bool VideoCall::DeliverPacket(const uint8_t* packet, size_t length) {
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bool Call::DeliverPacket(const uint8_t* packet, size_t length) {
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// TODO(pbos): ExtensionMap if there are extensions.
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if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)))
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return DeliverRtcp(packet, length);
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@ -7,8 +7,8 @@
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_ENGINE_VIDEO_CALL_IMPL_H_
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#define WEBRTC_VIDEO_ENGINE_VIDEO_CALL_IMPL_H_
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#ifndef WEBRTC_VIDEO_ENGINE_CALL_IMPL_H_
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#define WEBRTC_VIDEO_ENGINE_CALL_IMPL_H_
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#include <map>
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#include <vector>
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@ -18,7 +18,7 @@
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/video_engine/internal/video_receive_stream.h"
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#include "webrtc/video_engine/internal/video_send_stream.h"
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#include "webrtc/video_engine/new_include/video_call.h"
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#include "webrtc/video_engine/new_include/call.h"
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namespace webrtc {
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@ -30,11 +30,10 @@ namespace internal {
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// TODO(pbos): Split out the packet receiver, should be sharable between
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// VideoEngine and VoiceEngine.
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class VideoCall : public webrtc::VideoCall, public PacketReceiver {
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class Call : public webrtc::Call, public PacketReceiver {
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public:
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VideoCall(webrtc::VideoEngine* video_engine,
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const VideoCall::Config& config);
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virtual ~VideoCall();
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Call(webrtc::VideoEngine* video_engine, const Call::Config& config);
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virtual ~Call();
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virtual PacketReceiver* Receiver() OVERRIDE;
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virtual std::vector<VideoCodec> GetVideoCodecs() OVERRIDE;
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@ -52,8 +51,8 @@ class VideoCall : public webrtc::VideoCall, public PacketReceiver {
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virtual VideoReceiveStream* CreateReceiveStream(
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const VideoReceiveStream::Config& config) OVERRIDE;
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virtual void DestroyReceiveStream(webrtc::VideoReceiveStream* receive_stream)
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OVERRIDE;
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virtual void DestroyReceiveStream(
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webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
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virtual uint32_t SendBitrateEstimate() OVERRIDE;
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virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
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@ -66,7 +65,7 @@ class VideoCall : public webrtc::VideoCall, public PacketReceiver {
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const uint8_t* packet,
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size_t length);
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VideoCall::Config config_;
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Call::Config config_;
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std::map<uint32_t, VideoReceiveStream*> receive_ssrcs_;
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scoped_ptr<RWLockWrapper> receive_lock_;
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@ -80,9 +79,9 @@ class VideoCall : public webrtc::VideoCall, public PacketReceiver {
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ViERTP_RTCP* rtp_rtcp_;
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ViECodec* codec_;
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DISALLOW_COPY_AND_ASSIGN(VideoCall);
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DISALLOW_COPY_AND_ASSIGN(Call);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_INTERNAL_VIDEO_CALL_H_
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#endif // WEBRTC_VIDEO_ENGINE_INTERNAL_CALL_H_
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@ -7,8 +7,8 @@
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
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#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
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#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
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#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
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#include <string>
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#include <vector>
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@ -31,10 +31,10 @@ class PacketReceiver {
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virtual ~PacketReceiver() {}
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};
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// A VideoCall instance can contain several send and/or receive streams. All
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// streams are assumed to have the same remote endpoint and will share bitrate
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// estimates etc.
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class VideoCall {
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// A Call instance can contain several send and/or receive streams. All streams
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// are assumed to have the same remote endpoint and will share bitrate estimates
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// etc.
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class Call {
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public:
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struct Config {
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explicit Config(newapi::Transport* send_transport)
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@ -47,14 +47,14 @@ class VideoCall {
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newapi::Transport* send_transport;
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bool overuse_detection;
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// VoiceEngine used for audio/video synchronization for this VideoCall.
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// VoiceEngine used for audio/video synchronization for this Call.
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VoiceEngine* voice_engine;
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TraceCallback* trace_callback;
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uint32_t trace_filter;
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};
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static VideoCall* Create(const VideoCall::Config& config);
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static Call* Create(const Call::Config& config);
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virtual std::vector<VideoCodec> GetVideoCodecs() = 0;
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@ -76,7 +76,7 @@ class VideoCall {
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// All received RTP and RTCP packets for the call should be inserted to this
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// PacketReceiver. The PacketReceiver pointer is valid as long as the
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// VideoCall instance exists.
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// Call instance exists.
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virtual PacketReceiver* Receiver() = 0;
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// Returns the estimated total send bandwidth. Note: this can differ from the
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@ -87,8 +87,8 @@ class VideoCall {
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// differ from the actual receive bitrate.
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virtual uint32_t ReceiveBitrateEstimate() = 0;
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virtual ~VideoCall() {}
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virtual ~Call() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
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#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
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@ -10,7 +10,7 @@
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#include "webrtc/video_engine/test/common/direct_transport.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/video_engine/new_include/video_call.h"
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#include "webrtc/video_engine/new_include/call.h"
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namespace webrtc {
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namespace test {
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@ -21,7 +21,7 @@
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/event_wrapper.h"
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#include "webrtc/video_engine/new_include/video_call.h"
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#include "webrtc/video_engine/new_include/call.h"
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#include "webrtc/video_engine/test/common/direct_transport.h"
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#include "webrtc/video_engine/test/common/fake_decoder.h"
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#include "webrtc/video_engine/test/common/fake_encoder.h"
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@ -35,7 +35,8 @@ namespace webrtc {
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class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
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public:
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typedef std::map<uint32_t, int> BytesSentMap;
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StreamObserver(int num_expected_ssrcs, newapi::Transport* feedback_transport,
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StreamObserver(int num_expected_ssrcs,
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newapi::Transport* feedback_transport,
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Clock* clock)
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: critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
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all_ssrcs_sent_(EventWrapper::Create()),
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@ -65,21 +66,19 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
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CriticalSectionScoped lock(critical_section_.get());
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if (ssrcs.size() == num_expected_ssrcs_ && bitrate >= kExpectedBitrateBps)
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all_ssrcs_sent_->Set();
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rtp_rtcp_->SetREMBData(bitrate, static_cast<uint8_t>(ssrcs.size()),
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&ssrcs[0]);
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rtp_rtcp_->SetREMBData(
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bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]);
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rtp_rtcp_->Process();
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}
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virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
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CriticalSectionScoped lock(critical_section_.get());
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RTPHeader header;
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EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length),
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&header));
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EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
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receive_stats_->IncomingPacket(header, length, false);
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rtp_rtcp_->SetRemoteSSRC(header.ssrc);
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remote_bitrate_estimator_->IncomingPacket(clock_->TimeInMilliseconds(),
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static_cast<int>(length - 12),
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header);
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remote_bitrate_estimator_->IncomingPacket(
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clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
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if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
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remote_bitrate_estimator_->Process();
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}
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@ -90,9 +89,7 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
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return true;
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}
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EventTypeWrapper Wait() {
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return all_ssrcs_sent_->Wait(120 * 1000);
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}
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EventTypeWrapper Wait() { return all_ssrcs_sent_->Wait(120 * 1000); }
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private:
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class TransportWrapper : public webrtc::Transport {
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@ -101,14 +98,17 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
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: new_transport_(new_transport) {}
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virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
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return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len) ?
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len : -1;
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return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len)
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? len
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: -1;
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}
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virtual int SendRTCPPacket(int channel, const void *data,
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virtual int SendRTCPPacket(int channel,
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const void* data,
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int len) OVERRIDE {
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return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len) ?
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len : -1;
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return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len)
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? len
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: -1;
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}
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private:
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@ -130,9 +130,7 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
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class RampUpTest : public ::testing::TestWithParam<bool> {
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public:
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virtual void SetUp() {
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reserved_ssrcs_.clear();
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}
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virtual void SetUp() { reserved_ssrcs_.clear(); }
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protected:
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std::map<uint32_t, bool> reserved_ssrcs_;
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@ -140,12 +138,11 @@ class RampUpTest : public ::testing::TestWithParam<bool> {
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TEST_P(RampUpTest, RampUpWithPadding) {
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test::DirectTransport receiver_transport;
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StreamObserver stream_observer(3, &receiver_transport,
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Clock::GetRealTimeClock());
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VideoCall::Config call_config(&stream_observer);
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scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
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VideoSendStream::Config send_config =
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call->GetDefaultSendConfig();
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StreamObserver stream_observer(
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3, &receiver_transport, Clock::GetRealTimeClock());
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Call::Config call_config(&stream_observer);
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scoped_ptr<Call> call(Call::Create(call_config));
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VideoSendStream::Config send_config = call->GetDefaultSendConfig();
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receiver_transport.SetReceiver(call->Receiver());
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@ -157,21 +154,19 @@ TEST_P(RampUpTest, RampUpWithPadding) {
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test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
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VideoSendStream* send_stream =
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call->CreateSendStream(send_config);
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VideoSendStream* send_stream = call->CreateSendStream(send_config);
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VideoReceiveStream::Config receive_config;
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receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
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receive_config.rtp.nack.rtp_history_ms =
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send_config.rtp.nack.rtp_history_ms;
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VideoReceiveStream* receive_stream = call->CreateReceiveStream(
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receive_config);
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receive_config.rtp.nack.rtp_history_ms = send_config.rtp.nack.rtp_history_ms;
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VideoReceiveStream* receive_stream =
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call->CreateReceiveStream(receive_config);
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scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
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test::FrameGeneratorCapturer::Create(
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send_stream->Input(),
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test::FrameGenerator::Create(
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send_config.codec.width, send_config.codec.height,
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test::FrameGenerator::Create(send_config.codec.width,
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send_config.codec.height,
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Clock::GetRealTimeClock()),
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30));
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@ -213,10 +208,10 @@ class EngineTest : public ::testing::TestWithParam<EngineTestParams> {
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protected:
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void CreateCalls(newapi::Transport* sender_transport,
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newapi::Transport* receiver_transport) {
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VideoCall::Config sender_config(sender_transport);
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VideoCall::Config receiver_config(receiver_transport);
|
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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() {
|
||||
@ -279,8 +274,8 @@ class EngineTest : public ::testing::TestWithParam<EngineTestParams> {
|
||||
|
||||
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 @@ class NackObserver : public test::RtpRtcpObserver {
|
||||
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 @@ TEST_P(EngineTest, ReceivesAndRetransmitsNack) {
|
||||
|
||||
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 @@ TEST_P(EngineTest, SurvivesIncomingRtpPacketsToDestroyedReceiveStream) {
|
||||
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) {
|
||||
|
||||
@ -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"
|
||||
@ -47,13 +47,15 @@ struct FullStackTestParams {
|
||||
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,
|
||||
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> {
|
||||
@ -280,9 +282,9 @@ TEST_P(FullStackTest, DISABLED_NoPacketLoss) {
|
||||
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_P(FullStackTest, DISABLED_NoPacketLoss) {
|
||||
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;
|
||||
|
||||
|
||||
@ -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 @@ TEST_F(LoopbackTest, Test) {
|
||||
"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_F(LoopbackTest, Test) {
|
||||
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();
|
||||
|
||||
|
||||
@ -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 @@ class SendTransportObserver : public test::NullTransport {
|
||||
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 SendTransportObserver : public test::NullTransport {
|
||||
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 @@ class VideoSendStreamTest : public ::testing::Test {
|
||||
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 @@ TEST_F(VideoSendStreamTest, SendsSetSsrc) {
|
||||
}
|
||||
} 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 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
|
||||
}
|
||||
} 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);
|
||||
|
||||
@ -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',
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user