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Use make_shared where possible

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This commit is contained in:
Paul-Louis Ageneau
2021-04-20 19:01:41 +02:00
parent ce9fa374d0
commit cd68d1dba7
3 changed files with 7 additions and 7 deletions
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10
examples/streamer/main.cpp
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@ -218,11 +218,11 @@ shared_ptr<ClientTrackData> addVideo(const shared_ptr<PeerConnection> pc, const
video.addSSRC(ssrc, cname, msid, cname);
auto track = pc->addTrack(video);
// create RTP configuration
auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate));
auto rtpConfig = make_shared<RtpPacketizationConfig>(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate);
// create packetizer
auto packetizer = shared_ptr<H264RtpPacketizer>(new H264RtpPacketizer(H264RtpPacketizer::Separator::Length, rtpConfig));
auto packetizer = make_shared<H264RtpPacketizer>(H264RtpPacketizer::Separator::Length, rtpConfig);
// create H264 handler
shared_ptr<H264PacketizationHandler> h264Handler(new H264PacketizationHandler(packetizer));
auto h264Handler = make_shared<H264PacketizationHandler>(packetizer);
// add RTCP SR handler
auto srReporter = make_shared<RtcpSrReporter>(rtpConfig);
h264Handler->addToChain(srReporter);
@ -242,7 +242,7 @@ shared_ptr<ClientTrackData> addAudio(const shared_ptr<PeerConnection> pc, const
audio.addSSRC(ssrc, cname, msid, cname);
auto track = pc->addTrack(audio);
// create RTP configuration
auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate));
auto rtpConfig = make_shared<RtpPacketizationConfig>(ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate);
// create packetizer
auto packetizer = make_shared<OpusRtpPacketizer>(rtpConfig);
// create opus handler
@ -265,7 +265,7 @@ shared_ptr<Client> createPeerConnection(const Configuration &config,
weak_ptr<WebSocket> wws,
string id) {
auto pc = make_shared<PeerConnection>(config);
shared_ptr<Client> client(new Client(pc));
auto client = make_shared<Client>(pc);
pc->onStateChange([id](PeerConnection::State state) {
cout << "State: " << state << endl;

2
include/rtc/h264rtppacketizer.hpp
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@ -34,7 +34,7 @@ class RTC_CPP_EXPORT H264RtpPacketizer final : public RtpPacketizer, public Medi
public:
/// Default clock rate for H264 in RTP
static const auto defaultClockRate = 90 * 1000;
inline static const uint32_t defaultClockRate = 90 * 1000;
/// Nalunit separator
enum class Separator {

2
include/rtc/opusrtppacketizer.hpp
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@ -30,7 +30,7 @@ namespace rtc {
class RTC_CPP_EXPORT OpusRtpPacketizer final : public RtpPacketizer, public MediaHandlerRootElement {
public:
/// default clock rate used in opus RTP communication
static const uint32_t defaultClockRate = 48 * 1000;
inline static const uint32_t defaultClockRate = 48 * 1000;
/// Constructs opus packetizer with given RTP configuration.
/// @note RTP configuration is used in packetization process which may change some configuration