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Side by Side Diff: webrtc/media/base/mediachannel.h

Issue 1785713005: Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Feedback from tommi. Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
13 13
14 #include <memory> 14 #include <memory>
15 #include <string> 15 #include <string>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/basictypes.h" 18 #include "webrtc/base/basictypes.h"
19 #include "webrtc/base/buffer.h" 19 #include "webrtc/base/copyonwritebuffer.h"
20 #include "webrtc/base/dscp.h" 20 #include "webrtc/base/dscp.h"
21 #include "webrtc/base/logging.h" 21 #include "webrtc/base/logging.h"
22 #include "webrtc/base/optional.h" 22 #include "webrtc/base/optional.h"
23 #include "webrtc/base/sigslot.h" 23 #include "webrtc/base/sigslot.h"
24 #include "webrtc/base/socket.h" 24 #include "webrtc/base/socket.h"
25 #include "webrtc/base/window.h" 25 #include "webrtc/base/window.h"
26 #include "webrtc/media/base/codec.h" 26 #include "webrtc/media/base/codec.h"
27 #include "webrtc/media/base/mediaconstants.h" 27 #include "webrtc/media/base/mediaconstants.h"
28 #include "webrtc/media/base/streamparams.h" 28 #include "webrtc/media/base/streamparams.h"
29 #include "webrtc/media/base/videosinkinterface.h" 29 #include "webrtc/media/base/videosinkinterface.h"
(...skipping 307 matching lines...) Expand 10 before | Expand all | Expand 10 after
337 return &(*it); 337 return &(*it);
338 } 338 }
339 return NULL; 339 return NULL;
340 } 340 }
341 341
342 class MediaChannel : public sigslot::has_slots<> { 342 class MediaChannel : public sigslot::has_slots<> {
343 public: 343 public:
344 class NetworkInterface { 344 class NetworkInterface {
345 public: 345 public:
346 enum SocketType { ST_RTP, ST_RTCP }; 346 enum SocketType { ST_RTP, ST_RTCP };
347 virtual bool SendPacket(rtc::Buffer* packet, 347 virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
348 const rtc::PacketOptions& options) = 0; 348 const rtc::PacketOptions& options) = 0;
349 virtual bool SendRtcp(rtc::Buffer* packet, 349 virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
350 const rtc::PacketOptions& options) = 0; 350 const rtc::PacketOptions& options) = 0;
351 virtual int SetOption(SocketType type, rtc::Socket::Option opt, 351 virtual int SetOption(SocketType type, rtc::Socket::Option opt,
352 int option) = 0; 352 int option) = 0;
353 virtual ~NetworkInterface() {} 353 virtual ~NetworkInterface() {}
354 }; 354 };
355 355
356 MediaChannel(const MediaConfig& config) 356 MediaChannel(const MediaConfig& config)
357 : enable_dscp_(config.enable_dscp), network_interface_(NULL) {} 357 : enable_dscp_(config.enable_dscp), network_interface_(NULL) {}
358 MediaChannel() : enable_dscp_(false), network_interface_(NULL) {} 358 MediaChannel() : enable_dscp_(false), network_interface_(NULL) {}
359 virtual ~MediaChannel() {} 359 virtual ~MediaChannel() {}
360 360
361 // Sets the abstract interface class for sending RTP/RTCP data. 361 // Sets the abstract interface class for sending RTP/RTCP data.
362 virtual void SetInterface(NetworkInterface *iface) { 362 virtual void SetInterface(NetworkInterface *iface) {
363 rtc::CritScope cs(&network_interface_crit_); 363 rtc::CritScope cs(&network_interface_crit_);
364 network_interface_ = iface; 364 network_interface_ = iface;
365 SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT); 365 SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
366 } 366 }
367 virtual rtc::DiffServCodePoint PreferredDscp() const { 367 virtual rtc::DiffServCodePoint PreferredDscp() const {
368 return rtc::DSCP_DEFAULT; 368 return rtc::DSCP_DEFAULT;
369 } 369 }
370 // Called when a RTP packet is received. 370 // Called when a RTP packet is received.
371 virtual void OnPacketReceived(rtc::Buffer* packet, 371 virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
372 const rtc::PacketTime& packet_time) = 0; 372 const rtc::PacketTime& packet_time) = 0;
373 // Called when a RTCP packet is received. 373 // Called when a RTCP packet is received.
374 virtual void OnRtcpReceived(rtc::Buffer* packet, 374 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
375 const rtc::PacketTime& packet_time) = 0; 375 const rtc::PacketTime& packet_time) = 0;
376 // Called when the socket's ability to send has changed. 376 // Called when the socket's ability to send has changed.
377 virtual void OnReadyToSend(bool ready) = 0; 377 virtual void OnReadyToSend(bool ready) = 0;
378 // Creates a new outgoing media stream with SSRCs and CNAME as described 378 // Creates a new outgoing media stream with SSRCs and CNAME as described
379 // by sp. 379 // by sp.
380 virtual bool AddSendStream(const StreamParams& sp) = 0; 380 virtual bool AddSendStream(const StreamParams& sp) = 0;
381 // Removes an outgoing media stream. 381 // Removes an outgoing media stream.
382 // ssrc must be the first SSRC of the media stream if the stream uses 382 // ssrc must be the first SSRC of the media stream if the stream uses
383 // multiple SSRCs. 383 // multiple SSRCs.
384 virtual bool RemoveSendStream(uint32_t ssrc) = 0; 384 virtual bool RemoveSendStream(uint32_t ssrc) = 0;
385 // Creates a new incoming media stream with SSRCs and CNAME as described 385 // Creates a new incoming media stream with SSRCs and CNAME as described
386 // by sp. 386 // by sp.
387 virtual bool AddRecvStream(const StreamParams& sp) = 0; 387 virtual bool AddRecvStream(const StreamParams& sp) = 0;
388 // Removes an incoming media stream. 388 // Removes an incoming media stream.
389 // ssrc must be the first SSRC of the media stream if the stream uses 389 // ssrc must be the first SSRC of the media stream if the stream uses
390 // multiple SSRCs. 390 // multiple SSRCs.
391 virtual bool RemoveRecvStream(uint32_t ssrc) = 0; 391 virtual bool RemoveRecvStream(uint32_t ssrc) = 0;
392 392
393 // Returns the absoulte sendtime extension id value from media channel. 393 // Returns the absoulte sendtime extension id value from media channel.
394 virtual int GetRtpSendTimeExtnId() const { 394 virtual int GetRtpSendTimeExtnId() const {
395 return -1; 395 return -1;
396 } 396 }
397 397
398 // Base method to send packet using NetworkInterface. 398 // Base method to send packet using NetworkInterface.
399 bool SendPacket(rtc::Buffer* packet, const rtc::PacketOptions& options) { 399 bool SendPacket(rtc::CopyOnWriteBuffer* packet,
400 const rtc::PacketOptions& options) {
400 return DoSendPacket(packet, false, options); 401 return DoSendPacket(packet, false, options);
401 } 402 }
402 403
403 bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) { 404 bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
405 const rtc::PacketOptions& options) {
404 return DoSendPacket(packet, true, options); 406 return DoSendPacket(packet, true, options);
405 } 407 }
406 408
407 int SetOption(NetworkInterface::SocketType type, 409 int SetOption(NetworkInterface::SocketType type,
408 rtc::Socket::Option opt, 410 rtc::Socket::Option opt,
409 int option) { 411 int option) {
410 rtc::CritScope cs(&network_interface_crit_); 412 rtc::CritScope cs(&network_interface_crit_);
411 if (!network_interface_) 413 if (!network_interface_)
412 return -1; 414 return -1;
413 415
414 return network_interface_->SetOption(type, opt, option); 416 return network_interface_->SetOption(type, opt, option);
415 } 417 }
416 418
417 private: 419 private:
418 // This method sets DSCP |value| on both RTP and RTCP channels. 420 // This method sets DSCP |value| on both RTP and RTCP channels.
419 int SetDscp(rtc::DiffServCodePoint value) { 421 int SetDscp(rtc::DiffServCodePoint value) {
420 int ret; 422 int ret;
421 ret = SetOption(NetworkInterface::ST_RTP, 423 ret = SetOption(NetworkInterface::ST_RTP,
422 rtc::Socket::OPT_DSCP, 424 rtc::Socket::OPT_DSCP,
423 value); 425 value);
424 if (ret == 0) { 426 if (ret == 0) {
425 ret = SetOption(NetworkInterface::ST_RTCP, 427 ret = SetOption(NetworkInterface::ST_RTCP,
426 rtc::Socket::OPT_DSCP, 428 rtc::Socket::OPT_DSCP,
427 value); 429 value);
428 } 430 }
429 return ret; 431 return ret;
430 } 432 }
431 433
432 bool DoSendPacket(rtc::Buffer* packet, 434 bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
433 bool rtcp, 435 bool rtcp,
434 const rtc::PacketOptions& options) { 436 const rtc::PacketOptions& options) {
435 rtc::CritScope cs(&network_interface_crit_); 437 rtc::CritScope cs(&network_interface_crit_);
436 if (!network_interface_) 438 if (!network_interface_)
437 return false; 439 return false;
438 440
439 return (!rtcp) ? network_interface_->SendPacket(packet, options) 441 return (!rtcp) ? network_interface_->SendPacket(packet, options)
440 : network_interface_->SendRtcp(packet, options); 442 : network_interface_->SendRtcp(packet, options);
441 } 443 }
442 444
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1089 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0; 1091 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
1090 1092
1091 // TODO(pthatcher): Implement this. 1093 // TODO(pthatcher): Implement this.
1092 virtual bool GetStats(DataMediaInfo* info) { return true; } 1094 virtual bool GetStats(DataMediaInfo* info) { return true; }
1093 1095
1094 virtual bool SetSend(bool send) = 0; 1096 virtual bool SetSend(bool send) = 0;
1095 virtual bool SetReceive(bool receive) = 0; 1097 virtual bool SetReceive(bool receive) = 0;
1096 1098
1097 virtual bool SendData( 1099 virtual bool SendData(
1098 const SendDataParams& params, 1100 const SendDataParams& params,
1099 const rtc::Buffer& payload, 1101 const rtc::CopyOnWriteBuffer& payload,
1100 SendDataResult* result = NULL) = 0; 1102 SendDataResult* result = NULL) = 0;
1101 // Signals when data is received (params, data, len) 1103 // Signals when data is received (params, data, len)
1102 sigslot::signal3<const ReceiveDataParams&, 1104 sigslot::signal3<const ReceiveDataParams&,
1103 const char*, 1105 const char*,
1104 size_t> SignalDataReceived; 1106 size_t> SignalDataReceived;
1105 // Signal when the media channel is ready to send the stream. Arguments are: 1107 // Signal when the media channel is ready to send the stream. Arguments are:
1106 // writable(bool) 1108 // writable(bool)
1107 sigslot::signal1<bool> SignalReadyToSend; 1109 sigslot::signal1<bool> SignalReadyToSend;
1108 // Signal for notifying that the remote side has closed the DataChannel. 1110 // Signal for notifying that the remote side has closed the DataChannel.
1109 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1111 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1110 }; 1112 };
1111 1113
1112 } // namespace cricket 1114 } // namespace cricket
1113 1115
1114 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1116 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
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