Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(527)

Side by Side Diff: talk/media/webrtc/webrtcvoiceengine.h

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « no previous file | webrtc/common_types.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 322 matching lines...) Expand 10 before | Expand all | Expand 10 after
333 const rtc::PacketTime& packet_time) override; 333 const rtc::PacketTime& packet_time) override;
334 void OnReadyToSend(bool ready) override {} 334 void OnReadyToSend(bool ready) override {}
335 bool SetMaxSendBandwidth(int bps) override; 335 bool SetMaxSendBandwidth(int bps) override;
336 bool GetStats(VoiceMediaInfo* info) override; 336 bool GetStats(VoiceMediaInfo* info) override;
337 // Gets last reported error from WebRtc voice engine. This should be only 337 // Gets last reported error from WebRtc voice engine. This should be only
338 // called in response a failure. 338 // called in response a failure.
339 void GetLastMediaError(uint32* ssrc, 339 void GetLastMediaError(uint32* ssrc,
340 VoiceMediaChannel::Error* error) override; 340 VoiceMediaChannel::Error* error) override;
341 341
342 // implements Transport interface 342 // implements Transport interface
343 int SendPacket(int channel, const void* data, size_t len) override { 343 int SendPacket(const void* data, size_t len) override {
344 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 344 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
345 kMaxRtpPacketLen); 345 kMaxRtpPacketLen);
346 return VoiceMediaChannel::SendPacket(&packet) ? static_cast<int>(len) : -1; 346 return VoiceMediaChannel::SendPacket(&packet) ? static_cast<int>(len) : -1;
347 } 347 }
348 348
349 int SendRTCPPacket(int channel, const void* data, size_t len) override { 349 int SendRTCPPacket(const void* data, size_t len) override {
350 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 350 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
351 kMaxRtpPacketLen); 351 kMaxRtpPacketLen);
352 return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1; 352 return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1;
353 } 353 }
354 354
355 bool FindSsrc(int channel_num, uint32* ssrc); 355 bool FindSsrc(int channel_num, uint32* ssrc);
356 void OnError(uint32 ssrc, int error); 356 void OnError(uint32 ssrc, int error);
357 357
358 bool sending() const { return send_ != SEND_NOTHING; } 358 bool sending() const { return send_ != SEND_NOTHING; }
359 int GetReceiveChannelNum(uint32 ssrc) const; 359 int GetReceiveChannelNum(uint32 ssrc) const;
(...skipping 95 matching lines...) Expand 10 before | Expand all | Expand 10 after
455 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 455 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
456 456
457 // Do not lock this on the VoE media processor thread; potential for deadlock 457 // Do not lock this on the VoE media processor thread; potential for deadlock
458 // exists. 458 // exists.
459 mutable rtc::CriticalSection receive_channels_cs_; 459 mutable rtc::CriticalSection receive_channels_cs_;
460 }; 460 };
461 461
462 } // namespace cricket 462 } // namespace cricket
463 463
464 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 464 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/common_types.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698