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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, |
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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; |
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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_ |
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