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

Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 5 years, 2 months ago
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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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2668 if (which_channel == -1) { 2668 if (which_channel == -1) {
2669 which_channel = voe_channel(); 2669 which_channel = voe_channel();
2670 } 2670 }
2671 2671
2672 // Pass it off to the decoder. 2672 // Pass it off to the decoder.
2673 engine()->voe()->network()->ReceivedRTPPacket( 2673 engine()->voe()->network()->ReceivedRTPPacket(
2674 which_channel, packet->data(), packet->size(), 2674 which_channel, packet->data(), packet->size(),
2675 webrtc::PacketTime(packet_time.timestamp, packet_time.not_before)); 2675 webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
2676 } 2676 }
2677 2677
2678 void WebRtcVoiceMediaChannel::OnPacketSent(const rtc::SentPacket& packet_sent) {
2679 RTC_DCHECK(thread_checker_.CalledOnValidThread());
2680 // TODO(holmer): Hook up to call.
2681 }
2682
2678 void WebRtcVoiceMediaChannel::OnRtcpReceived( 2683 void WebRtcVoiceMediaChannel::OnRtcpReceived(
2679 rtc::Buffer* packet, const rtc::PacketTime& packet_time) { 2684 rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
2680 RTC_DCHECK(thread_checker_.CalledOnValidThread()); 2685 RTC_DCHECK(thread_checker_.CalledOnValidThread());
2681 2686
2682 // Forward packet to Call as well. 2687 // Forward packet to Call as well.
2683 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, 2688 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
2684 packet_time.not_before); 2689 packet_time.not_before);
2685 call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, 2690 call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
2686 reinterpret_cast<const uint8_t*>(packet->data()), packet->size(), 2691 reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
2687 webrtc_packet_time); 2692 webrtc_packet_time);
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3285 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); 3290 LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
3286 return false; 3291 return false;
3287 } 3292 }
3288 } 3293 }
3289 return true; 3294 return true;
3290 } 3295 }
3291 3296
3292 } // namespace cricket 3297 } // namespace cricket
3293 3298
3294 #endif // HAVE_WEBRTC_VOICE 3299 #endif // HAVE_WEBRTC_VOICE
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