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Side by Side Diff: webrtc/voice_engine/channel.cc

Issue 2082483003: voice_engine: Removed old variants of Channel constructor and CreateChannel (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 4 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 #include "webrtc/voice_engine/channel.h" 11 #include "webrtc/voice_engine/channel.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <utility> 14 #include <utility>
15 15
16 #include "webrtc/base/checks.h" 16 #include "webrtc/base/checks.h"
17 #include "webrtc/base/criticalsection.h" 17 #include "webrtc/base/criticalsection.h"
18 #include "webrtc/base/format_macros.h" 18 #include "webrtc/base/format_macros.h"
19 #include "webrtc/base/logging.h" 19 #include "webrtc/base/logging.h"
20 #include "webrtc/base/rate_limiter.h" 20 #include "webrtc/base/rate_limiter.h"
21 #include "webrtc/base/thread_checker.h" 21 #include "webrtc/base/thread_checker.h"
22 #include "webrtc/base/timeutils.h" 22 #include "webrtc/base/timeutils.h"
23 #include "webrtc/call/rtc_event_log.h" 23 #include "webrtc/call/rtc_event_log.h"
24 #include "webrtc/common.h" 24 #include "webrtc/common.h"
25 #include "webrtc/config.h" 25 #include "webrtc/config.h"
26 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
27 #include "webrtc/modules/audio_device/include/audio_device.h" 26 #include "webrtc/modules/audio_device/include/audio_device.h"
28 #include "webrtc/modules/audio_processing/include/audio_processing.h" 27 #include "webrtc/modules/audio_processing/include/audio_processing.h"
29 #include "webrtc/modules/include/module_common_types.h" 28 #include "webrtc/modules/include/module_common_types.h"
30 #include "webrtc/modules/pacing/packet_router.h" 29 #include "webrtc/modules/pacing/packet_router.h"
31 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" 30 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
32 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" 31 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
33 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 32 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
34 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" 33 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
35 #include "webrtc/modules/utility/include/audio_frame_operations.h" 34 #include "webrtc/modules/utility/include/audio_frame_operations.h"
36 #include "webrtc/modules/utility/include/process_thread.h" 35 #include "webrtc/modules/utility/include/process_thread.h"
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745 if (_outputFilePlayerPtr) { 744 if (_outputFilePlayerPtr) {
746 if (_outputFilePlayerPtr->Frequency() > highestNeeded) { 745 if (_outputFilePlayerPtr->Frequency() > highestNeeded) {
747 highestNeeded = _outputFilePlayerPtr->Frequency(); 746 highestNeeded = _outputFilePlayerPtr->Frequency();
748 } 747 }
749 } 748 }
750 } 749 }
751 750
752 return (highestNeeded); 751 return (highestNeeded);
753 } 752 }
754 753
755 int32_t Channel::CreateChannel(Channel*& channel,
756 int32_t channelId,
757 uint32_t instanceId,
758 const Config& config) {
759 return CreateChannel(channel, channelId, instanceId, config,
760 CreateBuiltinAudioDecoderFactory());
761 }
762
763 int32_t Channel::CreateChannel( 754 int32_t Channel::CreateChannel(
764 Channel*& channel, 755 Channel*& channel,
765 int32_t channelId, 756 int32_t channelId,
766 uint32_t instanceId, 757 uint32_t instanceId,
767 const Config& config, 758 const Config& config,
768 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) { 759 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
769 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId), 760 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
770 "Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId, 761 "Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
771 instanceId); 762 instanceId);
772 763
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3591 int64_t min_rtt = 0; 3582 int64_t min_rtt = 0;
3592 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != 3583 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
3593 0) { 3584 0) {
3594 return 0; 3585 return 0;
3595 } 3586 }
3596 return rtt; 3587 return rtt;
3597 } 3588 }
3598 3589
3599 } // namespace voe 3590 } // namespace voe
3600 } // namespace webrtc 3591 } // namespace webrtc
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