Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(636)

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

Issue 1728503002: Replace scoped_ptr with unique_ptr in webrtc/media/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@up1
Patch Set: Created 4 years, 10 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/media/base/fakevideocapturer.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 * 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_FAKEMEDIAENGINE_H_ 11 #ifndef WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
12 #define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ 12 #define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
13 13
14 #include <list> 14 #include <list>
15 #include <map> 15 #include <map>
16 #include <memory>
16 #include <set> 17 #include <set>
17 #include <string> 18 #include <string>
18 #include <vector> 19 #include <vector>
19 20
20 #include "webrtc/audio/audio_sink.h" 21 #include "webrtc/audio/audio_sink.h"
21 #include "webrtc/base/buffer.h" 22 #include "webrtc/base/buffer.h"
22 #include "webrtc/base/stringutils.h" 23 #include "webrtc/base/stringutils.h"
23 #include "webrtc/media/base/audiorenderer.h" 24 #include "webrtc/media/base/audiorenderer.h"
24 #include "webrtc/media/base/mediaengine.h" 25 #include "webrtc/media/base/mediaengine.h"
25 #include "webrtc/media/base/rtputils.h" 26 #include "webrtc/media/base/rtputils.h"
(...skipping 299 matching lines...) Expand 10 before | Expand all | Expand 10 after
325 if (output_scalings_.find(ssrc) == output_scalings_.end()) 326 if (output_scalings_.find(ssrc) == output_scalings_.end())
326 return false; 327 return false;
327 *volume = output_scalings_[ssrc]; 328 *volume = output_scalings_[ssrc];
328 return true; 329 return true;
329 } 330 }
330 331
331 virtual bool GetStats(VoiceMediaInfo* info) { return false; } 332 virtual bool GetStats(VoiceMediaInfo* info) { return false; }
332 333
333 virtual void SetRawAudioSink( 334 virtual void SetRawAudioSink(
334 uint32_t ssrc, 335 uint32_t ssrc,
335 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) { 336 std::unique_ptr<webrtc::AudioSinkInterface> sink) {
336 sink_ = std::move(sink); 337 sink_ = std::move(sink);
337 } 338 }
338 339
339 private: 340 private:
340 class VoiceChannelAudioSink : public AudioRenderer::Sink { 341 class VoiceChannelAudioSink : public AudioRenderer::Sink {
341 public: 342 public:
342 explicit VoiceChannelAudioSink(AudioRenderer* renderer) 343 explicit VoiceChannelAudioSink(AudioRenderer* renderer)
343 : renderer_(renderer) { 344 : renderer_(renderer) {
344 renderer_->SetSink(this); 345 renderer_->SetSink(this);
345 } 346 }
(...skipping 55 matching lines...) Expand 10 before | Expand all | Expand 10 after
401 } 402 }
402 403
403 FakeVoiceEngine* engine_; 404 FakeVoiceEngine* engine_;
404 std::vector<AudioCodec> recv_codecs_; 405 std::vector<AudioCodec> recv_codecs_;
405 std::vector<AudioCodec> send_codecs_; 406 std::vector<AudioCodec> send_codecs_;
406 std::map<uint32_t, double> output_scalings_; 407 std::map<uint32_t, double> output_scalings_;
407 std::vector<DtmfInfo> dtmf_info_queue_; 408 std::vector<DtmfInfo> dtmf_info_queue_;
408 int time_since_last_typing_; 409 int time_since_last_typing_;
409 AudioOptions options_; 410 AudioOptions options_;
410 std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_; 411 std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_;
411 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_; 412 std::unique_ptr<webrtc::AudioSinkInterface> sink_;
412 }; 413 };
413 414
414 // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo. 415 // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
415 inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info, 416 inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
416 uint32_t ssrc, 417 uint32_t ssrc,
417 int event_code, 418 int event_code,
418 int duration) { 419 int duration) {
419 return (info.duration == duration && info.event_code == event_code && 420 return (info.duration == duration && info.event_code == event_code &&
420 info.ssrc == ssrc); 421 info.ssrc == ssrc);
421 } 422 }
(...skipping 446 matching lines...) Expand 10 before | Expand all | Expand 10 after
868 869
869 private: 870 private:
870 std::vector<FakeDataMediaChannel*> channels_; 871 std::vector<FakeDataMediaChannel*> channels_;
871 std::vector<DataCodec> data_codecs_; 872 std::vector<DataCodec> data_codecs_;
872 DataChannelType last_channel_type_; 873 DataChannelType last_channel_type_;
873 }; 874 };
874 875
875 } // namespace cricket 876 } // namespace cricket
876 877
877 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ 878 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/media/base/fakevideocapturer.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698