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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.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
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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_ENGINE_WEBRTCVOICEENGINE_H_ 11 #ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
12 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 12 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
13 13
14 #include <map> 14 #include <map>
15 #include <memory>
15 #include <string> 16 #include <string>
16 #include <vector> 17 #include <vector>
17 18
18 #include "webrtc/audio_state.h" 19 #include "webrtc/audio_state.h"
19 #include "webrtc/base/buffer.h" 20 #include "webrtc/base/buffer.h"
20 #include "webrtc/base/scoped_ptr.h"
21 #include "webrtc/base/stream.h" 21 #include "webrtc/base/stream.h"
22 #include "webrtc/base/thread_checker.h" 22 #include "webrtc/base/thread_checker.h"
23 #include "webrtc/call.h" 23 #include "webrtc/call.h"
24 #include "webrtc/common.h" 24 #include "webrtc/common.h"
25 #include "webrtc/config.h" 25 #include "webrtc/config.h"
26 #include "webrtc/media/base/rtputils.h" 26 #include "webrtc/media/base/rtputils.h"
27 #include "webrtc/media/engine/webrtccommon.h" 27 #include "webrtc/media/engine/webrtccommon.h"
28 #include "webrtc/media/engine/webrtcvoe.h" 28 #include "webrtc/media/engine/webrtcvoe.h"
29 #include "webrtc/pc/channel.h" 29 #include "webrtc/pc/channel.h"
30 30
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106 // webrtc::TraceCallback: 106 // webrtc::TraceCallback:
107 void Print(webrtc::TraceLevel level, const char* trace, int length) override; 107 void Print(webrtc::TraceLevel level, const char* trace, int length) override;
108 108
109 void StartAecDump(const std::string& filename); 109 void StartAecDump(const std::string& filename);
110 int CreateVoEChannel(); 110 int CreateVoEChannel();
111 111
112 rtc::ThreadChecker signal_thread_checker_; 112 rtc::ThreadChecker signal_thread_checker_;
113 rtc::ThreadChecker worker_thread_checker_; 113 rtc::ThreadChecker worker_thread_checker_;
114 114
115 // The primary instance of WebRtc VoiceEngine. 115 // The primary instance of WebRtc VoiceEngine.
116 rtc::scoped_ptr<VoEWrapper> voe_wrapper_; 116 std::unique_ptr<VoEWrapper> voe_wrapper_;
117 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 117 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
118 // The external audio device manager 118 // The external audio device manager
119 webrtc::AudioDeviceModule* adm_ = nullptr; 119 webrtc::AudioDeviceModule* adm_ = nullptr;
120 std::vector<AudioCodec> codecs_; 120 std::vector<AudioCodec> codecs_;
121 std::vector<WebRtcVoiceMediaChannel*> channels_; 121 std::vector<WebRtcVoiceMediaChannel*> channels_;
122 webrtc::Config voe_config_; 122 webrtc::Config voe_config_;
123 bool initialized_ = false; 123 bool initialized_ = false;
124 bool is_dumping_aec_ = false; 124 bool is_dumping_aec_ = false;
125 125
126 webrtc::AgcConfig default_agc_config_; 126 webrtc::AgcConfig default_agc_config_;
(...skipping 54 matching lines...) Expand 10 before | Expand all | Expand 10 after
181 181
182 void OnPacketReceived(rtc::Buffer* packet, 182 void OnPacketReceived(rtc::Buffer* packet,
183 const rtc::PacketTime& packet_time) override; 183 const rtc::PacketTime& packet_time) override;
184 void OnRtcpReceived(rtc::Buffer* packet, 184 void OnRtcpReceived(rtc::Buffer* packet,
185 const rtc::PacketTime& packet_time) override; 185 const rtc::PacketTime& packet_time) override;
186 void OnReadyToSend(bool ready) override {} 186 void OnReadyToSend(bool ready) override {}
187 bool GetStats(VoiceMediaInfo* info) override; 187 bool GetStats(VoiceMediaInfo* info) override;
188 188
189 void SetRawAudioSink( 189 void SetRawAudioSink(
190 uint32_t ssrc, 190 uint32_t ssrc,
191 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override; 191 std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
192 192
193 // implements Transport interface 193 // implements Transport interface
194 bool SendRtp(const uint8_t* data, 194 bool SendRtp(const uint8_t* data,
195 size_t len, 195 size_t len,
196 const webrtc::PacketOptions& options) override { 196 const webrtc::PacketOptions& options) override {
197 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 197 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
198 kMaxRtpPacketLen); 198 kMaxRtpPacketLen);
199 rtc::PacketOptions rtc_options; 199 rtc::PacketOptions rtc_options;
200 rtc_options.packet_id = options.packet_id; 200 rtc_options.packet_id = options.packet_id;
201 return VoiceMediaChannel::SendPacket(&packet, rtc_options); 201 return VoiceMediaChannel::SendPacket(&packet, rtc_options);
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233 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); 233 return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
234 } 234 }
235 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs); 235 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
236 bool SetSendBitrateInternal(int bps); 236 bool SetSendBitrateInternal(int bps);
237 237
238 rtc::ThreadChecker worker_thread_checker_; 238 rtc::ThreadChecker worker_thread_checker_;
239 239
240 WebRtcVoiceEngine* const engine_ = nullptr; 240 WebRtcVoiceEngine* const engine_ = nullptr;
241 std::vector<AudioCodec> recv_codecs_; 241 std::vector<AudioCodec> recv_codecs_;
242 std::vector<AudioCodec> send_codecs_; 242 std::vector<AudioCodec> send_codecs_;
243 rtc::scoped_ptr<webrtc::CodecInst> send_codec_; 243 std::unique_ptr<webrtc::CodecInst> send_codec_;
244 bool send_bitrate_setting_ = false; 244 bool send_bitrate_setting_ = false;
245 int send_bitrate_bps_ = 0; 245 int send_bitrate_bps_ = 0;
246 AudioOptions options_; 246 AudioOptions options_;
247 rtc::Optional<int> dtmf_payload_type_; 247 rtc::Optional<int> dtmf_payload_type_;
248 bool desired_playout_ = false; 248 bool desired_playout_ = false;
249 bool nack_enabled_ = false; 249 bool nack_enabled_ = false;
250 bool transport_cc_enabled_ = false; 250 bool transport_cc_enabled_ = false;
251 bool playout_ = false; 251 bool playout_ = false;
252 SendFlags desired_send_ = SEND_NOTHING; 252 SendFlags desired_send_ = SEND_NOTHING;
253 SendFlags send_ = SEND_NOTHING; 253 SendFlags send_ = SEND_NOTHING;
254 webrtc::Call* const call_ = nullptr; 254 webrtc::Call* const call_ = nullptr;
255 255
256 // SSRC of unsignalled receive stream, or -1 if there isn't one. 256 // SSRC of unsignalled receive stream, or -1 if there isn't one.
257 int64_t default_recv_ssrc_ = -1; 257 int64_t default_recv_ssrc_ = -1;
258 // Volume for unsignalled stream, which may be set before the stream exists. 258 // Volume for unsignalled stream, which may be set before the stream exists.
259 double default_recv_volume_ = 1.0; 259 double default_recv_volume_ = 1.0;
260 // Sink for unsignalled stream, which may be set before the stream exists. 260 // Sink for unsignalled stream, which may be set before the stream exists.
261 rtc::scoped_ptr<webrtc::AudioSinkInterface> default_sink_; 261 std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
262 // Default SSRC to use for RTCP receiver reports in case of no signaled 262 // Default SSRC to use for RTCP receiver reports in case of no signaled
263 // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 263 // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
264 // and https://code.google.com/p/chromium/issues/detail?id=547661 264 // and https://code.google.com/p/chromium/issues/detail?id=547661
265 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; 265 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
266 266
267 class WebRtcAudioSendStream; 267 class WebRtcAudioSendStream;
268 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; 268 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
269 std::vector<webrtc::RtpExtension> send_rtp_extensions_; 269 std::vector<webrtc::RtpExtension> send_rtp_extensions_;
270 270
271 class WebRtcAudioReceiveStream; 271 class WebRtcAudioReceiveStream;
272 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; 272 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
273 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 273 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
274 274
275 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 275 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
276 }; 276 };
277 } // namespace cricket 277 } // namespace cricket
278 278
279 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 279 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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