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Side by Side Diff: webrtc/modules/audio_coding/acm2/acm_receiver.h

Issue 1613643004: Remove mutable from rtc::CriticalSection members. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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
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274 // Get statistics of calls to GetAudio(). 274 // Get statistics of calls to GetAudio().
275 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; 275 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
276 276
277 private: 277 private:
278 const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header, 278 const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
279 uint8_t payload_type) const 279 uint8_t payload_type) const
280 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); 280 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
281 281
282 uint32_t NowInTimestamp(int decoder_sampling_rate) const; 282 uint32_t NowInTimestamp(int decoder_sampling_rate) const;
283 283
284 mutable rtc::CriticalSection crit_sect_; 284 rtc::CriticalSection crit_sect_;
285 int id_; // TODO(henrik.lundin) Make const. 285 int id_; // TODO(henrik.lundin) Make const.
286 const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_); 286 const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
287 AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_); 287 AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
288 ACMResampler resampler_ GUARDED_BY(crit_sect_); 288 ACMResampler resampler_ GUARDED_BY(crit_sect_);
289 // Used in GetAudio, declared as member to avoid allocating every 10ms. 289 // Used in GetAudio, declared as member to avoid allocating every 10ms.
290 // TODO(henrik.lundin) Stack-allocate in GetAudio instead? 290 // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
291 rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_); 291 rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
292 rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_); 292 rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
293 CallStatistics call_stats_ GUARDED_BY(crit_sect_); 293 CallStatistics call_stats_ GUARDED_BY(crit_sect_);
294 NetEq* neteq_; 294 NetEq* neteq_;
295 // Decoders map is keyed by payload type 295 // Decoders map is keyed by payload type
296 std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_); 296 std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_);
297 bool vad_enabled_; 297 bool vad_enabled_;
298 Clock* clock_; // TODO(henrik.lundin) Make const if possible. 298 Clock* clock_; // TODO(henrik.lundin) Make const if possible.
299 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); 299 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
300 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); 300 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
301 }; 301 };
302 302
303 } // namespace acm2 303 } // namespace acm2
304 304
305 } // namespace webrtc 305 } // namespace webrtc
306 306
307 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ 307 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
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