Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/

Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}
This commit is contained in:
Byoungchan Lee
2022-01-21 09:49:39 +09:00
committed by WebRTC LUCI CQ
parent ce6170fcdf
commit 604fd2f1ab
127 changed files with 466 additions and 393 deletions

View File

@ -20,7 +20,6 @@
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/audio_vector.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -32,6 +31,9 @@ class SyncBuffer : public AudioMultiVector {
end_timestamp_(0),
dtmf_index_(0) {}
SyncBuffer(const SyncBuffer&) = delete;
SyncBuffer& operator=(const SyncBuffer&) = delete;
// Returns the number of samples yet to play out from the buffer.
size_t FutureLength() const;
@ -102,8 +104,6 @@ class SyncBuffer : public AudioMultiVector {
size_t next_index_;
uint32_t end_timestamp_; // The timestamp of the last sample in the buffer.
size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer.
RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
};
} // namespace webrtc