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Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/

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BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1696853004

Cr-Commit-Position: refs/heads/master@{#11613}
This commit is contained in:
kwiberg
2016-02-14 01:10:03 -08:00
committed by Commit bot
parent 3f5f5aa449
commit 91d9756bc6
16 changed files with 41 additions and 35 deletions
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10
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
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@ -11,6 +11,7 @@
#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include <algorithm>
#include <memory>
#include <limits>
namespace webrtc {
@ -19,12 +20,13 @@ namespace {
const int kMaxFrameSizeMs = 60;
rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> CreateCngInst(
std::unique_ptr<CNG_enc_inst, CngInstDeleter> CreateCngInst(
int sample_rate_hz,
int sid_frame_interval_ms,
int num_cng_coefficients) {
rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst;
RTC_CHECK_EQ(0, WebRtcCng_CreateEnc(cng_inst.accept()));
CNG_enc_inst* ci;
RTC_CHECK_EQ(0, WebRtcCng_CreateEnc(&ci));
std::unique_ptr<CNG_enc_inst, CngInstDeleter> cng_inst(ci);
RTC_CHECK_EQ(0,
WebRtcCng_InitEnc(cng_inst.get(), sample_rate_hz,
sid_frame_interval_ms, num_cng_coefficients));
@ -55,7 +57,7 @@ AudioEncoderCng::AudioEncoderCng(const Config& config)
num_cng_coefficients_(config.num_cng_coefficients),
sid_frame_interval_ms_(config.sid_frame_interval_ms),
last_frame_active_(true),
vad_(config.vad ? rtc_make_scoped_ptr(config.vad)
vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
: CreateVad(config.vad_mode)) {
RTC_CHECK(config.IsOk()) << "Invalid configuration.";
cng_inst_ = CreateCngInst(SampleRateHz(), sid_frame_interval_ms_,

9
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
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@ -11,16 +11,17 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
#include <memory>
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/common_audio/vad/include/vad.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
namespace webrtc {
// Deleter for use with scoped_ptr.
// Deleter for use with unique_ptr.
struct CngInstDeleter {
void operator()(CNG_enc_inst* ptr) const { WebRtcCng_FreeEnc(ptr); }
};
@ -84,8 +85,8 @@ class AudioEncoderCng final : public AudioEncoder {
std::vector<int16_t> speech_buffer_;
std::vector<uint32_t> rtp_timestamps_;
bool last_frame_active_;
rtc::scoped_ptr<Vad> vad_;
rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
std::unique_ptr<Vad> vad_;
std::unique_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCng);
};

4
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/vad/mock/mock_vad.h"
#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
@ -185,7 +185,7 @@ class AudioEncoderCngTest : public ::testing::Test {
}
AudioEncoderCng::Config config_;
rtc::scoped_ptr<AudioEncoderCng> cng_;
std::unique_ptr<AudioEncoderCng> cng_;
MockAudioEncoder mock_encoder_;
MockVad* mock_vad_; // Ownership is transferred to |cng_|.
uint32_t timestamp_;

2
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
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@ -13,7 +13,7 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
namespace webrtc {

7
webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
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@ -11,8 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
#include <memory>
#include "webrtc/base/buffer.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
@ -51,7 +52,7 @@ class AudioEncoderG722 final : public AudioEncoder {
// The encoder state for one channel.
struct EncoderState {
G722EncInst* encoder;
rtc::scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
rtc::Buffer encoded_buffer; // Already encoded.
EncoderState();
~EncoderState();
@ -64,7 +65,7 @@ class AudioEncoderG722 final : public AudioEncoder {
const size_t num_10ms_frames_per_packet_;
size_t num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
const rtc::scoped_ptr<EncoderState[]> encoders_;
const std::unique_ptr<EncoderState[]> encoders_;
rtc::Buffer interleave_buffer_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
};

2
webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
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@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"

1
webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
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@ -12,7 +12,6 @@
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"

5
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
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@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
@ -31,7 +32,7 @@ class AudioEncoderOpusTest : public ::testing::Test {
}
CodecInst codec_inst_ = kOpusSettings;
rtc::scoped_ptr<AudioEncoderOpus> encoder_;
std::unique_ptr<AudioEncoderOpus> encoder_;
};
TEST_F(AudioEncoderOpusTest, DefaultApplicationModeMono) {

9
webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
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@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "webrtc/test/testsupport/fileutils.h"
@ -61,9 +62,9 @@ class OpusFecTest : public TestWithParam<coding_param> {
string in_filename_;
rtc::scoped_ptr<int16_t[]> in_data_;
rtc::scoped_ptr<int16_t[]> out_data_;
rtc::scoped_ptr<uint8_t[]> bit_stream_;
std::unique_ptr<int16_t[]> in_data_;
std::unique_ptr<int16_t[]> out_data_;
std::unique_ptr<uint8_t[]> bit_stream_;
};
void OpusFecTest::SetUp() {

4
webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
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@ -7,6 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
@ -636,7 +638,7 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
// Encode & decode.
int16_t audio_type;
rtc::scoped_ptr<int16_t[]> output_data_decode(
std::unique_ptr<int16_t[]> output_data_decode(
new int16_t[kPackets * kOpus20msFrameSamples * channels_]);
OpusRepacketizer* rp = opus_repacketizer_create();

1
webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
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@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
namespace webrtc {

1
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
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@ -14,7 +14,6 @@
#include <vector>
#include "webrtc/base/buffer.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
namespace webrtc {

4
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
@ -68,7 +68,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
}
MockAudioEncoder mock_encoder_;
rtc::scoped_ptr<AudioEncoderCopyRed> red_;
std::unique_ptr<AudioEncoderCopyRed> red_;
uint32_t timestamp_;
int16_t audio_[kMaxNumSamples];
const int sample_rate_hz_;

8
webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
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@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
#include <memory>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -61,11 +61,11 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
// Expected output number of samples-per-channel in a frame.
size_t output_length_sample_;
rtc::scoped_ptr<int16_t[]> in_data_;
rtc::scoped_ptr<int16_t[]> out_data_;
std::unique_ptr<int16_t[]> in_data_;
std::unique_ptr<int16_t[]> out_data_;
size_t data_pointer_;
size_t loop_length_samples_;
rtc::scoped_ptr<uint8_t[]> bit_stream_;
std::unique_ptr<uint8_t[]> bit_stream_;
// Maximum number of bytes in output bitstream for a frame of audio.
size_t max_bytes_;