Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(358)

Side by Side Diff: webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc

Issue 1174813003: Prepare to convert various types to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Review comments + resync Created 5 years, 6 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
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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 #include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h" 11 #include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
12 12
13 #include <limits> 13 #include <limits>
14 #include "webrtc/base/checks.h" 14 #include "webrtc/base/checks.h"
15 #include "webrtc/common_types.h" 15 #include "webrtc/common_types.h"
16 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" 16 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
17 17
18 namespace webrtc { 18 namespace webrtc {
19 19
20 namespace { 20 namespace {
21 21
22 const int kSampleRateHz = 16000; 22 const int kSampleRateHz = 16000;
23 23
24 } // namespace 24 } // namespace
25 25
26 bool AudioEncoderG722::Config::IsOk() const { 26 bool AudioEncoderG722::Config::IsOk() const {
27 return (frame_size_ms % 10 == 0) && (num_channels >= 1); 27 return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
28 (num_channels >= 1);
28 } 29 }
29 30
30 AudioEncoderG722::EncoderState::EncoderState() { 31 AudioEncoderG722::EncoderState::EncoderState() {
31 CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder)); 32 CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
32 CHECK_EQ(0, WebRtcG722_EncoderInit(encoder)); 33 CHECK_EQ(0, WebRtcG722_EncoderInit(encoder));
33 } 34 }
34 35
35 AudioEncoderG722::EncoderState::~EncoderState() { 36 AudioEncoderG722::EncoderState::~EncoderState() {
36 CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder)); 37 CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
37 } 38 }
(...skipping 108 matching lines...) Expand 10 before | Expand all | Expand 10 after
146 config.payload_type = codec_inst.pltype; 147 config.payload_type = codec_inst.pltype;
147 return config; 148 return config;
148 } 149 }
149 } // namespace 150 } // namespace
150 151
151 AudioEncoderMutableG722::AudioEncoderMutableG722(const CodecInst& codec_inst) 152 AudioEncoderMutableG722::AudioEncoderMutableG722(const CodecInst& codec_inst)
152 : AudioEncoderMutableImpl<AudioEncoderG722>(CreateConfig(codec_inst)) { 153 : AudioEncoderMutableImpl<AudioEncoderG722>(CreateConfig(codec_inst)) {
153 } 154 }
154 155
155 } // namespace webrtc 156 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/codecs/g711/test/testG711.cc ('k') | webrtc/modules/audio_coding/codecs/g722/test/testG722.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698