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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 } |
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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 |
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