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1 /* | 1 /* |
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2017 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/api/audio_codecs/g722/audio_encoder_g722.h" | 11 #include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h" |
12 | 12 |
13 #include <memory> | 13 #include <memory> |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
16 #include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h" | 16 #include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h" |
17 #include "webrtc/rtc_base/ptr_util.h" | 17 #include "webrtc/rtc_base/ptr_util.h" |
18 #include "webrtc/rtc_base/safe_conversions.h" | 18 #include "webrtc/rtc_base/safe_conversions.h" |
19 | 19 |
20 namespace webrtc { | 20 namespace webrtc { |
21 | 21 |
22 rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig( | 22 rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig( |
23 const SdpAudioFormat& format) { | 23 const SdpAudioFormat& format) { |
24 return AudioEncoderG722Impl::SdpToConfig(format); | 24 return AudioEncoderG722Impl::SdpToConfig(format); |
25 } | 25 } |
26 | 26 |
27 void AudioEncoderG722::AppendSupportedEncoders( | 27 void AudioEncoderG722::AppendSupportedEncoders( |
28 std::vector<AudioCodecSpec>* specs) { | 28 std::vector<AudioCodecSpec>* specs) { |
29 const SdpAudioFormat fmt = {"g722", 8000, 1}; | 29 const SdpAudioFormat fmt = {"G722", 8000, 1}; |
30 const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); | 30 const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); |
31 specs->push_back({fmt, info}); | 31 specs->push_back({fmt, info}); |
32 } | 32 } |
33 | 33 |
34 AudioCodecInfo AudioEncoderG722::QueryAudioEncoder( | 34 AudioCodecInfo AudioEncoderG722::QueryAudioEncoder( |
35 const AudioEncoderG722Config& config) { | 35 const AudioEncoderG722Config& config) { |
36 RTC_DCHECK(config.IsOk()); | 36 RTC_DCHECK(config.IsOk()); |
37 return {16000, rtc::dchecked_cast<size_t>(config.num_channels), | 37 return {16000, rtc::dchecked_cast<size_t>(config.num_channels), |
38 64000 * config.num_channels}; | 38 64000 * config.num_channels}; |
39 } | 39 } |
40 | 40 |
41 std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder( | 41 std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder( |
42 const AudioEncoderG722Config& config, | 42 const AudioEncoderG722Config& config, |
43 int payload_type) { | 43 int payload_type) { |
44 RTC_DCHECK(config.IsOk()); | 44 RTC_DCHECK(config.IsOk()); |
45 return rtc::MakeUnique<AudioEncoderG722Impl>(config, payload_type); | 45 return rtc::MakeUnique<AudioEncoderG722Impl>(config, payload_type); |
46 } | 46 } |
47 | 47 |
48 } // namespace webrtc | 48 } // namespace webrtc |
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