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1 /* | |
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
3 * | |
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 | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ | |
12 #define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ | |
13 | |
14 #include <memory> | |
15 #include <vector> | |
16 | |
17 #include "webrtc/api/audio_codecs/audio_encoder.h" | |
18 #include "webrtc/api/audio_codecs/audio_format.h" | |
19 #include "webrtc/rtc_base/optional.h" | |
20 | |
21 namespace webrtc { | |
22 | |
23 // iSACfloat encoder API for use as a template parameter to | |
ossu
2017/08/15 14:04:19
Perhaps rephrase this one as "Floating point iSAC
kwiberg-webrtc
2017/08/17 13:32:40
Done.
| |
24 // CreateAudioEncoderFactory<...>(). | |
25 // | |
26 // NOTE: This struct is still under development and may change without notice. | |
27 struct AudioEncoderIsacFloat { | |
28 struct Config { | |
29 bool IsOk() const { | |
30 return (sample_rate_hz == 16000 && | |
31 (frame_size_ms == 30 || frame_size_ms == 60)) || | |
32 (sample_rate_hz == 32000 && frame_size_ms == 30); | |
33 } | |
34 int sample_rate_hz = 16000; | |
35 int frame_size_ms = 30; | |
36 }; | |
37 static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); | |
38 static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); | |
39 static AudioCodecInfo QueryAudioEncoder(const Config& config); | |
40 static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config, | |
41 int payload_type); | |
42 }; | |
43 | |
44 } // namespace webrtc | |
45 | |
46 #endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ | |
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