| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ |
| #define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ |
| |
| #include <memory> |
| #include <vector> |
| |
| #include "webrtc/api/audio_codecs/audio_encoder.h" |
| #include "webrtc/api/audio_codecs/audio_format.h" |
| #include "webrtc/api/optional.h" |
| |
| namespace webrtc { |
| |
| // iSAC encoder API (floating-point implementation) for use as a template |
| // parameter to CreateAudioEncoderFactory<...>(). |
| // |
| // NOTE: This struct is still under development and may change without notice. |
| struct AudioEncoderIsacFloat { |
| struct Config { |
| bool IsOk() const { |
| return (sample_rate_hz == 16000 && |
| (frame_size_ms == 30 || frame_size_ms == 60)) || |
| (sample_rate_hz == 32000 && frame_size_ms == 30); |
| } |
| int sample_rate_hz = 16000; |
| int frame_size_ms = 30; |
| }; |
| static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); |
| static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); |
| static AudioCodecInfo QueryAudioEncoder(const Config& config); |
| static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config, |
| int payload_type); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ |