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 API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ 12 #define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ 13 14 #include <memory> 15 #include <vector> 16 17 #include "absl/types/optional.h" 18 #include "api/audio_codecs/audio_codec_pair_id.h" 19 #include "api/audio_codecs/audio_encoder.h" 20 #include "api/audio_codecs/audio_format.h" 21 #include "rtc_base/system/rtc_export.h" 22 23 namespace webrtc { 24 25 // iSAC encoder API (floating-point implementation) for use as a template 26 // parameter to CreateAudioEncoderFactory<...>(). 27 struct RTC_EXPORT AudioEncoderIsacFloat { 28 struct Config { IsOkAudioEncoderIsacFloat::Config29 bool IsOk() const { 30 switch (sample_rate_hz) { 31 case 16000: 32 if (frame_size_ms != 30 && frame_size_ms != 60) { 33 return false; 34 } 35 if (bit_rate < 10000 || bit_rate > 32000) { 36 return false; 37 } 38 return true; 39 case 32000: 40 if (frame_size_ms != 30) { 41 return false; 42 } 43 if (bit_rate < 10000 || bit_rate > 56000) { 44 return false; 45 } 46 return true; 47 default: 48 return false; 49 } 50 } 51 int sample_rate_hz = 16000; 52 int frame_size_ms = 30; 53 int bit_rate = 32000; // Limit on short-term average bit rate, in bits/s. 54 }; 55 static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); 56 static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); 57 static AudioCodecInfo QueryAudioEncoder(const Config& config); 58 static std::unique_ptr<AudioEncoder> MakeAudioEncoder( 59 const Config& config, 60 int payload_type, 61 absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt); 62 }; 63 64 } // namespace webrtc 65 66 #endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ 67