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_L16_AUDIO_ENCODER_L16_H_ 12 #define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_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 "api/field_trials_view.h" 22 #include "rtc_base/system/rtc_export.h" 23 24 namespace webrtc { 25 26 // L16 encoder API for use as a template parameter to 27 // CreateAudioEncoderFactory<...>(). 28 struct RTC_EXPORT AudioEncoderL16 { 29 struct Config { IsOkAudioEncoderL16::Config30 bool IsOk() const { 31 return (sample_rate_hz == 8000 || sample_rate_hz == 16000 || 32 sample_rate_hz == 32000 || sample_rate_hz == 48000) && 33 num_channels >= 1 && 34 num_channels <= AudioEncoder::kMaxNumberOfChannels && 35 frame_size_ms > 0 && frame_size_ms <= 120 && 36 frame_size_ms % 10 == 0; 37 } 38 int sample_rate_hz = 8000; 39 int num_channels = 1; 40 int frame_size_ms = 10; 41 }; 42 static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); 43 static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); 44 static AudioCodecInfo QueryAudioEncoder(const Config& config); 45 static std::unique_ptr<AudioEncoder> MakeAudioEncoder( 46 const Config& config, 47 int payload_type, 48 absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, 49 const FieldTrialsView* field_trials = nullptr); 50 }; 51 52 } // namespace webrtc 53 54 #endif // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_ 55