/* * Copyright (c) 2014 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 MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_ #define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_ #include #include #include #include "absl/types/optional.h" #include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h" #include "api/units/time_delta.h" #include "modules/audio_coding/codecs/ilbc/ilbc.h" namespace webrtc { class AudioEncoderIlbcImpl final : public AudioEncoder { public: AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type); ~AudioEncoderIlbcImpl() override; AudioEncoderIlbcImpl(const AudioEncoderIlbcImpl&) = delete; AudioEncoderIlbcImpl& operator=(const AudioEncoderIlbcImpl&) = delete; int SampleRateHz() const override; size_t NumChannels() const override; size_t Num10MsFramesInNextPacket() const override; size_t Max10MsFramesInAPacket() const override; int GetTargetBitrate() const override; EncodedInfo EncodeImpl(uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) override; void Reset() override; absl::optional> GetFrameLengthRange() const override; private: size_t RequiredOutputSizeBytes() const; static constexpr size_t kMaxSamplesPerPacket = 480; const int frame_size_ms_; const int payload_type_; const size_t num_10ms_frames_per_packet_; size_t num_10ms_frames_buffered_; uint32_t first_timestamp_in_buffer_; int16_t input_buffer_[kMaxSamplesPerPacket]; IlbcEncoderInstance* encoder_; }; } // namespace webrtc #endif // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_