| /* |
| * Copyright (c) 2015 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_G722_AUDIO_DECODER_G722_H_ |
| #define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_ |
| |
| #include "api/audio_codecs/audio_decoder.h" |
| |
| typedef struct WebRtcG722DecInst G722DecInst; |
| |
| namespace webrtc { |
| |
| class AudioDecoderG722Impl final : public AudioDecoder { |
| public: |
| AudioDecoderG722Impl(); |
| ~AudioDecoderG722Impl() override; |
| |
| AudioDecoderG722Impl(const AudioDecoderG722Impl&) = delete; |
| AudioDecoderG722Impl& operator=(const AudioDecoderG722Impl&) = delete; |
| |
| bool HasDecodePlc() const override; |
| void Reset() override; |
| std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, |
| uint32_t timestamp) override; |
| int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override; |
| int PacketDurationRedundant(const uint8_t* encoded, |
| size_t encoded_len) const override; |
| int SampleRateHz() const override; |
| size_t Channels() const override; |
| |
| protected: |
| int DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) override; |
| |
| private: |
| G722DecInst* dec_state_; |
| }; |
| |
| class AudioDecoderG722StereoImpl final : public AudioDecoder { |
| public: |
| AudioDecoderG722StereoImpl(); |
| ~AudioDecoderG722StereoImpl() override; |
| |
| AudioDecoderG722StereoImpl(const AudioDecoderG722StereoImpl&) = delete; |
| AudioDecoderG722StereoImpl& operator=(const AudioDecoderG722StereoImpl&) = |
| delete; |
| |
| void Reset() override; |
| std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, |
| uint32_t timestamp) override; |
| int SampleRateHz() const override; |
| int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override; |
| size_t Channels() const override; |
| |
| protected: |
| int DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) override; |
| |
| private: |
| // Splits the stereo-interleaved payload in `encoded` into separate payloads |
| // for left and right channels. The separated payloads are written to |
| // `encoded_deinterleaved`, which must hold at least `encoded_len` samples. |
| // The left channel starts at offset 0, while the right channel starts at |
| // offset encoded_len / 2 into `encoded_deinterleaved`. |
| void SplitStereoPacket(const uint8_t* encoded, |
| size_t encoded_len, |
| uint8_t* encoded_deinterleaved); |
| |
| G722DecInst* dec_state_left_; |
| G722DecInst* dec_state_right_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_ |