| /* |
| * 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_ |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h" |
| #include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h" |
| #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h" |
| |
| namespace webrtc { |
| |
| struct IsacFix { |
| typedef ISACFIX_MainStruct instance_type; |
| static const bool has_swb = false; |
| static const uint16_t kFixSampleRate = 16000; |
| static inline int16_t Control(instance_type* inst, |
| int32_t rate, |
| int framesize) { |
| return WebRtcIsacfix_Control(inst, rate, framesize); |
| } |
| static inline int16_t ControlBwe(instance_type* inst, |
| int32_t rate_bps, |
| int frame_size_ms, |
| int16_t enforce_frame_size) { |
| return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms, |
| enforce_frame_size); |
| } |
| static inline int16_t Create(instance_type** inst) { |
| return WebRtcIsacfix_Create(inst); |
| } |
| static inline int DecodeInternal(instance_type* inst, |
| const uint8_t* encoded, |
| int16_t len, |
| int16_t* decoded, |
| int16_t* speech_type) { |
| return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type); |
| } |
| static inline int16_t DecodePlc(instance_type* inst, |
| int16_t* decoded, |
| int16_t num_lost_frames) { |
| return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames); |
| } |
| static inline int16_t DecoderInit(instance_type* inst) { |
| return WebRtcIsacfix_DecoderInit(inst); |
| } |
| static inline int Encode(instance_type* inst, |
| const int16_t* speech_in, |
| uint8_t* encoded) { |
| return WebRtcIsacfix_Encode(inst, speech_in, encoded); |
| } |
| static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) { |
| return WebRtcIsacfix_EncoderInit(inst, coding_mode); |
| } |
| static inline uint16_t EncSampRate(instance_type* inst) { |
| return kFixSampleRate; |
| } |
| |
| static inline int16_t Free(instance_type* inst) { |
| return WebRtcIsacfix_Free(inst); |
| } |
| static inline void GetBandwidthInfo(instance_type* inst, |
| IsacBandwidthInfo* bwinfo) { |
| WebRtcIsacfix_GetBandwidthInfo(inst, bwinfo); |
| } |
| static inline int16_t GetErrorCode(instance_type* inst) { |
| return WebRtcIsacfix_GetErrorCode(inst); |
| } |
| |
| static inline int16_t GetNewFrameLen(instance_type* inst) { |
| return WebRtcIsacfix_GetNewFrameLen(inst); |
| } |
| static inline void SetBandwidthInfo(instance_type* inst, |
| const IsacBandwidthInfo* bwinfo) { |
| WebRtcIsacfix_SetBandwidthInfo(inst, bwinfo); |
| } |
| static inline int16_t SetDecSampRate(instance_type* inst, |
| uint16_t sample_rate_hz) { |
| DCHECK_EQ(sample_rate_hz, kFixSampleRate); |
| return 0; |
| } |
| static inline int16_t SetEncSampRate(instance_type* inst, |
| uint16_t sample_rate_hz) { |
| DCHECK_EQ(sample_rate_hz, kFixSampleRate); |
| return 0; |
| } |
| static inline int16_t UpdateBwEstimate(instance_type* inst, |
| const uint8_t* encoded, |
| int32_t packet_size, |
| uint16_t rtp_seq_number, |
| uint32_t send_ts, |
| uint32_t arr_ts) { |
| return WebRtcIsacfix_UpdateBwEstimate(inst, encoded, packet_size, |
| rtp_seq_number, send_ts, arr_ts); |
| } |
| static inline int16_t SetMaxPayloadSize(instance_type* inst, |
| int16_t max_payload_size_bytes) { |
| return WebRtcIsacfix_SetMaxPayloadSize(inst, max_payload_size_bytes); |
| } |
| static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) { |
| return WebRtcIsacfix_SetMaxRate(inst, max_bit_rate); |
| } |
| }; |
| |
| typedef AudioEncoderDecoderIsacT<IsacFix> AudioEncoderDecoderIsacFix; |
| |
| struct CodecInst; |
| |
| class AudioEncoderDecoderMutableIsacFix |
| : public AudioEncoderMutableImpl<AudioEncoderDecoderIsacFix, |
| AudioEncoderDecoderMutableIsac> { |
| public: |
| explicit AudioEncoderDecoderMutableIsacFix(const CodecInst& codec_inst); |
| void UpdateSettings(const CodecInst& codec_inst) override; |
| void SetMaxPayloadSize(int max_payload_size_bytes) override; |
| void SetMaxRate(int max_rate_bps) override; |
| |
| // From AudioDecoder. |
| int Decode(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| size_t max_decoded_bytes, |
| int16_t* decoded, |
| SpeechType* speech_type) override; |
| int DecodeRedundant(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| size_t max_decoded_bytes, |
| int16_t* decoded, |
| SpeechType* speech_type) override; |
| bool HasDecodePlc() const override; |
| int DecodePlc(int num_frames, int16_t* decoded) override; |
| int Init() override; |
| int IncomingPacket(const uint8_t* payload, |
| size_t payload_len, |
| uint16_t rtp_sequence_number, |
| uint32_t rtp_timestamp, |
| uint32_t arrival_timestamp) override; |
| int ErrorCode() 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; |
| bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override; |
| size_t Channels() const override; |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_ |
