| /* |
| * 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_AUDIO_ENCODER_ISAC_T_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ |
| |
| #include <vector> |
| |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| |
| namespace webrtc { |
| |
| class CriticalSectionWrapper; |
| |
| template <typename T> |
| class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder { |
| public: |
| // For constructing an encoder in instantaneous mode. Allowed combinations |
| // are |
| // - 16000 Hz, 30 ms, 10000-32000 bps |
| // - 16000 Hz, 60 ms, 10000-32000 bps |
| // - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support) |
| // - 48000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support) |
| struct Config { |
| Config(); |
| bool IsOk() const; |
| int payload_type; |
| int red_payload_type; |
| int sample_rate_hz; |
| int frame_size_ms; |
| int bit_rate; // Limit on the short-term average bit rate, in bits/second. |
| int max_bit_rate; |
| int max_payload_size_bytes; |
| }; |
| |
| // For constructing an encoder in channel-adaptive mode. Allowed combinations |
| // are |
| // - 16000 Hz, 30 ms, 10000-32000 bps |
| // - 16000 Hz, 60 ms, 10000-32000 bps |
| // - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support) |
| // - 48000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support) |
| struct ConfigAdaptive { |
| ConfigAdaptive(); |
| bool IsOk() const; |
| int payload_type; |
| int red_payload_type; |
| int sample_rate_hz; |
| int initial_frame_size_ms; |
| int initial_bit_rate; |
| int max_bit_rate; |
| bool enforce_frame_size; // Prevent adaptive changes to the frame size? |
| int max_payload_size_bytes; |
| }; |
| |
| explicit AudioEncoderDecoderIsacT(const Config& config); |
| explicit AudioEncoderDecoderIsacT(const ConfigAdaptive& config); |
| virtual ~AudioEncoderDecoderIsacT() OVERRIDE; |
| |
| void UpdateDecoderSampleRate(int sample_rate_hz); |
| |
| // AudioEncoder public methods. |
| virtual int SampleRateHz() const OVERRIDE; |
| virtual int NumChannels() const OVERRIDE; |
| virtual int Num10MsFramesInNextPacket() const OVERRIDE; |
| virtual int Max10MsFramesInAPacket() const OVERRIDE; |
| |
| // AudioDecoder methods. |
| virtual int Decode(const uint8_t* encoded, |
| size_t encoded_len, |
| int16_t* decoded, |
| SpeechType* speech_type) OVERRIDE; |
| virtual int DecodeRedundant(const uint8_t* encoded, |
| size_t encoded_len, |
| int16_t* decoded, |
| SpeechType* speech_type) OVERRIDE; |
| virtual bool HasDecodePlc() const OVERRIDE; |
| virtual int DecodePlc(int num_frames, int16_t* decoded) OVERRIDE; |
| virtual int Init() OVERRIDE; |
| virtual int IncomingPacket(const uint8_t* payload, |
| size_t payload_len, |
| uint16_t rtp_sequence_number, |
| uint32_t rtp_timestamp, |
| uint32_t arrival_timestamp) OVERRIDE; |
| virtual int ErrorCode() OVERRIDE; |
| |
| protected: |
| // AudioEncoder protected method. |
| virtual bool EncodeInternal(uint32_t rtp_timestamp, |
| const int16_t* audio, |
| size_t max_encoded_bytes, |
| uint8_t* encoded, |
| EncodedInfo* info) OVERRIDE; |
| |
| private: |
| // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and |
| // STREAM_MAXW16_60MS for iSAC fix (60 ms). |
| static const size_t kSufficientEncodeBufferSizeBytes = 400; |
| |
| const int payload_type_; |
| const int red_payload_type_; |
| |
| // iSAC encoder/decoder state, guarded by a mutex to ensure that encode calls |
| // from one thread won't clash with decode calls from another thread. |
| // Note: PT_GUARDED_BY is disabled since it is not yet supported by clang. |
| const scoped_ptr<CriticalSectionWrapper> state_lock_; |
| typename T::instance_type* isac_state_ |
| GUARDED_BY(state_lock_) /* PT_GUARDED_BY(lock_)*/; |
| |
| // Must be acquired before state_lock_. |
| const scoped_ptr<CriticalSectionWrapper> lock_; |
| |
| // Have we accepted input but not yet emitted it in a packet? |
| bool packet_in_progress_ GUARDED_BY(lock_); |
| |
| // Timestamp of the first input of the currently in-progress packet. |
| uint32_t packet_timestamp_ GUARDED_BY(lock_); |
| |
| // Timestamp of the previously encoded packet. |
| uint32_t last_encoded_timestamp_ GUARDED_BY(lock_); |
| |
| // Redundant encoding from last time. |
| // Note: If has_redundant_encoder is false, we set the array length to 1, |
| // since zero-length arrays are not supported by all compilers. |
| uint8_t redundant_payload_[T::has_redundant_encoder |
| ? kSufficientEncodeBufferSizeBytes |
| : 1] GUARDED_BY(lock_); |
| size_t redundant_length_bytes_ GUARDED_BY(lock_); |
| |
| DISALLOW_COPY_AND_ASSIGN(AudioEncoderDecoderIsacT); |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ |