#ifndef DEEPSPEECH_H #define DEEPSPEECH_H #ifdef __ANDROID__ #define USE_TFLITE #endif #ifndef SWIG #if defined _MSC_VER #define DEEPSPEECH_EXPORT extern "C" __declspec(dllexport) #else /*End of _MSC_VER*/ #define DEEPSPEECH_EXPORT __attribute__ ((visibility("default"))) #endif /*End of SWIG*/ #else #define DEEPSPEECH_EXPORT #endif struct ModelState; struct StreamingState; /** * @brief An object providing an interface to a trained DeepSpeech model. * * @param aModelPath The path to the frozen model graph. * @param aNCep The number of cepstrum the model was trained with. * @param aNContext The context window the model was trained with. * @param aAlphabetConfigPath The path to the configuration file specifying * the alphabet used by the network. See alphabet.h. * @param aBeamWidth The beam width used by the decoder. A larger beam * width generates better results at the cost of decoding * time. * @param[out] retval a ModelState pointer * * @return Zero on success, non-zero on failure. */ DEEPSPEECH_EXPORT int DS_CreateModel(const char* aModelPath, unsigned int aNCep, unsigned int aNContext, const char* aAlphabetConfigPath, unsigned int aBeamWidth, ModelState** retval); /** * @brief Frees associated resources and destroys model object. */ DEEPSPEECH_EXPORT void DS_DestroyModel(ModelState* ctx); /** * @brief Enable decoding using beam scoring with a KenLM language model. * * @param aCtx The ModelState pointer for the model being changed. * @param aAlphabetConfigPath The path to the configuration file specifying * the alphabet used by the network. See alphabet.h. * @param aLMPath The path to the language model binary file. * @param aTriePath The path to the trie file build from the same vocabu- * lary as the language model binary. * @param aLMWeight The weight to give to language model results when sco- * ring. * @param aValidWordCountWeight The weight (bonus) to give to beams when * adding a new valid word to the decoding. * * @return Zero on success, non-zero on failure (invalid arguments). */ DEEPSPEECH_EXPORT int DS_EnableDecoderWithLM(ModelState* aCtx, const char* aAlphabetConfigPath, const char* aLMPath, const char* aTriePath, float aLMWeight, float aValidWordCountWeight); /** * @brief Use the DeepSpeech model to perform Speech-To-Text. * * @param aCtx The ModelState pointer for the model to use. * @param aBuffer A 16-bit, mono raw audio signal at the appropriate * sample rate. * @param aBufferSize The number of samples in the audio signal. * @param aSampleRate The sample-rate of the audio signal. * * @return The STT result. The user is responsible for freeing the string. * Returns NULL on error. */ DEEPSPEECH_EXPORT char* DS_SpeechToText(ModelState* aCtx, const short* aBuffer, unsigned int aBufferSize, unsigned int aSampleRate); /** * @brief Create a new streaming inference state. The streaming state returned * by this function can then be passed to {@link DS_FeedAudioContent()} * and {@link DS_FinishStream()}. * * @param aCtx The ModelState pointer for the model to use. * @param aPreAllocFrames Number of timestep frames to reserve. One timestep * is equivalent to two window lengths (20ms). If set to * 0 we reserve enough frames for 3 seconds of audio (150). * @param aSampleRate The sample-rate of the audio signal. * @param[out] retval an opaque pointer that represents the streaming state. Can * be NULL if an error occurs. * * @return Zero for success, non-zero on failure. */ DEEPSPEECH_EXPORT int DS_SetupStream(ModelState* aCtx, unsigned int aPreAllocFrames, unsigned int aSampleRate, StreamingState** retval); /** * @brief Feed audio samples to an ongoing streaming inference. * * @param aSctx A streaming state pointer returned by {@link DS_SetupStream()}. * @param aBuffer An array of 16-bit, mono raw audio samples at the * appropriate sample rate. * @param aBufferSize The number of samples in @p aBuffer. */ DEEPSPEECH_EXPORT void DS_FeedAudioContent(StreamingState* aSctx, const short* aBuffer, unsigned int aBufferSize); /** * @brief Compute the intermediate decoding of an ongoing streaming inference. * This is an expensive process as the decoder implementation isn't * currently capable of streaming, so it always starts from the beginning * of the audio. * * @param aSctx A streaming state pointer returned by {@link DS_SetupStream()}. * * @return The STT intermediate result. The user is responsible for freeing the * string. */ DEEPSPEECH_EXPORT char* DS_IntermediateDecode(StreamingState* aSctx); /** * @brief Signal the end of an audio signal to an ongoing streaming * inference, returns the STT result over the whole audio signal. * * @param aSctx A streaming state pointer returned by {@link DS_SetupStream()}. * * @return The STT result. The user is responsible for freeing the string. * * @note This method will free the state pointer (@p aSctx). */ DEEPSPEECH_EXPORT char* DS_FinishStream(StreamingState* aSctx); /** * @brief Destroy a streaming state without decoding the computed logits. This * can be used if you no longer need the result of an ongoing streaming * inference and don't want to perform a costly decode operation. * * @param aSctx A streaming state pointer returned by {@link DS_SetupStream()}. * * @note This method will free the state pointer (@p aSctx). */ DEEPSPEECH_EXPORT void DS_DiscardStream(StreamingState* aSctx); /** * @brief Given audio, return a vector suitable for input to a DeepSpeech * model trained with the given parameters. * * Extracts MFCC features from a given audio signal and adds the appropriate * amount of context to run inference on a DeepSpeech model trained with * the given parameters. * * @param aBuffer A 16-bit, mono raw audio signal at the appropriate sample * rate. * @param aBufferSize The sample-length of the audio signal. * @param aSampleRate The sample-rate of the audio signal. * @param aNCep The number of cepstrum. * @param aNContext The size of the context window. * @param[out] aMfcc An array containing features, of shape * (@p aNFrames, ncep * ncontext). The user is responsible * for freeing the array. * @param[out] aNFrames (optional) The number of frames in @p aMfcc. * @param[out] aFrameLen (optional) The length of each frame * (ncep * ncontext) in @p aMfcc. */ DEEPSPEECH_EXPORT void DS_AudioToInputVector(const short* aBuffer, unsigned int aBufferSize, unsigned int aSampleRate, unsigned int aNCep, unsigned int aNContext, float** aMfcc, int* aNFrames = NULL, int* aFrameLen = NULL); /** * @brief Print version of this library and of the linked TensorFlow library. */ DEEPSPEECH_EXPORT void DS_PrintVersions(); #undef DEEPSPEECH_EXPORT #endif /* DEEPSPEECH_H */