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