mirror of https://github.com/thesofproject/sof.git
70 lines
2.4 KiB
C
70 lines
2.4 KiB
C
// SPDX-License-Identifier: BSD-3-Clause
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//
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// Copyright(c) 2021 Google LLC.
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//
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// Author: Lionel Koenig <lionelk@google.com>
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#ifndef GOOGLE_RTC_AUDIO_PROCESSING_H
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#define GOOGLE_RTC_AUDIO_PROCESSING_H
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#include <stdint.h>
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#ifdef __cplusplus
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extern "C" {
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#endif
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// This define ensure that the linked library matches the header file.
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#define GoogleRtcAudioProcessingCreate GoogleRtcAudioProcessingCreate_v1
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typedef struct GoogleRtcAudioProcessingState GoogleRtcAudioProcessingState;
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// Creates an instance of GoogleRtcAudioProcessing with the tuning embedded in
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// the library.
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GoogleRtcAudioProcessingState *GoogleRtcAudioProcessingCreate(void);
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// Frees all allocated resources in `state`.
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void GoogleRtcAudioProcessingFree(GoogleRtcAudioProcessingState *state);
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// Returns the framesize used for processing.
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int GoogleRtcAudioProcessingGetFramesizeInMs(
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GoogleRtcAudioProcessingState *const state);
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// Reconfigure the audio processing.
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// Returns 0 if success and non zero if failure.
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int GoogleRtcAudioProcessingReconfigure(
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GoogleRtcAudioProcessingState *const state, const uint8_t *const config,
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int config_size);
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// Processes the microphone stream.
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// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
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// |src| points to an array of samples for the channel. At output, the channels
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// will be in |dest|.
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// Returns 0 if success and non zero if failure.
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int GoogleRtcAudioProcessingProcessCapture_float32(
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GoogleRtcAudioProcessingState *const state, const float *const *src,
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float *const *dest);
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// Accepts and and produces a frame of interleaved 16 bit integer audio. `src`
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// and `dest` may use the same memory, if desired.
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// Returns 0 if success and non zero if failure.
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int GoogleRtcAudioProcessingProcessCapture_int16(
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GoogleRtcAudioProcessingState *const state, const int16_t *const src,
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int16_t *const dest);
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// Analyzes the playback stream.
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// Accepts deinterleaved float audio with the range [-1, 1]. Each element
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// of |src| points to an array of samples for the channel.
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// Returns 0 if success and non zero if failure.
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int GoogleRtcAudioProcessingAnalyzeRender_float32(
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GoogleRtcAudioProcessingState *const state, const float *const *src);
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// Accepts interleaved int16 audio.
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// Returns 0 if success and non zero if failure.
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int GoogleRtcAudioProcessingAnalyzeRender_int16(
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GoogleRtcAudioProcessingState *const state, const int16_t *const src);
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#ifdef __cplusplus
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}
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#endif
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#endif // GOOGLE_RTC_AUDIO_PROCESSING_H
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