You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
133 lines
4.7 KiB
133 lines
4.7 KiB
4 months ago
|
/*
|
||
|
|
* Copyright (C) 2019 The Android Open Source Project
|
||
|
|
*
|
||
|
|
* Licensed under the Apache License, Version 2.0 (the "License");
|
||
|
|
* you may not use this file except in compliance with the License.
|
||
|
|
* You may obtain a copy of the License at
|
||
|
|
*
|
||
|
|
* http://www.apache.org/licenses/LICENSE-2.0
|
||
|
|
*
|
||
|
|
* Unless required by applicable law or agreed to in writing, software
|
||
|
|
* distributed under the License is distributed on an "AS IS" BASIS,
|
||
|
|
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
|
||
|
|
* See the License for the specific language governing permissions and
|
||
|
|
* limitations under the License.
|
||
|
|
*/
|
||
|
|
|
||
|
|
/*
|
||
|
|
* Definitions and interface related to HAL implementations of Acoustic Echo Canceller (AEC).
|
||
|
|
*
|
||
|
|
* AEC cleans the microphone signal by removing from it audio data corresponding to loudspeaker
|
||
|
|
* playback. Note that this process can be nonlinear.
|
||
|
|
*
|
||
|
|
*/
|
||
|
|
|
||
|
|
#ifndef _AUDIO_AEC_H_
|
||
|
|
#define _AUDIO_AEC_H_
|
||
|
|
|
||
|
|
#include <stdint.h>
|
||
|
|
#include <pthread.h>
|
||
|
|
#include <sys/time.h>
|
||
|
|
#include <hardware/audio.h>
|
||
|
|
#include <audio_utils/resampler.h>
|
||
|
|
#include "audio_hw.h"
|
||
|
|
#include "fifo_wrapper.h"
|
||
|
|
|
||
|
|
struct aec_t {
|
||
|
|
pthread_mutex_t lock;
|
||
|
|
size_t num_reference_channels;
|
||
|
|
bool mic_initialized;
|
||
|
|
int32_t *mic_buf;
|
||
|
|
size_t mic_num_channels;
|
||
|
|
size_t mic_buf_size_bytes;
|
||
|
|
size_t mic_frame_size_bytes;
|
||
|
|
uint32_t mic_sampling_rate;
|
||
|
|
struct aec_info last_mic_info;
|
||
|
|
bool spk_initialized;
|
||
|
|
int32_t *spk_buf;
|
||
|
|
size_t spk_num_channels;
|
||
|
|
size_t spk_buf_size_bytes;
|
||
|
|
size_t spk_frame_size_bytes;
|
||
|
|
uint32_t spk_sampling_rate;
|
||
|
|
struct aec_info last_spk_info;
|
||
|
|
int16_t *spk_buf_playback_format;
|
||
|
|
int16_t *spk_buf_resampler_out;
|
||
|
|
void *spk_fifo;
|
||
|
|
void *ts_fifo;
|
||
|
|
ssize_t read_write_diff_bytes;
|
||
|
|
struct resampler_itfe *spk_resampler;
|
||
|
|
bool spk_running;
|
||
|
|
bool prev_spk_running;
|
||
|
|
};
|
||
|
|
|
||
|
|
/* Initialize AEC object.
|
||
|
|
* This must be called when the audio device is opened.
|
||
|
|
* ALSA device mutex must be held before calling this API.
|
||
|
|
* Returns -EINVAL if AEC object fails to initialize, else returns 0. */
|
||
|
|
int init_aec (int sampling_rate, int num_reference_channels,
|
||
|
|
int num_microphone_channels, struct aec_t **);
|
||
|
|
|
||
|
|
/* Release AEC object.
|
||
|
|
* This must be called when the audio device is closed. */
|
||
|
|
void release_aec(struct aec_t* aec);
|
||
|
|
|
||
|
|
/* Initialize reference configuration for AEC.
|
||
|
|
* Must be called when a new output stream is opened.
|
||
|
|
* Returns -EINVAL if any processing block fails to initialize,
|
||
|
|
* else returns 0. */
|
||
|
|
int init_aec_reference_config (struct aec_t *aec, struct alsa_stream_out *out);
|
||
|
|
|
||
|
|
/* Clear reference configuration for AEC.
|
||
|
|
* Must be called when the output stream is closed. */
|
||
|
|
void destroy_aec_reference_config (struct aec_t *aec);
|
||
|
|
|
||
|
|
/* Initialize microphone configuration for AEC.
|
||
|
|
* Must be called when a new input stream is opened.
|
||
|
|
* Returns -EINVAL if any processing block fails to initialize,
|
||
|
|
* else returns 0. */
|
||
|
|
int init_aec_mic_config(struct aec_t* aec, struct alsa_stream_in* in);
|
||
|
|
|
||
|
|
/* Clear microphone configuration for AEC.
|
||
|
|
* Must be called when the input stream is closed. */
|
||
|
|
void destroy_aec_mic_config (struct aec_t *aec);
|
||
|
|
|
||
|
|
/* Used to communicate playback state (running or not) to AEC interface.
|
||
|
|
* This is used by process_aec() to determine if AEC processing is to be run. */
|
||
|
|
void aec_set_spk_running (struct aec_t *aec, bool state);
|
||
|
|
|
||
|
|
/* Used to communicate playback state (running or not) to the caller. */
|
||
|
|
bool aec_get_spk_running(struct aec_t* aec);
|
||
|
|
|
||
|
|
/* Write audio samples to AEC reference FIFO for use in AEC.
|
||
|
|
* Both audio samples and timestamps are added in FIFO fashion.
|
||
|
|
* Must be called after every write to PCM.
|
||
|
|
* Returns -ENOMEM if the write fails, else returns 0. */
|
||
|
|
int write_to_reference_fifo(struct aec_t* aec, void* buffer, struct aec_info* info);
|
||
|
|
|
||
|
|
/* Get reference audio samples + timestamp, in the format expected by AEC,
|
||
|
|
* i.e. same sample rate and bit rate as microphone audio.
|
||
|
|
* Timestamp is updated in field 'timestamp_usec', and not in 'timestamp'.
|
||
|
|
* Returns:
|
||
|
|
* -EINVAL if the AEC object is invalid.
|
||
|
|
* -ENOMEM if the reference FIFO overflows or is corrupted.
|
||
|
|
* -ETIMEDOUT if we timed out waiting for the requested number of bytes
|
||
|
|
* 0 otherwise */
|
||
|
|
int get_reference_samples(struct aec_t* aec, void* buffer, struct aec_info* info);
|
||
|
|
|
||
|
|
#ifdef AEC_HAL
|
||
|
|
|
||
|
|
/* Processing function call for AEC.
|
||
|
|
* AEC output is updated at location pointed to by 'buffer'.
|
||
|
|
* This function does not run AEC when there is no playback -
|
||
|
|
* as communicated to this AEC interface using aec_set_spk_running().
|
||
|
|
* Returns -EINVAL if processing fails, else returns 0. */
|
||
|
|
int process_aec(struct aec_t* aec, void* buffer, struct aec_info* info);
|
||
|
|
|
||
|
|
#else /* #ifdef AEC_HAL */
|
||
|
|
|
||
|
|
#define process_aec(...) ((int)0)
|
||
|
|
|
||
|
|
#endif /* #ifdef AEC_HAL */
|
||
|
|
|
||
|
|
#endif /* _AUDIO_AEC_H_ */
|