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+/*
+ * 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.
+ */
+
+#ifndef _YUKAWA_AUDIO_HW_H_
+#define _YUKAWA_AUDIO_HW_H_
+
+#include <hardware/audio.h>
+#include <tinyalsa/asoundlib.h>
+
+#include "fir_filter.h"
+
+#define PORT_HDMI 0
+#define PORT_INTERNAL_SPEAKER 1
+#define PORT_BUILTIN_MIC 3
+
+#define MIXER_XML_PATH "/vendor/etc/mixer_paths.xml"
+/* Minimum granularity - Arbitrary but small value */
+#define CODEC_BASE_FRAME_COUNT 32
+
+#define CHANNEL_STEREO 2
+
+#ifdef AEC_HAL
+#define NUM_AEC_REFERENCE_CHANNELS 1
+#else
+/* App AEC uses 2-channel reference */
+#define NUM_AEC_REFERENCE_CHANNELS 2
+#endif /* #ifdef AEC_HAL */
+
+#define DEBUG_AEC 0
+
+#define PCM_OPEN_RETRIES 100
+#define PCM_OPEN_WAIT_TIME_MS 20
+
+/* Capture codec parameters */
+/* Set up a capture period of 32 ms:
+ * CAPTURE_PERIOD = PERIOD_SIZE / SAMPLE_RATE, so (32e-3) = PERIOD_SIZE / (16e3)
+ * => PERIOD_SIZE = 512 frames, where each "frame" consists of 1 sample of every channel (here, 2ch) */
+#define CAPTURE_PERIOD_MULTIPLIER 16
+#define CAPTURE_PERIOD_SIZE (CODEC_BASE_FRAME_COUNT * CAPTURE_PERIOD_MULTIPLIER)
+#define CAPTURE_PERIOD_COUNT 4
+#define CAPTURE_PERIOD_START_THRESHOLD 0
+#define CAPTURE_CODEC_SAMPLING_RATE 16000
+
+/* Playback codec parameters */
+/* number of base blocks in a short period (low latency) */
+#define PLAYBACK_PERIOD_MULTIPLIER 32 /* 21 ms */
+/* number of frames per short period (low latency) */
+#define PLAYBACK_PERIOD_SIZE (CODEC_BASE_FRAME_COUNT * PLAYBACK_PERIOD_MULTIPLIER)
+/* number of pseudo periods for low latency playback */
+#define PLAYBACK_PERIOD_COUNT 4
+#define PLAYBACK_PERIOD_START_THRESHOLD 2
+#define PLAYBACK_CODEC_SAMPLING_RATE 48000
+#define MIN_WRITE_SLEEP_US 5000
+
+#define SPEAKER_EQ_FILE "/vendor/etc/speaker_eq.fir"
+#define SPEAKER_MAX_EQ_LENGTH 512
+
+struct alsa_audio_device {
+ struct audio_hw_device hw_device;
+
+ pthread_mutex_t lock; /* see notes in in_read/out_write on mutex acquisition order */
+ struct alsa_stream_in *active_input;
+ struct alsa_stream_out *active_output;
+ struct audio_route *audio_route;
+ struct mixer *mixer;
+ bool mic_mute;
+ struct aec_t *aec;
+};
+
+struct alsa_stream_in {
+ struct audio_stream_in stream;
+
+ pthread_mutex_t lock; /* see note in in_read() on mutex acquisition order */
+ audio_devices_t devices;
+ struct pcm_config config;
+ struct pcm *pcm;
+ bool unavailable;
+ bool standby;
+ struct alsa_audio_device *dev;
+ int read_threshold;
+ unsigned int frames_read;
+ uint64_t timestamp_nsec;
+ audio_source_t source;
+};
+
+struct alsa_stream_out {
+ struct audio_stream_out stream;
+
+ pthread_mutex_t lock; /* see note in out_write() on mutex acquisition order */
+ audio_devices_t devices;
+ struct pcm_config config;
+ struct pcm *pcm;
+ bool unavailable;
+ int standby;
+ struct alsa_audio_device *dev;
+ int write_threshold;
+ unsigned int frames_written;
+ struct timespec timestamp;
+ fir_filter_t* speaker_eq;
+};
+
+/* 'bytes' are the number of bytes written to audio FIFO, for which 'timestamp' is valid.
+ * 'available' is the number of frames available to read (for input) or yet to be played
+ * (for output) frames in the PCM buffer.
+ * timestamp and available are updated by pcm_get_htimestamp(), so they use the same
+ * datatypes as the corresponding arguments to that function. */
+struct aec_info {
+ struct timespec timestamp;
+ uint64_t timestamp_usec;
+ unsigned int available;
+ size_t bytes;
+};
+
+#endif /* #ifndef _YUKAWA_AUDIO_HW_H_ */