audio: Kang audio HAL from amlogic yukawa device

Copied from device/amlogic/generic/hal/audio/ minus the
speaker equalizer filter file.

It is a generic and more feature rich HAL, plus it fixes
the occasional static noise we hear on db845c and PocoF1.

Removed the mixer control settings from qcom/init.qcom.rc
and added HDMI out mixer control in etc/mixer_paths.xml
to start with. ** Analog audio out is not tested yet. **

ToDo:
1. Fallback to Analog audio port if the monitor doesn't
   support HDMI audio out.
2. Mic AEC support is integrated deep in this HAL, so I
   didn't remove it. Might as well test and use that
   feature on Pixel 3 and Poco F1. Same with speaker EQ,
   though we may end up never using this equalizer
   feature at all.

Signed-off-by: Amit Pundir <amit.pundir@linaro.org>
Change-Id: I85614abdd684ab67f405f4c0e48380668ade4e06

1 files changed, 129 insertions, 0 deletions

diff --git a/audio/audio_hw.h b/audio/audio_hw.h
new file mode 100644
index 0000000..3e8e27c
--- /dev/null
+++ b/audio/audio_hw.h
@@ -0,0 +1,129 @@
+/*
+ * 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 CARD_OUT 0
+#define PORT_HDMI 0
+#define PORT_INTERNAL_SPEAKER 1
+#define CARD_IN 0
+#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_ */