From 11ee3ea7d80c541bfeb9f7fe71d41e5bde507001 Mon Sep 17 00:00:00 2001
From: SungHwan Yu <Sh.Yu@amlogic.com>
Date: Thu, 4 Jan 2024 10:54:55 +0900
Subject: [PATCH] audio: add data handle process when pcm unstandby

PD#OTT-52950

Problem:
AUDIO-MEDIAVOL-HEAAC-TC3 wavform is not perfect

Solution:
add data handle process when pcm unstandby

Verify:
S905X2

Change-Id: I52e44f7e5bb7020c238a68151534e89602526414
Signed-off-by: zeming.huang <Zeming.Huang@amlogic.com>
---
 audio/audio_hal/audio_hw.c           |  14 +++
 audio/audio_hal/audio_hw.h           |  17 ++++
 audio/audio_hal/audio_hw_utils.c     | 140 +++++++++++++++++++++++++++
 audio/audio_hal/audio_hw_utils.h     |   2 +
 audio/audio_hal/sub_mixing_factory.c |   7 ++
 5 files changed, 180 insertions(+)

diff --git a/audio/audio_hal/audio_hw.c b/audio/audio_hal/audio_hw.c
index 649bc46..f31b268 100755
--- a/audio/audio_hal/audio_hw.c
+++ b/audio/audio_hal/audio_hw.c
@@ -10272,6 +10272,13 @@ int adev_open_output_stream_new(struct audio_hw_device *dev,
     adev->active_outputs[aml_out->usecase] = aml_out;
     pthread_mutex_unlock(&adev->lock);
 
+    /* init ease for stream */
+    if (aml_audio_ease_init(&aml_out->audio_stream_ease) < 0) {
+        ALOGE("%s  aml_audio_ease_init failed\n", __func__);
+        ret = -EINVAL;
+        goto AUDIO_EASE_INIT_FAIL;
+    }
+
     if (aml_getprop_bool("vendor.media.audio.hal.debug")) {
         aml_out->debug_stream = 1;
     }
@@ -10279,6 +10286,10 @@ int adev_open_output_stream_new(struct audio_hw_device *dev,
         aml_out, usecase2Str(aml_out->usecase), aml_out->card, aml_out->device);
 
     return 0;
+
+AUDIO_EASE_INIT_FAIL:
+    adev_close_output_stream(dev, *stream_out);
+    return ret;
 }
 
 void adev_close_output_stream_new(struct audio_hw_device *dev,
@@ -10291,6 +10302,9 @@ void adev_close_output_stream_new(struct audio_hw_device *dev,
     /* call legacy close to reuse codes */
     adev->active_outputs[aml_out->usecase] = NULL;
 
+    /* free stream ease resource  */
+    aml_audio_ease_close(aml_out->audio_stream_ease);
+
     if (adev->useSubMix) {
         if (aml_out->is_normal_pcm ||
             aml_out->usecase == STREAM_PCM_HWSYNC ||
diff --git a/audio/audio_hal/audio_hw.h b/audio/audio_hal/audio_hw.h
index b7be540..dcb882d 100755
--- a/audio/audio_hal/audio_hw.h
+++ b/audio/audio_hal/audio_hw.h
@@ -581,6 +581,19 @@ typedef union output_info {
 } output_info_t;
 
 
+typedef enum audio_data_handle_state {
+    AUDIO_DATA_HANDLE_NONE = 0,
+    AUDIO_DATA_HANDLE_START,
+    AUDIO_DATA_HANDLE_DETECT,
+    AUDIO_DATA_HANDLE_DETECTED,
+    AUDIO_DATA_HANDLE_EASE_CONFIG,
+    AUDIO_DATA_HANDLE_EASING,
+    AUDIO_DATA_HANDLE_FINISHED,
+
+    AUDIO_DATA_HANDLE_MAX
+} audio_data_handle_state_t;
+
+
 struct aml_stream_out {
     struct audio_stream_out stream;
     /* see note below on mutex acquisition order */
@@ -711,6 +724,10 @@ struct aml_stream_out {
     int dtvsync_enable;
     bool is_ms12_main_decoder;
     pthread_mutex_t apts_update_lock; /*SWPL-88828: Make sure audio timestamps and frame positions are updated synchronously.*/
+
+    aml_audio_ease_t  *audio_stream_ease;
+    audio_data_handle_state_t audio_data_handle_state;
+    uint16_t easing_time;
 };
 
 typedef ssize_t (*write_func)(struct audio_stream_out *stream, const void *buffer, size_t bytes);
diff --git a/audio/audio_hal/audio_hw_utils.c b/audio/audio_hal/audio_hw_utils.c
index ab66578..718fe1c 100755
--- a/audio/audio_hal/audio_hw_utils.c
+++ b/audio/audio_hal/audio_hw_utils.c
@@ -48,6 +48,7 @@
 #include "a2dp_hal.h"
 #include "aml_audio_avsync_table.h"
 #include "dolby_lib_api.h"
+#include <inttypes.h>
 
 #ifdef LOG_NDEBUG_FUNCTION
 #define LOGFUNC(...) ((void)0)
@@ -1458,3 +1459,142 @@ void UpdateDecodeInfo_ChannelConfiguration(char *sysfs_buf, int ch_num) {
     }
 }
 
+/*****************************************************************************
+*   Function Name:  aml_audio_data_detect
+*   Description:    accumulate buf in bytes, then compare with detect_value.
+*                   the purpose is to detect value of buf that is zero or not.
+*   Parameters:     int16_t *: the audio data buffer.
+*                   size_t bytes: the buffer length.
+*                   int: for compared value.
+*   Return value:   true if buf value is zero, or false.
+******************************************************************************/
+bool aml_audio_data_detect(int16_t *buf, size_t bytes, int detect_value)
+{
+    int ret = false;
+    uint64_t buf_value = 0;
+    uint32_t i = 0;
+    int8_t *temp_buf = (int8_t *)buf;
+
+    while (i < bytes) {
+        buf_value +=  abs(temp_buf[i++]);
+    };
+
+    ALOGV("%s bytes:%zu i:%u  buf_value:%" PRIu64 "  sizeof(uint64_t):%zu sizeof(size_t):%zu", __func__,
+                bytes, i, buf_value,  sizeof(uint64_t), sizeof(size_t));
+    if (buf_value <= detect_value) {
+        ret = true;
+    } else {
+        ret = false;
+    }
+
+    return ret;
+}
+
+static int mixer_aux_start_ease_in(struct aml_stream_out *aml_out) {
+    /*start ease in the audio*/
+    ease_setting_t ease_setting;
+    aml_out->audio_stream_ease->data_format.format = aml_out->hal_format;
+    aml_out->audio_stream_ease->data_format.ch = aml_out->hal_ch;
+    aml_out->audio_stream_ease->data_format.sr = aml_out->hal_rate;
+    aml_out->audio_stream_ease->ease_type = EaseLinear;
+    ease_setting.duration = 40;
+    ease_setting.start_volume = 0.0;
+    ease_setting.target_volume = 1.0;
+    aml_audio_ease_config(aml_out->audio_stream_ease, &ease_setting);
+
+    ALOGV("%s ", __func__);
+    return 0;
+}
+
+/*****************************************************************************
+*   Function Name:  aml_audio_data_handle
+*   Description:    handle audio data before send to driver or decoder.
+*                   the purpose is to detect and fade in.
+*   Parameters:     struct audio_stream_out: audio output stream pointer.
+*                   const void *: the buffer pointer.
+*                   size_t: the buffer length.
+*   Return value:   true if buf value is zero, or false.
+******************************************************************************/
+int aml_audio_data_handle(struct audio_stream_out *stream, const void* buffer, size_t bytes)
+{
+// 8ms audio data
+#define DETECT_AUDIO_TIME_UNIT (8)
+#define DETECT_AUDIO_DATA_UNIT (DETECT_AUDIO_TIME_UNIT * out->hal_frame_size * out->hal_rate / 1000)
+/* value 2000 for filter noise data,
+** this value is confirmed according logs.
+*/
+#define AML_DETECT_VALUE (2000)
+
+    struct aml_stream_out *out = (struct aml_stream_out *)stream;
+    struct aml_audio_device *adev = out->dev;
+    bool ret = false;
+    int unit_size = 0;
+    int detected_size = 0;
+    size_t remaining_size = bytes;
+    audio_data_handle_state_t data_handle_state = out->audio_data_handle_state;
+
+    AM_LOGV("out_stream usecase:%d-->%s, hal_format:%#x hal_ch:%u --> hal_frame_size:%u, hal_rate:%u, DETECT_AUDIO_DATA_UNIT:%u, bytes:%zu",
+          out->usecase, usecase2Str(out->usecase), out->hal_format, out->hal_ch, out->hal_frame_size, out->hal_rate, DETECT_AUDIO_DATA_UNIT, bytes);
+    if ((data_handle_state == AUDIO_DATA_HANDLE_NONE) || (data_handle_state == AUDIO_DATA_HANDLE_MAX)) {
+        AM_LOGE("invalid audio_data_handle_state %d", data_handle_state);
+        return -1;
+    }
+
+    while (out->audio_data_handle_state < AUDIO_DATA_HANDLE_FINISHED && remaining_size) {
+        AM_LOGD("remaining_size:%zu,  out->audio_data_handle_status:%u", remaining_size, out->audio_data_handle_state);
+        switch (out->audio_data_handle_state) {
+            case AUDIO_DATA_HANDLE_START:
+                FALLTHROUGH_INTENDED; /* [[fallthrough]] */
+            case AUDIO_DATA_HANDLE_DETECT:
+                out->audio_data_handle_state = AUDIO_DATA_HANDLE_DETECT;
+                while (remaining_size > 0) {
+                    if (remaining_size > DETECT_AUDIO_DATA_UNIT) {
+                        unit_size = DETECT_AUDIO_DATA_UNIT;
+                    } else {
+                        unit_size = remaining_size;
+                    }
+
+                    ret = aml_audio_data_detect((int16_t *)((int8_t *)buffer + detected_size), unit_size , AML_DETECT_VALUE);
+                    if (false == ret) {
+                        out->audio_data_handle_state = AUDIO_DATA_HANDLE_DETECTED;
+                        AM_LOGD("detected the nonzero data, remaining_size:%zu  detected_size:%u", remaining_size, detected_size);
+                        break;
+                    } else {
+                        // when ease_setting.duration = 0, aml_audio_ease_process will not do easing.
+                        memset((int8_t *)buffer + detected_size, 0, unit_size);
+                    }
+
+                    remaining_size -= unit_size;
+                    detected_size += unit_size;
+                }
+                break;
+            // detect finished, then do fade in.
+            case AUDIO_DATA_HANDLE_DETECTED:
+                out->audio_data_handle_state = AUDIO_DATA_HANDLE_EASE_CONFIG;
+                break;
+            case AUDIO_DATA_HANDLE_EASE_CONFIG:
+                mixer_aux_start_ease_in(out);
+                out->easing_time = 0;
+                out->audio_data_handle_state = AUDIO_DATA_HANDLE_EASING;
+                break;
+            case AUDIO_DATA_HANDLE_EASING:
+                aml_audio_ease_process(out->audio_stream_ease, (void *)((uint8_t *)buffer + detected_size), remaining_size);
+                out->easing_time += remaining_size/(out->hal_frame_size * out->hal_rate / 1000);
+                AM_LOGD("easing_time:%u, audio_stream_ease->ease_time:%u", out->easing_time, out->audio_stream_ease->ease_time);
+                remaining_size = 0;
+                if (out->easing_time >=  out->audio_stream_ease->ease_time) {
+                    out->audio_data_handle_state = AUDIO_DATA_HANDLE_FINISHED;
+                }
+                break;
+            case AUDIO_DATA_HANDLE_FINISHED:
+                out->audio_data_handle_state = AUDIO_DATA_HANDLE_FINISHED;
+                AM_LOGD("handle finished");
+                break;
+            default :
+                break;
+        };
+    }
+
+    return 0;
+}
+
diff --git a/audio/audio_hal/audio_hw_utils.h b/audio/audio_hal/audio_hw_utils.h
index ffe88ed..01939ab 100755
--- a/audio/audio_hal/audio_hw_utils.h
+++ b/audio/audio_hal/audio_hw_utils.h
@@ -156,4 +156,6 @@ void UpdateDecodedInfo_SampleRate_ChannelNum_ChannelConfiguration(int samplerate
 
 void UpdateDecodeInfo_ChannelConfiguration(char *sysfs_buf, int ch_num);
 
+int aml_audio_data_handle(struct audio_stream_out *stream, const void* buffer, size_t bytes);
+
 #endif
diff --git a/audio/audio_hal/sub_mixing_factory.c b/audio/audio_hal/sub_mixing_factory.c
index f998586..6c55e9e 100755
--- a/audio/audio_hal/sub_mixing_factory.c
+++ b/audio/audio_hal/sub_mixing_factory.c
@@ -150,6 +150,10 @@ static ssize_t aml_out_write_to_mixer(struct audio_stream_out *stream, const voi
     uint32_t latency_frames = 0;
     struct timespec ts;
 
+    if (adev->is_netflix && (STREAM_PCM_NORMAL == out->usecase || STREAM_PCM_HWSYNC == out->usecase)) {
+        aml_audio_data_handle(stream, buffer, bytes);
+    }
+
     do {
         ssize_t written = 0;
         ALOGV("%s(), stream usecase: %s, written_total %d, bytes %d",
@@ -391,6 +395,7 @@ static ssize_t out_write_hwsync_lpcm(struct audio_stream_out *stream, const void
                 get_input_port_type(&out->audioCfg, out->flags));
         out->standby = false;
         mixer_set_continuous_output(sm->mixerData, false);
+		out->audio_data_handle_state = AUDIO_DATA_HANDLE_START;
         /*wait video ready*/
         {
             int vframe_ready_cnt = 0;
@@ -564,6 +569,7 @@ static ssize_t out_write_direct_pcm(struct audio_stream_out *stream, const void
         ALOGI("[%s:%d] direct port:%s", __func__, __LINE__,
                 inportType2Str(get_input_port_type(&out->audioCfg, out->flags)));
         out->standby = false;
+		out->audio_data_handle_state = AUDIO_DATA_HANDLE_START;
     }
 
     clock_gettime(CLOCK_MONOTONIC, &tval);
@@ -1195,6 +1201,7 @@ ssize_t mixer_aux_buffer_write_sm(struct audio_stream_out *stream, const void *b
         if (aml_out->out_device & AUDIO_DEVICE_OUT_ALL_A2DP)
             padding_bytes = 0;
 
+		aml_out->audio_data_handle_state = AUDIO_DATA_HANDLE_START;
         //set_thread_affinity();
         init_mixer_input_port(sm->mixerData, &aml_out->audioCfg, aml_out->flags,
             on_notify_cbk, aml_out, on_input_avail_cbk, aml_out,
-- 
2.17.1