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-rw-r--r--audio_hw.c204
-rw-r--r--audio_hw.h16
-rw-r--r--audio_hw_profile.c67
-rw-r--r--audio_hw_profile.h20
-rw-r--r--audio_hw_utils.c157
-rw-r--r--audio_hw_utils.h20
-rw-r--r--audio_hwsync.c18
-rw-r--r--audio_hwsync.h18
8 files changed, 406 insertions, 114 deletions
diff --git a/audio_hw_utils.c b/audio_hw_utils.c
index 17febea..6eb4bd6 100644
--- a/audio_hw_utils.c
+++ b/audio_hw_utils.c
@@ -1,3 +1,21 @@
+/*
+ * Copyright (C) 2010 Amlogic Corporation.
+ *
+ * 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.
+ */
+
+
+
#define LOG_TAG "audio_hw_utils"
#define LOG_NDEBUG 0
@@ -24,6 +42,7 @@
#include "audio_hwsync.h"
#include "audio_hw.h"
+#include <audio_utils/primitives.h>
#ifdef LOG_NDEBUG_FUNCTION
#define LOGFUNC(...) ((void)0)
@@ -164,3 +183,141 @@ int getprop_bool (const char *path)
return 0;
}
+/*
+convert audio formats to supported audio format
+8 ch goes to 32 bit
+2 ch can be 16 bit or 32 bit
+@return input buffer used by alsa drivers to do the data write
+*/
+void *convert_audio_sample_for_output(int input_frames, int input_format, int input_ch, void *input_buf, int *out_size)
+{
+ float lvol = 1.0;
+ int *out_buf = NULL;
+ short *p16 = (short*)input_buf;
+ int *p32 = (int*)input_buf;
+ int max_ch = input_ch;
+ int i;
+ //ALOGV("intput frame %d,input ch %d,buf ptr %p,vol %f\n", input_frames, input_ch, input_buf, lvol);
+ ALOG_ASSERT(input_buf);
+ if (input_ch > 2)
+ max_ch = 8;
+ //our HW need round the frames to 8 channels
+ out_buf = malloc(sizeof(int) * max_ch * input_frames);
+ if (out_buf == NULL) {
+ ALOGE("malloc buffer failed\n");
+ return NULL;
+ }
+ switch (input_format) {
+ case AUDIO_FORMAT_PCM_16_BIT:
+ break;
+ case AUDIO_FORMAT_PCM_32_BIT:
+ break;
+ case AUDIO_FORMAT_PCM_8_24_BIT:
+ for (i = 0; i < input_frames * input_ch; i++) {
+ p32[i] = p32[i] << 8;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_i16_from_float(out_buf, input_buf, input_frames * input_ch);
+ memcpy(input_buf, out_buf, sizeof(short)*input_frames * input_ch);
+ break;
+ }
+ //current all the data are in the input buffer
+ if (input_ch == 8) {
+ short *p16_temp;
+ int i, NumSamps;
+ int *p32_temp = out_buf;
+ float m_vol = lvol;
+ NumSamps = input_frames * input_ch;
+ //here to swap the channnl data here
+ //actual now:L,missing,R,RS,RRS,,LS,LRS,missing
+ //expect L,C,R,RS,RRS,LRS,LS,LFE (LFE comes from to center)
+ //actual audio data layout L,R,C,none/LFE,LRS,RRS,LS,RS
+ if (input_format == AUDIO_FORMAT_PCM_16_BIT) {
+ p16_temp = (short*)out_buf;
+ for (i = 0; i < NumSamps; i = i + 8) {
+ p16_temp[0 + i]/*L*/ = m_vol * p16[0 + i];
+ p16_temp[1 + i]/*R*/ = m_vol * p16[1 + i];
+ p16_temp[2 + i] /*LFE*/ = m_vol * p16[3 + i];
+ p16_temp[3 + i] /*C*/ = m_vol * p16[2 + i];
+ p16_temp[4 + i] /*LS*/ = m_vol * p16[6 + i];
+ p16_temp[5 + i] /*RS*/ = m_vol * p16[7 + i];
+ p16_temp[6 + i] /*LRS*/ = m_vol * p16[4 + i];
+ p16_temp[7 + i]/*RRS*/ = m_vol * p16[5 + i];
+ }
+ memcpy(p16, p16_temp, NumSamps * sizeof(short));
+ for (i = 0; i < NumSamps; i++) { //suppose 16bit/8ch PCM
+ p32_temp[i] = p16[i] << 16;
+ }
+ } else {
+ p32_temp = out_buf;
+ for (i = 0; i < NumSamps; i = i + 8) {
+ p32_temp[0 + i]/*L*/ = m_vol * p32[0 + i];
+ p32_temp[1 + i]/*R*/ = m_vol * p32[1 + i];
+ p32_temp[2 + i] /*LFE*/ = m_vol * p32[3 + i];
+ p32_temp[3 + i] /*C*/ = m_vol * p32[2 + i];
+ p32_temp[4 + i] /*LS*/ = m_vol * p32[6 + i];
+ p32_temp[5 + i] /*RS*/ = m_vol * p32[7 + i];
+ p32_temp[6 + i] /*LRS*/ = m_vol * p32[4 + i];
+ p32_temp[7 + i]/*RRS*/ = m_vol * p32[5 + i];
+ }
+
+ }
+ *out_size = NumSamps * sizeof(int);
+
+ } else if (input_ch == 6) {
+ int j, NumSamps, real_samples;
+ short *p16_temp;
+ int *p32_temp = out_buf;
+ float m_vol = lvol;
+ NumSamps = input_frames * input_ch;
+ real_samples = NumSamps;
+ NumSamps = real_samples * 8 / 6;
+ //ALOGI("6ch to 8 ch real %d, to %d\n",real_samples,NumSamps);
+ if (input_format == AUDIO_FORMAT_PCM_16_BIT) {
+ p16_temp = (short*)out_buf;
+ for (i = 0; i < real_samples; i = i + 6) {
+ p16_temp[0 + i]/*L*/ = m_vol * p16[0 + i];
+ p16_temp[1 + i]/*R*/ = m_vol * p16[1 + i];
+ p16_temp[2 + i] /*LFE*/ = m_vol * p16[3 + i];
+ p16_temp[3 + i] /*C*/ = m_vol * p16[2 + i];
+ p16_temp[4 + i] /*LS*/ = m_vol * p16[4 + i];
+ p16_temp[5 + i] /*RS*/ = m_vol * p16[5 + i];
+ }
+ memcpy(p16, p16_temp, real_samples * sizeof(short));
+ memset(p32_temp, 0, NumSamps * sizeof(int));
+ for (i = 0, j = 0; j < NumSamps; i = i + 6, j = j + 8) { //suppose 16bit/8ch PCM
+ p32_temp[j + 0] = p16[i] << 16;
+ p32_temp[j + 1] = p16[i + 1] << 16;
+ p32_temp[j + 2] = p16[i + 2] << 16;
+ p32_temp[j + 3] = p16[i + 3] << 16;
+ p32_temp[j + 4] = p16[i + 4] << 16;
+ p32_temp[j + 5] = p16[i + 5] << 16;
+ }
+ } else {
+ p32_temp = out_buf;
+ memset(p32_temp, 0, NumSamps * sizeof(int));
+ for (i = 0, j = 0; j < NumSamps; i = i + 6, j = j + 8) { //suppose 16bit/8ch PCM
+ p32_temp[j + 0] = m_vol * p32[i + 0];
+ p32_temp[j + 1] = m_vol * p32[i + 1] ;
+ p32_temp[j + 2] = m_vol * p32[i + 2] ;
+ p32_temp[j + 3] = m_vol * p32[i + 3] ;
+ p32_temp[j + 4] = m_vol * p32[i + 4] ;
+ p32_temp[j + 5] = m_vol * p32[i + 5] ;
+ }
+ }
+ *out_size = NumSamps * sizeof(int);
+ } else {
+ //2ch with 24 bit/32/float audio
+ int *p32_temp = out_buf;
+ short *p16_temp = (short*)out_buf;
+ for (i = 0; i < input_frames; i++) {
+ p16_temp[2 * i + 0] = lvol * p16[2 * i + 0];
+ p16_temp[2 * i + 1] = lvol * p16[2 * i + 1];
+ }
+ *out_size = sizeof(short) * input_frames * input_ch;
+ }
+ return out_buf;
+
+}
+
generated by cgit at 2024-05-13 19:52:42 (GMT)