commit 837dc465d1ac780b2eb318976546a09bf3733ff9
author Manish Dewangan <manish@codeaurora.org> Wed May 27 10:17:41 2015 +0530
committer Gerrit - the friendly Code Review server <code-review@localhost> Thu Oct 29 02:45:42 2015 -0700
tree 6aa0e54714863e8c422bcbef61005fefce60c340
parent f9a1b8126a44ba7d673c67d4809f9d1b9f6a1f73

hal: Limit multichannel clips >48khz to 48Khz

-Configuring copp and backend at 48Khz for multichannel
clips > 48khz
-This is to address ADSP MIPS concern for playback of multichannel
 clip with sample rate > 48khz with pp (SA+ or SA+_HPX)

Change-Id: Ic010ce212011519e729601d067f8ddb6737c6b28

hal/audio_extn/audio_extn.h

diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 3b1d817..00ef85d 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -446,6 +446,7 @@
                                   audio_format_t format,
                                   uint32_t sample_rate,
                                   uint32_t bit_width,
+                                  audio_channel_mask_t channel_mask,
                                   struct stream_app_type_cfg *app_type_cfg);
 int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase);
 void audio_extn_utils_send_audio_calibration(struct audio_device *adev,

hal/audio_extn/utils.c

diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 531cf69..e917367 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -480,12 +480,14 @@
                                   audio_format_t format,
                                   uint32_t sample_rate,
                                   uint32_t bit_width,
+                                  audio_channel_mask_t channel_mask,
                                   struct stream_app_type_cfg *app_type_cfg)
 {
     struct listnode *node_i, *node_j, *node_k;
     struct streams_output_cfg *so_info;
     struct stream_format *sf_info;
     struct stream_sample_rate *ss_info;
+    char value[PROPERTY_VALUE_MAX] = {0};
 
     if ((24 == bit_width) &&
         (devices & AUDIO_DEVICE_OUT_SPEAKER)) {
@@ -496,6 +498,16 @@
         ALOGI("%s Allowing 24-bit playback on speaker ONLY at default sampling rate", __func__);
     }
 
+    property_get("audio.playback.mch.downsample",value,"");
+    if (!strncmp("true", value, sizeof("true"))) {
+        if ((popcount(channel_mask) > 2) &&
+                (sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
+                !(flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))  {
+                    sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                    ALOGD("%s: MCH session defaulting sample rate to %d",
+                               __func__, sample_rate);
+        }
+    }
     ALOGV("%s: flags: %x, format: %x sample_rate %d",
            __func__, flags, format, sample_rate);
     list_for_each(node_i, streams_output_cfg_list) {
@@ -537,6 +549,7 @@
     struct mixer_ctl *ctl;
     int pcm_device_id, acdb_dev_id, snd_device = usecase->out_snd_device;
     int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+    char value[PROPERTY_VALUE_MAX] = {0};
 
     ALOGV("%s", __func__);
 
@@ -589,6 +602,14 @@
     }
     sample_rate = out->app_type_cfg.sample_rate;
 
+    property_get("audio.playback.mch.downsample",value,"");
+    if (!strncmp("true", value, sizeof("true"))) {
+        if ((popcount(out->channel_mask) > 2) &&
+               (out->sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
+               !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
+           sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+    }
+
     app_type_cfg[len++] = out->app_type_cfg.app_type;
     app_type_cfg[len++] = acdb_dev_id;
     if (((out->format == AUDIO_FORMAT_E_AC3) ||