commit a959cbd062bbf32d16837554dd7d3b55fd69b321
author Linux Build Service Account <lnxbuild@localhost> Tue Sep 30 14:43:42 2014 -0700
committer Gerrit - the friendly Code Review server <code-review@localhost> Tue Sep 30 14:43:42 2014 -0700
tree a1b99a3d756499e5df13b9eeebd310f641075c0c
parent 93dd4e03134d378515ccd4f204d7df8dde2c18ee
parent a4a37d8c52a4cf45cbbef75247e9ff245bd136e8

Merge "hal: Fix backend for 24 bit playback on speaker"

hal/audio_extn/utils.c

diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index aa22193..5287ae8 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -472,6 +472,7 @@
     struct audio_device *adev;
     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;
 
     ALOGV("%s", __func__);
 
@@ -513,13 +514,21 @@
         rc = -EINVAL;
         goto exit_send_app_type_cfg;
     }
+
+    if ((24 == usecase->stream.out->bit_width) &&
+            (AUDIO_DEVICE_OUT_SPEAKER == snd_device)) {
+        sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+    } else {
+        sample_rate = out->app_type_cfg.sample_rate;
+    }
+
     app_type_cfg[len++] = out->app_type_cfg.app_type;
     app_type_cfg[len++] = acdb_dev_id;
-    app_type_cfg[len++] = out->app_type_cfg.sample_rate;
+    app_type_cfg[len++] = sample_rate;
 
     mixer_ctl_set_array(ctl, app_type_cfg, len);
     ALOGI("%s app_type %d, acdb_dev_id %d, sample_rate %d",
-           __func__, out->app_type_cfg.app_type, acdb_dev_id, out->app_type_cfg.sample_rate);
+           __func__, out->app_type_cfg.app_type, acdb_dev_id, sample_rate);
     rc = 0;
 exit_send_app_type_cfg:
     return rc;

hal/audio_hw.c

diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 57a1a1b..c5736fc 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -797,6 +797,27 @@
     usecase->in_snd_device = in_snd_device;
     usecase->out_snd_device = out_snd_device;
 
+    if (usecase->type == PCM_PLAYBACK) {
+        if ((24 == usecase->stream.out->bit_width) &&
+            (AUDIO_DEVICE_OUT_SPEAKER == usecase->stream.out->devices)) {
+            audio_extn_utils_update_stream_app_type_cfg(adev->platform,
+                                                &adev->streams_output_cfg_list,
+                                                usecase->stream.out->flags,
+                                                usecase->stream.out->format,
+                                                DEFAULT_OUTPUT_SAMPLING_RATE,
+                                                usecase->stream.out->bit_width,
+                                                &usecase->stream.out->app_type_cfg);
+        } else {
+            audio_extn_utils_update_stream_app_type_cfg(adev->platform,
+                                                &adev->streams_output_cfg_list,
+                                                usecase->stream.out->flags,
+                                                usecase->stream.out->format,
+                                                usecase->stream.out->sample_rate,
+                                                usecase->stream.out->bit_width,
+                                                &usecase->stream.out->app_type_cfg);
+        }
+    ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
+    }
     enable_audio_route(adev, usecase);
 
     /* Applicable only on the targets that has external modem.
@@ -2382,6 +2403,7 @@
     struct stream_out *out;
     int i, ret = 0;
     audio_format_t format;
+    int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
 
     *stream_out = NULL;
 
@@ -2567,11 +2589,20 @@
         out->sample_rate = out->config.rate;
     }
 
-    ALOGV("%s flags %x, format %x, out->sample_rate %d, out->bit_width %d",
-           __func__, flags, format, out->sample_rate, out->bit_width);
+    if ((24 == out->bit_width) &&
+        (devices == AUDIO_DEVICE_OUT_SPEAKER)) {
+        sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        ALOGI("%s 24-bit playback on Speaker is allowed ONLY at 48khz. Hence changing sample rate to: %d",
+               __func__, sample_rate);
+    } else {
+        sample_rate = out->sample_rate;
+    }
+
+    ALOGV("%s flags %x, format %x, sample_rate %d, out->bit_width %d",
+           __func__, flags, format, sample_rate, out->bit_width);
     audio_extn_utils_update_stream_app_type_cfg(adev->platform,
                                                 &adev->streams_output_cfg_list,
-                                                flags, format, out->sample_rate,
+                                                flags, format, sample_rate,
                                                 out->bit_width, &out->app_type_cfg);
     if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) ||
         (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {

hal/msm8974/platform.c

diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 9a2f87d..ff9076e 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -2560,8 +2560,10 @@
     }
 
     // 16 bit playbacks is allowed through 16 bit/48 khz backend only
-    if (16 == bit_width)
+    if ((16 == bit_width) ||
+        ((24 == bit_width) && (SND_DEVICE_OUT_SPEAKER == usecase->devices))) {
         sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+    }
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
     if ((bit_width != adev->cur_codec_backend_bit_width) ||