hal: get correspoding sample rate per device when sending cal data

For combo devices, each of the sound device may have different
sample rate, so calibration data should be sent with its own
sample rate value.
Use the sample rate per sound device for calibration.

Change-Id: If1989b8ceb166a905f3e438161765b4b3a6dc5f5
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 09921af..ea0d324 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -1025,6 +1025,87 @@
     return rc;
 }
 
+int audio_extn_utils_get_app_sample_rate_for_device(
+                              struct audio_device *adev,
+                              struct audio_usecase *usecase, int snd_device)
+{
+    char value[PROPERTY_VALUE_MAX] = {0};
+    int sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+
+    if ((usecase->type == PCM_PLAYBACK) && (usecase->stream.out != NULL)) {
+        property_get("vendor.audio.playback.mch.downsample",value,"");
+        if (!strncmp("true", value, sizeof("true"))) {
+            if ((popcount(usecase->stream.out->channel_mask) > 2) &&
+                (usecase->stream.out->app_type_cfg.sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
+                !(usecase->stream.out->flags &
+                            (audio_output_flags_t)AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
+               sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        }
+
+        if (usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
+            usecase->stream.out->app_type_cfg.sample_rate = usecase->stream.out->sample_rate;
+        } else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
+            usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        } else if ((snd_device == SND_DEVICE_OUT_HDMI ||
+                    snd_device == SND_DEVICE_OUT_USB_HEADSET ||
+                    snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
+                   (usecase->stream.out->sample_rate >= OUTPUT_SAMPLING_RATE_44100)) {
+             /*
+              * To best utlize DSP, check if the stream sample rate is supported/multiple of
+              * configured device sample rate, if not update the COPP rate to be equal to the
+              * device sample rate, else open COPP at stream sample rate
+              */
+              platform_check_and_update_copp_sample_rate(adev->platform, snd_device,
+                                      usecase->stream.out->sample_rate,
+                                      &usecase->stream.out->app_type_cfg.sample_rate);
+        } else if (((snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&
+                     !audio_is_this_native_usecase(usecase)) &&
+            usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) ||
+            (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
+            /* Reset to default if no native stream is active*/
+            usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        } else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+                 /*
+                  * For a2dp playback get encoder sampling rate and set copp sampling rate,
+                  * for bit width use the stream param only.
+                  */
+                   audio_extn_a2dp_get_enc_sample_rate(&usecase->stream.out->app_type_cfg.sample_rate);
+                   ALOGI("%s using %d sample rate rate for A2DP CoPP",
+                        __func__, usecase->stream.out->app_type_cfg.sample_rate);
+        }
+        audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.out->app_type_cfg.sample_rate);
+        sample_rate = usecase->stream.out->app_type_cfg.sample_rate;
+
+        if (((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
+            (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC) ||
+            (usecase->stream.out->format == AUDIO_FORMAT_DOLBY_TRUEHD))
+            && audio_extn_passthru_is_passthrough_stream(usecase->stream.out)
+            && !audio_extn_passthru_is_convert_supported(adev, usecase->stream.out)) {
+            sample_rate = sample_rate * 4;
+            if (sample_rate > HDMI_PASSTHROUGH_MAX_SAMPLE_RATE)
+                sample_rate = HDMI_PASSTHROUGH_MAX_SAMPLE_RATE;
+        }
+    } else if (usecase->type == PCM_CAPTURE) {
+        if (usecase->stream.in != NULL) {
+            if (usecase->id == USECASE_AUDIO_RECORD_VOIP)
+                usecase->stream.in->app_type_cfg.sample_rate = usecase->stream.in->sample_rate;
+            if (voice_is_in_call_rec_stream(usecase->stream.in)) {
+                audio_extn_btsco_get_sample_rate(usecase->in_snd_device,
+                                                 &usecase->stream.in->app_type_cfg.sample_rate);
+            } else {
+                audio_extn_btsco_get_sample_rate(snd_device,
+                                                 &usecase->stream.in->app_type_cfg.sample_rate);
+            }
+            sample_rate = usecase->stream.in->app_type_cfg.sample_rate;
+        } else if (usecase->id == USECASE_AUDIO_SPKR_CALIB_TX) {
+            sample_rate = SAMPLE_RATE_8000;
+        }
+    } else if (usecase->type == TRANSCODE_LOOPBACK_RX) {
+        sample_rate = usecase->stream.inout->out_config.sample_rate;
+    }
+    return sample_rate;
+}
+
 static int send_app_type_cfg_for_device(struct audio_device *adev,
                                         struct audio_usecase *usecase,
                                         int split_snd_device)
@@ -1036,7 +1117,6 @@
     int pcm_device_id = 0, acdb_dev_id, app_type;
     int snd_device = split_snd_device, snd_device_be_idx = -1;
     int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
-    char value[PROPERTY_VALUE_MAX] = {0};
     struct streams_io_cfg *s_info = NULL;
     struct listnode *node = NULL;
     int bd_app_type = 0;
@@ -1111,58 +1191,9 @@
               snd_device_be_idx);
     }
 
+    sample_rate = audio_extn_utils_get_app_sample_rate_for_device(adev, usecase, snd_device);
+
     if ((usecase->type == PCM_PLAYBACK) && (usecase->stream.out != NULL)) {
-
-        property_get("vendor.audio.playback.mch.downsample",value,"");
-        if (!strncmp("true", value, sizeof("true"))) {
-            if ((popcount(usecase->stream.out->channel_mask) > 2) &&
-                   (usecase->stream.out->app_type_cfg.sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
-                   !(usecase->stream.out->flags &
-                            (audio_output_flags_t)AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
-               sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-        }
-
-        if (usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
-            usecase->stream.out->app_type_cfg.sample_rate = usecase->stream.out->sample_rate;
-        } else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
-            if (platform_spkr_use_default_sample_rate(adev->platform)) {
-                 usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
-            } else {
-                 platform_check_and_update_copp_sample_rate(adev->platform, snd_device,
-                                      usecase->stream.out->sample_rate,
-                                      &usecase->stream.out->app_type_cfg.sample_rate);
-            }
-
-        } else if ((snd_device == SND_DEVICE_OUT_HDMI ||
-                    snd_device == SND_DEVICE_OUT_USB_HEADSET ||
-                    snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
-                   (usecase->stream.out->sample_rate >= OUTPUT_SAMPLING_RATE_44100)) {
-             /*
-              * To best utlize DSP, check if the stream sample rate is supported/multiple of
-              * configured device sample rate, if not update the COPP rate to be equal to the
-              * device sample rate, else open COPP at stream sample rate
-              */
-              platform_check_and_update_copp_sample_rate(adev->platform, snd_device,
-                                      usecase->stream.out->sample_rate,
-                                      &usecase->stream.out->app_type_cfg.sample_rate);
-        } else if (((snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&
-                     !audio_is_this_native_usecase(usecase)) &&
-            usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) ||
-            (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
-            /* Reset to default if no native stream is active*/
-            usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
-        } else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
-                 /*
-                  * For a2dp playback get encoder sampling rate and set copp sampling rate,
-                  * for bit width use the stream param only.
-                  */
-                   audio_extn_a2dp_get_enc_sample_rate(&usecase->stream.out->app_type_cfg.sample_rate);
-                   ALOGI("%s using %d sample rate rate for A2DP CoPP",
-                        __func__, usecase->stream.out->app_type_cfg.sample_rate);
-        }
-        audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.out->app_type_cfg.sample_rate);
-        sample_rate = usecase->stream.out->app_type_cfg.sample_rate;
-
         /* Interactive streams are supported with only direct app type id.
          * Get Direct profile app type and use it for interactive streams
          */
@@ -1179,16 +1210,6 @@
             app_type = usecase->stream.out->app_type_cfg.app_type;
         app_type_cfg[len++] = app_type;
         app_type_cfg[len++] = acdb_dev_id;
-        if (((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
-            (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC) ||
-            (usecase->stream.out->format == AUDIO_FORMAT_DOLBY_TRUEHD))
-            && audio_extn_passthru_is_passthrough_stream(usecase->stream.out)
-            && !audio_extn_passthru_is_convert_supported(adev, usecase->stream.out)) {
-
-            sample_rate = sample_rate * 4;
-            if (sample_rate > HDMI_PASSTHROUGH_MAX_SAMPLE_RATE)
-                sample_rate = HDMI_PASSTHROUGH_MAX_SAMPLE_RATE;
-        }
         app_type_cfg[len++] = sample_rate;
 
         if (snd_device_be_idx > 0)
@@ -1201,19 +1222,6 @@
         app_type = usecase->stream.in->app_type_cfg.app_type;
         app_type_cfg[len++] = app_type;
         app_type_cfg[len++] = acdb_dev_id;
-        if (usecase->id == USECASE_AUDIO_RECORD_VOIP)
-            usecase->stream.in->app_type_cfg.sample_rate = usecase->stream.in->sample_rate;
-        if (voice_is_in_call_rec_stream(usecase->stream.in)) {
-            audio_extn_btsco_get_sample_rate(usecase->in_snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
-        } else {
-            audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
-        }
-        if (usecase->stream.in->device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP & ~AUDIO_DEVICE_BIT_IN) {
-            audio_extn_a2dp_get_dec_sample_rate(&usecase->stream.in->app_type_cfg.sample_rate);
-            ALOGI("%s using %d sample rate rate for A2DP dec CoPP",
-                  __func__, usecase->stream.in->app_type_cfg.sample_rate);
-        }
-        sample_rate = usecase->stream.in->app_type_cfg.sample_rate;
         app_type_cfg[len++] = sample_rate;
         if (snd_device_be_idx > 0)
             app_type_cfg[len++] = snd_device_be_idx;
@@ -1222,7 +1230,6 @@
     } else {
         app_type = platform_get_default_app_type_v2(adev->platform, usecase->type);
         if(usecase->type == TRANSCODE_LOOPBACK_RX) {
-            sample_rate = usecase->stream.inout->out_config.sample_rate;
             app_type = usecase->stream.inout->out_app_type_cfg.app_type;
         }
         app_type_cfg[len++] = app_type;
@@ -1582,29 +1589,15 @@
     int type = usecase->type;
 
     if (type == PCM_PLAYBACK && usecase->stream.out != NULL) {
-        struct stream_out *out = usecase->stream.out;
-        int snd_device = usecase->out_snd_device;
-        snd_device = (snd_device == SND_DEVICE_OUT_SPEAKER) ?
-                     platform_get_spkr_prot_snd_device(snd_device) : snd_device;
         platform_send_audio_calibration(adev->platform, usecase,
-                                        out->app_type_cfg.app_type,
-                                        usecase->stream.out->app_type_cfg.sample_rate);
+                         usecase->stream.out->app_type_cfg.app_type);
     } else if (type == PCM_CAPTURE && usecase->stream.in != NULL) {
         platform_send_audio_calibration(adev->platform, usecase,
-                         usecase->stream.in->app_type_cfg.app_type,
-                         usecase->stream.in->app_type_cfg.sample_rate);
-    } else if (type == PCM_HFP_CALL || type == PCM_CAPTURE) {
-        /* when app type is default. the sample rate is not used to send cal */
+                         usecase->stream.in->app_type_cfg.app_type);
+    } else if ((type == PCM_HFP_CALL) || (type == PCM_CAPTURE) ||
+               (type == TRANSCODE_LOOPBACK_RX && usecase->stream.inout != NULL)) {
         platform_send_audio_calibration(adev->platform, usecase,
-                         platform_get_default_app_type_v2(adev->platform, usecase->type),
-                         48000);
-    } else if (type == TRANSCODE_LOOPBACK_RX && usecase->stream.inout != NULL) {
-        int snd_device = usecase->out_snd_device;
-        snd_device = (snd_device == SND_DEVICE_OUT_SPEAKER) ?
-                     platform_get_spkr_prot_snd_device(snd_device) : snd_device;
-        platform_send_audio_calibration(adev->platform, usecase,
-                         platform_get_default_app_type_v2(adev->platform, usecase->type),
-                         usecase->stream.inout->out_config.sample_rate);
+                         platform_get_default_app_type_v2(adev->platform, usecase->type));
     } else {
         /* No need to send audio calibration for voice and voip call usecases */
         if ((type != VOICE_CALL) && (type != VOIP_CALL))