Merge "hal: get correspoding sample rate per device when sending cal data" into audio-hal.lnx.6.0

hal/audio_extn/audio_extn.h

@@ -157,7 +157,7 @@ typedef int (*fp_platform_get_snd_device_name_extn_t)(void *platform, snd_device
                                       char *device_name);
 typedef int (*fp_platform_get_default_app_type_v2_t)(void *, usecase_type_t);
 typedef int (*fp_platform_send_audio_calibration_t)(void *, struct audio_usecase *,
-                                                   int, int);
+                                                   int);
 typedef int (*fp_platform_get_pcm_device_id_t)(audio_usecase_t, int);
 typedef const char *(*fp_platform_get_snd_device_name_t)(snd_device_t);
 typedef int (*fp_platform_spkr_prot_is_wsa_analog_mode_t)(void *);
@@ -801,6 +801,8 @@ int audio_extn_utils_get_sample_rate_from_string(const char *);
 int audio_extn_utils_get_channels_from_string(const char *);
 void audio_extn_utils_release_snd_device(snd_device_t snd_device);
 int audio_extn_utils_is_vendor_enhanced_fwk();
+int audio_extn_utils_get_app_sample_rate_for_device(struct audio_device *adev,
+                                    struct audio_usecase *usecase, int snd_device);
 
 #ifdef DS2_DOLBY_DAP_ENABLED
 #define LIB_DS2_DAP_HAL "vendor/lib/libhwdaphal.so"

hal/audio_extn/spkr_protection.c

@@ -1030,7 +1030,7 @@ exit:
             app_type = fp_platform_get_default_app_type_v2(adev->platform,
                                                 PCM_CAPTURE);
             fp_platform_send_audio_calibration(adev->platform, uc_info_tx,
-                                                    app_type, 8000);
+                                                    app_type);
         }
         if (!v_validation) {
             if (!status.status) {
@@ -2281,6 +2281,8 @@ int spkr_prot_start_processing(snd_device_t snd_device)
     if (!uc_info_tx) {
         return -ENOMEM;
     }
+    uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
+    uc_info_tx->type = PCM_CAPTURE;
 
     if (fp_platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0) {
         ALOGE("%s: Invalid sound device returned", __func__);
@@ -2293,8 +2295,6 @@ int spkr_prot_start_processing(snd_device_t snd_device)
 
     pthread_mutex_lock(&handle.mutex_spkr_prot);
     if (handle.spkr_processing_state == SPKR_PROCESSING_IN_IDLE) {
-        uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
-        uc_info_tx->type = PCM_CAPTURE;
         uc_info_tx->in_snd_device = in_snd_device;
         uc_info_tx->out_snd_device = SND_DEVICE_NONE;
         handle.pcm_tx = NULL;
@@ -2333,15 +2333,13 @@ exit:
         app_type = fp_platform_get_default_app_type_v2(adev->platform,
                                             PCM_CAPTURE);
         fp_platform_send_audio_calibration(adev->platform, uc_info_tx,
-                                                app_type, 8000);
+                                                app_type);
     }
     if (ret) {
         if (handle.pcm_tx)
             pcm_close(handle.pcm_tx);
         handle.pcm_tx = NULL;
         list_remove(&uc_info_tx->list);
-        uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
-        uc_info_tx->type = PCM_CAPTURE;
         uc_info_tx->in_snd_device = in_snd_device;
         uc_info_tx->out_snd_device = SND_DEVICE_NONE;
         fp_disable_snd_device(adev, in_snd_device);

hal/audio_extn/utils.c

@@ -1025,6 +1025,87 @@ exit_send_app_type_cfg:
     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 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
     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 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
               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 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
             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 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
         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 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
     } 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 @@ void audio_extn_utils_send_audio_calibration(struct audio_device *adev,
     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.app_type);
-                                        usecase->stream.out->app_type_cfg.sample_rate);
     } 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.app_type);
-                         usecase->stream.in->app_type_cfg.sample_rate);
+    } else if ((type == PCM_HFP_CALL) || (type == PCM_CAPTURE) ||
-    } else if (type == PCM_HFP_CALL || type == PCM_CAPTURE) {
+               (type == TRANSCODE_LOOPBACK_RX && usecase->stream.inout != NULL)) {
-        /* when app type is default. the sample rate is not used to send cal */
         platform_send_audio_calibration(adev->platform, usecase,
-                         platform_get_default_app_type_v2(adev->platform, usecase->type),
+                         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);
     } else {
         /* No need to send audio calibration for voice and voip call usecases */
         if ((type != VOICE_CALL) && (type != VOIP_CALL))

hal/msm8916/platform.c

@@ -3622,7 +3622,7 @@ int platform_get_backend_index(snd_device_t snd_device)
 }
 
 int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
-                                    int app_type, int sample_rate)
+                                    int app_type)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
     int acdb_dev_id, acdb_dev_type;
@@ -3631,6 +3631,7 @@ int platform_send_audio_calibration(void *platform, struct audio_usecase *usecas
     int i, num_devices = 1;
     bool is_incall_rec_usecase = false;
     snd_device_t incall_rec_device;
+    int sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
 
     if (voice_is_in_call(my_data->adev))
         is_incall_rec_usecase = voice_is_in_call_rec_stream(usecase->stream.in);
@@ -3660,11 +3661,16 @@ int platform_send_audio_calibration(void *platform, struct audio_usecase *usecas
     }
 
     for (i = 0; i < num_devices; i++) {
-        if (!is_incall_rec_usecase)
+        if (!is_incall_rec_usecase) {
             acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(new_snd_device[i])];
-        else
+            sample_rate = audio_extn_utils_get_app_sample_rate_for_device(my_data->adev, usecase,
+                                                          new_snd_device[i]);
+        } else {
             // Use in_call_rec snd_device to extract the ACDB device ID instead of split snd devices
             acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(snd_device)];
+            sample_rate = audio_extn_utils_get_app_sample_rate_for_device(my_data->adev, usecase,
+                                                          snd_device);
+        }
 
         // Do not use Rx path default app type for TX path
         if ((usecase->type == PCM_CAPTURE) && (app_type == DEFAULT_APP_TYPE_RX_PATH)) {

hal/msm8960/platform.c

@@ -522,7 +522,7 @@ int platform_get_default_app_type(void *platform __unused)
 }
 
 int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
-                                    int app_type __unused, int sample_rate __unused)
+                                    int app_type __unused)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
     int acdb_dev_id, acdb_dev_type;

hal/msm8974/platform.c

@@ -4719,7 +4719,7 @@ int platform_get_backend_index(snd_device_t snd_device)
 }
 
 int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
-                                    int app_type, int sample_rate)
+                                    int app_type)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
     int acdb_dev_id, acdb_dev_type;
@@ -4728,6 +4728,7 @@ int platform_send_audio_calibration(void *platform, struct audio_usecase *usecas
     int i, num_devices = 1;
     bool is_incall_rec_usecase = false;
     snd_device_t incall_rec_device;
+    int sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
 
     if (voice_is_in_call(my_data->adev))
         is_incall_rec_usecase = voice_is_in_call_rec_stream(usecase->stream.in);
@@ -4762,11 +4763,16 @@ int platform_send_audio_calibration(void *platform, struct audio_usecase *usecas
     }
 
     for (i = 0; i < num_devices; i++) {
-        if (!is_incall_rec_usecase)
+        if (!is_incall_rec_usecase) {
             acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(new_snd_device[i])];
-        else
+            sample_rate = audio_extn_utils_get_app_sample_rate_for_device(my_data->adev, usecase,
+                                                          new_snd_device[i]);
+        } else {
             // Use in_call_rec snd_device to extract the ACDB device ID instead of split snd devices
             acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(snd_device)];
+            sample_rate = audio_extn_utils_get_app_sample_rate_for_device(my_data->adev, usecase,
+                                                          snd_device);
+        }
 
         // Do not use Rx path default app type for TX path
         if ((usecase->type == PCM_CAPTURE) && (app_type == DEFAULT_APP_TYPE_RX_PATH)) {

hal/platform_api.h

@@ -155,7 +155,7 @@ int platform_get_meta_info_key_from_list(void *platform, char *mod_name);
 int platform_set_native_support(int na_mode);
 int platform_get_native_support();
 int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
-                                    int app_type, int sample_rate);
+                                    int app_type);
 int platform_get_default_app_type(void *platform);
 int platform_get_default_app_type_v2(void *platform, usecase_type_t  type);
 int platform_switch_voice_call_device_pre(void *platform);