Merge "hal: Add support for qcs605 device with tavil codec"
diff --git a/hal/audio_extn/audio_defs.h b/hal/audio_extn/audio_defs.h
index 0e1848e..a0b1949 100644
--- a/hal/audio_extn/audio_defs.h
+++ b/hal/audio_extn/audio_defs.h
@@ -306,4 +306,12 @@
AUDIO_EXTN_PARAM_LICENSE_PARAMS,
} audio_extn_param_id;
+typedef union {
+ struct audio_out_render_window_param render_window_params;
+} audio_extn_loopback_param_payload;
+
+typedef enum {
+ AUDIO_EXTN_PARAM_LOOPBACK_RENDER_WINDOW /* PARAM to set render window */
+} audio_extn_loopback_param_id;
+
#endif /* AUDIO_DEFS_H */
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index c6c0924..3eee428 100755
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -1478,6 +1478,33 @@
return ret;
}
+#ifdef AUDIO_HW_LOOPBACK_ENABLED
+int audio_extn_hw_loopback_set_param_data(audio_patch_handle_t handle,
+ audio_extn_loopback_param_id param_id,
+ audio_extn_loopback_param_payload *payload) {
+ int ret = -EINVAL;
+
+ if (!payload) {
+ ALOGE("%s:: Invalid Param",__func__);
+ return ret;
+ }
+
+ ALOGD("%d: %s: param id is %d\n", __LINE__, __func__, param_id);
+
+ switch(param_id) {
+ case AUDIO_EXTN_PARAM_LOOPBACK_RENDER_WINDOW:
+ ret = audio_extn_hw_loopback_set_render_window(handle, payload);
+ break;
+ default:
+ ALOGE("%s: unsupported param id %d", __func__, param_id);
+ break;
+ }
+
+ return ret;
+}
+#endif
+
+
/* API to get playback stream specific config parameters */
int audio_extn_out_get_param_data(struct stream_out *out,
audio_extn_param_id param_id,
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 6ec07b3..e158b0a 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -999,6 +999,14 @@
const struct audio_port_config *config);
int audio_extn_hw_loopback_get_audio_port(struct audio_hw_device *dev,
struct audio_port *port_in);
+
+int audio_extn_hw_loopback_set_param_data(audio_patch_handle_t handle,
+ audio_extn_loopback_param_id param_id,
+ audio_extn_loopback_param_payload *payload);
+
+int audio_extn_hw_loopback_set_render_window(audio_patch_handle_t handle,
+ struct audio_out_render_window_param *render_window);
+
int audio_extn_hw_loopback_init(struct audio_device *adev);
void audio_extn_hw_loopback_deinit(struct audio_device *adev);
#else
@@ -1026,6 +1034,18 @@
{
return -ENOSYS;
}
+static int __unused audio_extn_hw_loopback_set_param_data(audio_patch_handle_t handle __unused,
+ audio_extn_loopback_param_id param_id __unused,
+ audio_extn_loopback_param_payload *payload __unused)
+{
+ return -ENOSYS;
+}
+
+static int __unused audio_extn_hw_loopback_set_render_window(audio_patch_handle_t handle __unused,
+ struct audio_out_render_window_param *render_window __unused)
+{
+ return -ENOSYS;
+}
static int __unused audio_extn_hw_loopback_init(struct audio_device *adev __unused)
{
return -ENOSYS;
diff --git a/hal/audio_extn/hw_loopback.c b/hal/audio_extn/hw_loopback.c
index 990a283..7516717 100644
--- a/hal/audio_extn/hw_loopback.c
+++ b/hal/audio_extn/hw_loopback.c
@@ -357,6 +357,78 @@
return 0;
}
+#ifdef SNDRV_COMPRESS_RENDER_WINDOW
+static loopback_patch_t *get_active_loopback_patch(audio_patch_handle_t handle)
+{
+ int n = 0;
+ int patch_index = -1;
+ loopback_patch_t *active_loopback_patch = NULL;
+
+ for (n=0; n < MAX_NUM_PATCHES; n++) {
+ if (audio_loopback_mod->patch_db.num_patches > 0) {
+ if (audio_loopback_mod->patch_db.loopback_patch[n].patch_handle_id == handle) {
+ patch_index = n;
+ break;
+ }
+ } else {
+ ALOGE("%s, No active audio loopback patch", __func__);
+ return active_loopback_patch;
+ }
+ }
+
+ if ((patch_index > -1) && (patch_index < MAX_NUM_PATCHES))
+ active_loopback_patch = &(audio_loopback_mod->patch_db.loopback_patch[
+ patch_index]);
+ else
+ ALOGE("%s, Requested Patch handle does not exist", __func__);
+
+ return active_loopback_patch;
+}
+
+int audio_extn_hw_loopback_set_render_window(audio_patch_handle_t handle,
+ struct audio_out_render_window_param *render_window)
+{
+ struct snd_compr_metadata metadata = {0};
+ int ret = 0;
+ loopback_patch_t *active_loopback_patch = get_active_loopback_patch(handle);
+
+ if (active_loopback_patch == NULL) {
+ ALOGE("%s: Invalid patch handle", __func__);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ if (render_window == NULL) {
+ ALOGE("%s: Invalid render_window", __func__);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ metadata.key = SNDRV_COMPRESS_RENDER_WINDOW;
+ /*render window start value */
+ metadata.value[0] = 0xFFFFFFFF & render_window->render_ws; /* lsb */
+ metadata.value[1] = \
+ (0xFFFFFFFF00000000 & render_window->render_ws) >> 32; /* msb*/
+ /*render window end value */
+ metadata.value[2] = 0xFFFFFFFF & render_window->render_we; /* lsb */
+ metadata.value[3] = \
+ (0xFFFFFFFF00000000 & render_window->render_we) >> 32; /* msb*/
+
+ ret = compress_set_metadata(active_loopback_patch->sink_stream, &metadata);
+
+exit:
+ return ret;
+}
+#else
+int audio_extn_hw_loopback_set_render_window(struct audio_hw_device *dev,
+ audio_patch_handle_t handle __unused,
+ struct audio_out_render_window_param *render_window __unused)
+{
+ ALOGD("%s:: configuring render window not supported", __func__);
+ return 0;
+}
+#endif
+
#if defined SNDRV_COMPRESS_LATENCY_MODE
static void transcode_loopback_util_set_latency_mode(
loopback_patch_t *active_loopback_patch,
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index bd3fa7c..198d871 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -990,7 +990,14 @@
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;
+ 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) &&
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index cb2d786..ac4233d 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -2125,11 +2125,18 @@
if (out_snd_device == SND_DEVICE_NONE) {
out_snd_device = platform_get_output_snd_device(adev->platform,
usecase->stream.out);
- if (usecase->stream.out == adev->primary_output &&
- adev->active_input &&
- out_snd_device != usecase->out_snd_device) {
- select_devices(adev, adev->active_input->usecase);
- }
+ voip_usecase = get_usecase_from_list(adev, USECASE_AUDIO_PLAYBACK_VOIP);
+ if (voip_usecase == NULL && adev->primary_output && !adev->primary_output->standby)
+ voip_usecase = get_usecase_from_list(adev, adev->primary_output->usecase);
+
+ if ((usecase->stream.out != NULL &&
+ voip_usecase != NULL &&
+ usecase->stream.out->usecase == voip_usecase->id) &&
+ adev->active_input &&
+ adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
+ out_snd_device != usecase->out_snd_device) {
+ select_devices(adev, adev->active_input->usecase);
+ }
}
} else if (usecase->type == PCM_CAPTURE) {
if (usecase->stream.in == NULL) {
@@ -2143,9 +2150,12 @@
if (adev->active_input &&
(adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
(adev->mode == AUDIO_MODE_IN_COMMUNICATION &&
- adev->active_input->source == AUDIO_SOURCE_MIC)) &&
- adev->primary_output && !adev->primary_output->standby) {
- out_device = adev->primary_output->devices;
+ adev->active_input->source == AUDIO_SOURCE_MIC))) {
+ voip_usecase = get_usecase_from_list(adev, USECASE_AUDIO_PLAYBACK_VOIP);
+ if (voip_usecase != NULL && voip_usecase->stream.out != NULL)
+ out_device = voip_usecase->stream.out->devices;
+ else if (adev->primary_output && !adev->primary_output->standby)
+ out_device = adev->primary_output->devices;
platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
} else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
@@ -3309,6 +3319,8 @@
int channel_count,
bool is_low_latency)
{
+ int i = 0;
+ size_t frame_size = 0;
size_t size = 0;
if (check_input_parameters(sample_rate, format, channel_count) != 0)
@@ -3318,15 +3330,23 @@
if (is_low_latency)
size = configured_low_latency_capture_period_size;
- size *= audio_bytes_per_sample(format) * channel_count;
+ frame_size = audio_bytes_per_sample(format) * channel_count;
+ size *= frame_size;
- /* make sure the size is multiple of 32 bytes
+ /* make sure the size is multiple of 32 bytes and additionally multiple of
+ * the frame_size (required for 24bit samples and non-power-of-2 channel counts)
* At 48 kHz mono 16-bit PCM:
* 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
* 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
+ *
+ * The loop reaches result within 32 iterations, as initial size is
+ * already a multiple of frame_size
*/
- size += 0x1f;
- size &= ~0x1f;
+ for (i=0; i<32; i++) {
+ if ((size & 0x1f) == 0)
+ break;
+ size += frame_size;
+ }
return size;
}
@@ -4502,6 +4522,15 @@
audio_format_t dst_format = out->hal_op_format;
audio_format_t src_format = out->hal_ip_format;
+ /* prevent division-by-zero */
+ uint32_t bitwidth_src = format_to_bitwidth_table[src_format];
+ uint32_t bitwidth_dst = format_to_bitwidth_table[dst_format];
+ if ((bitwidth_src == 0) || (bitwidth_dst == 0)) {
+ ALOGE("%s: Error bitwidth == 0", __func__);
+ ATRACE_END();
+ return -EINVAL;
+ }
+
uint32_t frames = bytes / format_to_bitwidth_table[src_format];
uint32_t bytes_to_write = frames * format_to_bitwidth_table[dst_format];
@@ -4642,10 +4671,18 @@
out->standby = true;
}
out_on_error(&out->stream.common);
- if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
- usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
- out_get_sample_rate(&out->stream.common));
+ if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
+ /* prevent division-by-zero */
+ uint32_t stream_size = audio_stream_out_frame_size(stream);
+ uint32_t srate = out_get_sample_rate(&out->stream.common);
+ if ((stream_size == 0) || (srate == 0)) {
+ ALOGE("%s: stream_size= %d, srate = %d", __func__, stream_size, srate);
+ ATRACE_END();
+ return -EINVAL;
+ }
+ usleep((uint64_t)bytes * 1000000 / stream_size / srate);
+ }
if (audio_extn_passthru_is_passthrough_stream(out)) {
ALOGE("%s: write error, ret = %zd", __func__, ret);
ATRACE_END();
@@ -7079,6 +7116,13 @@
config->format,
channel_count,
is_low_latency);
+ /* prevent division-by-zero */
+ if (frame_size == 0) {
+ ALOGE("%s: Error frame_size==0", __func__);
+ ret = -EINVAL;
+ goto err_open;
+ }
+
in->config.period_size = buffer_size / frame_size;
if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
diff --git a/hal/audio_hw_extn_api.c b/hal/audio_hw_extn_api.c
index f5e0659..310b537 100644
--- a/hal/audio_hw_extn_api.c
+++ b/hal/audio_hw_extn_api.c
@@ -1,5 +1,5 @@
/*
-* Copyright (c) 2016-2017, The Linux Foundation. All rights reserved.
+* Copyright (c) 2016-2017, 2018, The Linux Foundation. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are
@@ -423,6 +423,19 @@
return ret;
}
+int qahwi_loopback_set_param_data(audio_patch_handle_t handle,
+ audio_extn_loopback_param_id param_id,
+ void *payload) {
+ int ret = 0;
+
+ ret = audio_extn_hw_loopback_set_param_data(
+ handle,
+ param_id,
+ (audio_extn_loopback_param_payload *)payload);
+
+ return ret;
+}
+
void qahwi_init(hw_device_t *device)
{
struct audio_device *adev = (struct audio_device *) device;
diff --git a/hal/edid.h b/hal/edid.h
index da5c592..f920a82 100644
--- a/hal/edid.h
+++ b/hal/edid.h
@@ -57,6 +57,27 @@
#define PCM_CHANNEL_FRC 14 /* Front right of center. */
#define PCM_CHANNEL_RLC 15 /* Rear left of center. */
#define PCM_CHANNEL_RRC 16 /* Rear right of center. */
+#define PCM_CHANNEL_LFE2 17 /* Second low frequency channel. */
+#define PCM_CHANNEL_SL 18 /* Side left channel. */
+#define PCM_CHANNEL_SR 19 /* Side right channel. */
+#define PCM_CHANNEL_TFL 20 /* Top front left channel. */
+#define PCM_CHANNEL_LVH 20 /* Left vertical height channel. */
+#define PCM_CHANNEL_TFR 21 /* Top front right channel. */
+#define PCM_CHANNEL_RVH 21 /* Right vertical height channel. */
+#define PCM_CHANNEL_TC 22 /* Top center channel. */
+#define PCM_CHANNEL_TBL 23 /* Top back left channel. */
+#define PCM_CHANNEL_TBR 24 /* Top back right channel. */
+#define PCM_CHANNEL_TSL 25 /* Top side left channel. */
+#define PCM_CHANNEL_TSR 26 /* Top side right channel. */
+#define PCM_CHANNEL_TBC 27 /* Top back center channel. */
+#define PCM_CHANNEL_BFC 28 /* Bottom front center channel. */
+#define PCM_CHANNEL_BFL 29 /* Bottom front left channel. */
+#define PCM_CHANNEL_BFR 30 /* Bottom front right channel. */
+#define PCM_CHANNEL_LW 31 /* Left wide channel. */
+#define PCM_CHANNEL_RW 32 /* Right wide channel. */
+#define PCM_CHANNEL_LSD 33 /* Left side direct channel. */
+#define PCM_CHANNEL_RSD 34 /* Right side direct channel. */
+
#define MAX_HDMI_CHANNEL_CNT 8
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
old mode 100755
new mode 100644
index 82fafc7..68ffd56
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -298,6 +298,7 @@
struct acdb_init_data_v4 acdb_init_data;
bool use_generic_handset;
struct spkr_device_chmap *spkr_ch_map;
+ bool use_sprk_default_sample_rate;
};
struct spkr_device_chmap {
@@ -2290,6 +2291,7 @@
my_data->hw_dep_fd = -1;
my_data->mono_speaker = SPKR_1;
my_data->spkr_ch_map = NULL;
+ my_data->use_sprk_default_sample_rate = true;
be_dai_name_table = NULL;
@@ -2831,6 +2833,9 @@
/* free acdb_meta_key_list */
platform_release_acdb_metainfo_key(platform);
+ if (my_data->acdb_deallocate)
+ my_data->acdb_deallocate();
+
free(platform);
/* deinit usb */
audio_extn_usb_deinit();
@@ -4688,6 +4693,16 @@
(mode == AUDIO_MODE_IN_COMMUNICATION)) {
if (out_device & AUDIO_DEVICE_OUT_SPEAKER)
in_device = AUDIO_DEVICE_IN_BACK_MIC;
+ else if (out_device & AUDIO_DEVICE_OUT_EARPIECE)
+ in_device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET)
+ in_device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ else if (out_device & AUDIO_DEVICE_OUT_USB_DEVICE)
+ in_device = AUDIO_DEVICE_IN_USB_DEVICE;
+
+ in_device = ((out_device == AUDIO_DEVICE_NONE) ?
+ AUDIO_DEVICE_IN_BUILTIN_MIC : in_device) & ~AUDIO_DEVICE_BIT_IN;
+
if (adev->active_input) {
snd_device = get_snd_device_for_voice_comm(my_data, out_device, in_device);
}
@@ -7385,6 +7400,11 @@
platform_get_edid_info(platform);
}
+bool platform_spkr_use_default_sample_rate(void *platform) {
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->use_sprk_default_sample_rate;
+}
+
void platform_invalidate_backend_config(void * platform,snd_device_t snd_device)
{
struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 128a458..0766311 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -65,6 +65,7 @@
#define MIXER_XML_PATH_I2S "/etc/mixer_paths_i2s.xml"
#define PLATFORM_INFO_XML_PATH_I2S "/etc/audio_platform_info_extcodec.xml"
#define PLATFORM_INFO_XML_PATH_WSA "/etc/audio_platform_info_wsa.xml"
+#define PLATFORM_INFO_XML_PATH_TDM "/etc/audio_platform_info_tdm.xml"
#else
#define MIXER_XML_BASE_STRING "/vendor/etc/mixer_paths"
#define MIXER_XML_DEFAULT_PATH "/vendor/etc/mixer_paths.xml"
@@ -76,6 +77,7 @@
#define MIXER_XML_PATH_I2S "/vendor/etc/mixer_paths_i2s.xml"
#define PLATFORM_INFO_XML_PATH_I2S "/vendor/etc/audio_platform_info_i2s.xml"
#define PLATFORM_INFO_XML_PATH_WSA "/vendor/etc/audio_platform_info_wsa.xml"
+#define PLATFORM_INFO_XML_PATH_TDM "/vendor/etc/audio_platform_info_tdm.xml"
#endif
#include <linux/msm_audio.h>
@@ -276,6 +278,7 @@
struct acdb_init_data_v4 acdb_init_data;
bool use_generic_handset;
struct spkr_device_chmap *spkr_ch_map;
+ bool use_sprk_default_sample_rate;
};
struct spkr_device_chmap {
@@ -2096,7 +2099,7 @@
my_data->mono_speaker = SPKR_1;
my_data->speaker_lr_swap = false;
my_data->spkr_ch_map = NULL;
-
+ my_data->use_sprk_default_sample_rate = true;
be_dai_name_table = NULL;
property_get("ro.vendor.audio.sdk.fluencetype", my_data->fluence_cap, "");
@@ -2177,11 +2180,23 @@
else if (!strncmp(snd_card_name, "qcs405-wsa-snd-card",
sizeof("qcs405-wsa-snd-card")))
platform_info_init(PLATFORM_INFO_XML_PATH_WSA, my_data, PLATFORM);
+ else if (!strncmp(snd_card_name, "qcs405-tdm-snd-card",
+ sizeof("qcs405-tdm-snd-card")))
+ platform_info_init(PLATFORM_INFO_XML_PATH_TDM, my_data, PLATFORM);
else if (my_data->is_internal_codec)
platform_info_init(PLATFORM_INFO_XML_PATH_INTCODEC, my_data, PLATFORM);
else
platform_info_init(PLATFORM_INFO_XML_PATH, my_data, PLATFORM);
+ /* CSRA devices support multiple sample rates via I2S at spkr out */
+ if (!strncmp(snd_card_name, "qcs405-csra", strlen("qcs405-csra"))) {
+ ALOGE("%s: soundcard: %s supports multiple sample rates", __func__, snd_card_name);
+ my_data->use_sprk_default_sample_rate = false;
+ } else {
+ my_data->use_sprk_default_sample_rate = true;
+ ALOGE("%s: soundcard: %s supports only default sample rate", __func__, snd_card_name);
+ }
+
my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
if (my_data->acdb_handle == NULL) {
@@ -2437,11 +2452,18 @@
} else {
if (!strncmp(snd_card_name, "qcs405", strlen("qcs405"))) {
- my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
- strdup("WSA_CDC_DMA_RX_0 Format");
- my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
- strdup("WSA_CDC_DMA_RX_0 SampleRate");
+ if (!strncmp(snd_card_name, "qcs405-csra", strlen("qcs405-csra"))) {
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ strdup("PRIM_MI2S_RX Format");
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ strdup("PRIM_MI2S_RX SampleRate");
+ } else {
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ strdup("WSA_CDC_DMA_RX_0 Format");
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ strdup("WSA_CDC_DMA_RX_0 SampleRate");
+ }
my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
strdup("VA_CDC_DMA_TX_0 Format");
my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
@@ -2672,6 +2694,9 @@
/* free acdb_meta_key_list */
platform_release_acdb_metainfo_key(platform);
+ if (my_data->acdb_deallocate)
+ my_data->acdb_deallocate();
+
free(platform);
/* deinit usb */
audio_extn_usb_deinit();
@@ -4524,6 +4549,16 @@
(mode == AUDIO_MODE_IN_COMMUNICATION)) {
if (out_device & AUDIO_DEVICE_OUT_SPEAKER)
in_device = AUDIO_DEVICE_IN_BACK_MIC;
+ else if (out_device & AUDIO_DEVICE_OUT_EARPIECE)
+ in_device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET)
+ in_device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ else if (out_device & AUDIO_DEVICE_OUT_USB_DEVICE)
+ in_device = AUDIO_DEVICE_IN_USB_DEVICE;
+
+ in_device = ((out_device == AUDIO_DEVICE_NONE) ?
+ AUDIO_DEVICE_IN_BUILTIN_MIC : in_device) & ~AUDIO_DEVICE_BIT_IN;
+
if (adev->active_input) {
snd_device = get_snd_device_for_voice_comm(my_data, out_device, in_device);
}
@@ -6278,9 +6313,15 @@
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
ALOGD("%s:becf: afe: reset to default bitwidth %d", __func__, bit_width);
}
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
+ /*
+ * In case of CSRA speaker out, all sample rates are supported, so
+ * check platform here
+ */
+ if (platform_spkr_use_default_sample_rate(adev->platform)) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
"default Sample Rate(48k)", __func__);
+ }
}
if (backend_idx == USB_AUDIO_RX_BACKEND) {
@@ -6951,6 +6992,40 @@
channel_map[6] = PCM_CHANNEL_LS;
channel_map[7] = PCM_CHANNEL_RS;
break;
+ case 12:
+ /* AUDIO_CHANNEL_OUT_7POINT1POINT4 */
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_FC;
+ channel_map[3] = PCM_CHANNEL_LFE;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ channel_map[8] = PCM_CHANNEL_TFL;
+ channel_map[9] = PCM_CHANNEL_TFR;
+ channel_map[10] = PCM_CHANNEL_TSL;
+ channel_map[11] = PCM_CHANNEL_TSR;
+ break;
+ case 16:
+ /* 16 channels */
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_FC;
+ channel_map[3] = PCM_CHANNEL_LFE;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ channel_map[8] = PCM_CHANNEL_TFL;
+ channel_map[9] = PCM_CHANNEL_TFR;
+ channel_map[10] = PCM_CHANNEL_TSL;
+ channel_map[11] = PCM_CHANNEL_TSR;
+ channel_map[12] = PCM_CHANNEL_FLC;
+ channel_map[13] = PCM_CHANNEL_FRC;
+ channel_map[14] = PCM_CHANNEL_RLC;
+ channel_map[15] = PCM_CHANNEL_RRC;
+ break;
default:
ALOGE("unsupported channels %d for setting channel map", channels);
return -1;
@@ -7075,12 +7150,21 @@
struct mixer_ctl *ctl;
char mixer_ctl_name[44] = {0}; // max length of name is 44 as defined
int ret;
- unsigned int i;
- long set_values[FCC_8] = {0};
+ unsigned int i=0, n=0;
+ long set_values[AUDIO_MAX_DSP_CHANNELS];
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
ALOGV("%s channel_count:%d",__func__, ch_count);
- if (NULL == ch_map || (ch_count < 1) || (ch_count > FCC_8)) {
+
+ /*
+ * FIXME:
+ * Currently the channel mask in audio.h is limited to 30 channels,
+ * (=AUDIO_CHANNEL_COUNT_MAX), whereas the mixer controls already
+ * allow up to AUDIO_MAX_DSP_CHANNELS channels as per final requirement.
+ * Until channel mask definition is not changed from a uint32_t value
+ * to something else, a sanity check is needed here.
+ */
+ if (NULL == ch_map || (ch_count < 1) || (ch_count > AUDIO_CHANNEL_COUNT_MAX)) {
ALOGE("%s: Invalid channel mapping or channel count value", __func__);
return -EINVAL;
}
@@ -7098,12 +7182,34 @@
ALOGD("%s mixer_ctl_name:%s", __func__, mixer_ctl_name);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
- for (i = 0; i < (unsigned int)ch_count; i++) {
+
+ /* find out how many values the control can set */
+ n = mixer_ctl_get_num_values(ctl);
+
+ if (n != ch_count)
+ ALOGV("%s chcnt %d != mixerctl elem size %d",__func__, ch_count, n);
+
+ if (n < ch_count) {
+ ALOGE("%s chcnt %d > mixerctl elem size %d",__func__, ch_count, n);
+ return -EINVAL;
+ }
+
+ if (n > AUDIO_MAX_DSP_CHANNELS) {
+ ALOGE("%s mixerctl elem size %d > AUDIO_MAX_DSP_CHANNELS %d",__func__, n, AUDIO_MAX_DSP_CHANNELS);
+ return -EINVAL;
+ }
+
+ /* initialize all set_values to zero */
+ memset (set_values, 0, sizeof(set_values));
+
+ /* copy only as many values as corresponding mixer_ctrl allows */
+ for (i = 0; i < ch_count; i++) {
set_values[i] = ch_map[i];
}
@@ -7111,7 +7217,8 @@
set_values[0], set_values[1], set_values[2], set_values[3], set_values[4],
set_values[5], set_values[6], set_values[7], ch_count);
- ret = mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+ ret = mixer_ctl_set_array(ctl, set_values, n);
+
if (ret < 0) {
ALOGE("%s: Could not set ctl, error:%d ch_count:%d",
__func__, ret, ch_count);
@@ -7276,6 +7383,11 @@
return 0;
}
+bool platform_spkr_use_default_sample_rate(void *platform) {
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->use_sprk_default_sample_rate;
+}
+
int platform_set_edid_channels_configuration(void *platform, int channels) {
struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index c8ddaec..e1f433c 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -280,6 +280,8 @@
#define AUDIO_PARAMETER_KEY_TRUE_32_BIT "true_32_bit"
+#define AUDIO_MAX_DSP_CHANNELS 32
+
#define ALL_SESSION_VSID 0xFFFFFFFF
#define DEFAULT_MUTE_RAMP_DURATION_MS 20
#define DEFAULT_VOLUME_RAMP_DURATION_MS 20
diff --git a/hal/platform_api.h b/hal/platform_api.h
old mode 100755
new mode 100644
index 09c69de..1563673
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -218,6 +218,7 @@
snd_device_t snd_device,
struct mix_matrix_params mm_params);
int platform_set_edid_channels_configuration(void *platform, int channels);
+bool platform_spkr_use_default_sample_rate(void *platform);
unsigned char platform_map_to_edid_format(int format);
bool platform_is_edid_supported_format(void *platform, int format);
bool platform_is_edid_supported_sample_rate(void *platform, int sample_rate);
diff --git a/hdmi_in_test/Makefile.am b/hdmi_in_test/Makefile.am
index 34e4ff5..eb74d21 100644
--- a/hdmi_in_test/Makefile.am
+++ b/hdmi_in_test/Makefile.am
@@ -1,9 +1,17 @@
ACLOCAL_AMFLAGS = -I m4
bin_PROGRAMS = hdmi_in_test
+bin_PROGRAMS += fmt_change_test
pkgconfigdir = $(libdir)/pkgconfig
+REC_INCLUDES = -I $(top_srcdir)/qahw_api/inc
+REC_INCLUDES += -I $(top_srcdir)/qahw/inc
+
hdmi_in_test_SOURCES = src/hdmi_in_event_test.c
hdmi_in_test_CFLAGS = $(CFLAGS) -Wno-sign-compare -Werror
hdmi_in_test_LDADD = -llog -lpthread
+fmt_change_test_SOURCES = src/fmt_change_test.c
+fmt_change_test_CFLAGS = $(CFLAGS) -Wno-sign-compare -Werror $(REC_INCLUDES)
+fmt_change_test_LDADD = -llog -lpthread ../qahw_api/libqahw.la
+
diff --git a/hdmi_in_test/src/fmt_change_test.c b/hdmi_in_test/src/fmt_change_test.c
new file mode 100644
index 0000000..6651c6d
--- /dev/null
+++ b/hdmi_in_test/src/fmt_change_test.c
@@ -0,0 +1,811 @@
+/*
+ * Copyright (c) 2016-2018, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2015 The Android Open Source Project *
+ * 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.
+ */
+
+/* Test app to capture event updates from kernel */
+/*#define LOG_NDEBUG 0*/
+#include <getopt.h>
+#include <fcntl.h>
+#include <linux/netlink.h>
+#include <pthread.h>
+#include <poll.h>
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+#include <sys/prctl.h>
+#include <sys/stat.h>
+#include <sys/socket.h>
+#include <sys/types.h>
+#include <utils/Log.h>
+#include <signal.h>
+#include <errno.h>
+#include "qahw_api.h"
+#include "qahw_defs.h"
+
+/* add local define to prevent compilation errors on other platforms */
+#ifndef AUDIO_DEVICE_IN_HDMI_ARC
+#define AUDIO_DEVICE_IN_HDMI_ARC (AUDIO_DEVICE_BIT_IN | 0x8000000)
+#endif
+
+static int sock_event_fd = -1;
+
+void *context = NULL;
+FILE * log_file = NULL;
+volatile bool stop_test = false;
+volatile bool stop_record = false;
+volatile bool record_active = false;
+
+#define HDMI_SYS_PATH "/sys/devices/platform/soc/78b7000.i2c/i2c-3/3-0064/"
+const char hdmi_in_audio_sys_path[] = HDMI_SYS_PATH "link_on0";
+const char hdmi_in_power_on_sys_path[] = HDMI_SYS_PATH "power_on";
+const char hdmi_in_audio_path_sys_path[] = HDMI_SYS_PATH "audio_path";
+const char hdmi_in_arc_enable_sys_path[] = HDMI_SYS_PATH "arc_enable";
+
+const char hdmi_in_audio_state_sys_path[] = HDMI_SYS_PATH "audio_state";
+const char hdmi_in_audio_format_sys_path[] = HDMI_SYS_PATH "audio_format";
+const char hdmi_in_audio_sample_rate_sys_path[] = HDMI_SYS_PATH "audio_rate";
+const char hdmi_in_audio_layout_sys_path[] = HDMI_SYS_PATH "audio_layout";
+
+#define SPDIF_SYS_PATH "/sys/devices/platform/soc/soc:qcom,msm-dai-q6-spdif-pri-tx/"
+const char spdif_in_audio_state_sys_path[] = SPDIF_SYS_PATH "audio_state";
+const char spdif_in_audio_format_sys_path[] = SPDIF_SYS_PATH "audio_format";
+const char spdif_in_audio_sample_rate_sys_path[] = SPDIF_SYS_PATH "audio_rate";
+
+#define SPDIF_ARC_SYS_PATH "/sys/devices/platform/soc/soc:qcom,msm-dai-q6-spdif-sec-tx/"
+const char spdif_arc_in_audio_state_sys_path[] = SPDIF_ARC_SYS_PATH "audio_state";
+const char spdif_arc_in_audio_format_sys_path[] = SPDIF_ARC_SYS_PATH "audio_format";
+const char spdif_arc_in_audio_sample_rate_sys_path[] = SPDIF_ARC_SYS_PATH "audio_rate";
+
+#define ID_RIFF 0x46464952
+#define ID_WAVE 0x45564157
+#define ID_FMT 0x20746d66
+#define ID_DATA 0x61746164
+
+#define FORMAT_PCM 1
+
+struct wav_header {
+ uint32_t riff_id;
+ uint32_t riff_sz;
+ uint32_t riff_fmt;
+ uint32_t fmt_id;
+ uint32_t fmt_sz;
+ uint16_t audio_format;
+ uint16_t num_channels;
+ uint32_t sample_rate;
+ uint32_t byte_rate; /* sample_rate * num_channels * bps / 8 */
+ uint16_t block_align; /* num_channels * bps / 8 */
+ uint16_t bits_per_sample;
+ uint32_t data_id;
+ uint32_t data_sz;
+};
+
+struct test_data {
+ qahw_module_handle_t *qahw_mod_handle;
+ audio_io_handle_t handle;
+ audio_devices_t input_device;
+ double record_length;
+ int rec_cnt;
+
+ char *audio_fmt_chg_text;
+ int audio_fmt_chg_len;
+ pthread_t record_th;
+ pthread_t poll_event_th;
+ pthread_attr_t poll_event_attr;
+
+ int bit_width;
+ audio_input_flags_t flags;
+ audio_config_t config;
+ audio_source_t source;
+
+ int spdif_audio_state;
+ int spdif_audio_mode;
+ int spdif_sample_rate;
+ int spdif_num_channels;
+
+ int hdmi_power_on;
+ int hdmi_audio_path;
+ int hdmi_arc_enable;
+
+ int hdmi_audio_state;
+ int hdmi_audio_mode;
+ int hdmi_audio_layout;
+ int hdmi_sample_rate;
+ int hdmi_num_channels;
+
+ int spdif_arc_audio_state;
+ int spdif_arc_audio_mode;
+ int spdif_arc_sample_rate;
+ int spdif_arc_num_channels;
+
+ audio_devices_t new_input_device;
+
+ audio_devices_t act_input_device; /* HDMI might use I2S and SPDIF */
+
+ int act_audio_state; /* audio active */
+ int act_audio_mode; /* 0=LPCM, 1=Compr */
+ int act_sample_rate; /* transmission sample rate */
+ int act_num_channels; /* transmission channels */
+};
+
+struct test_data tdata;
+
+void stop_signal_handler(int signal)
+{
+ stop_test = true;
+}
+
+void *start_input(void *thread_param) {
+ int rc = 0, ret = 0, count = 0;
+ ssize_t bytes_read = -1;
+ char file_name[256] = "/data/rec";
+ int data_sz = 0, name_len = strlen(file_name);
+ qahw_in_buffer_t in_buf;
+
+ qahw_module_handle_t *qahw_mod_handle = tdata.qahw_mod_handle;
+
+ /* convert/check params before use */
+ tdata.config.sample_rate = tdata.act_sample_rate;
+
+ if (tdata.act_audio_mode) {
+ tdata.config.format = AUDIO_FORMAT_IEC61937;
+ tdata.flags = QAHW_INPUT_FLAG_COMPRESS | QAHW_INPUT_FLAG_PASSTHROUGH;
+ } else {
+ if (tdata.bit_width == 32)
+ tdata.config.format = AUDIO_FORMAT_PCM_8_24_BIT;
+ else if (tdata.bit_width == 24)
+ tdata.config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ else
+ tdata.config.format = AUDIO_FORMAT_PCM_16_BIT;
+ tdata.flags = 0;
+ }
+
+ switch (tdata.act_num_channels) {
+ case 2:
+ tdata.config.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ break;
+ case 8:
+ tdata.config.channel_mask = AUDIO_CHANNEL_INDEX_MASK_8;
+ break;
+ default:
+ fprintf(log_file,
+ "ERROR :::: channel count %d not supported\n",
+ tdata.act_num_channels);
+ pthread_exit(0);
+ }
+ tdata.config.frame_count = 0;
+
+ /* Open audio input stream */
+ qahw_stream_handle_t* in_handle = NULL;
+
+ rc = qahw_open_input_stream(qahw_mod_handle, tdata.handle,
+ tdata.act_input_device, &tdata.config, &in_handle, tdata.flags,
+ "input_stream", tdata.source);
+ if (rc) {
+ fprintf(log_file,
+ "ERROR :::: Could not open input stream, handle(%d)\n",
+ tdata.handle);
+ pthread_exit(0);
+ }
+
+ /* Get buffer size to get upper bound on data to read from the HAL */
+ size_t buffer_size = qahw_in_get_buffer_size(in_handle);
+ char *buffer = (char *) calloc(1, buffer_size);
+ size_t written_size;
+ if (buffer == NULL) {
+ fprintf(log_file, "calloc failed!!, handle(%d)\n", tdata.handle);
+ pthread_exit(0);
+ }
+
+ fprintf(log_file, " input opened, buffer %p, size %zu, handle(%d)\n", buffer,
+ buffer_size, tdata.handle);
+
+ /* set profile for the recording session */
+ qahw_in_set_parameters(in_handle, "audio_stream_profile=record_unprocessed");
+
+ if (audio_is_linear_pcm(tdata.config.format))
+ snprintf(file_name + name_len, sizeof(file_name) - name_len, "%d.wav",
+ tdata.rec_cnt);
+ else
+ snprintf(file_name + name_len, sizeof(file_name) - name_len, "%d.raw",
+ tdata.rec_cnt);
+
+ tdata.rec_cnt++;
+
+ FILE *fd = fopen(file_name, "w");
+ if (fd == NULL) {
+ fprintf(log_file, "File open failed\n");
+ free(buffer);
+ pthread_exit(0);
+ }
+
+ int bps = 16;
+
+ switch (tdata.config.format) {
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ bps = 24;
+ break;
+ case AUDIO_FORMAT_PCM_8_24_BIT:
+ case AUDIO_FORMAT_PCM_32_BIT:
+ bps = 32;
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ default:
+ bps = 16;
+ }
+
+ struct wav_header hdr;
+ hdr.riff_id = ID_RIFF;
+ hdr.riff_sz = 0;
+ hdr.riff_fmt = ID_WAVE;
+ hdr.fmt_id = ID_FMT;
+ hdr.fmt_sz = 16;
+ hdr.audio_format = FORMAT_PCM;
+ hdr.num_channels = tdata.act_num_channels;
+ hdr.sample_rate = tdata.config.sample_rate;
+ hdr.byte_rate = hdr.sample_rate * hdr.num_channels * (bps / 8);
+ hdr.block_align = hdr.num_channels * (bps / 8);
+ hdr.bits_per_sample = bps;
+ hdr.data_id = ID_DATA;
+ hdr.data_sz = 0;
+ if (audio_is_linear_pcm(tdata.config.format))
+ fwrite(&hdr, 1, sizeof(hdr), fd);
+
+ memset(&in_buf, 0, sizeof(qahw_in_buffer_t));
+ while (true && !stop_record) {
+ in_buf.buffer = buffer;
+ in_buf.bytes = buffer_size;
+ bytes_read = qahw_in_read(in_handle, &in_buf);
+
+ written_size = fwrite(in_buf.buffer, 1, bytes_read, fd);
+ if (written_size < bytes_read) {
+ printf("Error in fwrite(%d)=%s\n", ferror(fd),
+ strerror(ferror(fd)));
+ break;
+ }
+ data_sz += bytes_read;
+ }
+
+ if (audio_is_linear_pcm(tdata.config.format)) {
+ /* update lengths in header */
+ hdr.data_sz = data_sz;
+ hdr.riff_sz = data_sz + 44 - 8;
+ fseek(fd, 0, SEEK_SET);
+ fwrite(&hdr, 1, sizeof(hdr), fd);
+ }
+ free(buffer);
+ fclose(fd);
+ fd = NULL;
+
+ fprintf(log_file, " closing input, handle(%d), written %d bytes", tdata.handle, data_sz);
+
+ /* Close input stream and device. */
+ rc = qahw_in_standby(in_handle);
+ if (rc) {
+ fprintf(log_file, "in standby failed %d, handle(%d)\n", rc,
+ tdata.handle);
+ }
+
+ rc = qahw_close_input_stream(in_handle);
+ if (rc) {
+ fprintf(log_file, "could not close input stream %d, handle(%d)\n", rc,
+ tdata.handle);
+ }
+
+ fprintf(log_file,
+ "\n\n The audio recording has been saved to %s.\n"
+ "The audio data has the following characteristics:\n Sample rate: %i\n Format: %d\n "
+ "Num channels: %i, handle(%d)\n\n", file_name,
+ tdata.config.sample_rate, tdata.config.format, tdata.act_num_channels,
+ tdata.handle);
+
+ return NULL;
+}
+
+void start_rec_thread(void)
+{
+ int ret = 0;
+
+ stop_record = false;
+ record_active = true;
+
+ fprintf(log_file, "\n Create record thread \n");
+ ret = pthread_create(&tdata.record_th, NULL, start_input, (void *)&tdata);
+ if (ret) {
+ fprintf(log_file, " Failed to create record thread\n");
+ exit(1);
+ }
+}
+
+void stop_rec_thread(void)
+{
+ if (record_active) {
+ record_active = false;
+ stop_record = true;
+ fprintf(log_file, "\n Stop record thread \n");
+ pthread_join(tdata.record_th, NULL);
+ }
+}
+
+
+void read_data_from_fd(const char* path, int *value)
+{
+ int fd = -1;
+ char buf[16];
+ int ret;
+
+ fd = open(path, O_RDONLY, 0);
+ if (fd < 0) {
+ ALOGE("Unable open fd for file %s", path);
+ return;
+ }
+
+ ret = read(fd, buf, 15);
+ if (ret < 0) {
+ ALOGE("File %s Data is empty", path);
+ close(fd);
+ return;
+ }
+
+ buf[ret] = '\0';
+ *value = atoi(buf);
+ close(fd);
+}
+
+void get_input_status()
+{
+ switch (tdata.input_device) {
+ case AUDIO_DEVICE_IN_SPDIF:
+ read_data_from_fd(spdif_in_audio_state_sys_path, &tdata.spdif_audio_state);
+ read_data_from_fd(spdif_in_audio_format_sys_path, &tdata.spdif_audio_mode);
+ read_data_from_fd(spdif_in_audio_sample_rate_sys_path, &tdata.spdif_sample_rate);
+ tdata.spdif_num_channels = 2;
+ tdata.new_input_device = AUDIO_DEVICE_IN_SPDIF;
+
+ fprintf(log_file, "spdif audio_state: %d, audio_format: %d, sample_rate: %d, num_channels: %d\n",
+ tdata.spdif_audio_state, tdata.spdif_audio_mode, tdata.spdif_sample_rate, tdata.spdif_num_channels);
+ break;
+ case AUDIO_DEVICE_IN_HDMI:
+ read_data_from_fd(hdmi_in_power_on_sys_path, &tdata.hdmi_power_on);
+ read_data_from_fd(hdmi_in_audio_path_sys_path, &tdata.hdmi_audio_path);
+ read_data_from_fd(hdmi_in_arc_enable_sys_path, &tdata.hdmi_arc_enable);
+
+ read_data_from_fd(hdmi_in_audio_state_sys_path, &tdata.hdmi_audio_state);
+ read_data_from_fd(hdmi_in_audio_format_sys_path, &tdata.hdmi_audio_mode);
+ read_data_from_fd(hdmi_in_audio_sample_rate_sys_path, &tdata.hdmi_sample_rate);
+ read_data_from_fd(hdmi_in_audio_layout_sys_path, &tdata.hdmi_audio_layout);
+ if (tdata.hdmi_audio_layout)
+ tdata.hdmi_num_channels = 8;
+ else
+ tdata.hdmi_num_channels = 2;
+ /* todo: read ch_count, ch_alloc */
+
+ read_data_from_fd(spdif_arc_in_audio_state_sys_path, &tdata.spdif_arc_audio_state);
+ read_data_from_fd(spdif_arc_in_audio_format_sys_path, &tdata.spdif_arc_audio_mode);
+ read_data_from_fd(spdif_arc_in_audio_sample_rate_sys_path, &tdata.spdif_arc_sample_rate);
+ tdata.spdif_arc_num_channels = 2;
+
+ if (tdata.hdmi_arc_enable ||
+ (tdata.hdmi_audio_state && (tdata.hdmi_audio_layout == 0) && tdata.hdmi_audio_mode)) {
+ tdata.new_input_device = AUDIO_DEVICE_IN_HDMI_ARC;
+ fprintf(log_file, "hdmi audio interface SPDIF_ARC\n");
+ } else {
+ tdata.new_input_device = AUDIO_DEVICE_IN_HDMI;
+ fprintf(log_file, "hdmi audio interface MI2S\n");
+ }
+
+ fprintf(log_file, "hdmi audio_state: %d, audio_format: %d, sample_rate: %d, num_channels: %d\n",
+ tdata.hdmi_audio_state, tdata.hdmi_audio_mode, tdata.hdmi_sample_rate, tdata.hdmi_num_channels);
+ fprintf(log_file, "arc audio_state: %d, audio_format: %d, sample_rate: %d, num_channels: %d\n",
+ tdata.spdif_arc_audio_state, tdata.spdif_arc_audio_mode, tdata.spdif_arc_sample_rate, tdata.spdif_arc_num_channels);
+ break;
+ }
+}
+
+void input_restart_check(void)
+{
+ get_input_status();
+
+ switch (tdata.input_device) {
+ case AUDIO_DEVICE_IN_SPDIF:
+ if ((tdata.act_input_device != tdata.new_input_device) ||
+ (tdata.spdif_audio_state == 2)) {
+ fprintf(log_file, "old audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.act_audio_state, tdata.act_audio_mode,
+ tdata.act_sample_rate, tdata.act_num_channels);
+ fprintf(log_file, "new spdif audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.spdif_audio_state, tdata.spdif_audio_mode,
+ tdata.spdif_sample_rate, tdata.spdif_num_channels);
+
+ stop_rec_thread();
+
+ tdata.act_input_device = AUDIO_DEVICE_IN_SPDIF;
+ tdata.act_audio_state = 1;
+ tdata.act_audio_mode = tdata.spdif_audio_mode;
+ tdata.act_sample_rate = tdata.spdif_sample_rate;
+ tdata.act_num_channels = tdata.spdif_num_channels;
+
+ start_rec_thread();
+ }
+ break;
+ case AUDIO_DEVICE_IN_HDMI:
+ if (tdata.act_input_device != tdata.new_input_device) {
+ stop_rec_thread();
+
+ if (tdata.new_input_device == AUDIO_DEVICE_IN_HDMI) {
+ fprintf(log_file, "old audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.act_audio_state, tdata.act_audio_mode,
+ tdata.act_sample_rate, tdata.act_num_channels);
+ fprintf(log_file, "new hdmi audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.hdmi_audio_state, tdata.hdmi_audio_mode,
+ tdata.hdmi_sample_rate, tdata.hdmi_num_channels);
+
+ tdata.act_input_device = AUDIO_DEVICE_IN_HDMI;
+ tdata.act_audio_state = tdata.hdmi_audio_state;
+ tdata.act_audio_mode = tdata.hdmi_audio_mode;
+ tdata.act_sample_rate = tdata.hdmi_sample_rate;
+ tdata.act_num_channels = tdata.hdmi_num_channels;
+
+ if (tdata.hdmi_audio_state)
+ start_rec_thread();
+ } else {
+ tdata.act_input_device = AUDIO_DEVICE_IN_HDMI_ARC;
+ if (tdata.hdmi_arc_enable) {
+ fprintf(log_file, "old audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.act_audio_state, tdata.act_audio_mode,
+ tdata.act_sample_rate, tdata.act_num_channels);
+ fprintf(log_file, "new arc audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.spdif_arc_audio_state, tdata.spdif_arc_audio_mode,
+ tdata.spdif_arc_sample_rate, tdata.spdif_arc_num_channels);
+
+ tdata.act_audio_state = 1;
+ tdata.act_audio_mode = tdata.spdif_arc_audio_mode;
+ tdata.act_sample_rate = tdata.spdif_arc_sample_rate;
+ tdata.act_num_channels = tdata.spdif_arc_num_channels;
+ } else {
+ fprintf(log_file, "old audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.act_audio_state, tdata.act_audio_mode,
+ tdata.act_sample_rate, tdata.act_num_channels);
+ fprintf(log_file, "new arc (from hdmi) audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.hdmi_audio_state, tdata.hdmi_audio_mode,
+ tdata.hdmi_sample_rate, tdata.hdmi_num_channels);
+
+ tdata.act_audio_state = 1;
+ tdata.act_audio_mode = tdata.hdmi_audio_mode;
+ tdata.act_sample_rate = tdata.hdmi_sample_rate;
+ tdata.act_num_channels = tdata.hdmi_num_channels;
+ }
+ start_rec_thread();
+ }
+ } else { /* check for change on same audio device */
+ if (tdata.new_input_device == AUDIO_DEVICE_IN_HDMI) {
+ if ((tdata.act_audio_state != tdata.hdmi_audio_state) ||
+ (tdata.act_audio_mode != tdata.hdmi_audio_mode) ||
+ (tdata.act_sample_rate != tdata.hdmi_sample_rate) ||
+ (tdata.act_num_channels != tdata.hdmi_num_channels)) {
+
+ fprintf(log_file, "old audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.act_audio_state, tdata.act_audio_mode,
+ tdata.act_sample_rate, tdata.act_num_channels);
+ fprintf(log_file, "new hdmi audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.hdmi_audio_state, tdata.hdmi_audio_mode,
+ tdata.hdmi_sample_rate, tdata.hdmi_num_channels);
+
+ stop_rec_thread();
+
+ tdata.act_audio_state = tdata.hdmi_audio_state;
+ tdata.act_audio_mode = tdata.hdmi_audio_mode;
+ tdata.act_sample_rate = tdata.hdmi_sample_rate;
+ tdata.act_num_channels = tdata.hdmi_num_channels;
+
+ if (tdata.hdmi_audio_state)
+ start_rec_thread();
+ }
+ } else {
+ if (tdata.spdif_arc_audio_state == 2) {
+ fprintf(log_file, "old audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.act_audio_state, tdata.act_audio_mode,
+ tdata.act_sample_rate, tdata.act_num_channels);
+ fprintf(log_file, "new arc audio_state: %d, audio_format: %d, rate: %d, channels: %d\n",
+ tdata.spdif_arc_audio_state, tdata.spdif_arc_audio_mode,
+ tdata.spdif_arc_sample_rate, tdata.spdif_arc_num_channels);
+
+ stop_rec_thread();
+
+ tdata.act_audio_state = 1;
+ tdata.act_audio_mode = tdata.spdif_arc_audio_mode;
+ tdata.act_sample_rate = tdata.spdif_arc_sample_rate;
+ tdata.act_num_channels = tdata.spdif_arc_num_channels;
+
+ start_rec_thread();
+ }
+ }
+ }
+ break;
+ }
+}
+
+int poll_event_init()
+{
+ struct sockaddr_nl sock_addr;
+ int sz = (64*1024);
+ int soc;
+
+ memset(&sock_addr, 0, sizeof(sock_addr));
+ sock_addr.nl_family = AF_NETLINK;
+ sock_addr.nl_pid = getpid();
+ sock_addr.nl_groups = 0xffffffff;
+
+ soc = socket(PF_NETLINK, SOCK_DGRAM, NETLINK_KOBJECT_UEVENT);
+ if (soc < 0) {
+ return 0;
+ }
+
+ setsockopt(soc, SOL_SOCKET, SO_RCVBUFFORCE, &sz, sizeof(sz));
+
+ if (bind(soc, (struct sockaddr*) &sock_addr, sizeof(sock_addr)) < 0) {
+ close(soc);
+ return 0;
+ }
+
+ sock_event_fd = soc;
+
+ return (soc > 0);
+}
+
+void* listen_uevent()
+{
+ char buffer[64*1024];
+ struct pollfd fds;
+ int i, count;
+ int j;
+ char *dev_path = NULL;
+ char *switch_state = NULL;
+ char *switch_name = NULL;
+ int audio_changed;
+
+ input_restart_check();
+
+ while(!stop_test) {
+
+ fds.fd = sock_event_fd;
+ fds.events = POLLIN;
+ fds.revents = 0;
+ i = poll(&fds, 1, 5); /* wait 5 msec */
+
+ if (i > 0 && (fds.revents & POLLIN)) {
+ count = recv(sock_event_fd, buffer, (64*1024), 0 );
+ if (count > 0) {
+ buffer[count] = '\0';
+ audio_changed = 0;
+ j = 0;
+ while(j < count) {
+ if (strncmp(&buffer[j], "DEVPATH=", 8) == 0) {
+ dev_path = &buffer[j+8];
+ j += 8;
+ continue;
+ } else if (tdata.input_device == AUDIO_DEVICE_IN_SPDIF) {
+ if (strncmp(&buffer[j], "PRI_SPDIF_TX=MEDIA_CONFIG_CHANGE", strlen("PRI_SPDIF_TX=MEDIA_CONFIG_CHANGE")) == 0) {
+ audio_changed = 1;
+ ALOGI("AUDIO CHANGE EVENT: %s\n", &buffer[j]);
+ j += strlen("PRI_SPDIF_TX=MEDIA_CONFIG_CHANGE");
+ continue;
+ }
+ } else if (tdata.input_device == AUDIO_DEVICE_IN_HDMI) {
+ if (strncmp(&buffer[j], "EP92EVT_AUDIO=MEDIA_CONFIG_CHANGE", strlen("EP92EVT_AUDIO=MEDIA_CONFIG_CHANGE")) == 0) {
+ audio_changed = 1;
+ ALOGI("AUDIO CHANGE EVENT: %s\n", &buffer[j]);
+ j += strlen("EP92EVT_AUDIO=MEDIA_CONFIG_CHANGE");
+ continue;
+ } else if (strncmp(&buffer[j], "SEC_SPDIF_TX=MEDIA_CONFIG_CHANGE", strlen("SEC_SPDIF_TX=MEDIA_CONFIG_CHANGE")) == 0) {
+ audio_changed = 1;
+ ALOGI("AUDIO CHANGE EVENT: %s\n", &buffer[j]);
+ j += strlen("SEC_SPDIF_TX=MEDIA_CONFIG_CHANGE");
+ continue;
+ } else if (strncmp(&buffer[j], "EP92EVT_", 8) == 0) {
+ ALOGI("EVENT: %s\n", &buffer[j]);
+ j += 8;
+ continue;
+ }
+ }
+ j++;
+ }
+
+ if (audio_changed)
+ input_restart_check();
+ }
+ } else {
+ ALOGV("NO Data\n");
+ }
+ }
+
+ stop_rec_thread();
+}
+
+void fill_default_params(struct test_data *tdata) {
+ memset(tdata, 0, sizeof(struct test_data));
+
+ tdata->input_device = AUDIO_DEVICE_IN_SPDIF;
+ tdata->bit_width = 24;
+ tdata->source = AUDIO_SOURCE_UNPROCESSED;
+ tdata->record_length = 8 /*sec*/;
+
+ tdata->handle = 0x99A;
+}
+
+void usage() {
+ printf(" \n Command \n");
+ printf(" \n fmt_change_test <options>\n");
+ printf(" \n Options\n");
+ printf(" -d --device <int> - see system/media/audio/include/system/audio.h for device values\n");
+ printf(" spdif_in 2147549184, hdmi_in 2147483680\n");
+ printf(" Optional Argument and Default value is spdif_in\n\n");
+ printf(" -b --bits <int> - Bitwidth in PCM mode (16, 24 or 32), Default is 24\n\n");
+ printf(" -F --flags <int> - Integer value of flags to be used for opening input stream\n\n");
+ printf(" -t --recording-time <in seconds> - Time duration for the recording\n\n");
+ printf(" -l --log-file <FILEPATH> - File path for debug msg, to print\n");
+ printf(" on console use stdout or 1 \n\n");
+ printf(" -h --help - Show this help\n\n");
+ printf(" \n Examples \n");
+ printf(" hdmi_in_event_test -> start a recording stream with default configurations\n\n");
+ printf(" hdmi_in_event_test -d 2147483680 -t 20 -> start a recording session, with device hdmi_in,\n");
+ printf(" record data for 20 secs.\n\n");
+}
+
+static void qti_audio_server_death_notify_cb(void *ctxt) {
+ fprintf(log_file, "qas died\n");
+ fprintf(stderr, "qas died\n");
+ stop_test = true;
+ stop_record = true;
+}
+
+int main(int argc, char* argv[])
+{
+ qahw_module_handle_t *qahw_mod_handle;
+ const char *mod_name = "audio.primary";
+
+ char log_filename[256] = "stdout";
+ int i;
+ int ret = -1;
+
+ log_file = stdout;
+ fill_default_params(&tdata);
+ struct option long_options[] = {
+ /* These options set a flag. */
+ {"device", required_argument, 0, 'd'},
+ {"bits", required_argument, 0, 'b'},
+ {"flags", required_argument, 0, 'F'},
+ {"recording-time", required_argument, 0, 't'},
+ {"log-file", required_argument, 0, 'l'},
+ {"help", no_argument, 0, 'h'},
+ {0, 0, 0, 0}
+ };
+
+ int opt = 0;
+ int option_index = 0;
+ while ((opt = getopt_long(argc,
+ argv,
+ "-d:b:F:t:l:h",
+ long_options,
+ &option_index)) != -1) {
+ switch (opt) {
+ case 'd':
+ tdata.input_device = atoll(optarg);
+ break;
+ case 'b':
+ tdata.bit_width = atoll(optarg);
+ break;
+ case 'F':
+ tdata.flags = atoll(optarg);
+ break;
+ case 't':
+ tdata.record_length = atoi(optarg);
+ break;
+ case 'l':
+ snprintf(log_filename, sizeof(log_filename), "%s", optarg);
+ break;
+ case 'h':
+ usage();
+ return 0;
+ break;
+ }
+ }
+ fprintf(log_file, "registering qas callback");
+ qahw_register_qas_death_notify_cb((audio_error_callback)qti_audio_server_death_notify_cb, context);
+
+ switch (tdata.input_device) {
+ case AUDIO_DEVICE_IN_SPDIF:
+ break;
+ case AUDIO_DEVICE_IN_HDMI:
+ break;
+ default:
+ fprintf(log_file, "device %d not supported\n", tdata.input_device);
+ return -1;
+ }
+
+ switch (tdata.bit_width) {
+ case 16:
+ case 24:
+ case 32:
+ break;
+ default:
+ fprintf(log_file, "bitwidth %d not supported\n", tdata.bit_width);
+ return -1;
+ }
+
+ qahw_mod_handle = qahw_load_module(mod_name);
+ if(qahw_mod_handle == NULL) {
+ fprintf(log_file, " qahw_load_module failed");
+ return -1;
+ }
+ fprintf(log_file, " Starting audio recording test. \n");
+ if (strcasecmp(log_filename, "stdout") && strcasecmp(log_filename, "1")) {
+ if ((log_file = fopen(log_filename,"wb"))== NULL) {
+ fprintf(stderr, "Cannot open log file %s\n", log_filename);
+ /* continue to log to std out */
+ log_file = stdout;
+ }
+ }
+
+ tdata.qahw_mod_handle = qahw_mod_handle;
+
+ /* Register the SIGINT to close the App properly */
+ if (signal(SIGINT, stop_signal_handler) == SIG_ERR)
+ fprintf(log_file, "Failed to register SIGINT:%d\n", errno);
+
+ /* Register the SIGTERM to close the App properly */
+ if (signal(SIGTERM, stop_signal_handler) == SIG_ERR)
+ fprintf(log_file, "Failed to register SIGTERM:%d\n", errno);
+
+ time_t start_time = time(0);
+ double time_elapsed = 0;
+
+ pthread_attr_init(&tdata.poll_event_attr);
+ pthread_attr_setdetachstate(&tdata.poll_event_attr, PTHREAD_CREATE_JOINABLE);
+ poll_event_init();
+ pthread_create(&tdata.poll_event_th, &tdata.poll_event_attr,
+ (void *) listen_uevent, NULL);
+
+ while(true && !stop_test) {
+ time_elapsed = difftime(time(0), start_time);
+ if (tdata.record_length && (time_elapsed > tdata.record_length)) {
+ fprintf(log_file, "\n Test completed.\n");
+ stop_test = true;
+ break;
+ }
+ }
+
+ fprintf(log_file, "\n Stop test \n");
+
+ pthread_join(tdata.poll_event_th, NULL);
+
+ fprintf(log_file, "\n Unload HAL\n");
+
+ ret = qahw_unload_module(qahw_mod_handle);
+ if (ret) {
+ fprintf(log_file, "could not unload hal %d\n", ret);
+ }
+
+ fprintf(log_file, "Done with hal record test\n");
+ if (log_file != stdout) {
+ if (log_file) {
+ fclose(log_file);
+ log_file = NULL;
+ }
+ }
+
+ return 0;
+}
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 5da769c..4441ab0 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -39,6 +39,10 @@
LOCAL_SRC_FILES += asphere.c
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_INSTANCE_ID)), true)
+ LOCAL_CFLAGS += -DINSTANCE_ID_ENABLED
+endif
+
LOCAL_CFLAGS+= -O2 -fvisibility=hidden
ifneq ($(strip $(AUDIO_FEATURE_DISABLED_DTS_EAGLE)),true)
@@ -110,10 +114,6 @@
LOCAL_CFLAGS += -DHW_ACC_HPX
endif
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_INSTANCE_ID)), true)
- LOCAL_CFLAGS += -DINSTANCE_ID_ENABLED
-endif
-
LOCAL_MODULE:= libhwacceffectswrapper
LOCAL_VENDOR_MODULE := true
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index 68cd46f..02c68d4 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -32,6 +32,8 @@
#include "effect_api.h"
#include "bass_boost.h"
+#define BASSBOOST_MAX_LATENCY 30
+
/* Offload bassboost UUID: 2c4a8c24-1581-487f-94f6-0002a5d5c51b */
const effect_descriptor_t bassboost_descriptor = {
{0x0634f220, 0xddd4, 0x11db, 0xa0fc, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b }},
@@ -101,6 +103,11 @@
p->status = -EINVAL;
p->vsize = sizeof(int16_t);
break;
+ case BASSBOOST_PARAM_LATENCY:
+ if (p->vsize < sizeof(uint32_t))
+ p->status = -EINVAL;
+ p->vsize = sizeof(uint32_t);
+ break;
default:
p->status = -EINVAL;
}
@@ -127,6 +134,10 @@
*(int16_t *)value = 0;
break;
+ case BASSBOOST_PARAM_LATENCY:
+ *(uint32_t *)value = BASSBOOST_MAX_LATENCY;
+ break;
+
default:
p->status = -EINVAL;
break;
diff --git a/post_proc/bass_boost.h b/post_proc/bass_boost.h
index 8bf51d3..ff674d4 100644
--- a/post_proc/bass_boost.h
+++ b/post_proc/bass_boost.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, 2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -20,6 +20,8 @@
#ifndef OFFLOAD_EFFECT_BASS_BOOST_H_
#define OFFLOAD_EFFECT_BASS_BOOST_H_
+#define BASSBOOST_PARAM_LATENCY 0x80000000
+
#include "bundle.h"
enum {
diff --git a/post_proc/equalizer.c b/post_proc/equalizer.c
index c1c1303..479f848 100644
--- a/post_proc/equalizer.c
+++ b/post_proc/equalizer.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, 2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, 2017-2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -29,6 +29,8 @@
#include "effect_api.h"
#include "equalizer.h"
+#define EQUALIZER_MAX_LATENCY 0
+
/* Offload equalizer UUID: a0dac280-401c-11e3-9379-0002a5d5c51b */
const effect_descriptor_t equalizer_descriptor = {
{0x0bed4300, 0xddd6, 0x11db, 0x8f34, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // type
@@ -253,6 +255,12 @@
p->vsize = (2 + NUM_EQ_BANDS) * sizeof(uint16_t);
break;
+ case EQ_PARAM_LATENCY:
+ if (p->vsize < sizeof(uint32_t))
+ p->status = -EINVAL;
+ p->vsize = sizeof(uint32_t);
+ break;
+
default:
p->status = -EINVAL;
}
@@ -352,6 +360,10 @@
}
} break;
+ case EQ_PARAM_LATENCY:
+ *(uint32_t *)value = EQUALIZER_MAX_LATENCY;
+ break;
+
default:
p->status = -EINVAL;
break;
diff --git a/post_proc/equalizer.h b/post_proc/equalizer.h
index 7fec058..2cd06c2 100644
--- a/post_proc/equalizer.h
+++ b/post_proc/equalizer.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, 2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -26,6 +26,8 @@
#define INVALID_PRESET -2
#define PRESET_CUSTOM -1
+#define EQ_PARAM_LATENCY 0x80000000
+
extern const effect_descriptor_t equalizer_descriptor;
typedef struct equalizer_context_s {
diff --git a/post_proc/reverb.c b/post_proc/reverb.c
index e97b651..a0a0441 100644
--- a/post_proc/reverb.c
+++ b/post_proc/reverb.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013 - 2014, 2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013 - 2014, 2017-2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -30,6 +30,8 @@
#include "effect_api.h"
#include "reverb.h"
+#define REVERB_MAX_LATENCY 100
+
/* Offload auxiliary environmental reverb UUID: 79a18026-18fd-4185-8233-0002a5d5c51b */
const effect_descriptor_t aux_env_reverb_descriptor = {
{ 0xc2e5d5f0, 0x94bd, 0x4763, 0x9cac, { 0x4e, 0x23, 0x4d, 0x06, 0x83, 0x9e } },
@@ -522,6 +524,11 @@
p->status = -EINVAL;
p->vsize = sizeof(reverb_settings_t);
break;
+ case REVERB_PARAM_LATENCY:
+ if (p->vsize < sizeof(uint32_t))
+ return -EINVAL;
+ p->vsize = sizeof(uint32_t);
+ break;
default:
p->status = -EINVAL;
}
@@ -575,6 +582,9 @@
reverb_settings->diffusion = reverb_get_diffusion(reverb_ctxt);
reverb_settings->density = reverb_get_density(reverb_ctxt);
break;
+ case REVERB_PARAM_LATENCY:
+ *(uint16_t *)value = REVERB_MAX_LATENCY;
+ break;
default:
p->status = -EINVAL;
break;
diff --git a/post_proc/reverb.h b/post_proc/reverb.h
index 3bdd9af..cc11c46 100644
--- a/post_proc/reverb.h
+++ b/post_proc/reverb.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, 2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -24,6 +24,8 @@
#define REVERB_DEFAULT_PRESET REVERB_PRESET_NONE
+#define REVERB_PARAM_LATENCY 0x80000000
+
extern const effect_descriptor_t aux_env_reverb_descriptor;
extern const effect_descriptor_t ins_env_reverb_descriptor;
extern const effect_descriptor_t aux_preset_reverb_descriptor;
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index dfa7691..578cf0b 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2015, 2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, 2017-2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -29,6 +29,8 @@
#include "effect_api.h"
#include "virtualizer.h"
+#define VIRUALIZER_MAX_LATENCY 30
+
/* Offload Virtualizer UUID: 509a4498-561a-4bea-b3b1-0002a5d5c51b */
const effect_descriptor_t virtualizer_descriptor = {
{0x37cc2c00, 0xdddd, 0x11db, 0x8577, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
@@ -304,6 +306,11 @@
p->status = -EINVAL;
p->vsize = sizeof(uint32_t);
break;
+ case VIRTUALIZER_PARAM_LATENCY:
+ if (p->vsize < sizeof(uint32_t))
+ p->status = -EINVAL;
+ p->vsize = sizeof(uint32_t);
+ break;
default:
p->status = -EINVAL;
}
@@ -347,6 +354,10 @@
*(uint32_t *)value = (uint32_t) virtualizer_get_virtualization_mode(virt_ctxt);
break;
+ case VIRTUALIZER_PARAM_LATENCY:
+ *(uint32_t *)value = VIRUALIZER_MAX_LATENCY;
+ break;
+
default:
p->status = -EINVAL;
break;
diff --git a/post_proc/virtualizer.h b/post_proc/virtualizer.h
index 904a0c6..c0e6a87 100644
--- a/post_proc/virtualizer.h
+++ b/post_proc/virtualizer.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, 2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -20,6 +20,8 @@
#ifndef OFFLOAD_VIRTUALIZER_H_
#define OFFLOAD_VIRTUALIZER_H_
+#define VIRTUALIZER_PARAM_LATENCY 0x80000000
+
#include "bundle.h"
extern const effect_descriptor_t virtualizer_descriptor;
diff --git a/qahw/inc/qahw.h b/qahw/inc/qahw.h
index e91fd00..dd5b403 100644
--- a/qahw/inc/qahw.h
+++ b/qahw/inc/qahw.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2016-2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2016-2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2011 The Android Open Source Project *
@@ -463,6 +463,13 @@
/* Release an audio patch */
int qahw_release_audio_patch_l(qahw_module_handle_t *hw_module,
audio_patch_handle_t handle);
+
+/* API to set loopback stream specific config parameters. */
+int qahw_loopback_set_param_data_l(qahw_module_handle_t *hw_module,
+ audio_patch_handle_t handle,
+ qahw_loopback_param_id param_id,
+ qahw_loopback_param_payload *payload);
+
/* Fills the list of supported attributes for a given audio port.
* As input, "port" contains the information (type, role, address etc...)
* needed by the HAL to identify the port.
diff --git a/qahw/inc/qahw_defs.h b/qahw/inc/qahw_defs.h
index 4e7faff..755553b 100644
--- a/qahw/inc/qahw_defs.h
+++ b/qahw/inc/qahw_defs.h
@@ -417,6 +417,14 @@
QAHW_PARAM_LICENSE_PARAMS,
} qahw_param_id;
+typedef union {
+ struct qahw_out_render_window_param render_window_params;
+} qahw_loopback_param_payload;
+
+typedef enum {
+ QAHW_PARAM_LOOPBACK_RENDER_WINDOW /* PARAM to set render window */
+} qahw_loopback_param_id;
+
__END_DECLS
#endif // QTI_AUDIO_HAL_DEFS_H
diff --git a/qahw/src/qahw.c b/qahw/src/qahw.c
index 0c00158..126f794 100644
--- a/qahw/src/qahw.c
+++ b/qahw/src/qahw.c
@@ -69,6 +69,10 @@
qahw_param_id param_id,
qahw_param_payload *payload);
+typedef int (*qahwi_loopback_set_param_data_t)(audio_patch_handle_t patch_handle,
+ qahw_param_id param_id,
+ qahw_param_payload *payload);
+
typedef struct {
audio_hw_device_t *audio_device;
char module_name[MAX_MODULE_NAME_LENGTH];
@@ -80,6 +84,7 @@
const hw_module_t* module;
qahwi_get_param_data_t qahwi_get_param_data;
qahwi_set_param_data_t qahwi_set_param_data;
+ qahwi_loopback_set_param_data_t qahwi_loopback_set_param_data;
} qahw_module_t;
typedef struct {
@@ -1438,6 +1443,34 @@
return ret;
}
+int qahw_loopback_set_param_data_l(qahw_module_handle_t *hw_module,
+ audio_patch_handle_t handle,
+ qahw_loopback_param_id param_id,
+ qahw_loopback_param_payload *payload)
+
+{
+ int ret = -EINVAL;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+
+ if (!payload) {
+ ALOGE("%s:: invalid param", __func__);
+ goto exit;
+ }
+
+ if (qahw_module->qahwi_loopback_set_param_data) {
+ ret = qahw_module->qahwi_loopback_set_param_data(handle,
+ param_id,
+ (void *)payload);
+ } else {
+ ret = -ENOSYS;
+ ALOGE("%s not supported\n", __func__);
+ }
+
+exit:
+ return ret;
+
+}
+
/* Fills the list of supported attributes for a given audio port.
* As input, "port" contains the information (type, role, address etc...)
* needed by the HAL to identify the port.
@@ -1889,6 +1922,12 @@
if (!qahw_module->qahwi_set_param_data)
ALOGD("%s::qahwi_set_param_data api is not defined\n",__func__);
+ qahw_module->qahwi_loopback_set_param_data = (qahwi_loopback_set_param_data_t)
+ dlsym(module->dso,
+ "qahwi_loopback_set_param_data");
+ if (!qahw_module->qahwi_loopback_set_param_data)
+ ALOGD("%s::qahwi_loopback_set_param_data api is not defined\n", __func__);
+
if (!qahw_list_count)
list_init(&qahw_module_list);
qahw_list_count++;
diff --git a/qahw_api/inc/qahw_api.h b/qahw_api/inc/qahw_api.h
index 0aa3c79..823c6bb 100644
--- a/qahw_api/inc/qahw_api.h
+++ b/qahw_api/inc/qahw_api.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2016-2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2016-2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2011 The Android Open Source Project *
@@ -459,6 +459,13 @@
/* Release an audio patch */
int qahw_release_audio_patch(qahw_module_handle_t *hw_module,
audio_patch_handle_t handle);
+
+/* API to set loopback stream specific config parameters */
+int qahw_loopback_set_param_data(qahw_module_handle_t *hw_module,
+ audio_patch_handle_t handle,
+ qahw_loopback_param_id param_id,
+ qahw_loopback_param_payload *payload);
+
/* Fills the list of supported attributes for a given audio port.
* As input, "port" contains the information (type, role, address etc...)
* needed by the HAL to identify the port.
diff --git a/qahw_api/inc/qahw_defs.h b/qahw_api/inc/qahw_defs.h
index c6d42ca..7c01c57 100644
--- a/qahw_api/inc/qahw_defs.h
+++ b/qahw_api/inc/qahw_defs.h
@@ -399,6 +399,15 @@
QAHW_PARAM_LICENSE_PARAMS,
} qahw_param_id;
+
+typedef union {
+ struct qahw_out_render_window_param render_window_params;
+} qahw_loopback_param_payload;
+
+typedef enum {
+ QAHW_PARAM_LOOPBACK_RENDER_WINDOW /* PARAM to set render window */
+} qahw_loopback_param_id;
+
__END_DECLS
#endif // QTI_AUDIO_HAL_DEFS_H
diff --git a/qahw_api/inc/qahw_effect_bassboost.h b/qahw_api/inc/qahw_effect_bassboost.h
index 2ca8409..b397f21 100644
--- a/qahw_api/inc/qahw_effect_bassboost.h
+++ b/qahw_api/inc/qahw_effect_bassboost.h
@@ -40,7 +40,9 @@
typedef enum
{
BASSBOOST_PARAM_STRENGTH_SUPPORTED,
- BASSBOOST_PARAM_STRENGTH
+ BASSBOOST_PARAM_STRENGTH,
+ BASSBOOST_PARAM_LATENCY = 0x80000000 // Internal paramter specific to qahw.
+ // Used to get latency introduced by bassboost effect.
} qahw_bassboost_params;
#ifdef __cplusplus
diff --git a/qahw_api/inc/qahw_effect_environmentalreverb.h b/qahw_api/inc/qahw_effect_environmentalreverb.h
index a47eb28..61ef39e 100644
--- a/qahw_api/inc/qahw_effect_environmentalreverb.h
+++ b/qahw_api/inc/qahw_effect_environmentalreverb.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2017-2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2011 The Android Open Source Project
@@ -22,7 +22,7 @@
#include <qahw_effect_api.h>
-#if __cplusplus
+#ifdef __cplusplus
extern "C" {
#endif
@@ -55,7 +55,9 @@
REVERB_PARAM_DIFFUSION, // in permilles, range 0 to 1000
REVERB_PARAM_DENSITY, // in permilles, range 0 to 1000
REVERB_PARAM_PROPERTIES,
- REVERB_PARAM_BYPASS
+ REVERB_PARAM_BYPASS,
+ REVERB_PARAM_LATENCY = 0x80000000 // Internal paramter specific to qahw.
+ // Used to get latency introduced by reverb effect.
} qahw_env_reverb_params;
//qahw_reverb_settings is equal to SLEnvironmentalReverbSettings defined in OpenSL ES specification.
@@ -73,7 +75,7 @@
} __attribute__((packed)) qahw_reverb_settings;
-#if __cplusplus
+#ifdef __cplusplus
} // extern "C"
#endif
diff --git a/qahw_api/inc/qahw_effect_equalizer.h b/qahw_api/inc/qahw_effect_equalizer.h
index fd71c4c..e4d6c5b 100644
--- a/qahw_api/inc/qahw_effect_equalizer.h
+++ b/qahw_api/inc/qahw_effect_equalizer.h
@@ -50,7 +50,9 @@
EQ_PARAM_CUR_PRESET, // Gets/Sets the current preset.
EQ_PARAM_GET_NUM_OF_PRESETS, // Gets the total number of presets the equalizer supports.
EQ_PARAM_GET_PRESET_NAME, // Gets the preset name based on the index.
- EQ_PARAM_PROPERTIES // Gets/Sets all parameters at a time.
+ EQ_PARAM_PROPERTIES, // Gets/Sets all parameters at a time.
+ EQ_PARAM_LATENCY = 0x80000000 // Internal paramter specific to qahw.
+ // Used to get latency introduced by equalizer effect.
} qahw_equalizer_params;
enum
diff --git a/qahw_api/inc/qahw_effect_virtualizer.h b/qahw_api/inc/qahw_effect_virtualizer.h
index 5ff95ce..481f0ef 100644
--- a/qahw_api/inc/qahw_effect_virtualizer.h
+++ b/qahw_api/inc/qahw_effect_virtualizer.h
@@ -75,7 +75,10 @@
// AUDIO_DEVICE_NONE when not virtualizing
// status int -EINVAL if an error occurred
// 0 if the output value is successfully retrieved
- VIRTUALIZER_PARAM_VIRTUALIZATION_MODE
+ VIRTUALIZER_PARAM_VIRTUALIZATION_MODE,
+ // Internal paramter specific to qahw.
+ // Used to get latency introduced by virtuaizer effect.
+ VIRTUALIZER_PARAM_LATENCY = 0x80000000
} qahw_virtualizer_params;
#ifdef __cplusplus
diff --git a/qahw_api/src/qahw_api.cpp b/qahw_api/src/qahw_api.cpp
index cbd9041..f1c75f4 100644
--- a/qahw_api/src/qahw_api.cpp
+++ b/qahw_api/src/qahw_api.cpp
@@ -1,5 +1,5 @@
/*
-* Copyright (c) 2016-2017, The Linux Foundation. All rights reserved.
+* Copyright (c) 2016-2018, The Linux Foundation. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are
@@ -915,6 +915,15 @@
}
}
+int qahw_loopback_set_param_data(qahw_module_handle_t *hw_module __unused,
+ audio_patch_handle_t handle __unused,
+ qahw_loopback_param_id param_id __unused,
+ qahw_loopback_param_payload *payload __unused)
+{
+ ALOGD("%d:%s", __LINE__, __func__);
+ return -ENOSYS;
+}
+
int qahw_get_audio_port(qahw_module_handle_t *hw_module,
struct audio_port *port)
{
@@ -1699,6 +1708,15 @@
return qahw_release_audio_patch_l(hw_module, handle);
}
+int qahw_loopback_set_param_data(qahw_module_handle_t *hw_module,
+ audio_patch_handle_t handle,
+ qahw_loopback_param_id param_id,
+ qahw_loopback_param_payload *payload)
+{
+ ALOGV("%d:%s\n", __LINE__, __func__);
+ return qahw_loopback_set_param_data_l(hw_module, handle, param_id, payload);
+}
+
int qahw_get_audio_port(qahw_module_handle_t *hw_module,
struct audio_port *port)
{
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
index 253d59e..556f520 100644
--- a/qahw_api/test/qahw_playback_test.c
+++ b/qahw_api/test/qahw_playback_test.c
@@ -2571,6 +2571,7 @@
fprintf(log_file, "stream %d: Output Flags:%d\n", stream->stream_index, stream->flags);
fprintf(log_file, "stream %d: Sample Rate:%d\n", stream->stream_index, stream->config.offload_info.sample_rate);
fprintf(log_file, "stream %d: Channels:%d\n", stream->stream_index, stream->channels);
+ fprintf(log_file, "stream %d: Channel Mask:%x\n", stream->stream_index, stream->config.channel_mask);
fprintf(log_file, "stream %d: Bitwidth:%d\n", stream->stream_index, stream->config.offload_info.bit_width);
fprintf(log_file, "stream %d: AAC Format Type:%d\n", stream->stream_index, stream->aac_fmt_type);
fprintf(log_file, "stream %d: Kvpair Values:%s\n", stream->stream_index, stream->kvpair_values);