hal: Add platform specific code for 8909
Add 8909 specific platform specific code for:
-platform initialization
-pcm device selection
-pcm stream configuration
-sound device selection
-acdb ID selection
Change-Id: I88a50b758311a8a77669434a96f738afa8ef39e0
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
new file mode 100644
index 0000000..a386652
--- /dev/null
+++ b/hal/msm8916/platform.h
@@ -0,0 +1,271 @@
+/*
+ *
+ * Copyright (C) 2013 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.
+ */
+
+#ifndef QCOM_AUDIO_PLATFORM_H
+#define QCOM_AUDIO_PLATFORM_H
+
+enum {
+ FLUENCE_NONE,
+ FLUENCE_DUAL_MIC = 0x1,
+ FLUENCE_QUAD_MIC = 0x2,
+};
+
+enum {
+ FLUENCE_ENDFIRE = 0x1,
+ FLUENCE_BROADSIDE = 0x2,
+};
+
+/*
+ * Below are the devices for which is back end is same, SLIMBUS_0_RX.
+ * All these devices are handled by the internal HW codec. We can
+ * enable any one of these devices at any time. An exception here is
+ * 44.1k headphone which uses different backend. This is filtered
+ * as different hal internal device in the code but remains same
+ * as standard android device AUDIO_DEVICE_OUT_WIRED_HEADPHONE
+ * for other layers.
+ */
+#define AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND \
+ (AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \
+ AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE|\
+ AUDIO_DEVICE_OUT_LINE)
+
+/* Sound devices specific to the platform
+ * The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound
+ * devices to enable corresponding mixer paths
+ */
+enum {
+ SND_DEVICE_NONE = 0,
+
+ /* Playback devices */
+ SND_DEVICE_MIN,
+ SND_DEVICE_OUT_BEGIN = SND_DEVICE_MIN,
+ SND_DEVICE_OUT_HANDSET = SND_DEVICE_OUT_BEGIN,
+ SND_DEVICE_OUT_SPEAKER,
+ SND_DEVICE_OUT_SPEAKER_REVERSE,
+ SND_DEVICE_OUT_LINE,
+ SND_DEVICE_OUT_HEADPHONES,
+ SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+ SND_DEVICE_OUT_SPEAKER_AND_LINE,
+ SND_DEVICE_OUT_VOICE_HANDSET,
+ SND_DEVICE_OUT_VOICE_HAC_HANDSET,
+ SND_DEVICE_OUT_VOICE_SPEAKER,
+ SND_DEVICE_OUT_VOICE_HEADPHONES,
+ SND_DEVICE_OUT_VOICE_LINE,
+ SND_DEVICE_OUT_HDMI,
+ SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+ SND_DEVICE_OUT_BT_SCO,
+ SND_DEVICE_OUT_BT_SCO_WB,
+ SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
+ SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
+ SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
+ SND_DEVICE_OUT_VOICE_TX,
+ SND_DEVICE_OUT_AFE_PROXY,
+ SND_DEVICE_OUT_USB_HEADSET,
+ SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET,
+ SND_DEVICE_OUT_ANC_HEADSET,
+ SND_DEVICE_OUT_ANC_FB_HEADSET,
+ SND_DEVICE_OUT_VOICE_ANC_HEADSET,
+ SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET,
+ SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+ SND_DEVICE_OUT_ANC_HANDSET,
+ SND_DEVICE_OUT_SPEAKER_PROTECTED,
+ SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+ SND_DEVICE_OUT_END,
+
+ /*
+ * Note: IN_BEGIN should be same as OUT_END because total number of devices
+ * SND_DEVICES_MAX should not exceed MAX_RX + MAX_TX devices.
+ */
+ /* Capture devices */
+ SND_DEVICE_IN_BEGIN = SND_DEVICE_OUT_END,
+ SND_DEVICE_IN_HANDSET_MIC = SND_DEVICE_IN_BEGIN,
+ SND_DEVICE_IN_HANDSET_MIC_EXTERNAL,
+ SND_DEVICE_IN_HANDSET_MIC_AEC,
+ SND_DEVICE_IN_HANDSET_MIC_NS,
+ SND_DEVICE_IN_HANDSET_MIC_AEC_NS,
+ SND_DEVICE_IN_HANDSET_DMIC,
+ SND_DEVICE_IN_HANDSET_DMIC_AEC,
+ SND_DEVICE_IN_HANDSET_DMIC_NS,
+ SND_DEVICE_IN_HANDSET_DMIC_AEC_NS,
+ SND_DEVICE_IN_SPEAKER_MIC,
+ SND_DEVICE_IN_SPEAKER_MIC_AEC,
+ SND_DEVICE_IN_SPEAKER_MIC_NS,
+ SND_DEVICE_IN_SPEAKER_MIC_AEC_NS,
+ SND_DEVICE_IN_SPEAKER_DMIC,
+ SND_DEVICE_IN_SPEAKER_DMIC_AEC,
+ SND_DEVICE_IN_SPEAKER_DMIC_NS,
+ SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS,
+ SND_DEVICE_IN_HEADSET_MIC,
+ SND_DEVICE_IN_HEADSET_MIC_FLUENCE,
+ SND_DEVICE_IN_VOICE_SPEAKER_MIC,
+ SND_DEVICE_IN_VOICE_HEADSET_MIC,
+ SND_DEVICE_IN_HDMI_MIC,
+ SND_DEVICE_IN_BT_SCO_MIC,
+ SND_DEVICE_IN_BT_SCO_MIC_NREC,
+ SND_DEVICE_IN_BT_SCO_MIC_WB,
+ SND_DEVICE_IN_BT_SCO_MIC_WB_NREC,
+ SND_DEVICE_IN_CAMCORDER_MIC,
+ SND_DEVICE_IN_VOICE_DMIC,
+ SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
+ SND_DEVICE_IN_VOICE_SPEAKER_QMIC,
+ SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC,
+ SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC,
+ SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC,
+ SND_DEVICE_IN_VOICE_REC_MIC,
+ SND_DEVICE_IN_VOICE_REC_MIC_NS,
+ SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
+ SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
+ SND_DEVICE_IN_VOICE_RX,
+ SND_DEVICE_IN_USB_HEADSET_MIC,
+ SND_DEVICE_IN_CAPTURE_FM,
+ SND_DEVICE_IN_AANC_HANDSET_MIC,
+ SND_DEVICE_IN_QUAD_MIC,
+ SND_DEVICE_IN_HANDSET_STEREO_DMIC,
+ SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
+ SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE,
+ SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE,
+ SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE,
+ SND_DEVICE_IN_SPEAKER_DMIC_NS_BROADSIDE,
+ SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS_BROADSIDE,
+ SND_DEVICE_IN_VOICE_FLUENCE_DMIC_AANC,
+ SND_DEVICE_IN_HANDSET_QMIC,
+ SND_DEVICE_IN_SPEAKER_QMIC_AEC,
+ SND_DEVICE_IN_SPEAKER_QMIC_NS,
+ SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS,
+ SND_DEVICE_IN_END,
+
+ SND_DEVICE_MAX = SND_DEVICE_IN_END,
+
+};
+
+#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+
+#define ALL_SESSION_VSID 0xFFFFFFFF
+#define DEFAULT_MUTE_RAMP_DURATION_MS 20
+#define DEFAULT_VOLUME_RAMP_DURATION_MS 20
+#define MIXER_PATH_MAX_LENGTH 100
+
+#define MAX_VOL_INDEX 5
+#define MIN_VOL_INDEX 0
+#define percent_to_index(val, min, max) \
+ ((val) * ((max) - (min)) * 0.01 + (min) + .5)
+
+/*
+ * tinyAlsa library interprets period size as number of frames
+ * one frame = channel_count * sizeof (pcm sample)
+ * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
+ * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
+ * We should take care of returning proper size when AudioFlinger queries for
+ * the buffer size of an input/output stream
+ */
+#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
+#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
+#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
+
+#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
+#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
+#define LOW_LATENCY_CAPTURE_USE_CASE 1
+
+#define HDMI_MULTI_PERIOD_SIZE 336
+#define HDMI_MULTI_PERIOD_COUNT 8
+#define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
+#define HDMI_MULTI_PERIOD_BYTES (HDMI_MULTI_PERIOD_SIZE * HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)
+
+#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
+#define AUDIO_CAPTURE_PERIOD_COUNT 2
+
+#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
+#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
+#define LOW_LATENCY_CAPTURE_USE_CASE 1
+
+#define DEVICE_NAME_MAX_SIZE 128
+#define HW_INFO_ARRAY_MAX_SIZE 32
+
+#define DEEP_BUFFER_PCM_DEVICE 0
+#define AUDIO_RECORD_PCM_DEVICE 0
+#define MULTIMEDIA2_PCM_DEVICE 1
+#define MULTIMEDIA3_PCM_DEVICE 4
+#define FM_PLAYBACK_PCM_DEVICE 5
+#define FM_CAPTURE_PCM_DEVICE 6
+#define HFP_PCM_RX 5
+#define HFP_SCO_RX 17
+#define HFP_ASM_RX_TX 18
+
+#define INCALL_MUSIC_UPLINK_PCM_DEVICE 1
+#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
+#define SPKR_PROT_CALIB_RX_PCM_DEVICE 5
+#define SPKR_PROT_CALIB_TX_PCM_DEVICE 26
+#define PLAYBACK_OFFLOAD_DEVICE 9
+#define PLAYBACK_OFFLOAD_DEVICE2 24
+
+/* Define macro for Internal FM volume mixer */
+#define FM_RX_VOLUME "Internal FM RX Volume"
+
+#define LOWLATENCY_PCM_DEVICE 12
+#define EC_REF_RX "I2S_RX"
+
+#define VOICE_CALL_PCM_DEVICE 2
+#define VOICE2_CALL_PCM_DEVICE 13
+#define VOLTE_CALL_PCM_DEVICE 15
+#define QCHAT_CALL_PCM_DEVICE 26
+#define QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC 28
+#define VOWLAN_CALL_PCM_DEVICE 16
+
+#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
+#define AFE_PROXY_RECORD_PCM_DEVICE 8
+
+#define LIB_CSD_CLIENT "libcsd-client.so"
+/* CSD-CLIENT related functions */
+typedef int (*init_t)();
+typedef int (*deinit_t)();
+typedef int (*disable_device_t)();
+typedef int (*enable_device_config_t)(int, int);
+typedef int (*enable_device_t)(int, int, uint32_t);
+typedef int (*volume_t)(uint32_t, int, uint16_t);
+typedef int (*mic_mute_t)(uint32_t, int, uint16_t);
+typedef int (*slow_talk_t)(uint32_t, uint8_t);
+typedef int (*start_voice_t)(uint32_t);
+typedef int (*stop_voice_t)(uint32_t);
+typedef int (*start_playback_t)(uint32_t);
+typedef int (*stop_playback_t)(uint32_t);
+typedef int (*start_record_t)(uint32_t, int);
+typedef int (*stop_record_t)(uint32_t);
+/* CSD Client structure */
+struct csd_data {
+ void *csd_client;
+ init_t init;
+ deinit_t deinit;
+ disable_device_t disable_device;
+ enable_device_config_t enable_device_config;
+ enable_device_t enable_device;
+ volume_t volume;
+ mic_mute_t mic_mute;
+ slow_talk_t slow_talk;
+ start_voice_t start_voice;
+ stop_voice_t stop_voice;
+ start_playback_t start_playback;
+ stop_playback_t stop_playback;
+ start_record_t start_record;
+ stop_record_t stop_record;
+};
+
+#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
+#define PLATFORM_INFO_XML_BASE_STRING "/system/etc/audio_platform_info"
+
+#endif // QCOM_AUDIO_PLATFORM_H