ASoC: msm: Add incall recording support
Add two backend DAIs, and add two mixer controls to route
capture stream to the new backend DAIs. Start/stop record
commands will be sent to CVD if starting/stoping recording,
after voice call is established.
Change-Id: Iabff5390ca8c50ba05ae2dfa31fa3f1a8797f1db
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
diff --git a/arch/arm/mach-msm/board-8960.c b/arch/arm/mach-msm/board-8960.c
index bd39579..e6ca8e0c 100644
--- a/arch/arm/mach-msm/board-8960.c
+++ b/arch/arm/mach-msm/board-8960.c
@@ -1772,6 +1772,8 @@
&msm_lpa_pcm,
&msm_compr_dsp,
&msm_cpudai_incall_music_rx,
+ &msm_cpudai_incall_record_rx,
+ &msm_cpudai_incall_record_tx,
#if defined(CONFIG_CRYPTO_DEV_QCRYPTO) || \
defined(CONFIG_CRYPTO_DEV_QCRYPTO_MODULE)
@@ -1833,6 +1835,8 @@
&msm_pcm_afe,
&msm_compr_dsp,
&msm_cpudai_incall_music_rx,
+ &msm_cpudai_incall_record_rx,
+ &msm_cpudai_incall_record_tx,
&msm_pcm_hostless,
&msm_bus_apps_fabric,
&msm_bus_sys_fabric,
diff --git a/arch/arm/mach-msm/devices-8960.c b/arch/arm/mach-msm/devices-8960.c
index 0bb159a..2d06d97 100644
--- a/arch/arm/mach-msm/devices-8960.c
+++ b/arch/arm/mach-msm/devices-8960.c
@@ -1425,6 +1425,16 @@
.id = 0x8005,
};
+struct platform_device msm_cpudai_incall_record_rx = {
+ .name = "msm-dai-q6",
+ .id = 0x8004,
+};
+
+struct platform_device msm_cpudai_incall_record_tx = {
+ .name = "msm-dai-q6",
+ .id = 0x8003,
+};
+
/*
* Machine specific data for AUX PCM Interface
* which the driver will be unware of.
diff --git a/arch/arm/mach-msm/devices.h b/arch/arm/mach-msm/devices.h
index 384aced..0d6aa09 100644
--- a/arch/arm/mach-msm/devices.h
+++ b/arch/arm/mach-msm/devices.h
@@ -174,6 +174,8 @@
extern struct platform_device msm_pcm_afe;
extern struct platform_device msm_compr_dsp;
extern struct platform_device msm_cpudai_incall_music_rx;
+extern struct platform_device msm_cpudai_incall_record_rx;
+extern struct platform_device msm_cpudai_incall_record_tx;
extern struct platform_device msm_pil_q6v3;
extern struct platform_device msm_8960_q6_lpass;
diff --git a/sound/soc/msm/msm-dai-q6.c b/sound/soc/msm/msm-dai-q6.c
index 00dc0b7..4af4f06 100644
--- a/sound/soc/msm/msm-dai-q6.c
+++ b/sound/soc/msm/msm-dai-q6.c
@@ -301,6 +301,8 @@
rc = msm_dai_q6_afe_rtproxy_hw_params(params, dai);
break;
case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_RX:
+ case VOICE_RECORD_TX:
rc = 0;
break;
default:
@@ -348,6 +350,8 @@
if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
switch (dai->id) {
case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
pr_debug("%s, stop pseudo port:%d\n",
__func__, dai->id);
rc = afe_stop_pseudo_port(dai->id);
@@ -492,6 +496,8 @@
if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
switch (dai->id) {
case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
afe_pseudo_port_start_nowait(dai->id);
break;
default:
@@ -509,6 +515,8 @@
if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
switch (dai->id) {
case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
afe_pseudo_port_stop_nowait(dai->id);
break;
default:
@@ -614,6 +622,8 @@
if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
switch (dai->id) {
case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
pr_debug("%s, stop pseudo port:%d\n",
__func__, dai->id);
rc = afe_stop_pseudo_port(dai->id);
@@ -784,6 +794,21 @@
.remove = msm_dai_q6_dai_remove,
};
+static struct snd_soc_dai_driver msm_dai_q6_incall_record_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
static struct snd_soc_dai_driver msm_dai_q6_bt_sco_rx_dai = {
.playback = {
.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
@@ -925,6 +950,11 @@
rc = snd_soc_register_dai(&pdev->dev,
&msm_dai_q6_voice_playback_tx_dai);
break;
+ case VOICE_RECORD_RX:
+ case VOICE_RECORD_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_incall_record_dai);
+ break;
default:
rc = -ENODEV;
break;
diff --git a/sound/soc/msm/msm-pcm-routing.c b/sound/soc/msm/msm-pcm-routing.c
index 933e4a8..453bce4 100644
--- a/sound/soc/msm/msm-pcm-routing.c
+++ b/sound/soc/msm/msm-pcm-routing.c
@@ -115,6 +115,8 @@
{ PCM_RX, 0, NULL, 0, 0},
{ PCM_TX, 0, NULL, 0, 0},
{ VOICE_PLAYBACK_TX, 0, NULL, 0, 0},
+ { VOICE_RECORD_RX, 0, NULL, 0, 0},
+ { VOICE_RECORD_TX, 0, NULL, 0, 0},
};
@@ -286,6 +288,9 @@
fe_dai_map[val][session_type], path_type);
}
}
+ if ((msm_bedais[reg].port_id == VOICE_RECORD_RX)
+ || (msm_bedais[reg].port_id == VOICE_RECORD_TX))
+ voc_start_record(msm_bedais[reg].port_id, set);
mutex_unlock(&routing_lock);
}
@@ -780,6 +785,12 @@
SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX,
MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new mmul2_mixer_controls[] = {
@@ -1124,6 +1135,10 @@
/* incall */
SND_SOC_DAPM_AIF_OUT("VOICE_PLAYBACK_TX", "Voice Farend Playback",
0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("INCALL_RECORD_TX", "Voice Uplink Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("INCALL_RECORD_RX", "Voice Downlink Capture",
+ 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("AUX_PCM_RX", "AUX PCM Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("AUX_PCM_TX", "AUX PCM Capture", 0, 0, 0, 0),
/* Switch Definitions */
@@ -1210,6 +1225,10 @@
{"Incall_Music Audio Mixer", "MultiMedia1", "MM_DL1"},
{"Incall_Music Audio Mixer", "MultiMedia2", "MM_DL2"},
{"VOICE_PLAYBACK_TX", NULL, "Incall_Music Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "VOC_REC_UL", "INCALL_RECORD_TX"},
+ {"MultiMedia1 Mixer", "VOC_REC_DL", "INCALL_RECORD_RX"},
+
{"MultiMedia1 Mixer", "PRI_TX", "PRI_I2S_TX"},
{"MultiMedia1 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
{"MultiMedia1 Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"},
diff --git a/sound/soc/msm/msm-pcm-routing.h b/sound/soc/msm/msm-pcm-routing.h
index f45af05..2020939 100644
--- a/sound/soc/msm/msm-pcm-routing.h
+++ b/sound/soc/msm/msm-pcm-routing.h
@@ -27,6 +27,9 @@
#define LPASS_BE_AUXPCM_RX "(Backend) AUX_PCM_RX"
#define LPASS_BE_AUXPCM_TX "(Backend) AUX_PCM_TX"
#define LPASS_BE_VOICE_PLAYBACK_TX "(Backend) VOICE_PLAYBACK_TX"
+#define LPASS_BE_INCALL_RECORD_RX "(Backend) INCALL_RECORD_TX"
+#define LPASS_BE_INCALL_RECORD_TX "(Backend) INCALL_RECORD_RX"
+
/* For multimedia front-ends, asm session is allocated dynamically.
* Hence, asm session/multimedia front-end mapping has to be maintained.
@@ -64,6 +67,8 @@
MSM_BACKEND_DAI_AUXPCM_RX,
MSM_BACKEND_DAI_AUXPCM_TX,
MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ MSM_BACKEND_DAI_INCALL_RECORD_TX,
MSM_BACKEND_DAI_MAX,
};
diff --git a/sound/soc/msm/msm8960.c b/sound/soc/msm/msm8960.c
index 228047e..dd8224c 100644
--- a/sound/soc/msm/msm8960.c
+++ b/sound/soc/msm/msm8960.c
@@ -1074,6 +1074,32 @@
.be_id = MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
.be_hw_params_fixup = msm8960_be_hw_params_fixup,
},
+ /* Incall Record Uplink BACK END DAI Link */
+ {
+ .name = LPASS_BE_INCALL_RECORD_TX,
+ .stream_name = "Voice Uplink Capture",
+ .cpu_dai_name = "msm-dai-q6.32772",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .no_codec = 1,
+ .be_id = MSM_BACKEND_DAI_INCALL_RECORD_TX,
+ .be_hw_params_fixup = msm8960_be_hw_params_fixup,
+ },
+ /* Incall Record Downlink BACK END DAI Link */
+ {
+ .name = LPASS_BE_INCALL_RECORD_RX,
+ .stream_name = "Voice Downlink Capture",
+ .cpu_dai_name = "msm-dai-q6.32771",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .no_codec = 1,
+ .be_id = MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ .be_hw_params_fixup = msm8960_be_hw_params_fixup,
+ },
};
struct snd_soc_card snd_soc_card_msm8960 = {
diff --git a/sound/soc/msm/qdsp6/q6voice.c b/sound/soc/msm/qdsp6/q6voice.c
index 6676d4d..de63fa0 100644
--- a/sound/soc/msm/qdsp6/q6voice.c
+++ b/sound/soc/msm/qdsp6/q6voice.c
@@ -56,6 +56,8 @@
static int voice_cvs_stop_playback(struct voice_data *v);
static int voice_cvs_start_playback(struct voice_data *v);
+static int voice_cvs_start_record(struct voice_data *v, uint32_t rec_mode);
+static int voice_cvs_stop_record(struct voice_data *v);
static int32_t qdsp_mvm_callback(struct apr_client_data *data, void *priv);
static int32_t qdsp_cvs_callback(struct apr_client_data *data, void *priv);
@@ -1921,10 +1923,14 @@
if (is_voip_session(v->session_id))
voice_send_netid_timing_cmd(v);
- /* Start in-call music delivery if it's enabled */
+ /* Start in-call music delivery if this feature is enabled */
if (v->music_info.play_enable)
voice_cvs_start_playback(v);
+ /* Start in-call recording if this feature is enabled */
+ if (v->rec_info.rec_enable)
+ voice_cvs_start_record(v, v->rec_info.rec_mode);
+
rtac_add_voice(voice_get_cvs_handle(v),
voice_get_cvp_handle(v),
v->dev_rx.port_id, v->dev_tx.port_id,
@@ -2156,10 +2162,10 @@
mvm_handle = voice_get_mvm_handle(v);
cvp_handle = voice_get_cvp_handle(v);
- /* stop playback */
+ /* stop playback or recording */
v->music_info.force = 1;
voice_cvs_stop_playback(v);
-
+ voice_cvs_stop_record(v);
/* send stop voice cmd */
voice_send_stop_voice_cmd(v);
@@ -2332,6 +2338,262 @@
return 0;
}
+static int voice_cvs_start_record(struct voice_data *v, uint32_t rec_mode)
+{
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+
+ struct cvs_start_record_cmd cvs_start_record;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ if (!v->rec_info.recording) {
+ cvs_start_record.hdr.hdr_field = APR_HDR_FIELD(
+ APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_start_record.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_start_record) - APR_HDR_SIZE);
+ cvs_start_record.hdr.src_port = v->session_id;
+ cvs_start_record.hdr.dest_port = cvs_handle;
+ cvs_start_record.hdr.token = 0;
+ cvs_start_record.hdr.opcode = VSS_ISTREAM_CMD_START_RECORD;
+
+ if (rec_mode == VOC_REC_UPLINK) {
+ cvs_start_record.rec_mode.rx_tap_point =
+ VSS_TAP_POINT_NONE;
+ cvs_start_record.rec_mode.tx_tap_point =
+ VSS_TAP_POINT_STREAM_END;
+ } else if (rec_mode == VOC_REC_DOWNLINK) {
+ cvs_start_record.rec_mode.rx_tap_point =
+ VSS_TAP_POINT_STREAM_END;
+ cvs_start_record.rec_mode.tx_tap_point =
+ VSS_TAP_POINT_NONE;
+ } else if (rec_mode == VOC_REC_BOTH) {
+ cvs_start_record.rec_mode.rx_tap_point =
+ VSS_TAP_POINT_STREAM_END;
+ cvs_start_record.rec_mode.tx_tap_point =
+ VSS_TAP_POINT_STREAM_END;
+ } else {
+ pr_err("%s: Invalid in-call rec_mode %d\n", __func__,
+ rec_mode);
+
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_start_record);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending START_RECORD\n", __func__,
+ ret);
+
+ goto fail;
+ }
+
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ goto fail;
+ }
+ v->rec_info.recording = 1;
+ } else {
+ pr_debug("%s: Start record already sent\n", __func__);
+ }
+
+ return 0;
+
+fail:
+ return ret;
+}
+
+static int voice_cvs_stop_record(struct voice_data *v)
+{
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+ struct apr_hdr cvs_stop_record;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ if (v->rec_info.recording) {
+ cvs_stop_record.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvs_stop_record.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_stop_record) - APR_HDR_SIZE);
+ cvs_stop_record.src_port = v->session_id;
+ cvs_stop_record.dest_port = cvs_handle;
+ cvs_stop_record.token = 0;
+ cvs_stop_record.opcode = VSS_ISTREAM_CMD_STOP_RECORD;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_stop_record);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending STOP_RECORD\n",
+ __func__, ret);
+
+ goto fail;
+ }
+
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ goto fail;
+ }
+ v->rec_info.recording = 0;
+ } else {
+ pr_debug("%s: Stop record already sent\n", __func__);
+ }
+
+ return 0;
+
+fail:
+
+ return ret;
+}
+
+int voc_start_record(uint32_t port_id, uint32_t set)
+{
+ int ret = 0;
+ int rec_mode = 0;
+ u16 cvs_handle;
+ int i, rec_set = 0;
+
+ for (i = 0; i < MAX_VOC_SESSIONS; i++) {
+ struct voice_data *v = &common.voice[i];
+ pr_debug("%s: i:%d port_id: %d, set: %d\n",
+ __func__, i, port_id, set);
+
+ mutex_lock(&v->lock);
+ rec_mode = v->rec_info.rec_mode;
+ rec_set = set;
+ if (set) {
+ if ((v->rec_route_state.ul_flag != 0) &&
+ (v->rec_route_state.dl_flag != 0)) {
+ pr_debug("%s: i=%d, rec mode already set.\n",
+ __func__, i);
+ mutex_unlock(&v->lock);
+ if (i < MAX_VOC_SESSIONS)
+ continue;
+ else
+ return 0;
+ }
+
+ if (port_id == VOICE_RECORD_TX) {
+ if ((v->rec_route_state.ul_flag == 0)
+ && (v->rec_route_state.dl_flag == 0)) {
+ rec_mode = VOC_REC_UPLINK;
+ v->rec_route_state.ul_flag = 1;
+ } else if ((v->rec_route_state.ul_flag == 0)
+ && (v->rec_route_state.dl_flag != 0)) {
+ voice_cvs_stop_record(v);
+ rec_mode = VOC_REC_BOTH;
+ v->rec_route_state.ul_flag = 1;
+ }
+ } else if (port_id == VOICE_RECORD_RX) {
+ if ((v->rec_route_state.ul_flag == 0)
+ && (v->rec_route_state.dl_flag == 0)) {
+ rec_mode = VOC_REC_DOWNLINK;
+ v->rec_route_state.dl_flag = 1;
+ } else if ((v->rec_route_state.ul_flag != 0)
+ && (v->rec_route_state.dl_flag == 0)) {
+ voice_cvs_stop_record(v);
+ rec_mode = VOC_REC_BOTH;
+ v->rec_route_state.dl_flag = 1;
+ }
+ }
+ rec_set = 1;
+ } else {
+ if ((v->rec_route_state.ul_flag == 0) &&
+ (v->rec_route_state.dl_flag == 0)) {
+ pr_debug("%s: i=%d, rec already stops.\n",
+ __func__, i);
+ mutex_unlock(&v->lock);
+ if (i < MAX_VOC_SESSIONS)
+ continue;
+ else
+ return 0;
+ }
+
+ if (port_id == VOICE_RECORD_TX) {
+ if ((v->rec_route_state.ul_flag != 0)
+ && (v->rec_route_state.dl_flag == 0)) {
+ v->rec_route_state.ul_flag = 0;
+ rec_set = 0;
+ } else if ((v->rec_route_state.ul_flag != 0)
+ && (v->rec_route_state.dl_flag != 0)) {
+ voice_cvs_stop_record(v);
+ v->rec_route_state.ul_flag = 0;
+ rec_mode = VOC_REC_DOWNLINK;
+ rec_set = 1;
+ }
+ } else if (port_id == VOICE_RECORD_RX) {
+ if ((v->rec_route_state.ul_flag == 0)
+ && (v->rec_route_state.dl_flag != 0)) {
+ v->rec_route_state.dl_flag = 0;
+ rec_set = 0;
+ } else if ((v->rec_route_state.ul_flag != 0)
+ && (v->rec_route_state.dl_flag != 0)) {
+ voice_cvs_stop_record(v);
+ v->rec_route_state.dl_flag = 0;
+ rec_mode = VOC_REC_UPLINK;
+ rec_set = 1;
+ }
+ }
+ }
+ pr_debug("%s: i=%d, mode =%d, set =%d\n", __func__,
+ i, rec_mode, rec_set);
+ cvs_handle = voice_get_cvs_handle(v);
+
+ if (cvs_handle != 0) {
+ if (rec_set)
+ ret = voice_cvs_start_record(v, rec_mode);
+ else
+ ret = voice_cvs_stop_record(v);
+ }
+
+ /* Cache the value */
+ v->rec_info.rec_enable = rec_set;
+ v->rec_info.rec_mode = rec_mode;
+
+ mutex_unlock(&v->lock);
+ }
+
+ return ret;
+}
+
static int voice_cvs_start_playback(struct voice_data *v)
{
int ret = 0;
@@ -3110,6 +3372,8 @@
case VSS_ICOMMON_CMD_SET_UI_PROPERTY:
case VSS_ISTREAM_CMD_START_PLAYBACK:
case VSS_ISTREAM_CMD_STOP_PLAYBACK:
+ case VSS_ISTREAM_CMD_START_RECORD:
+ case VSS_ISTREAM_CMD_STOP_RECORD:
pr_debug("%s: cmd = 0x%x\n", __func__, ptr[0]);
v->cvs_state = CMD_STATUS_SUCCESS;
wake_up(&v->cvs_wait);
diff --git a/sound/soc/msm/qdsp6/q6voice.h b/sound/soc/msm/qdsp6/q6voice.h
index 03993f6..d330ada 100644
--- a/sound/soc/msm/qdsp6/q6voice.h
+++ b/sound/soc/msm/qdsp6/q6voice.h
@@ -17,6 +17,10 @@
#define MAX_VOC_PKT_SIZE 642
#define SESSION_NAME_LEN 20
+#define VOC_REC_UPLINK 0x00
+#define VOC_REC_DOWNLINK 0x01
+#define VOC_REC_BOTH 0x02
+
struct voice_header {
uint32_t id;
uint32_t data_len;
@@ -43,6 +47,11 @@
u16 tx_route_flag;
};
+struct voice_rec_route_state {
+ u16 ul_flag;
+ u16 dl_flag;
+};
+
enum {
VOC_INIT = 0,
VOC_RUN,
@@ -324,6 +333,30 @@
#define VSS_ISTREAM_CMD_STOP_PLAYBACK 0x00011239
/* Stop the in-call music delivery on the Tx voice path. */
+#define VSS_ISTREAM_CMD_START_RECORD 0x00011236
+/* Start in-call conversation recording. */
+#define VSS_ISTREAM_CMD_STOP_RECORD 0x00011237
+/* Stop in-call conversation recording. */
+
+#define VSS_TAP_POINT_NONE 0x00010F78
+/* Indicates no tapping for specified path. */
+
+#define VSS_TAP_POINT_STREAM_END 0x00010F79
+/* Indicates that specified path should be tapped at the end of the stream. */
+
+struct vss_istream_cmd_start_record_t {
+ uint32_t rx_tap_point;
+ /* Tap point to use on the Rx path. Supported values are:
+ * VSS_TAP_POINT_NONE : Do not record Rx path.
+ * VSS_TAP_POINT_STREAM_END : Rx tap point is at the end of the stream.
+ */
+ uint32_t tx_tap_point;
+ /* Tap point to use on the Tx path. Supported values are:
+ * VSS_TAP_POINT_NONE : Do not record tx path.
+ * VSS_TAP_POINT_STREAM_END : Tx tap point is at the end of the stream.
+ */
+} __packed;
+
struct vss_istream_cmd_create_passive_control_session_t {
char name[SESSION_NAME_LEN];
/**<
@@ -544,6 +577,10 @@
struct apr_hdr hdr;
struct vss_icommon_cmd_set_ui_property_st_enable_t vss_set_st;
} __packed;
+struct cvs_start_record_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_start_record_t rec_mode;
+} __packed;
/* TO CVP commands */
@@ -753,8 +790,9 @@
};
struct incall_rec_info {
- uint32_t pending;
+ uint32_t rec_enable;
uint32_t rec_mode;
+ uint32_t recording;
};
struct incall_music_info {
@@ -798,7 +836,12 @@
struct voice_dev_route_state voc_route_state;
u16 session_id;
+
+ struct incall_rec_info rec_info;
+
struct incall_music_info music_info;
+
+ struct voice_rec_route_state rec_route_state;
};
#define MAX_VOC_SESSIONS 2
@@ -866,4 +909,5 @@
uint16_t voc_get_session_id(char *name);
int voc_start_playback(uint32_t set);
+int voc_start_record(uint32_t port_id, uint32_t set);
#endif