ASoC: msm: Add low latency playback and capture support
ADSP has support for low latency playback and capture.
Enhance msm alsa drivers for B-family to make use of
low latency mode.
Change-Id: Ie35c4697a5d6e3af12ef6681180b551c080ad182
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
diff --git a/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt b/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt
index 232ddec..fed8cb4 100644
--- a/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt
+++ b/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt
@@ -6,6 +6,21 @@
- compatible : "qcom,msm-pcm-dsp"
+ - qcom,msm-pcm-dsp-id : device node id
+
+* msm-pcm-low-latency
+
+Required properties:
+
+ - compatible : "qcom,msm-pcm-dsp"
+
+ - qcom,msm-pcm-dsp-id : device node id
+
+ Optional properties
+
+ - qcom,msm-pcm-low-latency : Flag indicating whether
+ the device node is of type low latency.
+
* msm-pcm-routing
Required properties:
@@ -212,7 +227,14 @@
Example:
qcom,msm-pcm {
- compatible = "qcom,msm-pcm-dsp";
+ compatible = "qcom,msm-pcm-dsp";
+ qcom,msm-pcm-dsp-id = <0>;
+ };
+
+ qcom,msm-pcm-low-latency {
+ compatible = "qcom,msm-pcm-dsp";
+ qcom,msm-pcm-dsp-id = <1>;
+ qcom,msm-pcm-low-latency;
};
qcom,msm-pcm-routing {
diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
index cad737a..93f0aa90 100644
--- a/include/sound/apr_audio-v2.h
+++ b/include/sound/apr_audio-v2.h
@@ -82,6 +82,14 @@
*/
#define ADM_CMD_DEVICE_OPEN_V5 0x00010326
+#define ADM_BIT_SHIFT_DEVICE_PERF_MODE_FLAG 13
+
+/* Definition for a legacy device session. */
+#define ADM_LEGACY_DEVICE_SESSION 0
+
+/* Definition for a low latency stream session. */
+#define ADM_LOW_LATENCY_DEVICE_SESSION 1
+
/* Indicates that endpoint_id_2 is to be ignored.*/
#define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE 0xFFFF
@@ -3794,8 +3802,15 @@
/* adsp_asm_session_command.h*/
#define ASM_STREAM_CMD_OPEN_WRITE_V2 0x00010D8F
+#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
-struct asm_stream_cmd_open_write_v2 {
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_WRITE 28
+
+#define ASM_LEGACY_STREAM_SESSION 0
+
+#define ASM_LOW_LATENCY_STREAM_SESSION 1
+
+struct asm_stream_cmd_open_write_v3 {
struct apr_hdr hdr;
uint32_t mode_flags;
/* Mode flags that configure the stream to notify the client
@@ -3878,6 +3893,9 @@
} __packed;
#define ASM_STREAM_CMD_OPEN_READ_V2 0x00010D8C
+
+#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
+
/* Definition of the timestamp type flag bitmask */
#define ASM_BIT_MASKIMESTAMPYPE_FLAG (0x00000020UL)
@@ -3890,8 +3908,10 @@
/* Absolute timestamp is identified by this value.*/
#define ASM_ABSOLUTEIMESTAMP 1
+/* Bit shift for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29
-struct asm_stream_cmd_open_read_v2 {
+struct asm_stream_cmd_open_read_v3 {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode flags that indicate whether meta information per encoded
diff --git a/include/sound/q6adm-v2.h b/include/sound/q6adm-v2.h
index 6e5e649..699e63a 100644
--- a/include/sound/q6adm-v2.h
+++ b/include/sound/q6adm-v2.h
@@ -29,10 +29,11 @@
int srs_trumedia_open(int port_id, int srs_tech_id, void *srs_params);
-int adm_open(int port, int path, int rate, int mode, int topology);
+int adm_open(int port, int path, int rate, int mode, int topology,
+ bool perf_mode);
int adm_multi_ch_copp_open(int port, int path, int rate, int mode,
- int topology);
+ int topology, bool perf_mode);
int adm_memory_map_regions(int port_id, uint32_t *buf_add, uint32_t mempool_id,
uint32_t *bufsz, uint32_t bufcnt);
diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h
index 6891b14..553b33b 100644
--- a/include/sound/q6asm-v2.h
+++ b/include/sound/q6asm-v2.h
@@ -151,6 +151,7 @@
struct audio_port_data port[2];
wait_queue_head_t cmd_wait;
wait_queue_head_t time_wait;
+ bool perf_mode;
};
void q6asm_audio_client_free(struct audio_client *ac);
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
index 6df2fb93..581d8ad 100644
--- a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
@@ -527,6 +527,7 @@
return -ENOMEM;
}
+ prtd->audio_client->perf_mode = false;
pr_info("%s: session ID %d\n", __func__, prtd->audio_client->session);
prtd->session_id = prtd->audio_client->session;
@@ -723,6 +724,7 @@
}
msm_pcm_routing_reg_phy_stream(
soc_prtd->dai_link->be_id,
+ prtd->audio_client->perf_mode,
prtd->session_id,
substream->stream);
@@ -754,6 +756,7 @@
pr_debug("msm_pcm_routing_reg_phy_stream\n");
msm_pcm_routing_reg_phy_stream(
soc_prtd->dai_link->be_id,
+ prtd->audio_client->perf_mode,
prtd->session_id, substream->stream);
break;
default:
diff --git a/sound/soc/msm/qdsp6v2/msm-multi-ch-pcm-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-multi-ch-pcm-q6-v2.c
index 827aaa3..3c9190a 100644
--- a/sound/soc/msm/qdsp6v2/msm-multi-ch-pcm-q6-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-multi-ch-pcm-q6-v2.c
@@ -172,8 +172,7 @@
break;
}
if (prtd->mmap_flag) {
- pr_debug("%s:writing %d bytes"
- " of buffer to dsp\n",
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
__func__,
prtd->pcm_count);
q6asm_write_nolock(prtd->audio_client,
@@ -181,8 +180,7 @@
0, 0, NO_TIMESTAMP);
} else {
while (atomic_read(&prtd->out_needed)) {
- pr_debug("%s:writing %d bytes"
- " of buffer to dsp\n",
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
__func__,
prtd->pcm_count);
q6asm_write_nolock(prtd->audio_client,
@@ -336,6 +334,7 @@
kfree(prtd);
return -ENOMEM;
}
+ prtd->audio_client->perf_mode = false;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
runtime->hw = msm_pcm_hardware_playback;
ret = q6asm_open_write(prtd->audio_client,
@@ -363,6 +362,7 @@
prtd->session_id = prtd->audio_client->session;
msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->audio_client->perf_mode,
prtd->session_id, substream->stream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -412,8 +412,8 @@
rc = q6asm_set_volume(multi_ch_pcm_audio.prtd->audio_client,
volume);
if (rc < 0) {
- pr_err("%s: Send Volume command failed"
- " rc=%d\n", __func__, rc);
+ pr_err("%s: Send Volume command failed rc=%d\n",
+ __func__, rc);
}
}
multi_ch_pcm_audio.volume = volume;
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c
index 4d88246..6491cc5 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c
@@ -293,6 +293,7 @@
kfree(prtd);
return -ENOMEM;
}
+ prtd->audio_client->perf_mode = false;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM);
if (ret < 0) {
@@ -315,6 +316,7 @@
pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
prtd->session_id = prtd->audio_client->session;
msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->audio_client->perf_mode,
prtd->session_id, substream->stream);
ret = snd_pcm_hw_constraint_list(runtime, 0,
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c
index 34b38a6..c5f500a 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c
@@ -48,7 +48,7 @@
#define PLAYBACK_NUM_PERIODS 8
#define PLAYBACK_MAX_PERIOD_SIZE 12288
-#define PLAYBACK_MIN_PERIOD_SIZE 2048
+#define PLAYBACK_MIN_PERIOD_SIZE 1024
#define CAPTURE_NUM_PERIODS 16
#define CAPTURE_MAX_PERIOD_SIZE 4096
#define CAPTURE_MIN_PERIOD_SIZE 512
@@ -311,9 +311,15 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct msm_audio *prtd;
+ struct msm_plat_data *pdata;
int ret = 0;
- pr_debug("%s\n", __func__);
+ pdata = (struct msm_plat_data *)
+ dev_get_drvdata(soc_prtd->platform->dev);
+ if (!pdata) {
+ pr_err("%s: platform data not populated\n", __func__);
+ return -EINVAL;
+ }
prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
if (prtd == NULL) {
pr_err("Failed to allocate memory for msm_audio\n");
@@ -327,6 +333,8 @@
kfree(prtd);
return -ENOMEM;
}
+ prtd->audio_client->perf_mode = pdata->perf_mode;
+ pr_debug("%s: perf: %x\n", __func__, pdata->perf_mode);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
runtime->hw = msm_pcm_hardware_playback;
ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM);
@@ -341,6 +349,7 @@
prtd->audio_client->session);
prtd->session_id = prtd->audio_client->session;
msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->audio_client->perf_mode,
prtd->session_id, substream->stream);
prtd->cmd_ack = 1;
@@ -669,14 +678,15 @@
prtd->audio_client = NULL;
return -ENOMEM;
}
+
+ pr_debug("%s: session ID %d\n",
+ __func__, prtd->audio_client->session);
+ prtd->session_id = prtd->audio_client->session;
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->audio_client->perf_mode,
+ prtd->session_id, substream->stream);
}
- pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
- prtd->session_id = prtd->audio_client->session;
- msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
- prtd->session_id, substream->stream);
-
-
ret = q6asm_audio_client_buf_alloc_contiguous(dir,
prtd->audio_client,
(params_buffer_bytes(params) / params_periods(params)),
@@ -769,16 +779,46 @@
static __devinit int msm_pcm_probe(struct platform_device *pdev)
{
- if (pdev->dev.of_node)
- dev_set_name(&pdev->dev, "%s", "msm-pcm-dsp");
+ int rc;
+ int id;
+ struct msm_plat_data *pdata;
- pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ rc = of_property_read_u32(pdev->dev.of_node,
+ "qcom,msm-pcm-dsp-id", &id);
+ if (rc) {
+ dev_err(&pdev->dev, "%s: qcom,msm-pcm-dsp-id missing in DT node\n",
+ __func__);
+ return rc;
+ }
+
+ pdata = kzalloc(sizeof(struct msm_plat_data), GFP_KERNEL);
+ if (!pdata) {
+ dev_err(&pdev->dev, "Failed to allocate memory for platform data\n");
+ return -ENOMEM;
+ }
+
+ if (of_property_read_bool(pdev->dev.of_node,
+ "qcom,msm-pcm-low-latency"))
+ pdata->perf_mode = 1;
+ else
+ pdata->perf_mode = 0;
+
+ dev_set_drvdata(&pdev->dev, pdata);
+
+ dev_set_name(&pdev->dev, "%s.%d", "msm-pcm-dsp", id);
+
+ dev_dbg(&pdev->dev, "%s: dev name %s\n",
+ __func__, dev_name(&pdev->dev));
return snd_soc_register_platform(&pdev->dev,
&msm_soc_platform);
}
static int msm_pcm_remove(struct platform_device *pdev)
{
+ struct msm_plat_data *pdata;
+
+ pdata = dev_get_drvdata(&pdev->dev);
+ kfree(pdata);
snd_soc_unregister_platform(&pdev->dev);
return 0;
}
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h
index 01ed41f..4b3cfe7 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h
@@ -95,4 +95,8 @@
uint32_t reserved[12];
};
+struct msm_plat_data {
+ int perf_mode;
+};
+
#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c
index 8a5abc9..f168dc1 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c
@@ -40,6 +40,7 @@
unsigned long port_sessions; /* track Tx BE ports -> Rx BE */
unsigned int sample_rate;
unsigned int channel;
+ bool perf_mode;
};
#define INVALID_SESSION -1
@@ -71,6 +72,9 @@
static const DECLARE_TLV_DB_LINEAR(compressed_rx_vol_gain, 0,
INT_RX_VOL_MAX_STEPS);
+static int msm_route_multimedia5_vol_control;
+static const DECLARE_TLV_DB_LINEAR(multimedia5_rx_vol_gain, 0,
+ INT_RX_VOL_MAX_STEPS);
/* Equal to Frontend after last of the MULTIMEDIA SESSIONS */
@@ -208,6 +212,8 @@
{INVALID_SESSION, INVALID_SESSION},
/* MULTIMEDIA4 */
{INVALID_SESSION, INVALID_SESSION},
+ /* MULTIMEDIA5 */
+ {INVALID_SESSION, INVALID_SESSION},
};
static uint8_t is_be_dai_extproc(int be_dai)
@@ -283,7 +289,8 @@
mutex_unlock(&routing_lock);
}
-void msm_pcm_routing_reg_phy_stream(int fedai_id, int dspst_id, int stream_type)
+void msm_pcm_routing_reg_phy_stream(int fedai_id, bool perf_mode,
+ int dspst_id, int stream_type)
{
int i, session_type, path_type, port_type;
struct route_payload payload;
@@ -313,6 +320,8 @@
if (eq_data[fedai_id].enable)
msm_send_eq_values(fedai_id);
for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if (test_bit(fedai_id, &msm_bedais[i].fe_sessions))
+ msm_bedais[i].perf_mode = perf_mode;
if (!is_be_dai_extproc(i) &&
(afe_get_port_type(msm_bedais[i].port_id) == port_type) &&
(msm_bedais[i].active) &&
@@ -321,18 +330,28 @@
channels = msm_bedais[i].channel;
if ((stream_type == SNDRV_PCM_STREAM_PLAYBACK) &&
+ ((channels == 1) || (channels == 2)) &&
+ msm_bedais[i].perf_mode) {
+ pr_debug("%s configure COPP to lowlatency mode",
+ __func__);
+ adm_multi_ch_copp_open(msm_bedais[i].port_id,
+ path_type,
+ msm_bedais[i].sample_rate,
+ msm_bedais[i].channel,
+ DEFAULT_COPP_TOPOLOGY, msm_bedais[i].perf_mode);
+ } else if ((stream_type == SNDRV_PCM_STREAM_PLAYBACK) &&
(channels > 2))
adm_multi_ch_copp_open(msm_bedais[i].port_id,
path_type,
msm_bedais[i].sample_rate,
msm_bedais[i].channel,
- DEFAULT_COPP_TOPOLOGY);
+ DEFAULT_COPP_TOPOLOGY, msm_bedais[i].perf_mode);
else
adm_open(msm_bedais[i].port_id,
path_type,
msm_bedais[i].sample_rate,
msm_bedais[i].channel,
- DEFAULT_COPP_TOPOLOGY);
+ DEFAULT_COPP_TOPOLOGY, false);
payload.copp_ids[payload.num_copps++] =
msm_bedais[i].port_id;
@@ -431,17 +450,30 @@
channels = msm_bedais[reg].channel;
- if ((session_type == SESSION_TYPE_RX) && (channels > 2))
+ if ((session_type == SESSION_TYPE_RX) &&
+ ((channels == 1) || (channels == 2))
+ && msm_bedais[reg].perf_mode) {
adm_multi_ch_copp_open(msm_bedais[reg].port_id,
path_type,
msm_bedais[reg].sample_rate,
channels,
- DEFAULT_COPP_TOPOLOGY);
+ DEFAULT_COPP_TOPOLOGY,
+ msm_bedais[reg].perf_mode);
+ pr_debug("%s:configure COPP to lowlatency mode",
+ __func__);
+ } else if ((session_type == SESSION_TYPE_RX)
+ && (channels > 2))
+ adm_multi_ch_copp_open(msm_bedais[reg].port_id,
+ path_type,
+ msm_bedais[reg].sample_rate,
+ channels,
+ DEFAULT_COPP_TOPOLOGY,
+ msm_bedais[reg].perf_mode);
else
adm_open(msm_bedais[reg].port_id,
path_type,
msm_bedais[reg].sample_rate, channels,
- DEFAULT_COPP_TOPOLOGY);
+ DEFAULT_COPP_TOPOLOGY, false);
msm_pcm_routing_build_matrix(val,
fe_dai_map[val][session_type], path_type);
@@ -787,6 +819,25 @@
return 0;
}
+static int msm_routing_get_multimedia5_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ ucontrol->value.integer.value[0] = msm_route_multimedia5_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_multimedia5_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ if (!multi_ch_pcm_set_volume(ucontrol->value.integer.value[0]))
+ msm_route_multimedia5_vol_control =
+ ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
static int msm_routing_set_multimedia2_vol_mixer(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1082,6 +1133,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_PRI_I2S_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = {
@@ -1097,6 +1151,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_SEC_I2S_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
@@ -1112,6 +1169,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_SLIMBUS_0_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = {
@@ -1127,6 +1187,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_MI2S_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = {
@@ -1187,6 +1250,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_HDMI_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
/* incall music delivery mixer */
static const struct snd_kcontrol_new incall_music_delivery_mixer_controls[] = {
@@ -1220,6 +1286,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_INT_BT_SCO_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = {
@@ -1235,6 +1304,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_INT_FM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = {
@@ -1250,6 +1322,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_AFE_PCM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = {
@@ -1265,6 +1340,9 @@
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_AUXPCM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new mmul1_mixer_controls[] = {
@@ -1318,6 +1396,27 @@
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new mmul5_mixer_controls[] = {
+ SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_TX", MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
static const struct snd_kcontrol_new pri_rx_voice_mixer_controls[] = {
SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_PRI_I2S_RX,
MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
@@ -1658,6 +1757,12 @@
msm_routing_set_multimedia2_vol_mixer, multimedia2_rx_vol_gain),
};
+static const struct snd_kcontrol_new multimedia5_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("HIFI3 RX Volume", SND_SOC_NOPM, 0,
+ INT_RX_VOL_GAIN, 0, msm_routing_get_multimedia5_vol_mixer,
+ msm_routing_set_multimedia5_vol_mixer, multimedia5_rx_vol_gain),
+};
+
static const struct snd_kcontrol_new compressed_vol_mixer_controls[] = {
SOC_SINGLE_EXT_TLV("COMPRESSED RX Volume", SND_SOC_NOPM, 0,
INT_RX_VOL_GAIN, 0, msm_routing_get_compressed_vol_mixer,
@@ -1913,9 +2018,11 @@
SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("VOIP_DL", "VoIP Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("CS-VOICE_DL1", "CS-VOICE Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("CS-VOICE_UL1", "CS-VOICE Capture", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("VoLTE_DL", "VoLTE Playback", 0, 0, 0, 0),
@@ -2048,6 +2155,8 @@
mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)),
SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0,
mmul2_mixer_controls, ARRAY_SIZE(mmul2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia5 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul5_mixer_controls, ARRAY_SIZE(mmul5_mixer_controls)),
SND_SOC_DAPM_MIXER("AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
auxpcm_rx_mixer_controls, ARRAY_SIZE(auxpcm_rx_mixer_controls)),
/* incall */
@@ -2152,24 +2261,28 @@
{"PRI_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"PRI_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"PRI_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"PRI_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
{"PRI_I2S_RX", NULL, "PRI_RX Audio Mixer"},
{"SEC_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"SEC_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"SEC_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"SEC_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SEC_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
{"SEC_I2S_RX", NULL, "SEC_RX Audio Mixer"},
{"SLIMBUS_0_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"SLIMBUS_0_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"SLIMBUS_0_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"SLIMBUS_0_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
{"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Audio Mixer"},
{"HDMI Mixer", "MultiMedia1", "MM_DL1"},
{"HDMI Mixer", "MultiMedia2", "MM_DL2"},
{"HDMI Mixer", "MultiMedia3", "MM_DL3"},
{"HDMI Mixer", "MultiMedia4", "MM_DL4"},
+ {"HDMI Mixer", "MultiMedia5", "MM_DL5"},
{"HDMI", NULL, "HDMI Mixer"},
/* incall */
@@ -2183,10 +2296,12 @@
{"MultiMedia1 Mixer", "VOC_REC_UL", "INCALL_RECORD_TX"},
{"MultiMedia1 Mixer", "VOC_REC_DL", "INCALL_RECORD_RX"},
{"MultiMedia1 Mixer", "SLIM_4_TX", "SLIMBUS_4_TX"},
+ {"MultiMedia5 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
{"MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
{"MI2S_RX", NULL, "MI2S_RX Audio Mixer"},
{"QUAT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
@@ -2214,6 +2329,7 @@
{"MultiMedia1 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
{"MultiMedia1 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
{"MultiMedia1 Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"},
+ {"MultiMedia5 Mixer", "AUX_PCM_TX", "AUX_PCM_TX"},
{"MultiMedia2 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
{"MultiMedia1 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
@@ -2221,32 +2337,40 @@
{"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
{"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX Audio Mixer"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
{"INT_FM_RX", NULL, "INTERNAL_FM_RX Audio Mixer"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
{"PCM_RX", NULL, "AFE_PCM_RX Audio Mixer"},
{"MultiMedia1 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+ {"MultiMedia5 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
{"MultiMedia1 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+ {"MultiMedia5 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
{"MultiMedia1 Mixer", "AFE_PCM_TX", "PCM_TX"},
+ {"MultiMedia5 Mixer", "AFE_PCM_TX", "PCM_TX"},
{"MM_UL1", NULL, "MultiMedia1 Mixer"},
{"MultiMedia2 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
{"MM_UL2", NULL, "MultiMedia2 Mixer"},
+ {"MM_UL5", NULL, "MultiMedia5 Mixer"},
{"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"AUX_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"AUX_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
{"AUX_PCM_RX", NULL, "AUX_PCM_RX Audio Mixer"},
{"MI2S_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
@@ -2483,6 +2607,7 @@
bedai->active = 0;
bedai->sample_rate = 0;
bedai->channel = 0;
+ bedai->perf_mode = false;
mutex_unlock(&routing_lock);
return 0;
@@ -2528,18 +2653,30 @@
channels = bedai->channel;
if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ && ((channels == 2) || (channels == 1)) &&
+ bedai->perf_mode) {
+ adm_multi_ch_copp_open(bedai->port_id,
+ path_type,
+ bedai->sample_rate,
+ channels,
+ DEFAULT_COPP_TOPOLOGY, bedai->perf_mode);
+ pr_debug("%s:configure COPP to lowlatency mode",
+ __func__);
+ } else if ((substream->stream ==
+ SNDRV_PCM_STREAM_PLAYBACK ||
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE)
&& (channels > 2))
adm_multi_ch_copp_open(bedai->port_id,
path_type,
bedai->sample_rate,
channels,
- DEFAULT_COPP_TOPOLOGY);
+ DEFAULT_COPP_TOPOLOGY, bedai->perf_mode);
else
adm_open(bedai->port_id,
path_type,
bedai->sample_rate,
channels,
- DEFAULT_COPP_TOPOLOGY);
+ DEFAULT_COPP_TOPOLOGY, false);
msm_pcm_routing_build_matrix(i,
fe_dai_map[i][session_type], path_type);
@@ -2610,6 +2747,10 @@
ARRAY_SIZE(multimedia2_vol_mixer_controls));
snd_soc_add_platform_controls(platform,
+ multimedia5_vol_mixer_controls,
+ ARRAY_SIZE(multimedia5_vol_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
compressed_vol_mixer_controls,
ARRAY_SIZE(compressed_vol_mixer_controls));
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h
index cf24f9a..7ecdff3 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h
@@ -63,6 +63,7 @@
MSM_FRONTEND_DAI_MULTIMEDIA2,
MSM_FRONTEND_DAI_MULTIMEDIA3,
MSM_FRONTEND_DAI_MULTIMEDIA4,
+ MSM_FRONTEND_DAI_MULTIMEDIA5,
MSM_FRONTEND_DAI_CS_VOICE,
MSM_FRONTEND_DAI_VOIP,
MSM_FRONTEND_DAI_AFE_RX,
@@ -73,8 +74,8 @@
MSM_FRONTEND_DAI_MAX,
};
-#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA4 + 1)
-#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA4
+#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA5 + 1)
+#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA5
enum {
MSM_BACKEND_DAI_PRI_I2S_RX = 0,
@@ -121,7 +122,7 @@
* dspst_id: DSP audio stream ID
* stream_type: playback or capture
*/
-void msm_pcm_routing_reg_phy_stream(int fedai_id, int dspst_id,
+void msm_pcm_routing_reg_phy_stream(int fedai_id, bool perf_mode, int dspst_id,
int stream_type);
void msm_pcm_routing_reg_psthr_stream(int fedai_id, int dspst_id,
int stream_type);
diff --git a/sound/soc/msm/qdsp6v2/q6adm.c b/sound/soc/msm/qdsp6v2/q6adm.c
index 685deef..e9c2735 100644
--- a/sound/soc/msm/qdsp6v2/q6adm.c
+++ b/sound/soc/msm/qdsp6v2/q6adm.c
@@ -740,7 +740,8 @@
return ret;
}
-int adm_open(int port_id, int path, int rate, int channel_mode, int topology)
+int adm_open(int port_id, int path, int rate, int channel_mode, int topology,
+ bool perf_mode)
{
struct adm_cmd_device_open_v5 open;
int ret = 0;
@@ -787,10 +788,16 @@
open.hdr.dest_port = tmp_port;
open.hdr.token = port_id;
open.hdr.opcode = ADM_CMD_DEVICE_OPEN_V5;
+ open.flags = 0x00;
+ if (perf_mode) {
+ open.flags |= ADM_LOW_LATENCY_DEVICE_SESSION <<
+ ADM_BIT_SHIFT_DEVICE_PERF_MODE_FLAG;
+ } else {
+ open.flags |= ADM_LEGACY_DEVICE_SESSION <<
+ ADM_BIT_SHIFT_DEVICE_PERF_MODE_FLAG;
+ }
open.mode_of_operation = path;
- /* Reserved for future use, need to set this to 0 */
- open.flags = 0x00;
open.endpoint_id_1 = tmp_port;
open.endpoint_id_2 = 0xFFFF;
@@ -879,8 +886,8 @@
atomic_read(&this_adm.copp_stat[index]),
msecs_to_jiffies(TIMEOUT_MS));
if (!ret) {
- pr_err("%s ADM open failed for port %#x"
- "for [%d]\n", __func__, tmp_port, port_id);
+ pr_err("%s ADM open failed for port %#x for [%d]\n",
+ __func__, tmp_port, port_id);
ret = -EINVAL;
goto fail_cmd;
}
@@ -895,11 +902,11 @@
int adm_multi_ch_copp_open(int port_id, int path, int rate, int channel_mode,
- int topology)
+ int topology, bool perf_mode)
{
int ret = 0;
- ret = adm_open(port_id, path, rate, channel_mode, topology);
+ ret = adm_open(port_id, path, rate, channel_mode, topology, perf_mode);
return ret;
}
diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c
index 0c1bb209..14202a8 100644
--- a/sound/soc/msm/qdsp6v2/q6asm.c
+++ b/sound/soc/msm/qdsp6v2/q6asm.c
@@ -341,6 +341,7 @@
session[ac->session] = 0;
mutex_unlock(&session_lock);
ac->session = 0;
+ ac->perf_mode = 0;
return;
}
@@ -622,6 +623,7 @@
ac->cb = cb;
ac->priv = priv;
ac->io_mode = SYNC_IO_MODE;
+ ac->perf_mode = false;
ac->apr = apr_register("ADSP", "ASM", \
(apr_fn)q6asm_callback,\
((ac->session) << 8 | 0x0001),\
@@ -1087,8 +1089,8 @@
__func__, token, ac->session);
return -EINVAL;
}
- case ASM_STREAM_CMD_OPEN_READ_V2:
- case ASM_STREAM_CMD_OPEN_WRITE_V2:
+ case ASM_STREAM_CMD_OPEN_READ_V3:
+ case ASM_STREAM_CMD_OPEN_WRITE_V3:
case ASM_STREAM_CMD_OPEN_READWRITE_V2:
case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
@@ -1433,7 +1435,7 @@
uint32_t format)
{
int rc = 0x00;
- struct asm_stream_cmd_open_read_v2 open;
+ struct asm_stream_cmd_open_read_v3 open;
uint16_t bits_per_sample = 16;
@@ -1447,7 +1449,7 @@
pr_debug("%s:session[%d]", __func__, ac->session);
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
- open.hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V2;
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
/* Stream prio : High, provide meta info with encoded frames */
open.src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
@@ -1455,30 +1457,39 @@
if (open.preprocopo_id == 0)
open.preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT;
open.bits_per_sample = bits_per_sample;
+ open.mode_flags = 0x0;
+
+ if (ac->perf_mode) {
+ open.mode_flags |= ASM_LOW_LATENCY_STREAM_SESSION <<
+ ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
+ } else {
+ open.mode_flags |= ASM_LEGACY_STREAM_SESSION <<
+ ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
+ }
switch (format) {
case FORMAT_LINEAR_PCM:
- open.mode_flags = 0x00;
+ open.mode_flags |= 0x00;
open.enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
case FORMAT_MPEG4_AAC:
- open.mode_flags = BUFFER_META_ENABLE;
+ open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_AAC_V2;
break;
case FORMAT_V13K:
- open.mode_flags = BUFFER_META_ENABLE;
+ open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_V13K_FS;
break;
case FORMAT_EVRC:
- open.mode_flags = BUFFER_META_ENABLE;
+ open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_EVRC_FS;
break;
case FORMAT_AMRNB:
- open.mode_flags = BUFFER_META_ENABLE ;
+ open.mode_flags |= BUFFER_META_ENABLE ;
open.enc_cfg_id = ASM_MEDIA_FMT_AMRNB_FS;
break;
case FORMAT_AMRWB:
- open.mode_flags = BUFFER_META_ENABLE ;
+ open.mode_flags |= BUFFER_META_ENABLE ;
open.enc_cfg_id = ASM_MEDIA_FMT_AMRWB_FS;
break;
default:
@@ -1505,7 +1516,7 @@
int q6asm_open_write(struct audio_client *ac, uint32_t format)
{
int rc = 0x00;
- struct asm_stream_cmd_open_write_v2 open;
+ struct asm_stream_cmd_open_write_v3 open;
if ((ac == NULL) || (ac->apr == NULL)) {
pr_err("%s: APR handle NULL\n", __func__);
@@ -1516,8 +1527,15 @@
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
- open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V2;
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
open.mode_flags = 0x00;
+ if (ac->perf_mode)
+ open.mode_flags |= (ASM_LOW_LATENCY_STREAM_SESSION <<
+ ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_WRITE);
+ else
+ open.mode_flags |= (ASM_LEGACY_STREAM_SESSION <<
+ ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_WRITE);
+
/* source endpoint : matrix */
open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
open.bits_per_sample = 16;