ASoC: msm: Add support to have FM playback through ASM loopback mode
- Add support to provide PP on FM playback.
- Add PCM platform driver which enables ASM loopback mode
Define MultiMedia6 front-end dai
Change-Id: Ibf9d2645e706f9a3027dd356223a2135224a0337
Signed-off-by: Damir Didjusto <damird@codeaurora.org>
diff --git a/include/sound/apr_audio.h b/include/sound/apr_audio.h
index 40b0e1e..81636a3 100644
--- a/include/sound/apr_audio.h
+++ b/include/sound/apr_audio.h
@@ -1453,6 +1453,23 @@
u32 read_format;
} __attribute__((packed));
+#define ASM_STREAM_CMD_OPEN_LOOPBACK 0x00010D6E
+struct asm_stream_cmd_open_loopback {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags.
+ * Bit 0-31: reserved; client should set these bits to 0
+ */
+ u16 src_endpointype;
+ /* Endpoint type. 0 = Tx Matrix */
+ u16 sink_endpointype;
+ /* Endpoint type. 0 = Rx Matrix */
+ u32 postprocopo_id;
+/* Postprocessor topology ID. Specifies the topology of
+ * postprocessing algorithms.
+ */
+} __packed;
+
#define ADM_CMD_CONNECT_AFE_PORT 0x00010320
#define ADM_CMD_DISCONNECT_AFE_PORT 0x00010321
@@ -1909,5 +1926,4 @@
int srs_ss3d_open(int port_id, int srs_tech_id, void *srs_params);
/* SRS Studio Sound 3D end */
-
#endif /*_APR_AUDIO_H_*/
diff --git a/include/sound/q6asm.h b/include/sound/q6asm.h
index 406407d..41f875b 100644
--- a/include/sound/q6asm.h
+++ b/include/sound/q6asm.h
@@ -207,6 +207,8 @@
uint32_t rd_format,
uint32_t wr_format);
+int q6asm_open_loopack(struct audio_client *ac);
+
int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t flags);
int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
diff --git a/sound/soc/msm/Makefile b/sound/soc/msm/Makefile
index ebde90b..7ab4811 100644
--- a/sound/soc/msm/Makefile
+++ b/sound/soc/msm/Makefile
@@ -55,8 +55,7 @@
# for MSM 8960 sound card driver
obj-$(CONFIG_SND_SOC_MSM_QDSP6_INTF) += qdsp6/
-
-snd-soc-qdsp6-objs := msm-dai-q6.o msm-pcm-q6.o msm-multi-ch-pcm-q6.o msm-lowlatency-pcm-q6.o msm-pcm-routing.o msm-dai-fe.o msm-compr-q6.o msm-dai-stub.o
+snd-soc-qdsp6-objs := msm-dai-q6.o msm-pcm-q6.o msm-multi-ch-pcm-q6.o msm-lowlatency-pcm-q6.o msm-pcm-loopback.o msm-pcm-routing.o msm-dai-fe.o msm-compr-q6.o msm-dai-stub.o
obj-$(CONFIG_SND_SOC_MSM_QDSP6_HDMI_AUDIO) += msm-dai-q6-hdmi.o
obj-$(CONFIG_SND_SOC_VOICE) += msm-pcm-voice.o msm-pcm-voip.o msm-pcm-dtmf.o msm-pcm-host-voice.o
snd-soc-qdsp6-objs += msm-pcm-lpa.o msm-pcm-afe.o
diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c
index 8db13f6..ffb0f3b 100644
--- a/sound/soc/msm/msm-dai-fe.c
+++ b/sound/soc/msm/msm-dai-fe.c
@@ -236,6 +236,17 @@
.rate_min = 8000,
.rate_max = 192000,
},
+ .capture = {
+ .stream_name = "MultiMedia6 Capture",
+ .aif_name = "MM_UL6",
+ .rates = (SNDRV_PCM_RATE_8000_48000|
+ SNDRV_PCM_RATE_KNOT),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
.ops = &msm_fe_Multimedia_dai_ops,
.name = "MultiMedia6",
},
diff --git a/sound/soc/msm/msm-pcm-loopback.c b/sound/soc/msm/msm-pcm-loopback.c
new file mode 100644
index 0000000..16b6f09
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-loopback.c
@@ -0,0 +1,339 @@
+/* Copyright (c) 2013, The Linux Foundation. All rights reserved.
+
+* This program is free software; you can redistribute it and/or modify
+* it under the terms of the GNU General Public License version 2 and
+* only version 2 as published by the Free Software Foundation.
+
+* This program is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+* GNU General Public License for more details.
+*/
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/apr_audio.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/q6asm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+
+#include "msm-pcm-routing.h"
+
+struct msm_pcm_loopback {
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ int instance;
+
+ struct mutex lock;
+
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+
+ int playback_start;
+ int capture_start;
+ int session_id;
+ struct audio_client *audio_client;
+};
+
+static void stop_pcm(struct msm_pcm_loopback *pcm);
+
+static const struct snd_pcm_hardware dummy_pcm_hardware = {
+ .formats = 0xffffffff,
+ .channels_min = 1,
+ .channels_max = UINT_MAX,
+
+ /* Random values to keep userspace happy when checking constraints */
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .buffer_bytes_max = 128*1024,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 1024*2,
+ .periods_min = 2,
+ .periods_max = 128,
+};
+
+static void msm_pcm_loopback_event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ pr_debug("%s\n", __func__);
+ switch (opcode) {
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ break;
+ default:
+ break;
+ }
+ }
+ break;
+ default:
+ pr_err("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+ struct msm_pcm_loopback *pcm;
+ int ret = 0;
+
+ pcm = dev_get_drvdata(rtd->platform->dev);
+ mutex_lock(&pcm->lock);
+
+ snd_soc_set_runtime_hwparams(substream, &dummy_pcm_hardware);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ pcm->playback_substream = substream;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ pcm->capture_substream = substream;
+
+ pcm->instance++;
+ dev_dbg(rtd->platform->dev, "%s: pcm out open: %d,%d\n", __func__,
+ pcm->instance, substream->stream);
+ if (pcm->instance == 2) {
+ struct snd_soc_pcm_runtime *soc_pcm_rx =
+ pcm->playback_substream->private_data;
+ struct snd_soc_pcm_runtime *soc_pcm_tx =
+ pcm->capture_substream->private_data;
+ if (pcm->audio_client != NULL)
+ stop_pcm(pcm);
+
+ pcm->audio_client = q6asm_audio_client_alloc(
+ (app_cb)msm_pcm_loopback_event_handler, pcm);
+ if (!pcm->audio_client) {
+ dev_err(rtd->platform->dev,
+ "%s: Could not allocate memory\n", __func__);
+ mutex_unlock(&pcm->lock);
+ return -ENOMEM;
+ }
+ pcm->session_id = pcm->audio_client->session;
+ pcm->audio_client->perf_mode = false;
+ ret = q6asm_open_loopack(pcm->audio_client);
+ if (ret < 0) {
+ dev_err(rtd->platform->dev,
+ "%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(pcm->audio_client);
+ mutex_unlock(&pcm->lock);
+ return -ENOMEM;
+ }
+ msm_pcm_routing_reg_phy_stream(soc_pcm_tx->dai_link->be_id,
+ pcm->audio_client->perf_mode,
+ pcm->session_id, pcm->capture_substream->stream);
+ msm_pcm_routing_reg_phy_stream(soc_pcm_rx->dai_link->be_id,
+ pcm->audio_client->perf_mode,
+ pcm->session_id, pcm->playback_substream->stream);
+ }
+ dev_info(rtd->platform->dev, "%s: Instance = %d, Stream ID = %s\n",
+ __func__ , pcm->instance, substream->pcm->id);
+ runtime->private_data = pcm;
+
+ mutex_unlock(&pcm->lock);
+
+ return 0;
+}
+
+static void stop_pcm(struct msm_pcm_loopback *pcm)
+{
+ struct snd_soc_pcm_runtime *soc_pcm_rx =
+ pcm->playback_substream->private_data;
+ struct snd_soc_pcm_runtime *soc_pcm_tx =
+ pcm->capture_substream->private_data;
+
+ if (pcm->audio_client == NULL)
+ return;
+ q6asm_cmd(pcm->audio_client, CMD_CLOSE);
+
+ msm_pcm_routing_dereg_phy_stream(soc_pcm_rx->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ msm_pcm_routing_dereg_phy_stream(soc_pcm_tx->dai_link->be_id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ q6asm_audio_client_free(pcm->audio_client);
+ pcm->audio_client = NULL;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_pcm_loopback *pcm = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+ int ret = 0;
+
+ mutex_lock(&pcm->lock);
+
+ dev_dbg(rtd->platform->dev, "%s: end pcm call:%d\n",
+ __func__, substream->stream);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ pcm->playback_start = 0;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ pcm->capture_start = 0;
+
+ pcm->instance--;
+ if (!pcm->playback_start || !pcm->capture_start) {
+ dev_dbg(rtd->platform->dev, "%s: end pcm call\n", __func__);
+ stop_pcm(pcm);
+ }
+
+ mutex_unlock(&pcm->lock);
+ return ret;
+}
+
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_pcm_loopback *pcm = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+
+ mutex_lock(&pcm->lock);
+
+ dev_dbg(rtd->platform->dev, "%s: ASM loopback stream:%d\n",
+ __func__, substream->stream);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (!pcm->playback_start)
+ pcm->playback_start = 1;
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (!pcm->capture_start)
+ pcm->capture_start = 1;
+ }
+ mutex_unlock(&pcm->lock);
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_pcm_loopback *pcm = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dev_dbg(rtd->platform->dev,
+ "%s: playback_start:%d,capture_start:%d\n", __func__,
+ pcm->playback_start, pcm->capture_start);
+ if (pcm->playback_start && pcm->capture_start)
+ q6asm_run_nowait(pcm->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_STOP:
+ dev_dbg(rtd->platform->dev,
+ "%s:Pause/Stop - playback_start:%d,capture_start:%d\n",
+ __func__, pcm->playback_start, pcm->capture_start);
+ if (pcm->playback_start && pcm->capture_start)
+ q6asm_cmd_nowait(pcm->audio_client, CMD_PAUSE);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+
+ dev_dbg(rtd->platform->dev, "%s: ASM loopback\n", __func__);
+
+ return snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(params));
+}
+
+static int msm_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .hw_params = msm_pcm_hw_params,
+ .hw_free = msm_pcm_hw_free,
+ .close = msm_pcm_close,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+};
+
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ struct msm_pcm_loopback *pcm;
+
+ dev_dbg(&pdev->dev, "%s: dev name %s\n",
+ __func__, dev_name(&pdev->dev));
+
+ pcm = kzalloc(sizeof(struct msm_pcm_loopback), GFP_KERNEL);
+ if (!pcm) {
+ dev_err(&pdev->dev, "%s Failed to allocate memory for pcm\n",
+ __func__);
+ return -ENOMEM;
+ } else {
+ mutex_init(&pcm->lock);
+ dev_set_drvdata(&pdev->dev, pcm);
+ }
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ struct msm_pcm_loopback *pcm;
+
+ pcm = dev_get_drvdata(&pdev->dev);
+ mutex_destroy(&pcm->lock);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-loopback",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM loopback platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm-routing.c b/sound/soc/msm/msm-pcm-routing.c
index e74a0dd..b56067f 100644
--- a/sound/soc/msm/msm-pcm-routing.c
+++ b/sound/soc/msm/msm-pcm-routing.c
@@ -976,10 +976,8 @@
static int msm_routing_set_fm_vol_mixer(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- afe_loopback_gain(INT_FM_TX , ucontrol->value.integer.value[0]);
-
+ afe_loopback_gain(INT_FM_TX, ucontrol->value.integer.value[0]);
msm_route_fm_vol_control = ucontrol->value.integer.value[0];
-
return 0;
}
@@ -1660,6 +1658,9 @@
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),
+ SOC_SINGLE_EXT("MultiMedia6", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = {
@@ -1696,6 +1697,9 @@
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),
+ SOC_SINGLE_EXT("MultiMedia6", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = {
@@ -1714,6 +1718,9 @@
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),
+ SOC_SINGLE_EXT("MultiMedia6", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = {
@@ -1810,6 +1817,12 @@
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new mmul6_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 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,
@@ -2715,6 +2728,7 @@
SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 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),
@@ -2828,6 +2842,8 @@
mmul4_mixer_controls, ARRAY_SIZE(mmul4_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("MultiMedia6 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul6_mixer_controls, ARRAY_SIZE(mmul6_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)),
SND_SOC_DAPM_MIXER("SEC_AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
@@ -3008,6 +3024,7 @@
{"MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
{"MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"MI2S_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
{"MI2S_RX", NULL, "MI2S_RX Audio Mixer"},
{"MultiMedia1 Mixer", "PRI_TX", "PRI_I2S_TX"},
@@ -3026,6 +3043,7 @@
{"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"},
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
{"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX Audio Mixer"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
@@ -3040,12 +3058,14 @@
{"AFE_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
{"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"},
+ {"MultiMedia6 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
{"MultiMedia1 Mixer", "AFE_PCM_TX", "PCM_TX"},
{"MultiMedia5 Mixer", "AFE_PCM_TX", "PCM_TX"},
@@ -3054,12 +3074,14 @@
{"MM_UL2", NULL, "MultiMedia2 Mixer"},
{"MM_UL4", NULL, "MultiMedia4 Mixer"},
{"MM_UL5", NULL, "MultiMedia5 Mixer"},
+ {"MM_UL6", NULL, "MultiMedia6 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 Audio Mixer", "MultiMedia6", "MM_DL6"},
{"AUX_PCM_RX", NULL, "AUX_PCM_RX Audio Mixer"},
{"SEC_AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
diff --git a/sound/soc/msm/qdsp6/q6asm.c b/sound/soc/msm/qdsp6/q6asm.c
index f15f4d1..659d5a2 100644
--- a/sound/soc/msm/qdsp6/q6asm.c
+++ b/sound/soc/msm/qdsp6/q6asm.c
@@ -912,6 +912,7 @@
case ASM_STREAM_CMD_OPEN_WRITE:
case ASM_STREAM_CMD_OPEN_WRITE_V2_1:
case ASM_STREAM_CMD_OPEN_READWRITE:
+ case ASM_STREAM_CMD_OPEN_LOOPBACK:
case ASM_DATA_CMD_MEDIA_FORMAT_UPDATE:
case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
case ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED:
@@ -1852,6 +1853,45 @@
return -EINVAL;
}
+int q6asm_open_loopack(struct audio_client *ac)
+{
+ int rc = 0x00;
+ struct asm_stream_cmd_open_loopback open;
+
+ if ((ac == NULL) || (ac->apr == NULL)) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d]", __func__, ac->session);
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_LOOPBACK;
+
+ open.mode_flags = 0;
+ open.src_endpointype = 0;
+ open.sink_endpointype = 0;
+ /* source endpoint : matrix */
+ open.postprocopo_id = get_asm_topology();
+ if (open.postprocopo_id == 0)
+ open.postprocopo_id = DEFAULT_POPP_TOPOLOGY;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+ if (rc < 0) {
+ pr_err("open failed op[0x%x]rc[%d]\n", \
+ open.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for OPEN_WRITE rc[%d]\n", rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
int q6asm_run(struct audio_client *ac, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts)
{