ASoC: msm: HDMI PCM 6 channel support
HDMI 1.3 supports Multi channel PCM up to 8 channels with sample
size of 16/20/24 bits and sample rate of 32, 44.1, 48, 96, 176.4,
192K. This patch add supports for 6 channel PCM at 48K sample rate
with sample size of 16 bits.
Change-Id: Id09f1f9d7ef2e2444c8c1b661bfc5b3b4c1e66a6
Signed-off-by: Kiran Kandi <kkandi@codeaurora.org>
diff --git a/arch/arm/mach-msm/board-8960.c b/arch/arm/mach-msm/board-8960.c
index 49539ef..76a27ff 100644
--- a/arch/arm/mach-msm/board-8960.c
+++ b/arch/arm/mach-msm/board-8960.c
@@ -1958,6 +1958,7 @@
&msm_bus_sys_fpb,
&msm_bus_cpss_fpb,
&msm_pcm,
+ &msm_multi_ch_pcm,
&msm_pcm_routing,
&msm_cpudai0,
&msm_cpudai1,
@@ -2009,6 +2010,7 @@
&msm_device_hsusb_host,
&android_usb_device,
&msm_pcm,
+ &msm_multi_ch_pcm,
&msm_pcm_routing,
&msm_cpudai0,
&msm_cpudai1,
diff --git a/arch/arm/mach-msm/devices-8960.c b/arch/arm/mach-msm/devices-8960.c
index 872d9d4..cefa0c4 100644
--- a/arch/arm/mach-msm/devices-8960.c
+++ b/arch/arm/mach-msm/devices-8960.c
@@ -1389,6 +1389,11 @@
.id = -1,
};
+struct platform_device msm_multi_ch_pcm = {
+ .name = "msm-multi-ch-pcm-dsp",
+ .id = -1,
+};
+
struct platform_device msm_pcm_routing = {
.name = "msm-pcm-routing",
.id = -1,
@@ -1405,7 +1410,7 @@
};
struct platform_device msm_cpudai_hdmi_rx = {
- .name = "msm-dai-q6",
+ .name = "msm-dai-q6-hdmi",
.id = 8,
};
diff --git a/arch/arm/mach-msm/devices.h b/arch/arm/mach-msm/devices.h
index e3c875b..7037617 100644
--- a/arch/arm/mach-msm/devices.h
+++ b/arch/arm/mach-msm/devices.h
@@ -163,6 +163,7 @@
extern struct platform_device msm_device_vidc_720p;
extern struct platform_device msm_pcm;
+extern struct platform_device msm_multi_ch_pcm;
extern struct platform_device msm_pcm_routing;
extern struct platform_device msm_cpudai0;
extern struct platform_device msm_cpudai1;
diff --git a/include/sound/apr_audio.h b/include/sound/apr_audio.h
index 30f1a7c..87bafed 100644
--- a/include/sound/apr_audio.h
+++ b/include/sound/apr_audio.h
@@ -234,10 +234,18 @@
/* HDMI_5Point1 (6-ch) = 2 */
/* HDMI_6Point1 (8-ch) = 3 */
u16 data_type; /* HDMI_Linear = 0 */
- /* HDMI_non_Linaer = 1 */
+ /* HDMI_non_Linear = 1 */
} __attribute__ ((packed));
+struct afe_port_hdmi_multi_ch_cfg {
+ u16 data_type; /* HDMI_Linear = 0 */
+ /* HDMI_non_Linear = 1 */
+ u16 channel_allocation; /* The default is 0 (Stereo) */
+ u16 reserved; /* must be set to 0 */
+} __packed;
+
+
/* Slimbus Device Ids */
#define AFE_SLIMBUS_DEVICE_1 0x0
#define AFE_SLIMBUS_DEVICE_2 0x1
@@ -276,14 +284,16 @@
int num_ch; /* 1 to 8 */
} __packed;
-#define AFE_PORT_AUDIO_IF_CONFIG 0x000100d3
+#define AFE_PORT_AUDIO_IF_CONFIG 0x000100d3
+#define AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG 0x000100D9
union afe_port_config {
- struct afe_port_pcm_cfg pcm;
- struct afe_port_mi2s_cfg mi2s;
- struct afe_port_hdmi_cfg hdmi;
- struct afe_port_slimbus_cfg slimbus;
- struct afe_port_rtproxy_cfg rtproxy;
+ struct afe_port_pcm_cfg pcm;
+ struct afe_port_mi2s_cfg mi2s;
+ struct afe_port_hdmi_cfg hdmi;
+ struct afe_port_hdmi_multi_ch_cfg hdmi_multi_ch;
+ struct afe_port_slimbus_cfg slimbus;
+ struct afe_port_rtproxy_cfg rtproxy;
} __attribute__((packed));
struct afe_audioif_config_command {
@@ -482,6 +492,20 @@
#define ADM_CMD_COPP_CLOSE 0x00010305
+#define ADM_CMD_MULTI_CHANNEL_COPP_OPEN 0x00010310
+struct adm_multi_ch_copp_open_command {
+ struct apr_hdr hdr;
+ u16 flags;
+ u16 mode; /* 1-RX, 2-Live TX, 3-Non Live TX */
+ u16 endpoint_id1;
+ u16 endpoint_id2;
+ u32 topology_id;
+ u16 channel_config;
+ u16 reserved;
+ u32 rate;
+ u8 dev_channel_mapping[8];
+} __packed;
+
#define ADM_CMD_MEMORY_MAP 0x00010C30
struct adm_cmd_memory_map{
struct apr_hdr hdr;
@@ -635,6 +659,9 @@
u16 reserved;
} __attribute__ ((packed));
+#define ADM_CMDRSP_MULTI_CHANNEL_COPP_OPEN 0x00010311
+
+
#define ASM_STREAM_PRIORITY_NORMAL 0
#define ASM_STREAM_PRIORITY_LOW 1
#define ASM_STREAM_PRIORITY_HIGH 2
@@ -676,6 +703,125 @@
u16 interleaved;
};
+#define PCM_CHANNEL_NULL 0
+
+/* Front left channel. */
+#define PCM_CHANNEL_FL 1
+
+/* Front right channel. */
+#define PCM_CHANNEL_FR 2
+
+/* Front center channel. */
+#define PCM_CHANNEL_FC 3
+
+/* Left surround channel.*/
+#define PCM_CHANNEL_LS 4
+
+/* Right surround channel.*/
+#define PCM_CHANNEL_RS 5
+
+/* Low frequency effect channel. */
+#define PCM_CHANNEL_LFE 6
+
+/* Center surround channel; Rear center channel. */
+#define PCM_CHANNEL_CS 7
+
+/* Left back channel; Rear left channel. */
+#define PCM_CHANNEL_LB 8
+
+/* Right back channel; Rear right channel. */
+#define PCM_CHANNEL_RB 9
+
+/* Top surround channel. */
+#define PCM_CHANNEL_TS 10
+
+/* Center vertical height channel.*/
+#define PCM_CHANNEL_CVH 11
+
+/* Mono surround channel.*/
+#define PCM_CHANNEL_MS 12
+
+/* Front left of center. */
+#define PCM_CHANNEL_FLC 13
+
+/* Front right of center. */
+#define PCM_CHANNEL_FRC 14
+
+/* Rear left of center. */
+#define PCM_CHANNEL_RLC 15
+
+/* Rear right of center. */
+#define PCM_CHANNEL_RRC 16
+
+#define PCM_FORMAT_MAX_NUM_CHANNEL 8
+
+
+/*
+ * Multiple-channel PCM decoder format block structure used in the
+ * #ASM_STREAM_CMD_OPEN_WRITE command.
+ * The data must be in little-endian format.
+ */
+struct asm_multi_channel_pcm_fmt_blk {
+
+ u16 num_channels; /*
+ * Number of channels.
+ * Supported values:1 to 8
+ */
+
+ u16 bits_per_sample; /*
+ * Number of bits per sample per channel.
+ * Supported values: 16, 24 When used for
+ * playback, the client must send 24-bit
+ * samples packed in 32-bit words. The
+ * 24-bit samples must be placed in the most
+ * significant 24 bits of the 32-bit word. When
+ * used for recording, the aDSP sends 24-bit
+ * samples packed in 32-bit words. The 24-bit
+ * samples are placed in the most significant
+ * 24 bits of the 32-bit word.
+ */
+
+ u32 sample_rate; /*
+ * Number of samples per second
+ * (in Hertz). Supported values:
+ * 2000 to 48000
+ */
+
+ u16 is_signed; /*
+ * Flag that indicates the samples
+ * are signed (1).
+ */
+
+ u16 is_interleaved; /*
+ * Flag that indicates whether the channels are
+ * de-interleaved (0) or interleaved (1).
+ * Interleaved format means corresponding
+ * samples from the left and right channels are
+ * interleaved within the buffer.
+ * De-interleaved format means samples from
+ * each channel are contiguous in the buffer.
+ * The samples from one channel immediately
+ * follow those of the previous channel.
+ */
+
+ u8 channel_mapping[8]; /*
+ * Supported values:
+ * PCM_CHANNEL_NULL, PCM_CHANNEL_FL,
+ * PCM_CHANNEL_FR, PCM_CHANNEL_FC,
+ * PCM_CHANNEL_LS, PCM_CHANNEL_RS,
+ * PCM_CHANNEL_LFE, PCM_CHANNEL_CS,
+ * PCM_CHANNEL_LB, PCM_CHANNEL_RB,
+ * PCM_CHANNEL_TS, PCM_CHANNEL_CVH,
+ * PCM_CHANNEL_MS, PCM_CHANNEL_FLC,
+ * PCM_CHANNEL_FRC, PCM_CHANNEL_RLC,
+ * PCM_CHANNEL_RRC.
+ * Channel[i] mapping describes channel I. Each
+ * element i of the array describes channel I
+ * inside the buffer where I < num_channels.
+ * An unused channel is set to zero.
+ */
+};
+
struct asm_adpcm_cfg {
u16 ch_cfg;
u16 bits_per_sample;
@@ -878,6 +1024,7 @@
#define MPEG4_MULTI_AAC 0x00010D86
#define US_POINT_EPOS_FORMAT 0x00012310
#define US_RAW_FORMAT 0x0001127C
+#define MULTI_CHANNEL_PCM 0x00010C66
#define ASM_ENCDEC_SBCRATE 0x00010C13
#define ASM_ENCDEC_IMMDIATE_DECODE 0x00010C14
@@ -1059,6 +1206,7 @@
struct asm_aac_cfg aac_cfg;
struct asm_flac_cfg flac_cfg;
struct asm_vorbis_cfg vorbis_cfg;
+ struct asm_multi_channel_pcm_fmt_blk multi_ch_pcm_cfg;
} __attribute__((packed)) write_cfg;
} __attribute__((packed));
diff --git a/include/sound/q6adm.h b/include/sound/q6adm.h
index 80374c5..fe25d22 100644
--- a/include/sound/q6adm.h
+++ b/include/sound/q6adm.h
@@ -1,4 +1,4 @@
-/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+/* Copyright (c) 2010-2012, Code Aurora Forum. 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
@@ -26,6 +26,9 @@
int adm_open(int port, int path, int rate, int mode, int topology);
+int adm_multi_ch_copp_open(int port, int path, int rate, int mode,
+ int topology);
+
int adm_memory_map_regions(uint32_t *buf_add, uint32_t mempool_id,
uint32_t *bufsz, uint32_t bufcnt);
diff --git a/include/sound/q6asm.h b/include/sound/q6asm.h
index 16439e8..d08f528 100644
--- a/include/sound/q6asm.h
+++ b/include/sound/q6asm.h
@@ -1,4 +1,4 @@
-/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+/* Copyright (c) 2010-2012, Code Aurora Forum. 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
@@ -39,6 +39,7 @@
#define FORMAT_WMA_V9 0x000f
#define FORMAT_AMR_WB_PLUS 0x0010
#define FORMAT_MPEG4_MULTI_AAC 0x0011
+#define FORMAT_MULTI_CHANNEL_LINEAR_PCM 0x0012
#define ENCDEC_SBCBITRATE 0x0001
#define ENCDEC_IMMEDIATE_DECODE 0x0002
@@ -244,6 +245,9 @@
int q6asm_media_format_block_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels);
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels);
+
int q6asm_media_format_block_aac(struct audio_client *ac,
struct asm_aac_cfg *cfg);
diff --git a/sound/soc/msm/Kconfig b/sound/soc/msm/Kconfig
index 9e0549b..1ed5f74 100644
--- a/sound/soc/msm/Kconfig
+++ b/sound/soc/msm/Kconfig
@@ -80,6 +80,13 @@
config SND_VOIP_PCM
tristate
+config SND_SOC_MSM_QDSP6_HDMI_AUDIO
+ tristate "Soc QDSP6 HDMI Audio DAI driver"
+ depends on FB_MSM_HDMI_MSM_PANEL
+ default n
+ help
+ To support HDMI Audio on MSM8960 over QDSP6.
+
config MSM_8x60_VOIP
tristate "SoC Machine driver for voip"
depends on SND_SOC_MSM8X60
@@ -120,6 +127,7 @@
select SND_SOC_MSM_STUB
select SND_SOC_WCD9310
select SND_SOC_MSM_HOSTLESS_PCM
+ select SND_SOC_MSM_QDSP6_HDMI_AUDIO
default n
help
To add support for SoC audio on MSM8960 and APQ8064 boards
diff --git a/sound/soc/msm/Makefile b/sound/soc/msm/Makefile
index c583ce2..1b3014e 100644
--- a/sound/soc/msm/Makefile
+++ b/sound/soc/msm/Makefile
@@ -56,7 +56,8 @@
obj-$(CONFIG_SND_SOC_MSM_QDSP6_INTF) += qdsp6/
-snd-soc-qdsp6-objs := msm-dai-q6.o msm-pcm-q6.o msm-pcm-routing.o msm-dai-fe.o msm-compr-q6.o
+snd-soc-qdsp6-objs := msm-dai-q6.o msm-pcm-q6.o msm-multi-ch-pcm-q6.o msm-pcm-routing.o msm-dai-fe.o msm-compr-q6.o
+obj-$(CONFIG_SND_SOC_MSM_QDSP6_HDMI_AUDIO) += msm-dai-q6-hdmi.o
obj-$(CONFIG_SND_SOC_VOICE) += msm-pcm-voice.o msm-pcm-voip.o
snd-soc-qdsp6-objs += msm-pcm-lpa.o msm-pcm-afe.o
obj-$(CONFIG_SND_SOC_QDSP6) += snd-soc-qdsp6.o
diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c
index 42e7935..8f71e83 100644
--- a/sound/soc/msm/msm-dai-fe.c
+++ b/sound/soc/msm/msm-dai-fe.c
@@ -75,7 +75,7 @@
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 6,
.rate_min = 8000,
.rate_max = 48000,
},
diff --git a/sound/soc/msm/msm-dai-q6-hdmi.c b/sound/soc/msm/msm-dai-q6-hdmi.c
new file mode 100644
index 0000000..6907ded
--- /dev/null
+++ b/sound/soc/msm/msm-dai-q6-hdmi.c
@@ -0,0 +1,283 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/wcd9310/core.h>
+#include <linux/bitops.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/apr_audio.h>
+#include <sound/q6afe.h>
+#include <sound/q6adm.h>
+#include <sound/msm-dai-q6.h>
+#include <mach/clk.h>
+#include <mach/msm_hdmi_audio.h>
+
+
+enum {
+ STATUS_PORT_STARTED, /* track if AFE port has started */
+ STATUS_MAX
+};
+
+struct msm_dai_q6_hdmi_dai_data {
+ DECLARE_BITMAP(status_mask, STATUS_MAX);
+ u32 rate;
+ u32 channels;
+ union afe_port_config port_config;
+};
+
+
+/* Current implementation assumes hw_param is called once
+ * This may not be the case but what to do when ADM and AFE
+ * port are already opened and parameter changes
+ */
+static int msm_dai_q6_hdmi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ u32 channel_allocation = 0;
+ u32 level_shift = 0; /* 0dB */
+ bool down_mix = FALSE;
+
+ dai_data->channels = params_channels(params);
+ dai_data->rate = params_rate(params);
+ dai_data->port_config.hdmi_multi_ch.data_type = 0;
+ dai_data->port_config.hdmi_multi_ch.reserved = 0;
+
+ switch (dai_data->channels) {
+ case 2:
+ channel_allocation = 0;
+ hdmi_msm_audio_info_setup(1, MSM_HDMI_AUDIO_CHANNEL_2,
+ channel_allocation, level_shift, down_mix);
+ dai_data->port_config.hdmi_multi_ch.channel_allocation =
+ channel_allocation;
+ break;
+ case 6:
+ channel_allocation = 0x0B;
+ hdmi_msm_audio_info_setup(1, MSM_HDMI_AUDIO_CHANNEL_6,
+ channel_allocation, level_shift, down_mix);
+ dai_data->port_config.hdmi_multi_ch.channel_allocation =
+ channel_allocation;
+ break;
+ default:
+ dev_err(dai->dev, "invalid Channels = %u\n",
+ dai_data->channels);
+ return -EINVAL;
+ }
+ dev_dbg(dai->dev, "%s() num_ch = %u rate =%u"
+ " channel_allocation = %u\n", __func__, dai_data->channels,
+ dai_data->rate,
+ dai_data->port_config.hdmi_multi_ch.channel_allocation);
+
+ return 0;
+}
+
+
+static void msm_dai_q6_hdmi_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ pr_info("%s: afe port not started. dai_data->status_mask"
+ " = %ld\n", __func__, *dai_data->status_mask);
+ return;
+ }
+
+ rc = afe_close(dai->id); /* can block */
+
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+
+ pr_debug("%s: dai_data->status_mask = %ld\n", __func__,
+ *dai_data->status_mask);
+
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+}
+
+
+static int msm_dai_q6_hdmi_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ /* PORT START should be set if prepare called in active state */
+ rc = afe_q6_interface_prepare();
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to open AFE APR\n");
+ }
+ return rc;
+}
+
+static int msm_dai_q6_hdmi_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ /* Start/stop port without waiting for Q6 AFE response. Need to have
+ * native q6 AFE driver propagates AFE response in order to handle
+ * port start/stop command error properly if error does arise.
+ */
+ pr_debug("%s:port:%d cmd:%d dai_data->status_mask = %ld",
+ __func__, dai->id, cmd, *dai_data->status_mask);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ afe_port_start_nowait(dai->id, &dai_data->port_config,
+ dai_data->rate);
+
+ set_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ afe_port_stop_nowait(dai->id);
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+ break;
+
+ default:
+ dev_err(dai->dev, "invalid Trigger command = %d\n", cmd);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int msm_dai_q6_hdmi_dai_probe(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data;
+ int rc = 0;
+
+ dai_data = kzalloc(sizeof(struct msm_dai_q6_hdmi_dai_data),
+ GFP_KERNEL);
+
+ if (!dai_data) {
+ dev_err(dai->dev, "DAI-%d: fail to allocate dai data\n",
+ dai->id);
+ rc = -ENOMEM;
+ } else
+ dev_set_drvdata(dai->dev, dai_data);
+
+ return rc;
+}
+
+static int msm_dai_q6_hdmi_dai_remove(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data;
+ int rc;
+
+ dai_data = dev_get_drvdata(dai->dev);
+
+ /* If AFE port is still up, close it */
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ rc = afe_close(dai->id); /* can block */
+
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+ kfree(dai_data);
+ snd_soc_unregister_dai(dai->dev);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops msm_dai_q6_hdmi_ops = {
+ .prepare = msm_dai_q6_hdmi_prepare,
+ .trigger = msm_dai_q6_hdmi_trigger,
+ .hw_params = msm_dai_q6_hdmi_hw_params,
+ .shutdown = msm_dai_q6_hdmi_shutdown,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_hdmi_hdmi_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 2,
+ .channels_max = 6,
+ .rate_max = 48000,
+ .rate_min = 48000,
+ },
+ .ops = &msm_dai_q6_hdmi_ops,
+ .probe = msm_dai_q6_hdmi_dai_probe,
+ .remove = msm_dai_q6_hdmi_dai_remove,
+};
+
+
+/* To do: change to register DAIs as batch */
+static __devinit int msm_dai_q6_hdmi_dev_probe(struct platform_device *pdev)
+{
+ int rc = 0;
+
+ dev_dbg(&pdev->dev, "dev name %s dev-id %d\n",
+ dev_name(&pdev->dev), pdev->id);
+
+ switch (pdev->id) {
+ case HDMI_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_hdmi_hdmi_rx_dai);
+ break;
+ default:
+ dev_err(&pdev->dev, "invalid device ID %d\n", pdev->id);
+ rc = -ENODEV;
+ break;
+ }
+ return rc;
+}
+
+static __devexit int msm_dai_q6_hdmi_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_dai_q6_hdmi_driver = {
+ .probe = msm_dai_q6_hdmi_dev_probe,
+ .remove = msm_dai_q6_hdmi_dev_remove,
+ .driver = {
+ .name = "msm-dai-q6-hdmi",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_dai_q6_hdmi_init(void)
+{
+ return platform_driver_register(&msm_dai_q6_hdmi_driver);
+}
+module_init(msm_dai_q6_hdmi_init);
+
+static void __exit msm_dai_q6_hdmi_exit(void)
+{
+ platform_driver_unregister(&msm_dai_q6_hdmi_driver);
+}
+module_exit(msm_dai_q6_hdmi_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM DSP HDMI DAI driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-dai-q6.c b/sound/soc/msm/msm-dai-q6.c
index c7d7004..0951795 100644
--- a/sound/soc/msm/msm-dai-q6.c
+++ b/sound/soc/msm/msm-dai-q6.c
@@ -206,30 +206,6 @@
return 0;
}
-static int msm_dai_q6_hdmi_hw_params(struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
-{
- struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
-
- dev_dbg(dai->dev, "%s start HDMI port\n", __func__);
-
- dai_data->channels = params_channels(params);
- switch (dai_data->channels) {
- case 2:
- dai_data->port_config.hdmi.channel_mode = 0; /* Put in macro */
- break;
- default:
- return -EINVAL;
- break;
- }
-
- /* Q6 only supports 16 as now */
- dai_data->port_config.hdmi.bitwidth = 16;
- dai_data->port_config.hdmi.data_type = 0;
- dai_data->rate = params_rate(params);
-
- return 0;
-}
static int msm_dai_q6_slim_bus_hw_params(struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai, int stream)
@@ -425,9 +401,6 @@
case MI2S_RX:
rc = msm_dai_q6_mi2s_hw_params(params, dai, substream->stream);
break;
- case HDMI_RX:
- rc = msm_dai_q6_hdmi_hw_params(params, dai);
- break;
case SLIMBUS_0_RX:
case SLIMBUS_0_TX:
rc = msm_dai_q6_slim_bus_hw_params(params, dai,
@@ -903,20 +876,6 @@
.remove = msm_dai_q6_dai_remove,
};
-static struct snd_soc_dai_driver msm_dai_q6_hdmi_rx_dai = {
- .playback = {
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- .channels_min = 2,
- .channels_max = 2,
- .rate_max = 48000,
- .rate_min = 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_voice_playback_tx_dai = {
.playback = {
.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
@@ -1101,9 +1060,6 @@
rc = snd_soc_register_dai(&pdev->dev,
&msm_dai_q6_mi2s_rx_dai);
break;
- case HDMI_RX:
- rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_hdmi_rx_dai);
- break;
case SLIMBUS_0_RX:
rc = snd_soc_register_dai(&pdev->dev,
&msm_dai_q6_slimbus_rx_dai);
diff --git a/sound/soc/msm/msm-multi-ch-pcm-q6.c b/sound/soc/msm/msm-multi-ch-pcm-q6.c
new file mode 100644
index 0000000..1dac5d2
--- /dev/null
+++ b/sound/soc/msm/msm-multi-ch-pcm-q6.c
@@ -0,0 +1,723 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. 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/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+
+#include "msm-pcm-q6.h"
+#include "msm-pcm-routing.h"
+
+static struct audio_locks the_locks;
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+#define PLAYBACK_NUM_PERIODS 8
+#define PLAYBACK_PERIOD_SIZE 4032
+#define CAPTURE_NUM_PERIODS 16
+#define CAPTURE_PERIOD_SIZE 320
+
+static struct snd_pcm_hardware msm_pcm_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = CAPTURE_NUM_PERIODS * CAPTURE_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_PERIOD_SIZE,
+ .periods_min = CAPTURE_NUM_PERIODS,
+ .periods_max = CAPTURE_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_pcm_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 6,
+ .buffer_bytes_max = PLAYBACK_NUM_PERIODS * PLAYBACK_PERIOD_SIZE,
+ .period_bytes_min = PLAYBACK_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_PERIOD_SIZE,
+ .periods_min = PLAYBACK_NUM_PERIODS,
+ .periods_max = PLAYBACK_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static uint32_t in_frame_info[CAPTURE_NUM_PERIODS][2];
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ uint32_t *ptrmem = (uint32_t *)payload;
+ int i = 0;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ pr_debug("%s\n", __func__);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE: {
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start))
+ break;
+ if (!prtd->mmap_flag)
+ break;
+ if (q6asm_is_cpu_buf_avail_nolock(IN,
+ prtd->audio_client,
+ &size, &idx)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ }
+ break;
+ }
+ case ASM_DATA_CMDRSP_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case ASM_DATA_EVENT_READ_DONE: {
+ pr_debug("ASM_DATA_EVENT_READ_DONE\n");
+ pr_debug("token = 0x%08x\n", token);
+ for (i = 0; i < 8; i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ in_frame_info[token][0] = payload[2];
+ in_frame_info[token][1] = payload[3];
+ prtd->pcm_irq_pos += in_frame_info[token][0];
+ pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ if (atomic_read(&prtd->in_count) <= prtd->periods)
+ atomic_inc(&prtd->in_count);
+ wake_up(&the_locks.read_wait);
+ if (prtd->mmap_flag
+ && q6asm_is_cpu_buf_avail_nolock(OUT,
+ prtd->audio_client,
+ &size, &idx))
+ q6asm_read_nolock(prtd->audio_client);
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN:
+ if (substream->stream
+ != SNDRV_PCM_STREAM_PLAYBACK) {
+ atomic_set(&prtd->start, 1);
+ break;
+ }
+ if (prtd->mmap_flag) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ } else {
+ while (atomic_read(&prtd->out_needed)) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ atomic_dec(&prtd->out_needed);
+ wake_up(&the_locks.write_wait);
+ };
+ }
+ atomic_set(&prtd->start, 1);
+ break;
+ default:
+ break;
+ }
+ }
+ break;
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ if (prtd->enabled)
+ return 0;
+
+ ret = q6asm_media_format_block_multi_ch_pcm(prtd->audio_client,
+ runtime->rate, runtime->channels);
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret = 0;
+ int i = 0;
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+
+ if (prtd->enabled)
+ return 0;
+
+ pr_debug("Samp_rate = %d\n", prtd->samp_rate);
+ pr_debug("Channel = %d\n", prtd->channel_mode);
+ ret = q6asm_enc_cfg_blk_pcm(prtd->audio_client, prtd->samp_rate,
+ prtd->channel_mode);
+ if (ret < 0)
+ pr_debug("%s: cmd cfg pcm was block failed", __func__);
+
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client);
+ prtd->periods = runtime->periods;
+
+ prtd->enabled = 1;
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_err("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw = msm_pcm_hardware_playback;
+ ret = q6asm_open_write(prtd->audio_client,
+ FORMAT_MULTI_CHANNEL_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw = msm_pcm_hardware_capture;
+ ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm in open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ 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->session_id, substream->stream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->cmd_ack = 1;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ runtime->private_data = prtd;
+
+ return 0;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer = 0;
+ char *bufptr = NULL;
+ void *data = NULL;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ pr_debug("%s: prtd->out_count = %d\n",
+ __func__, atomic_read(&prtd->out_count));
+ ret = wait_event_timeout(the_locks.write_wait,
+ (atomic_read(&prtd->out_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_err("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+
+ if (!atomic_read(&prtd->out_count)) {
+ pr_err("%s: pcm stopped out_count 0\n", __func__);
+ return 0;
+ }
+
+ data = q6asm_is_cpu_buf_avail(IN, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ if (bufptr) {
+ pr_debug("%s:fbytes =%d: xfer=%d size=%d\n",
+ __func__, fbytes, xfer, size);
+ xfer = fbytes;
+ if (copy_from_user(bufptr, buf, xfer)) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ buf += xfer;
+ fbytes -= xfer;
+ pr_debug("%s:fbytes = %d: xfer=%d\n", __func__, fbytes, xfer);
+ if (atomic_read(&prtd->start)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__, xfer);
+ ret = q6asm_write(prtd->audio_client, xfer,
+ 0, 0, NO_TIMESTAMP);
+ if (ret < 0) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ atomic_inc(&prtd->out_needed);
+ atomic_dec(&prtd->out_count);
+ }
+fail:
+ return ret;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = 0;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ ret = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (ret < 0)
+ pr_err("%s: CMD_EOS failed\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer;
+ char *bufptr;
+ void *data = NULL;
+ static uint32_t idx;
+ static uint32_t size;
+ uint32_t offset = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+
+ pr_debug("%s\n", __func__);
+ fbytes = frames_to_bytes(runtime, frames);
+
+ pr_debug("appl_ptr %d\n", (int)runtime->control->appl_ptr);
+ pr_debug("hw_ptr %d\n", (int)runtime->status->hw_ptr);
+ pr_debug("avail_min %d\n", (int)runtime->control->avail_min);
+
+ ret = wait_event_timeout(the_locks.read_wait,
+ (atomic_read(&prtd->in_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_debug("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+ if (!atomic_read(&prtd->in_count)) {
+ pr_debug("%s: pcm stopped in_count 0\n", __func__);
+ return 0;
+ }
+ pr_debug("Checking if valid buffer is available...%08x\n",
+ (unsigned int) data);
+ data = q6asm_is_cpu_buf_avail(OUT, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ pr_debug("Size = %d\n", size);
+ pr_debug("fbytes = %d\n", fbytes);
+ pr_debug("idx = %d\n", idx);
+ if (bufptr) {
+ xfer = fbytes;
+ if (xfer > size)
+ xfer = size;
+ offset = in_frame_info[idx][1];
+ pr_debug("Offset value = %d\n", offset);
+ if (copy_to_user(buf, bufptr+offset, xfer)) {
+ pr_err("Failed to copy buf to user\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ fbytes -= xfer;
+ size -= xfer;
+ in_frame_info[idx][1] += xfer;
+ pr_debug("%s:fbytes = %d: size=%d: xfer=%d\n",
+ __func__, fbytes, size, xfer);
+ pr_debug(" Sending next buffer to dsp\n");
+ memset(&in_frame_info[idx], 0,
+ sizeof(uint32_t) * 2);
+ atomic_dec(&prtd->in_count);
+ ret = q6asm_read(prtd->audio_client);
+ if (ret < 0) {
+ pr_err("q6asm read failed\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ pr_err("No valid buffer\n");
+
+ pr_debug("Returning from capture_copy... %d\n", ret);
+fail:
+ return ret;
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = OUT;
+
+ pr_debug("%s\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+
+ return result;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ dir = OUT;
+
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+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)
+{
+ pr_info("%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)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-multi-ch-pcm-dsp",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ 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("Multi channel PCM module 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 2b4999f..8b6b5f1 100644
--- a/sound/soc/msm/msm-pcm-routing.c
+++ b/sound/soc/msm/msm-pcm-routing.c
@@ -162,6 +162,7 @@
{
int i, session_type, path_type, port_type;
struct route_payload payload;
+ u32 channels;
if (fedai_id > MSM_FRONTEND_DAI_MM_MAX_ID) {
/* bad ID assigned in machine driver */
@@ -191,11 +192,23 @@
port_type) && msm_bedais[i].active &&
(test_bit(fedai_id,
&msm_bedais[i].fe_sessions))) {
- adm_open(msm_bedais[i].port_id,
+
+ channels = params_channels(msm_bedais[i].hw_params);
+
+ if ((stream_type == SNDRV_PCM_STREAM_PLAYBACK) &&
+ (channels > 2))
+ adm_multi_ch_copp_open(msm_bedais[i].port_id,
+ path_type,
+ params_rate(msm_bedais[i].hw_params),
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(msm_bedais[i].port_id,
path_type,
params_rate(msm_bedais[i].hw_params),
params_channels(msm_bedais[i].hw_params),
DEFAULT_COPP_TOPOLOGY);
+
payload.copp_ids[payload.num_copps++] =
msm_bedais[i].port_id;
}
@@ -243,6 +256,7 @@
static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
{
int session_type, path_type;
+ u32 channels;
pr_debug("%s: reg %x val %x set %x\n", __func__, reg, val, set);
@@ -271,10 +285,22 @@
set_bit(val, &msm_bedais[reg].fe_sessions);
if (msm_bedais[reg].active && fe_dai_map[val][session_type] !=
INVALID_SESSION) {
- adm_open(msm_bedais[reg].port_id, path_type,
+
+ channels = params_channels(msm_bedais[reg].hw_params);
+
+ if ((session_type == SESSION_TYPE_RX) && (channels > 2))
+ adm_multi_ch_copp_open(msm_bedais[reg].port_id,
+ path_type,
+ params_rate(msm_bedais[reg].hw_params),
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(msm_bedais[reg].port_id,
+ path_type,
params_rate(msm_bedais[reg].hw_params),
params_channels(msm_bedais[reg].hw_params),
DEFAULT_COPP_TOPOLOGY);
+
msm_pcm_routing_build_matrix(val,
fe_dai_map[val][session_type], path_type);
}
@@ -1463,6 +1489,7 @@
unsigned int be_id = rtd->dai_link->be_id;
int i, path_type, session_type;
struct msm_pcm_routing_bdai_data *bedai;
+ u32 channels;
if (be_id >= MSM_BACKEND_DAI_MAX) {
pr_err("%s: unexpected be_id %d\n", __func__, be_id);
@@ -1500,10 +1527,22 @@
for_each_set_bit(i, &bedai->fe_sessions, MSM_FRONTEND_DAI_MM_SIZE) {
if (fe_dai_map[i][session_type] != INVALID_SESSION) {
- adm_open(bedai->port_id, path_type,
+
+ channels = params_channels(bedai->hw_params);
+ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) &&
+ (channels > 2))
+ adm_multi_ch_copp_open(bedai->port_id,
+ path_type,
+ params_rate(bedai->hw_params),
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(bedai->port_id,
+ path_type,
params_rate(bedai->hw_params),
params_channels(bedai->hw_params),
DEFAULT_COPP_TOPOLOGY);
+
msm_pcm_routing_build_matrix(i,
fe_dai_map[i][session_type], path_type);
}
diff --git a/sound/soc/msm/msm8960.c b/sound/soc/msm/msm8960.c
index 578f819..1ed73e2 100644
--- a/sound/soc/msm/msm8960.c
+++ b/sound/soc/msm/msm8960.c
@@ -787,8 +787,10 @@
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
+ pr_debug("%s channels->min %u channels->max %u ()\n", __func__,
+ channels->min, channels->max);
+
rate->min = rate->max = 48000;
- channels->min = channels->max = 2;
return 0;
}
@@ -936,7 +938,7 @@
.name = "MSM8960 Media2",
.stream_name = "MultiMedia2",
.cpu_dai_name = "MultiMedia2",
- .platform_name = "msm-pcm-dsp",
+ .platform_name = "msm-multi-ch-pcm-dsp",
.dynamic = 1,
.dsp_link = &fe_media,
.be_id = MSM_FRONTEND_DAI_MULTIMEDIA2,
@@ -1118,7 +1120,7 @@
{
.name = LPASS_BE_HDMI,
.stream_name = "HDMI Playback",
- .cpu_dai_name = "msm-dai-q6.8",
+ .cpu_dai_name = "msm-dai-q6-hdmi.8",
.platform_name = "msm-pcm-routing",
.codec_name = "msm-stub-codec.1",
.codec_dai_name = "msm-stub-rx",
diff --git a/sound/soc/msm/qdsp6/q6adm.c b/sound/soc/msm/qdsp6/q6adm.c
index 177e1d8..2710fbb 100644
--- a/sound/soc/msm/qdsp6/q6adm.c
+++ b/sound/soc/msm/qdsp6/q6adm.c
@@ -103,7 +103,8 @@
}
switch (data->opcode) {
- case ADM_CMDRSP_COPP_OPEN: {
+ case ADM_CMDRSP_COPP_OPEN:
+ case ADM_CMDRSP_MULTI_CHANNEL_COPP_OPEN: {
struct adm_copp_open_respond *open = data->payload;
if (open->copp_id == INVALID_COPP_ID) {
pr_err("%s: invalid coppid rxed %d\n",
@@ -360,6 +361,133 @@
return ret;
}
+
+int adm_multi_ch_copp_open(int port_id, int path, int rate, int channel_mode,
+ int topology)
+{
+ struct adm_multi_ch_copp_open_command open;
+ int ret = 0;
+ int index;
+
+ pr_debug("%s: port %d path:%d rate:%d channel :%d\n", __func__,
+ port_id, path, rate, channel_mode);
+
+ port_id = afe_convert_virtual_to_portid(port_id);
+
+ if (afe_validate_port(port_id) < 0) {
+ pr_err("%s port idi[%d] is invalid\n", __func__, port_id);
+ return -ENODEV;
+ }
+
+ index = afe_get_port_index(port_id);
+ pr_debug("%s: Port ID %d, index %d\n", __func__, port_id, index);
+
+ if (this_adm.apr == NULL) {
+ this_adm.apr = apr_register("ADSP", "ADM", adm_callback,
+ 0xFFFFFFFF, &this_adm);
+ if (this_adm.apr == NULL) {
+ pr_err("%s: Unable to register ADM\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ rtac_set_adm_handle(this_adm.apr);
+ }
+
+ /* Create a COPP if port id are not enabled */
+ if (atomic_read(&this_adm.copp_cnt[index]) == 0) {
+
+ open.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+
+ open.hdr.pkt_size =
+ sizeof(struct adm_multi_ch_copp_open_command);
+ open.hdr.opcode = ADM_CMD_MULTI_CHANNEL_COPP_OPEN;
+ memset(open.dev_channel_mapping, 0, 8);
+
+ if (channel_mode == 1) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FC;
+ } else if (channel_mode == 2) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FL;
+ open.dev_channel_mapping[1] = PCM_CHANNEL_FR;
+ } else if (channel_mode == 6) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FL;
+ open.dev_channel_mapping[1] = PCM_CHANNEL_FR;
+ open.dev_channel_mapping[2] = PCM_CHANNEL_LFE;
+ open.dev_channel_mapping[3] = PCM_CHANNEL_FC;
+ open.dev_channel_mapping[4] = PCM_CHANNEL_LB;
+ open.dev_channel_mapping[5] = PCM_CHANNEL_RB;
+ } else {
+ pr_err("%s invalid num_chan %d\n", __func__,
+ channel_mode);
+ return -EINVAL;
+ }
+
+
+ open.hdr.src_svc = APR_SVC_ADM;
+ open.hdr.src_domain = APR_DOMAIN_APPS;
+ open.hdr.src_port = port_id;
+ open.hdr.dest_svc = APR_SVC_ADM;
+ open.hdr.dest_domain = APR_DOMAIN_ADSP;
+ open.hdr.dest_port = port_id;
+ open.hdr.token = port_id;
+
+ open.mode = path;
+ open.endpoint_id1 = port_id;
+ open.endpoint_id2 = 0xFFFF;
+
+ /* convert path to acdb path */
+ if (path == ADM_PATH_PLAYBACK)
+ open.topology_id = get_adm_rx_topology();
+ else {
+ open.topology_id = get_adm_tx_topology();
+ if ((open.topology_id ==
+ VPM_TX_SM_ECNS_COPP_TOPOLOGY) ||
+ (open.topology_id ==
+ VPM_TX_DM_FLUENCE_COPP_TOPOLOGY) ||
+ (open.topology_id ==
+ VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY))
+ rate = 16000;
+ }
+
+ if (open.topology_id == 0)
+ open.topology_id = topology;
+
+ open.channel_config = channel_mode & 0x00FF;
+ open.rate = rate;
+
+ pr_debug("%s: channel_config=%d port_id=%d rate=%d"
+ " topology_id=0x%X\n", __func__, open.channel_config,
+ open.endpoint_id1, open.rate,
+ open.topology_id);
+
+ atomic_set(&this_adm.copp_stat[index], 0);
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)&open);
+ if (ret < 0) {
+ pr_err("%s:ADM enable for port %d failed\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ /* Wait for the callback with copp id */
+ ret = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s ADM open failed for port %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ }
+ atomic_inc(&this_adm.copp_cnt[index]);
+ return 0;
+
+fail_cmd:
+
+ return ret;
+}
+
int adm_matrix_map(int session_id, int path, int num_copps,
unsigned int *port_id, int copp_id)
{
diff --git a/sound/soc/msm/qdsp6/q6afe.c b/sound/soc/msm/qdsp6/q6afe.c
index 302ef57..ef01fb3 100644
--- a/sound/soc/msm/qdsp6/q6afe.c
+++ b/sound/soc/msm/qdsp6/q6afe.c
@@ -77,6 +77,7 @@
if (data->opcode == APR_BASIC_RSP_RESULT) {
switch (payload[0]) {
case AFE_PORT_AUDIO_IF_CONFIG:
+ case AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG:
case AFE_PORT_CMD_STOP:
case AFE_PORT_CMD_START:
case AFE_PORT_CMD_LOOPBACK:
@@ -280,7 +281,7 @@
ret_size = SIZEOF_CFG_CMD(afe_port_mi2s_cfg);
break;
case HDMI_RX:
- ret_size = SIZEOF_CFG_CMD(afe_port_hdmi_cfg);
+ ret_size = SIZEOF_CFG_CMD(afe_port_hdmi_multi_ch_cfg);
break;
case SLIMBUS_0_RX:
case SLIMBUS_0_TX:
@@ -400,13 +401,25 @@
ret = -ENODEV;
return ret;
}
- config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+
+ if (port_id == HDMI_RX) {
+ config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
- config.hdr.pkt_size = afe_sizeof_cfg_cmd(port_id);
- config.hdr.src_port = 0;
- config.hdr.dest_port = 0;
- config.hdr.token = 0;
- config.hdr.opcode = AFE_PORT_AUDIO_IF_CONFIG;
+ config.hdr.pkt_size = afe_sizeof_cfg_cmd(port_id);
+ config.hdr.src_port = 0;
+ config.hdr.dest_port = 0;
+ config.hdr.token = 0;
+ config.hdr.opcode = AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG;
+ } else {
+
+ config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ config.hdr.pkt_size = afe_sizeof_cfg_cmd(port_id);
+ config.hdr.src_port = 0;
+ config.hdr.dest_port = 0;
+ config.hdr.token = 0;
+ config.hdr.opcode = AFE_PORT_AUDIO_IF_CONFIG;
+ }
if (afe_validate_port(port_id) < 0) {
diff --git a/sound/soc/msm/qdsp6/q6asm.c b/sound/soc/msm/qdsp6/q6asm.c
index 62168d2..dc49f12 100644
--- a/sound/soc/msm/qdsp6/q6asm.c
+++ b/sound/soc/msm/qdsp6/q6asm.c
@@ -1,6 +1,6 @@
/*
- * Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ * Copyright (c) 2010-2012, Code Aurora Forum. All rights reserved.
* Author: Brian Swetland <swetland@google.com>
*
* This software is licensed under the terms of the GNU General Public
@@ -1193,6 +1193,9 @@
case FORMAT_LINEAR_PCM:
open.format = LINEAR_PCM;
break;
+ case FORMAT_MULTI_CHANNEL_LINEAR_PCM:
+ open.format = MULTI_CHANNEL_PCM;
+ break;
case FORMAT_MPEG4_AAC:
open.format = MPEG4_AAC;
break;
@@ -1761,6 +1764,66 @@
return -EINVAL;
}
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels)
+{
+ struct asm_stream_media_format_update fmt;
+ u8 *channel_mapping;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]rate[%d]ch[%d]\n", __func__, ac->session, rate,
+ channels);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
+
+ fmt.format = MULTI_CHANNEL_PCM;
+ fmt.cfg_size = sizeof(struct asm_multi_channel_pcm_fmt_blk);
+ fmt.write_cfg.multi_ch_pcm_cfg.num_channels = channels;
+ fmt.write_cfg.multi_ch_pcm_cfg.bits_per_sample = 16;
+ fmt.write_cfg.multi_ch_pcm_cfg.sample_rate = rate;
+ fmt.write_cfg.multi_ch_pcm_cfg.is_signed = 1;
+ fmt.write_cfg.multi_ch_pcm_cfg.is_interleaved = 1;
+ channel_mapping =
+ fmt.write_cfg.multi_ch_pcm_cfg.channel_mapping;
+
+ memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
+
+ if (channels == 1) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ } else if (channels == 2) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ channel_mapping[1] = PCM_CHANNEL_FR;
+ } else if (channels == 6) {
+ channel_mapping[0] = PCM_CHANNEL_FC;
+ channel_mapping[1] = PCM_CHANNEL_FL;
+ channel_mapping[2] = PCM_CHANNEL_FR;
+ channel_mapping[3] = PCM_CHANNEL_LB;
+ channel_mapping[4] = PCM_CHANNEL_RB;
+ channel_mapping[5] = PCM_CHANNEL_LFE;
+ } else {
+ pr_err("%s: ERROR.unsupported num_ch = %u\n", __func__,
+ channels);
+ return -EINVAL;
+ }
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
int q6asm_media_format_block_aac(struct audio_client *ac,
struct asm_aac_cfg *cfg)
{