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Merge changes I40b4684e,I9548b2da into msm-3.0

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* changes:
  ASoC: msm: HDMI PCM 6 channel support
  msm_fb: HDMI: Export interface for setting Audio info frame
This commit is contained in:
Linux Build Service Account
2012-02-10 21:50:50 -08:00
committed by QuIC Gerrit Code Review
parent e92ab0d569 5e809b01da
commit c538fbde96
20 changed files with 1494 additions and 81 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
arch/arm/mach-msm/board-8960.c
View File

@@ -1958,6 +1958,7 @@ static struct platform_device *sim_devices[] __initdata = {
&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 @@ static struct platform_device *cdp_devices[] __initdata = {
&msm_device_hsusb_host,
&android_usb_device,
&msm_pcm,
&msm_multi_ch_pcm,
&msm_pcm_routing,
&msm_cpudai0,
&msm_cpudai1,

7
arch/arm/mach-msm/devices-8960.c
View File

@@ -1389,6 +1389,11 @@ struct platform_device msm_pcm = {
.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_cpudai1 = {
};
struct platform_device msm_cpudai_hdmi_rx = {
.name = "msm-dai-q6",
.name = "msm-dai-q6-hdmi",
.id = 8,
};

1
arch/arm/mach-msm/devices.h
View File

@@ -163,6 +163,7 @@ extern struct platform_device msm_gsbi1_qup_spi_device;
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;

13
arch/arm/mach-msm/include/mach/msm_hdmi_audio.h
View File

@@ -1,4 +1,4 @@
/* Copyright (c) 2011, Code Aurora Forum. All rights reserved.
/* 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
@@ -13,6 +13,15 @@
#ifndef __MSM_HDMI_AUDIO_H
#define __MSM_HDMI_AUDIO_H
/* Supported HDMI Audio channels */
#define MSM_HDMI_AUDIO_CHANNEL_2 0
#define MSM_HDMI_AUDIO_CHANNEL_4 1
#define MSM_HDMI_AUDIO_CHANNEL_6 2
#define MSM_HDMI_AUDIO_CHANNEL_8 3
#define TRUE 1
#define FALSE 0
enum hdmi_supported_sample_rates {
HDMI_SAMPLE_RATE_32KHZ,
HDMI_SAMPLE_RATE_44_1KHZ,
@@ -27,5 +36,7 @@ int hdmi_audio_enable(bool on , u32 fifo_water_mark);
int hdmi_audio_packet_enable(bool on);
void hdmi_msm_audio_sample_rate_reset(int rate);
int hdmi_msm_audio_get_sample_rate(void);
int hdmi_msm_audio_info_setup(bool enabled, u32 num_of_channels,
u32 channel_allocation, u32 level_shift, bool down_mix);
#endif /* __MSM_HDMI_AUDIO_H*/

7
drivers/video/msm/external_common.c
View File

@@ -1,4 +1,4 @@
/* Copyright (c) 2010, 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
@@ -1000,14 +1000,15 @@ static void hdmi_edid_extract_speaker_allocation_data(const uint8 *in_buf)
return;
external_common_state->speaker_allocation_block = sad[1];
DEV_DBG("EDID: speaker allocation data=%s%s%s%s%s%s%s\n",
DEV_DBG("EDID: speaker allocation data SP byte = %08x %s%s%s%s%s%s%s\n",
sad[1],
(sad[1] & BIT(0)) ? "FL/FR," : "",
(sad[1] & BIT(1)) ? "LFE," : "",
(sad[1] & BIT(2)) ? "FC," : "",
(sad[1] & BIT(3)) ? "RL/RR," : "",
(sad[1] & BIT(4)) ? "RC," : "",
(sad[1] & BIT(5)) ? "FLC/FRC," : "",
(sad[1] & BIT(6)) ? "LFE," : "");
(sad[1] & BIT(6)) ? "RLC/RRC," : "");
}
static void hdmi_edid_extract_audio_data_blocks(const uint8 *in_buf)

34
drivers/video/msm/hdmi_msm.c
View File

@@ -64,6 +64,8 @@ DEFINE_MUTEX(hdmi_msm_state_mutex);
EXPORT_SYMBOL(hdmi_msm_state_mutex);
static DEFINE_MUTEX(hdcp_auth_state_mutex);
static void hdmi_msm_dump_regs(const char *prefix);
#ifdef CONFIG_FB_MSM_HDMI_MSM_PANEL_HDCP_SUPPORT
static void hdmi_msm_hdcp_enable(void);
#else
@@ -3280,14 +3282,23 @@ int hdmi_audio_packet_enable(bool on)
}
EXPORT_SYMBOL(hdmi_audio_packet_enable);
static void hdmi_msm_audio_info_setup(boolean enabled, int num_of_channels,
int level_shift, boolean down_mix)
/* TO-DO: return -EINVAL when num_of_channels and channel_allocation
* does not match CEA 861-D spec.
*/
int hdmi_msm_audio_info_setup(bool enabled, u32 num_of_channels,
u32 channel_allocation, u32 level_shift, bool down_mix)
{
uint32 channel_allocation = 0; /* Default to FR,FL */
uint32 channel_count = 1; /* Default to 2 channels
-> See Table 17 in CEA-D spec */
uint32 check_sum, audio_info_0_reg, audio_info_1_reg;
uint32 audio_info_ctrl_reg;
u32 aud_pck_ctrl_2_reg;
u32 layout;
layout = (MSM_HDMI_AUDIO_CHANNEL_2 == num_of_channels) ? 0 : 1;
aud_pck_ctrl_2_reg = 1 | (layout << 1);
HDMI_OUTP(0x00044, aud_pck_ctrl_2_reg);
/* Please see table 20 Audio InfoFrame in HDMI spec
FL = front left
@@ -3310,6 +3321,7 @@ static void hdmi_msm_audio_info_setup(boolean enabled, int num_of_channels,
if (enabled) {
switch (num_of_channels) {
case MSM_HDMI_AUDIO_CHANNEL_2:
channel_allocation = 0; /* Default to FR,FL */
break;
case MSM_HDMI_AUDIO_CHANNEL_4:
channel_count = 3;
@@ -3327,6 +3339,9 @@ static void hdmi_msm_audio_info_setup(boolean enabled, int num_of_channels,
channel_allocation = 0x1f;
break;
default:
pr_err("%s(): Unsupported num_of_channels = %u\n",
__func__, num_of_channels);
return -EINVAL;
break;
}
@@ -3382,7 +3397,14 @@ static void hdmi_msm_audio_info_setup(boolean enabled, int num_of_channels,
}
/* HDMI_INFOFRAME_CTRL0[0x002C] */
HDMI_OUTP(0x002C, audio_info_ctrl_reg);
hdmi_msm_dump_regs("HDMI-AUDIO-ON: ");
return 0;
}
EXPORT_SYMBOL(hdmi_msm_audio_info_setup);
static void hdmi_msm_en_gc_packet(boolean av_mute_is_requested)
{
@@ -3488,7 +3510,7 @@ static void hdmi_msm_audio_setup(void)
hdmi_msm_audio_acr_setup(TRUE,
external_common_state->video_resolution,
msm_hdmi_sample_rate, channels);
hdmi_msm_audio_info_setup(TRUE, channels, 0, FALSE);
hdmi_msm_audio_info_setup(TRUE, channels, 0, 0, FALSE);
/* Turn on Audio FIFO and SAM DROP ISR */
HDMI_OUTP(0x02CC, HDMI_INP(0x02CC) | BIT(1) | BIT(3));
@@ -3519,7 +3541,7 @@ static int hdmi_msm_audio_off(void)
return -ETIMEDOUT;
}
}
hdmi_msm_audio_info_setup(FALSE, 0, 0, FALSE);
hdmi_msm_audio_info_setup(FALSE, 0, 0, 0, FALSE);
hdmi_msm_audio_acr_setup(FALSE, 0, 0, 0);
DEV_INFO("HDMI Audio: Disabled\n");
return 0;
@@ -4042,7 +4064,7 @@ static int hdmi_msm_power_on(struct platform_device *pdev)
return rc;
}
}
hdmi_msm_audio_info_setup(TRUE, 0, 0, FALSE);
hdmi_msm_audio_info_setup(TRUE, 0, 0, 0, FALSE);
mutex_lock(&external_common_state_hpd_mutex);
hdmi_msm_state->panel_power_on = TRUE;

162
include/sound/apr_audio.h
View File

@@ -234,10 +234,18 @@ struct afe_port_hdmi_cfg {
/* 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 @@ struct afe_port_rtproxy_cfg {
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 @@ struct adm_copp_open_command {
#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 @@ struct adm_copp_open_respond {
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 @@ struct asm_pcm_cfg {
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 @@ struct asm_stream_cmd_open_read {
#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_stream_media_format_update{
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));

5
include/sound/q6adm.h
View File

@@ -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 @@ struct route_payload {
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);

6
include/sound/q6asm.h
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@@ -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_enc_cfg_blk_amrwb(struct audio_client *ac, uint32_t frames_per_buf,
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);

8
sound/soc/msm/Kconfig
View File

@@ -80,6 +80,13 @@ config SND_SOC_MSM8660_LPAIF
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 @@ config SND_SOC_MSM8960
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

3
sound/soc/msm/Makefile
View File

@@ -56,7 +56,8 @@ obj-$(CONFIG_SND_SOC_MSM8X60) += snd-soc-lpass-dma.o
obj-$(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

2
sound/soc/msm/msm-dai-fe.c
View File

@@ -75,7 +75,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.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,
},

283
sound/soc/msm/msm-dai-q6-hdmi.c Normal file
View File

@@ -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");

44
sound/soc/msm/msm-dai-q6.c
View File

@@ -206,30 +206,6 @@ static int msm_dai_q6_cdc_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
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 @@ static int msm_dai_q6_hw_params(struct snd_pcm_substream *substream,
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 @@ static struct snd_soc_dai_driver msm_dai_q6_afe_tx_dai = {
.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 @@ static __devinit int msm_dai_q6_dev_probe(struct platform_device *pdev)
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);

723
sound/soc/msm/msm-multi-ch-pcm-q6.c Normal file
View File

@@ -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");

45
sound/soc/msm/msm-pcm-routing.c
View File

@@ -162,6 +162,7 @@ void msm_pcm_routing_reg_phy_stream(int fedai_id, int dspst_id, int stream_type)
{
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 @@ void msm_pcm_routing_reg_phy_stream(int fedai_id, int dspst_id, int stream_type)
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 @@ void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type)
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 @@ static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
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 @@ static int msm_pcm_routing_prepare(struct snd_pcm_substream *substream)
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 @@ static int msm_pcm_routing_prepare(struct snd_pcm_substream *substream)
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);
}

8
sound/soc/msm/msm8960.c
View File

@@ -787,8 +787,10 @@ static int msm8960_hdmi_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
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 @@ static struct snd_soc_dai_link msm8960_dai[] = {
.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 @@ static struct snd_soc_dai_link msm8960_dai[] = {
{
.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",

130
sound/soc/msm/qdsp6/q6adm.c
View File

@@ -103,7 +103,8 @@ static int32_t adm_callback(struct apr_client_data *data, void *priv)
}
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 @@ fail_cmd:
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)
{

27
sound/soc/msm/qdsp6/q6afe.c
View File

@@ -77,6 +77,7 @@ static int32_t afe_callback(struct apr_client_data *data, void *priv)
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 @@ int afe_sizeof_cfg_cmd(u16 port_id)
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 @@ int afe_port_start_nowait(u16 port_id, union afe_port_config *afe_config,
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) {

65
sound/soc/msm/qdsp6/q6asm.c
View File

@@ -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 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format)
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 @@ fail_cmd:
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)
{