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转载：https://blog.csdn.net/u010481276/article/details/81122226
音频调试工具：audacity ,cool edit, gold wav Sonic Visualiser
http://www.360doc.com/content/18/0314/12/32862269_736896674.shtml
./hardware/qcom/audio/configs/msm8909/mixer_paths_msm8909_pm8916.xml中查看 ，这个配置文件是根据平台声卡名得来的
hardware/qcom/audio/hal/msm8916/platform.c
msm8909:/dev/snd # cat /proc/asound/cards
0 [msm8909pm8916sn]: msm8909-pm8916- - msm8909-pm8916-snd-card
msm8909-pm8916-snd-card
} else if (!strncmp(snd_card_name, "msm8909-pm8916-snd-card",
sizeof("msm8909-pm8916-snd-card"))) {
strlcpy(mixer_xml_path, MIXER_XML_PATH_MSM8909_PM8916,
sizeof(MIXER_XML_PATH_MSM8909_PM8916));
msm_device_to_be_id = msm_device_to_be_id_internal_codec;
msm_be_id_array_len =
sizeof(msm_device_to_be_id_internal_codec) / sizeof(msm_device_to_be_id_internal_codec[0]);
}
msm8909:/dev/snd # cat /proc/asound/cards
0 [msm8909pm8916sn]: msm8909-pm8916- - msm8909-pm8916-snd-card
msm8909-pm8916-snd-card
xml包含codec切换path的配置，如下面配有切换至speaker所需要的参数设置（通过tinymix进行配置）
驱动代码放在如下位置, 里面包含audio path切换的"audio_map"表格，
本平台为msm8909
注意，mixer_paths_msm8909_pm8916.xml中如下，只需要一个空格，否则无效
//headset
<path name="handset">
<ctl name="RX1 MIX1 INP1" value="RX1" />
<ctl name="RDAC2 MUX" value="RX1" />
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="EAR PA Gain" value="POS_6_DB" />
<ctl name="EAR_S" value="Switch" />
</path>
tinymix “PRI_MI2S_RX Audio Mixer MultiMedia1” “1”
tinymix “RX1 MIX1 INP1” “RX1”
tinymix “RDAC2 MUX” “RX1”
tinymix “RX1 Digital Volume” “84”
tinymix “EAR PA Gain” “POS_6_DB”
tinymix “EAR_S” “Switch”
tinyplay xxx.wav
tinymix ‘MI2S_RX Format’ ‘1’ 设置成24bit，i2s传输,设备也需要切到对应的i2s格式
//headphones
<path name="headphones">
<ctl name="MI2S_RX Channels" value="Two" />
<ctl name="RX1 MIX1 INP1" value="RX1" />
<ctl name="RX2 MIX1 INP1" value="RX2" />
<ctl name="RDAC2 MUX" value="RX2" />
<ctl name="HPHL" value="Switch" />
<ctl name="HPHR" value="Switch" />
</path>
tinymix “PRI_MI2S_RX Audio Mixer MultiMedia1” “1”
SND_SOC_DAPM_AIF_OUT("PRI_MI2S_RX", "Primary MI2S Playback",
0, 0, 0, 0),snd_soc_dapm_aif_out
实际为snd_soc_dapm_aif_out对应一个数字音频输出接口，比如I2S接口的输出端。如一组i2s的输出端
msm8909:/ # tinymix |grep "PRI_MI2S_RX Audio Mixer MultiMedia1"
1460 BOOL 1 PRI_MI2S_RX Audio Mixer MultiMedia1 Off
1468 BOOL 1 PRI_MI2S_RX Audio Mixer MultiMedia10 Off
1469 BOOL 1 PRI_MI2S_RX Audio Mixer MultiMedia11 Off
1470 BOOL 1 PRI_MI2S_RX Audio Mixer MultiMedia12 Off
1471 BOOL 1 PRI_MI2S_RX Audio Mixer MultiMedia13 Off
1472 BOOL 1 PRI_MI2S_RX Audio Mixer MultiMedia14 Off
1473 BOOL 1 PRI_MI2S_RX Audio Mixer MultiMedia15 Off
1474 BOOL 1 PRI_MI2S_RX Audio Mixer MultiMedia16 Off
tinymix “MI2S_RX Channels” “Two”
tinymix “RX1 MIX1 INP1” “RX1”
tinymix “RX2 MIX1 INP1” “RX2”
tinymix “RDAC2 MUX” “RX2”
tinymix “HPHL” “Switch”
tinymix “HPHR” “Switch”
tinymix “RX2 Digital Volume” “84”
//mic adc1 如上面mic为adc1那路，那只找 adc1即可
<path name="adc1">
<ctl name="ADC1 Volume" value="6" />
<ctl name="DEC1 MUX" value="ADC1" />
</path>
tinymix “MultiMedia1 Mixer TERT_MI2S_TX” 1
tinymix “ADC1 Volume” 6
tinymix “DEC1 MUX” “ADC1”
tinycap /data/mic1.wav
//speaker
<path name="speaker">
<ctl name="RX3 MIX1 INP1" value="RX1" />
<ctl name="SPK" value="Switch" />
</path>
tinymix ‘PRI_MI2S_RX Audio Mixer MultiMedia1’ 1
tinymix ‘RX3 MIX1 INP1’ ‘RX1’
tinymix “RX1 Digital Volume” “84”
tinymix “RX1 Digital Volume” “60”
tinymix “SPK” “Switch”
tinyplay /data/01_1KHz_0dB.wav
以上的控制功能都是由kcontrol提供的如pm8916上的kcontrol
static const struct snd_kcontrol_new msm8x16_wcd_snd_controls[] = {
SOC_ENUM_EXT("RX HPH Mode", msm8x16_wcd_hph_mode_ctl_enum[0],
msm8x16_wcd_hph_mode_get, msm8x16_wcd_hph_mode_set),
SOC_ENUM_EXT("Boost Option", msm8x16_wcd_boost_option_ctl_enum[0],
msm8x16_wcd_boost_option_get, msm8x16_wcd_boost_option_set),
SOC_ENUM_EXT("EAR PA Boost", msm8x16_wcd_ear_pa_boost_ctl_enum[0],
msm8x16_wcd_ear_pa_boost_get, msm8x16_wcd_ear_pa_boost_set),
SOC_ENUM_EXT("EAR PA Gain", msm8x16_wcd_ear_pa_gain_enum[0],
msm8x16_wcd_pa_gain_get, msm8x16_wcd_pa_gain_put),
SOC_ENUM_EXT("Speaker Boost", msm8x16_wcd_spk_boost_ctl_enum[0],
msm8x16_wcd_spk_boost_get, msm8x16_wcd_spk_boost_set),
SOC_ENUM_EXT("Ext Spk Boost", msm8x16_wcd_ext_spk_boost_ctl_enum[0],
msm8x16_wcd_ext_spk_boost_get, msm8x16_wcd_ext_spk_boost_set),
SOC_ENUM_EXT("LOOPBACK Mode", msm8x16_wcd_loopback_mode_ctl_enum[0],
msm8x16_wcd_loopback_mode_get, msm8x16_wcd_loopback_mode_put),
SOC_SINGLE_TLV("ADC1 Volume", MSM8X16_WCD_A_ANALOG_TX_1_EN, 3,
8, 0, analog_gain),
SOC_SINGLE_TLV("ADC2 Volume", MSM8X16_WCD_A_ANALOG_TX_2_EN, 3,
8, 0, analog_gain),
SOC_SINGLE_TLV("ADC3 Volume", MSM8X16_WCD_A_ANALOG_TX_3_EN, 3,
8, 0, analog_gain),
SOC_ENUM_EXT("EAR PA Gain", msm8x16_wcd_ear_pa_gain_enum[0],
msm8x16_wcd_pa_gain_get, msm8x16_wcd_pa_gain_put),
如ear的增益
"EAR PA Gain"是暴露给tinymix的属性名字
msm8x16_wcd_ear_pa_gain_enum[0]是对应可供选择的值
“POS_1P5_DB”, “POS_6_DB”
msm8x16_wcd_pa_gain_get, msm8x16_wcd_pa_gain_put是对应的操作函数
static const char * const msm8x16_wcd_ear_pa_gain_text[] = {
"POS_1P5_DB", "POS_6_DB"};
static const struct soc_enum msm8x16_wcd_ear_pa_gain_enum[] = {
SOC_ENUM_SINGLE_EXT(2, msm8x16_wcd_ear_pa_gain_text),
};
对应的操作接口set
static int msm8x16_wcd_pa_gain_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
u8 ear_pa_gain;
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
dev_dbg(codec->dev, "%s: ucontrol->value.integer.value[0] = %ld\n",
__func__, ucontrol->value.integer.value[0]);
switch (ucontrol->value.integer.value[0]) {
case 0: //对应的枚举值
ear_pa_gain = 0x00;
break;
case 1:
ear_pa_gain = 0x20;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, MSM8X16_WCD_A_ANALOG_RX_EAR_CTL,
0x20, ear_pa_gain);
return 0;
}
ls -al /dev/snd/下面的文件
音频设备的命名规则为 [device type]C[card index]D[device index][capture/playback]，即名字中含有4部分的信息：
device type
设备类型，通常只有compr/hw/pcm这3种。从上图可以看到声卡会管理很多设备，PCM设备只是其中的一种设备。
card index
声卡的id，代表第几块声卡。通常都是0，代表第一块声卡。手机上通常都只有一块声卡。
device index
设备的id，代表这个设备是声卡上的第几个设备。设备的ID只和驱动中配置的DAI link的次序有关。如果驱动没有改变，那么这些ID就是固定的。
msm8909:/dev/snd # cat /sys/kernel/debug/asoc/dais
cajon_vifeedback
msm8x16_wcd_i2s_tx1
msm8x16_wcd_i2s_rx1
tas5751-i2s
MultiMedia29
MultiMedia28
MultiMedia21
MultiMedia20
MultiMedia19
MultiMedia18
MultiMedia17
VoiceMMode2
VoiceMMode1
MultiMedia16
MultiMedia15
MultiMedia14
MultiMedia13
MultiMedia12
MultiMedia11
MultiMedia10
VoWLAN
LSM8
LSM7
LSM6
LSM5
LSM4
LSM3
LSM2
LSM1
前面已经创建了control设备，现在soc_probe_link_dais调用soc_new_pcm创建pcm设备。
1）设置pcm native中要使用的pcm操作函数，这些函数用于操作音频物理设备，包括machine、codec_dai、cpu_dai、platform；
2）调用snd_pcm_new()创建pcm设备，回放子流实例和录制子流实例都在这里创建；
3）回调platform驱动的pcm_new()，完成音频dma设备初始化和dma buffer内存分配；
soc-core.c
soc_probe
platform_get_drvdata(pdev)//获取声卡
snd_soc_register_card
snd_soc_instantiate_card
soc_probe_link_dais
soc-pcm.c
soc_new_pcm
capture/playback
只有PCM设备才有这部分，只有c和p两种。c代表capture，说明这是一个提供录音的设备，p代表palyback，说明这是一个提供播放的设备。
系统会在/proc/asound/pcm文件中列出所有的音频设备的信息，如果是肉眼查看，/proc/asound/pcm中的信息会更直观一些
调用流程：
tinyplay调用流程
从pcm的操作函数开始
const struct file_operations snd_pcm_f_ops[2] = {
{
.owner = THIS_MODULE,
.write = snd_pcm_write,
.aio_write = snd_pcm_aio_write,
.open = snd_pcm_playback_open,
.release = snd_pcm_release,
.llseek = no_llseek,
.poll = snd_pcm_playback_poll,
.unlocked_ioctl = snd_pcm_playback_ioctl,
.compat_ioctl = snd_pcm_ioctl_compat,
.mmap = snd_pcm_mmap,
.fasync = snd_pcm_fasync,
.get_unmapped_area = snd_pcm_get_unmapped_area,
},
{
.owner = THIS_MODULE,
.read = snd_pcm_read,
.aio_read = snd_pcm_aio_read,
.open = snd_pcm_capture_open,
.release = snd_pcm_release,
.llseek = no_llseek,
.poll = snd_pcm_capture_poll,
.unlocked_ioctl = snd_pcm_capture_ioctl,
.compat_ioctl = snd_pcm_ioctl_compat,
.mmap = snd_pcm_mmap,
.fasync = snd_pcm_fasync,
.get_unmapped_area = snd_pcm_get_unmapped_area,
}
};
snd_pcm_playback_open
snd_pcm_open
snd_pcm_open_file
snd_pcm_open_substream-》substream->ops->open在
soc_new_pcm中注册回调的
/* ASoC PCM operations */
if (rtd->dai_link->dynamic) {
rtd->ops.open = dpcm_fe_dai_open;
rtd->ops.hw_params = dpcm_fe_dai_hw_params;
rtd->ops.prepare = dpcm_fe_dai_prepare;
rtd->ops.trigger = dpcm_fe_dai_trigger;
rtd->ops.hw_free = dpcm_fe_dai_hw_free;
rtd->ops.close = dpcm_fe_dai_close;
rtd->ops.pointer = soc_pcm_pointer;
rtd->ops.delay_blk = soc_pcm_delay_blk;
rtd->ops.ioctl = soc_pcm_ioctl;
rtd->ops.compat_ioctl = soc_pcm_compat_ioctl;
} else {
rtd->ops.open = soc_pcm_open;
rtd->ops.hw_params = soc_pcm_hw_params;
rtd->ops.prepare = soc_pcm_prepare;
rtd->ops.trigger = soc_pcm_trigger;
rtd->ops.hw_free = soc_pcm_hw_free;
rtd->ops.close = soc_pcm_close;
rtd->ops.pointer = soc_pcm_pointer;
rtd->ops.delay_blk = soc_pcm_delay_blk;
rtd->ops.ioctl = soc_pcm_ioctl;
rtd->ops.compat_ioctl = soc_pcm_compat_ioctl;
}
if (platform->driver->ops) {
rtd->ops.ack = platform->driver->ops->ack;
rtd->ops.copy = platform->driver->ops->copy;
rtd->ops.silence = platform->driver->ops->silence;
rtd->ops.page = platform->driver->ops->page;
rtd->ops.mmap = platform->driver->ops->mmap;
rtd->ops.restart = platform->driver->ops->restart;
}
soc_pcm_open
1
/*
* Called by ALSA when a PCM substream is opened, the runtime->hw record is
* then initialized and any private data can be allocated. This also calls
* startup for the cpu DAI, platform, machine and codec DAI.
*/
static int soc_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai;
const char *codec_dai_name = "multicodec";
int i, ret = 0;
pinctrl_pm_select_default_state(cpu_dai->dev);
for (i = 0; i < rtd->num_codecs; i++)
pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev);
pm_runtime_get_sync(cpu_dai->dev);
for (i = 0; i < rtd->num_codecs; i++)
pm_runtime_get_sync(rtd->codec_dais[i]->dev);
pm_runtime_get_sync(platform->dev);
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
if (rtd->dai_link->no_host_mode == SND_SOC_DAI_LINK_NO_HOST)
snd_soc_set_runtime_hwparams(substream, &no_host_hardware);
/* startup the audio subsystem */操作cpu dai的startup接口
if (cpu_dai->driver->ops && cpu_dai->driver->ops->startup) {
ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
if (ret < 0) {
dev_err(cpu_dai->dev, "ASoC: can't open interface"
" %s: %d\n", cpu_dai->name, ret);
goto out;
}
}
操作platform的open接口
if (platform->driver->ops && platform->driver->ops->open) {
ret = platform->driver->ops->open(substream);
if (ret < 0) {
dev_err(platform->dev, "ASoC: can't open platform"
" %s: %d\n", platform->component.name, ret);
goto platform_err;
}
}
操作codec dai的startup接口
for (i = 0; i < rtd->num_codecs; i++) {
codec_dai = rtd->codec_dais[i];
if (codec_dai->driver->ops && codec_dai->driver->ops->startup) {
ret = codec_dai->driver->ops->startup(substream,
codec_dai);
if (ret < 0) {
dev_err(codec_dai->dev,
"ASoC: can't open codec %s: %d\n",
codec_dai->name, ret);
goto codec_dai_err;
}
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
codec_dai->tx_mask = 0;
else
codec_dai->rx_mask = 0;
}
操作 dai link的startup接口
if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
ret = rtd->dai_link->ops->startup(substream);
if (ret < 0) {
pr_err("ASoC: %s startup failed: %d\n",
rtd->dai_link->name, ret);
goto machine_err;
}
}
/* Dynamic PCM DAI links compat checks use dynamic capabilities */
if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm)
goto dynamic;
/* Check that the codec and cpu DAIs are compatible */
soc_pcm_init_runtime_hw(substream);
if (rtd->num_codecs == 1)
codec_dai_name = rtd->codec_dai->name;
if (soc_pcm_has_symmetry(substream))
runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
ret = -EINVAL;
if (!runtime->hw.rates) {
printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
codec_dai_name, cpu_dai->name);
goto config_err;
}
if (!runtime->hw.formats) {
printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n",
codec_dai_name, cpu_dai->name);
goto config_err;
}
if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
runtime->hw.channels_min > runtime->hw.channels_max) {
printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n",
codec_dai_name, cpu_dai->name);
goto config_err;
}
soc_pcm_apply_msb(substream);
/* Symmetry only applies if we've already got an active stream. */
if (cpu_dai->active) {
ret = soc_pcm_apply_symmetry(substream, cpu_dai);
if (ret != 0)
goto config_err;
}
for (i = 0; i < rtd->num_codecs; i++) {
if (rtd->codec_dais[i]->active) {
ret = soc_pcm_apply_symmetry(substream,
rtd->codec_dais[i]);
if (ret != 0)
goto config_err;
}
}
pr_debug("ASoC: %s <-> %s info:\n",
codec_dai_name, cpu_dai->name);
pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates);
pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min,
runtime->hw.channels_max);
pr_debug("ASoC: min rate %d max rate %d\n", runtime->hw.rate_min,
runtime->hw.rate_max);
dynamic:
snd_soc_runtime_activate(rtd, substream->stream);
mutex_unlock(&rtd->pcm_mutex);
return 0;
config_err://出错处理，shutdown dai link
if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
rtd->dai_link->ops->shutdown(substream);
machine_err:
i = rtd->num_codecs;
codec_dai_err:
while (--i >= 0) {//出错处理，shutdown codec dai
codec_dai = rtd->codec_dais[i];
if (codec_dai->driver->ops->shutdown)
codec_dai->driver->ops->shutdown(substream, codec_dai);
}
//出错处理，close platform
if (platform->driver->ops && platform->driver->ops->close)
platform->driver->ops->close(substream);
platform_err://出错处理，shutdown cpu dai
if (cpu_dai->driver->ops && cpu_dai->driver->ops->shutdown)
cpu_dai->driver->ops->shutdown(substream, cpu_dai);
out:
mutex_unlock(&rtd->pcm_mutex);
pm_runtime_put(platform->dev);
for (i = 0; i < rtd->num_codecs; i++)
pm_runtime_put(rtd->codec_dais[i]->dev);
pm_runtime_put(cpu_dai->dev);
for (i = 0; i < rtd->num_codecs; i++) {
if (!rtd->codec_dais[i]->active)
pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
}
if (!cpu_dai->active)
pinctrl_pm_select_sleep_state(cpu_dai->dev);
return ret;
}
.name = LPASS_BE_QUAT_MI2S_TX,
.stream_name = "Quaternary MI2S Capture",
.cpu_dai_name = "msm-dai-q6-mi2s.3",
.platform_name = "msm-pcm-routing",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.no_pcm = 1,
.dpcm_capture = 1,
.be_id = MSM_BACKEND_DAI_QUATERNARY_MI2S_TX,
.be_hw_params_fixup = msm_be_hw_params_fixup,
.ops = &msm8952_quat_mi2s_be_ops,
.ignore_suspend = 1,
cpu_dai_name = "msm-dai-q6-mi2s.3",
{
.playback = {
.stream_name = "Quaternary MI2S Playback",
.aif_name = "QUAT_MI2S_RX",
.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
SNDRV_PCM_RATE_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rate_min = 8000,
.rate_max = 192000,
},
.capture = {
.stream_name = "Quaternary MI2S Capture",
.aif_name = "QUAT_MI2S_TX",
.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rate_min = 8000,
.rate_max = 48000,
},
.ops = &msm_dai_q6_mi2s_ops,
.id = MSM_QUAT_MI2S,
.probe = msm_dai_q6_dai_mi2s_probe,
.remove = msm_dai_q6_dai_mi2s_remove,
}
static struct snd_soc_dai_ops msm_dai_q6_mi2s_ops = {
.startup = msm_dai_q6_mi2s_startup,
.prepare = msm_dai_q6_mi2s_prepare,
.hw_params = msm_dai_q6_mi2s_hw_params,
.hw_free = msm_dai_q6_mi2s_hw_free,
.set_fmt = msm_dai_q6_mi2s_set_fmt,
.shutdown = msm_dai_q6_mi2s_shutdown,
};
msm-pcm-routing
static struct snd_soc_platform_driver msm_soc_routing_platform = {
.ops = &msm_routing_pcm_ops,
.probe = msm_routing_probe,
.pcm_new = msm_routing_pcm_new,
.pcm_free = msm_routing_pcm_free,
};
static struct snd_pcm_ops msm_routing_pcm_ops = {
.hw_params = msm_pcm_routing_hw_params,
.close = msm_pcm_routing_close,
.prepare = msm_pcm_routing_prepare,
};
snd-soc-dummy-dai
static struct snd_soc_dai_driver dummy_dai = {
.name = "snd-soc-dummy-dai",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 384,
.rates = STUB_RATES,
.formats = STUB_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 384,
.rates = STUB_RATES,
.formats = STUB_FORMATS,
},
};
static struct snd_pcm_ops dummy_dma_ops = {
.open = dummy_dma_open,
.ioctl = snd_pcm_lib_ioctl,
};
static struct snd_soc_platform_driver dummy_platform = {
.ops = &dummy_dma_ops,
};
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
const struct snd_pcm_hardware *hw)
{
struct snd_pcm_runtime *runtime = substream->runtime;
if (!runtime)
return 0;
runtime->hw.info = hw->info;
runtime->hw.formats = hw->formats;
runtime->hw.period_bytes_min = hw->period_bytes_min;
runtime->hw.period_bytes_max = hw->period_bytes_max;
runtime->hw.periods_min = hw->periods_min;
runtime->hw.periods_max = hw->periods_max;
runtime->hw.buffer_bytes_max = hw->buffer_bytes_max;
runtime->hw.fifo_size = hw->fifo_size;
return 0;
}
dai_link:
{
.name = LPASS_BE_QUAT_MI2S_TX,
.stream_name = "Quaternary MI2S Capture",
.cpu_dai_name = "msm-dai-q6-mi2s.3",
.platform_name = "msm-pcm-routing",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.no_pcm = 1,
.dpcm_capture = 1,
.be_id = MSM_BACKEND_DAI_QUATERNARY_MI2S_TX,
.be_hw_params_fixup = msm_be_hw_params_fixup,
.ops = &msm8952_quat_mi2s_be_ops,
.ignore_suspend = 1,
static struct snd_soc_ops msm8952_quat_mi2s_be_ops = {
.startup = msm_quat_mi2s_snd_startup,
.hw_params = msm_mi2s_snd_hw_params,
.shutdown = msm_quat_mi2s_snd_shutdown,
};
static int msm_quat_mi2s_snd_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_card *card = rtd->card;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct msm8916_asoc_mach_data *pdata =
snd_soc_card_get_drvdata(card);
int ret = 0, val = 0;
pr_debug("%s(): substream = %s stream = %d\n", __func__,
substream->name, substream->stream);
if (!q6core_is_adsp_ready()) {
pr_err("%s(): adsp not ready\n", __func__);
return -EINVAL;
}
if (pdata->vaddr_gpio_mux_mic_ctl) {
val = ioread32(pdata->vaddr_gpio_mux_mic_ctl);
val = val | 0x02020002;
iowrite32(val, pdata->vaddr_gpio_mux_mic_ctl);
}
ret = msm_mi2s_sclk_ctl(substream, true);
if (ret < 0) {
pr_err("failed to enable sclk\n");
return ret;
}
ret = msm_gpioset_activate(CLIENT_WCD_INT, "quat_i2s");
if (ret < 0) {
pr_err("failed to enable codec gpios\n");
goto err;
}
if (atomic_inc_return(&quat_mi2s_clk_ref) == 1) {
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
pr_err("%s: set fmt cpu dai failed\n", __func__);
}
return ret;
err:
ret = msm_mi2s_sclk_ctl(substream, false);
if (ret < 0)
pr_err("failed to disable sclk\n");
return ret;
}
tinymix ‘QUAT_MI2S_RX Audio Mixer MultiMedia1’ 1
tinyplay /data/1.wav
QUAT_MI2S_RX Audio Mixer是一个widget ,这里其实和MultiMedia1就决定了后端的端口号,即将要打开的port_id号
MultiMedia1是一个snd_kcontrol_new,根据情况分析,这里为对应的前端,一般MultiMedia1对应alsa前端即上层打开的设备号,对应的声卡设备如下,MultiMedia1 为c0d0
且tinyplay tinymix里面程序默认打开是c0d0p因此能播放,若是MultiMedia5刚不是c0d0p,刚tinymix tinyplay要指定c号d号是多少,详细参见tinyplay命令
程序中用pcm_open打开需要带device id号,如muutimedia6 card是0,device是18
00-18: MultiMedia6 (*) : : playback 1 : capture 1
struct pcm *pcm_open(unsigned int card, unsigned int device, unsigned int flags, struct pcm_config *config);
则device要为18
这里的c号d号为codec在snd_soc_dai_link的数组下标,MultiMedia6为结构中数组中的cpu_dai_name名字
msm8909:/proc/asound # cat pcm
cat pcm
00-00: MultiMedia1 (*) : : playback 1 : capture 1
00-01: MultiMedia2 (*) : : playback 1 : capture 1
00-02: CS-Voice (*) : : playback 1 : capture 1
00-03: VoIP (*) : : playback 1 : capture 1
00-04: ULL (*) : : playback 1
00-05: Primary MI2S_RX Hostless (*) : : playback 1
00-06: INT_FM Hostless (*) : : capture 1
00-07: AFE-PROXY RX msm-stub-rx-7 : : playback 1
00-08: AFE-PROXY TX msm-stub-tx-8 : : capture 1
00-09: (Compress1) : : playback 1 : capture 1
00-10: AUXPCM Hostless (*) : : playback 1 : capture 1
00-11: Tertiary MI2S_TX Hostless (*) : : capture 1
00-12: MultiMedia5 (*) : : playback 1 : capture 1
00-13: Voice2 (*) : : playback 1 : capture 1
00-14: MultiMedia9 (*) : : playback 1 : capture 1
00-15: VoLTE (*) : : playback 1 : capture 1
00-16: VoWLAN (*) : : playback 1 : capture 1
00-17: INT_HFP_BT Hostless (*) : : playback 1 : capture 1
00-18: MultiMedia6 (*) : : playback 1 : capture 1
00-19: Listen 1 Audio Service (*) : : capture 1
00-20: Listen 2 Audio Service (*) : : capture 1
00-21: Listen 3 Audio Service (*) : : capture 1
00-22: Listen 4 Audio Service (*) : : capture 1
00-23: Listen 5 Audio Service (*) : : capture 1
00-24: (Compress2) : : playback 1
00-25: QUAT_MI2S Hostless (*) : : playback 1
00-26: Senary_mi2s Capture cajon_vifeedback-26 : : capture 1
00-27: (Compress3) : : playback 1
00-28: (Compress4) : : playback 1
00-29: (Compress5) : : playback 1
00-30: (Compress6) : : playback 1
00-31: (Compress7) : : playback 1
00-32: (Compress8) : : playback 1
00-33: (Compress9) : : playback 1
00-34: VoiceMMode1 (*) : : playback 1 : capture 1
00-35: VoiceMMode2 (*) : : playback 1 : capture 1
00-36: MultiMedia8 (*) : : playback 1 : capture 1
00-37: QCHAT (*) : : playback 1 : capture 1
00-38: (Compress10) : : capture 1
00-39: (Compress11) : : capture 1
00-40: (Compress12) : : capture 1
00-41: (Compress13) : : capture 1
00-42: (Compress14) : : capture 1
00-43: (Primary MI2S Playback) : : playback 1
00-44: (Secondary MI2S Playback) : : playback 1
00-45: (Tertiary MI2S Capture) : : capture 1
00-46: (Quaternary MI2S Playback) : : playback 1
00-47: (Quaternary MI2S Capture) : : capture 1
00-48: (AUX PCM Playback) : : playback 1
00-49: (AUX PCM Capture) : : capture 1
00-50: (Internal BT-SCO Playback) : : playback 1
00-51: (Internal BT-SCO Capture) : : capture 1
00-52: (Internal FM Playback) : : playback 1
00-53: (Internal FM Capture) : : capture 1
00-54: (AFE Playback) : : playback 1
00-55: (AFE Capture) : : capture 1
00-56: (Voice Uplink Capture) : : capture 1
00-57: (Voice Downlink Capture) : : capture 1
00-58: (Voice Farend Playback) : : playback 1
00-59: (Voice2 Farend Playback) : : playback 1
00-60: (Quinary MI2S Capture) : : capture 1
00-61: (Quinary MI2S Playback) : : playback 1
00-62: (Internal BT-A2DP Playback) : : playback 1
msm8909:/proc/asound # cat pcm|grep BT
cat pcm|grep BT
00-17: INT_HFP_BT Hostless (*) : : playback 1 : capture 1
00-50: (Internal BT-SCO Playback) : : playback 1
00-51: (Internal BT-SCO Capture) : : capture 1
00-62: (Internal BT-A2DP Playback) : : playback 1
---------------------
#define SND_SOC_DAPM_SIGGEN(wname) \
{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM }
#define SND_SOC_DAPM_INPUT(wname) \
{ .id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM }
#define SND_SOC_DAPM_OUTPUT(wname) \
{ .id = snd_soc_dapm_output, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM }
#define SND_SOC_DAPM_MIC(wname, wevent) \
{ .id = snd_soc_dapm_mic, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
#define SND_SOC_DAPM_HP(wname, wevent) \
{ .id = snd_soc_dapm_hp, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
#define SND_SOC_DAPM_SPK(wname, wevent) \
{ .id = snd_soc_dapm_spk, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
#define SND_SOC_DAPM_LINE(wname, wevent) \
{ .id = snd_soc_dapm_line, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
以上这些widget分别对应信号发生器，输入引脚，输出引脚，麦克风，耳机，扬声器，线路输入接口。其中的reg字段被设置为SND_SOC_NOPM（-1），表明这些widget是没有寄存器控制位来控制widget的电源状态的。麦克风，耳机，扬声器，线路输入接口这几种widget，还可以定义一个dapm事件回调函数wevent，从event_flags字段的设置可以看出，他们只会响应SND_SOC_DAPM_POST_PMU（上电后）和SND_SOC_DAPM_PMD（下电前）事件，这几个widget通常会在machine驱动中定义，而SND_SOC_DAPM_INPUT和SND_SOC_DAPM_OUTPUT则用来定义codec芯片的输出输入脚，通常在codec驱动中定义，最后，在machine驱动中增加相应的route，把麦克风和耳机等widget与相应的codec输入输出引脚的widget连接起来
根据打印，先执行get,获取到值，1，再执行put，将值用于设置连接
将snd_soc_dapm_widget与snd_kcontrol_new连接
inymix 'QUAT_MI2S_RX Audio Mixer MultiMedia1' 1 <
[ 340.617577] msm_routing_get_audio_mixer: reg 1e shift 0 val 0
[ 340.622612] msm_pcm_routing_process_audio: reg 1e val 0 set 1
msm_pcm_routing_process_audio
snd_soc_dapm_mixer_update_power
soc_dpcm_runtime_update
dpcm_process_paths[[[[[[— Create any new FE <–> BE connections–dpcm_be_connect-> list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients);]]]]]]
dpcm_run_new_update
dpcm_run_update_startup
dpcm_be_dai_startup
soc_pcm_open
struct snd_soc_dapm_route {
const char *sink;
const char *control;
const char *source;
int (*connected)(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink);
};
注意该结构体的注释， sink 是目的部件， source 是源部件， control 是目的部件定义的
kcontrol ；通过 control 可以选择 source 作为 sink 的输入源
ink指向到达端widget的名字字符串，source指向起始端widget的名字字符串，control指向负责控制该连接所对应的kcontrol名字字符串，connected回调则定义了上一节所提到的自定义连接检查回调函数。该结构的意义很明显就是：source通过一个kcontrol，和sink连接在一起，现在是否处于连接状态，请调用connected回调函数检查。
struct snd_soc_dapm_path {
const char *name;
/* source (input) and sink (output) widgets */
struct snd_soc_dapm_widget *source;
struct snd_soc_dapm_widget *sink;
如上数据结构可以看出path是sink与source是两个widget
下面snd_soc_dapm_route中
{“QUAT_MI2S_RX Port Mixer”, “INTERNAL_BT_SCO_TX”,“INT_BT_SCO_TX”},
"QUAT_MI2S_RX Port Mixer"为widget的名字
"INT_BT_SCO_TX"为widget的名字,代表一个输入设备,数据流入此设备为in
SND_SOC_DAPM_AIF_IN("INT_BT_SCO_TX", "Internal BT-SCO Capture",
0, 0, 0, 0),
"INTERNAL_BT_SCO_TX"为负责控制该连接所对应的kcontrol名字字符串
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