Alsa项目的官方网址：http://www.alsa-project.org/
详细插件详细解释：https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
Alsa LIB API Reference：http://www.alsa-project.org/alsa-doc/alsa-lib/
配置文件的语法： http://www.alsa-project.org/alsa-doc/alsa-lib/conf.html
Asoundrc的官方说明文档：http://www.alsa-project.org/main/index.php/
Asoundrcarchlinux文档：https://wiki.archlinux.org/title/Advanced_Linux_Sound_Architecture/Configuration_examples
配置文件的位置
配置文件的位置是由Configure阶段的选项来决定的，不过多数时候，Alsa的配置文件位于：/usr/share/alsa目录下，主要配置文件为/usr/share/alsa/alsa.conf 其它文件是否需要，位置在哪，都是由alsa.conf来决定的。
通常会有/usr/share/alsa/card和/usr/share/alsa/pcm两个子目录，用于设置Card相关的参数，别名以及一些PCM默认设置。
此外，在alsa.conf中，通常还会引用 /etc/asound.conf和 ~/.asoundrc这两个配置文件，这两个文件通常是放置你个人需要特殊设置的相关参数。按照Alsa官方文档的说法，1.0.9版本以后，这两个文件就不再是必要的，甚至是不应该需要的。至少是不推荐使用吧。不过，对于我来说，在嵌入式系统中使用，为了简单和方便测试，恰恰是需要修改这两个文件
Alsa插件目录
/ # ls /usr/lib/alsa-lib/libasound_module_ctl_arcam_av.solibasound_module_ctl_oss.solibasound_module_pcm_oss.solibasound_module_pcm_upmix.solibasound_module_pcm_usb_stream.solibasound_module_pcm_vdownmix.solibasound_module_rate_speexrate.solibasound_module_rate_speexrate_best.solibasound_module_rate_speexrate_medium.so
Alsa.conf
Alsa.conf中主要的一些内容包括：用hook读取了/etc/asound.conf 和 ~/.asoundrc这两个配置文件：
@hooks [ { func load files [ "/etc/asound.conf" "~/.asoundrc" ] errors false }]
设置了default pcm的一些默认参数，如，默认使用Card 0 ，Device 0作为音频设备等等。
defaults.ctl.card 0defaults.pcm.card 0defaults.pcm.device 0defaults.pcm.subdevice -1defaults.pcm.nonblock 1defaults.pcm.ipc_key 5678293
设置了Alsa 内置的一些plugin的接口参数，例如file：
pcm.file { @args [ FILE FORMAT ] @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm null file $FILE format $FORMAT }
File plugin的作用是将PCM数据流存储到文件中。此外，通常alsa.conf还会载入/cards/aliases.conf ，设置一些声卡的别名等，这个我是不需要了。在aliases.conf 的结尾还有以下一段：
<confdir:pcm/default.conf> <confdir:pcm/dmix.conf> <confdir:pcm/dsnoop.conf>
用来读入/usr/share/alsa/pcm目录下所列的那3个文件分别设置 默认PCM设备的相关参数，dmix是用来实现播放时软件混音的内建plugin,dsnoop则是用来实现录音时多路分发的内建plugin。
一些配置和使用实例
使用蓝牙设备
- 在
/etc/asound.conf中添加下列一项用来使用蓝牙的A2DP设备
#device for bluetoothpcm.bluetooth{ type bluetooth device 00:02:5B:00:C1:A0}
然后调用 aplay –D bluetooth sample.wav播放。需要注意，为了使用该设备，你需要/usr/lib/alsa-lib/libasound_module_pcm_bluetooth.so 这一个蓝牙plugin的库文件。这是在Bluez相关的包里，和Alsa本身没有关系。从这里，我们也可以看出alsa的外部plugin和配置文件之间的名字关系规则：libasound_module_pcm_####.so 这里的#### 就是你再conf文件中pcm.xxxx 里所写的名字。
使用指定节点
在我的板子上，Buildin的Audio硬件在Alsa子系统中实现了两个硬件通道，一个是HIFI通道，另一个是语音通道，所以我添加了如下配置：
#device for voice channelpcm.voice{ type plug slave{ pcm "hw:0,1" }}
通过语音通道播放声音的调用的方式： aplay –D voice sample.wav这样的写法说明我通过plug这plugin对音频数据进行自动的采样率，通道等调整后，将数据送到我的第0个card的序号为1的device上。实际上，如果不写上述配置文件，用aplay -D "plug:SLAVE='hw:0,1'" sample.wav也可以得到同样的结果。
Hifi通道播放声音直接使用aplay sample.wav即可 也就是aplay –D default sample.wav
S16_LE转换S32_LE
通过arecord -D ref -c2 -r32000 -fS32_LE /data/rec.wav命令可以采集到S32_LE数据，hw:0,0只支持format S16_LE，slave指定使用S16_LE采集（必须指定否则会根据命令行参数使用S32_LE进行采集），通过plugin转成S32_LE
pcm.ref { type plug slave { pcm "hw:0,0" format S16_LE }}
多通道组合虚拟声卡
使用aplay -Dhw:0,0 /data/cn_32k.wav &和arecord -Dpcm.mu -f S32_LE -r 32000 -c 8 /data/rec_8ch.wav 可以将回采信号和mic array信号复合到一个声卡中，并以32bit进行采集，回采信号不支持32bit，使用refplugin转到了32bit。mic array使用32bit采集。通过arecord -Dpcm.mu -f S32_LE -r 32000 -c 8 /data/rec_8ch.wav -v可以清晰看到pipeline
pcm.mu { type multi slaves.a.pcm "hw:1,0" slaves.b.pcm ref slaves.a.channels 8 slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave a bindings.2.channel 2 bindings.3.slave a bindings.3.channel 3 bindings.4.slave a bindings.4.channel 4 bindings.5.slave a bindings.5.channel 5 bindings.6.slave b bindings.6.channel 0 bindings.7.slave b bindings.7.channel 1}# Sends to the two dmix interfacespcm.quad { type multi# Necessary to have both slaves be dmix; both as hw doesn't give errors, but wouldn't slaves.a.pcm "dmixerout" slaves.a.channels 2 slaves.b.pcm "dmixerloop" slaves.b.channels 2 bindings { 0 { slave a; channel 0; } 1 { slave a; channel 1; } 2 { slave b; channel 0; } 3 { slave b; channel 1; } }}
softvol：
实现软件调节音量
pcm.softvol { type softvol slave.pcm "playback" control { name "playback_softvol" card 2 } min_dB -40.0 max_dB 0.0 resolution 100}
创建了一个softvolPCM设备，其音量由名为playback_softvol的新音量控件控制音量改变的音频流将被传送到playback设备。由于该插件没有任何变化，因此新设备的音量，采样格式，采样率 和通道数与从设备的值相同。
- 首次使用新定义的设备只有使用一次新PCM设备后，才会出现
playback_softvol控件，同时也可以通过alsamixer amixer调节音量
aplay -D softvol /data/test.wavamixer contents -c2amixer cset -c2 numid=53,iface=MIXER,name='playback_softvol' 10
dmix插件：
dmix插件只能用于播放aplay -D real_playback sample.wav &你可以通过多次调用上述命令来测试多个音频数据的混音。也可以使用aplay -D dmix sample.wav &调用内置dmix
pcm.real_playback { type dmix ipc_key 5978293# must be unique for all dmix plugins!!!! ipc_key_add_uid yes slave { pcm "hw:0,0" channels 2 rate 48000 period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 }}
- Dump音频数据：
aplay -D "plug:'file:FILE=/tmp/dump.bin'" sample.wav
定义一组输入输出设备
pcm.uac { type asym playback.pcm { type plug slave.pcm "uac_playback" } capture.pcm { type plug slave.pcm "uac_capture" }}
DownMix 插件
使用-v 可以看到详细的Transformation table
pcm.uac_capture_mono { type vdownmix slave.pcm "uac_capture" slave.channels 2 ttable { 0.0 0.5 0.1 0.5 }}/ # arecord -D uac_capture_mono -c1 -r48000 -fS16_LE /data/rec.wav -vRecording WAVE '/data/rec.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, MonoRoute conversion PCM Transformation table: 0 <- 0*0.5 + 1*0.5 1 <- 0*0.5Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 48000
Route 插件
arecord
与downmix类似将2通道音频downmix到1通道
pcm.uac_capture_mono { type route slave.pcm "uac_capture" slave.channels 2 ttable { 0.0 0.5 0.1 0.5 }} ttable { master.slave gain master.slave gain }-v 可以查看详细转换关系：Slave #1: Route conversion PCM Transformation table: 0 <- 0*0.5 + 1*0.5
aplay
pcm.uac_playback_mono { type route slave.pcm "uac_playback" slave.channels 2 ttable { 0.0 1 0.1 1 }}Slave #0: Route conversion PCM Transformation table: 0 <- 0 * 1 1 <- 0 * 1
route plugin将2ch数据映射输出到4ch的multi插件中
# Duplicates to quad, use this to output to loopback & externalpcm.stereo2quad { type route slave.pcm "quad" # ttable.A.B G # where A - master channel # B - slave channel # G - volume gain (1.0 = original) ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1}
pcm.sub0_route { type route slave.pcm sub0_multi ttable [ [1 0 1 0] # ch0 映射到 ch0/ch2 [0 1 0 1] # ch1 映射到 ch1/ch3 ]}pcm.sub0_multi { type multi slaves.a.pcm main_proxy slaves.b.pcm sub0_dst slaves.a.channels 2 slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1}
dsnoop 插件
dsnoop插件用于录音场景，将一个捕获流拆分为多个捕获流。它的工作方式与dmix插件相反，多个客户端同时读取共享捕获缓冲区。以下参数的含义与dmix插件几乎相同。
pcm.cap_share { type dsnoop ipc_key 112233 slave { pcm "hw:0,5" channels 8 rate 32000 }}
resample 插件
# defaults.pcm.rate_converter "speexrate_medium"pcm.uac_playback { type rate # 定义使用采样率转换插件 slave { pcm "hw:2,0" # 硬件设备 rate 48000 # 表示从src_rate转成48k最终播放出去 }}
需要编译alsa-plugins，
ls /usr/lib64/alsa-lib/libasound_module_libasound_module_ctl_arcam_av.solibasound_module_ctl_oss.solibasound_module_pcm_oss.solibasound_module_pcm_upmix.solibasound_module_pcm_usb_stream.solibasound_module_pcm_vdownmix.solibasound_module_rate_speexrate.solibasound_module_rate_speexrate_best.solibasound_module_rate_speexrate_medium.so
另外card_no可以使用card_name替换
pcm.wireless_playback { type rate slave { pcm "hw:u2wl,0" rate 48000 }}pcm.wireless_capture { type rate slave { pcm "hw:u2wl,0" rate 48000 }}pcm.softvol { type softvol slave.pcm "playback" control { name "playback_softvol" card rockchipak7755 } min_dB -50.0 max_dB 0.0 resolution 100}
获取card***_no, card_name方式***
arecord -l
aplay -l
cat /proc/asound/cards
# cat /proc/asound/cards 0 [rockchiprk3308a]: rockchip_rk3308 - rockchip,rk3308-acodec rockchip,rk3308-acodec 1 [rockchippdmmica]: rockchip_pdm-mi - rockchip,pdm-mic-array rockchip,pdm-mic-array 2 [u2wl ]: u2-wl - u2-wl u2-wl 3 [rockchipak7755 ]: rockchip_ak7755 - rockchip,ak7755 rockchip,ak7755 4 [UAC1Gadget ]: UAC1_Gadget - UAC1_Gadget UAC1_Gadget 0 7 [Loopback ]: Loopback - Loopback Loopback 1
ALSA External PCM plugin SDK （ALSA扩展插件）
- alsa从
/usr/lib/alsa-lib/目录查找plugin，将libasound_module_pcm_fellow.so copy到/usr/lib/alsa-lib;并将/usr/lib/alsa-lib加到/etc/ld.config,否则有可能在运行时，找不到so - alsa plugin命名方式
libasound_module_pcm_XXX.so，名为XXX的插件
相关资料：
ALSA扩展插件分为2类：
External Filter plugin
https://www.cnblogs.com/fellow1988/p/12377713.html
- 滤波器类型插件，用于转换输入的PCM信号并输出到Slave。因此，该插件必须有Slave PCM 作为其输出。
- 该插件可以修改输入/输出PCM的
format和channel。但是 不能 修改rate，因为只是过滤型plugin - 在调用
snd_pcm_extplug_create()之前，必须在extplug记录中填写以下字段 ：version、name、callback。其他字段(private_data、parms)是可选的，应初始化为零. - 必须将常量
SND_PCM_EXTPLUG_VERSION传递给 alsa-lib 中的版本检查的版本字段version。 - 必须将非 NULL ASCII 字符串传递给名称字段
name。 callback字段包含此插件的回调函数表（定义为snd_pcm_extplug_callback_t）。private_data 用户私有数据，在回调中引用私有数据
#include <stdio.h>#include <string.h>#include <unistd.h>#include <alsa/asoundlib.h>#include <alsa/pcm_external.h>struct ctx_parms { int frames; int enable_dump; FILE *dump_fp; float gain;};typedef struct { snd_pcm_extplug_t ext; struct ctx_parms parms; short *buf; short *outbuf; unsigned int filled; unsigned int processed;}snd_pcm_fellowext_t;static inline void *area_addr(const snd_pcm_channel_area_t *area, snd_pcm_uframes_t offset){ unsigned int bitofs = area->first + area->step * offset; return (char *)area->addr + bitofs / 8;}static void process(snd_pcm_fellowext_t *ctx){ int frames = ctx->parms.frames; short *inbuf = ctx->buf; short *outbuf= ctx->outbuf; int channels= ctx->ext.channels; int ch_idx = 0, frame_idx = 0; for (frame_idx = 0; frame_idx < frames; frame_idx++) { for (ch_idx = 0; ch_idx < channels; ch_idx++) { outbuf[frame_idx * channels + ch_idx] = (short)((float)inbuf[frame_idx * channels + ch_idx] * ctx->parms.gain); } }}static snd_pcm_sframes_t fellowext_transfer(snd_pcm_extplug_t *ext, const snd_pcm_channel_area_t *dst_areas, snd_pcm_uframes_t dst_offset, snd_pcm_channel_area_t *src_areas, snd_pcm_uframes_t src_offset, snd_pcm_uframes_t size){ snd_pcm_fellowext_t *ctx = (snd_pcm_fellowext_t *)ext; short *src = area_addr(src_areas, src_offset); short *dst = area_addr(dst_areas, dst_offset); unsigned int count = size; int channels = ctx->ext.channels; int bytes_per_frame = 2 * channels; while (cout > 0) { unsigned int chunk; if (ctx->filled + count > ctx->parms.frames) chunk = ctx->parms.frames - ctx->filled; else chunk = count; if (ctx->processed) memcpy(dst, ctx->outbuf + ctx->filled * channels, chunk * bytes_per_frame); else memset(dst, 0, chunk * bytes_per_frame); if (ctx->parms.enable_dump) fwrite(dst, 1, chunk * bytes_per_frame, ctx->parms.dump_fp); dst += chunk * channels; memcpy(ctx->buf + ctx->filled * channels, src, chunk * bytes_per_frame); ctx->filled += chunk; if (ctx->filled == ctx->parms.frames) { process(ctx); ctx->processed = 1; } src += chunk * channels; count -= chunk; } return size;}static int fellowext_init(snd_pcm_extplug_t *ext){ snd_pcm_fellowext_t *ctx = (snd_pcm_fellowext_t *)ext; int channels = ctx->ext.channels; ctx->filled = 0; ctx->processed = 0; if (!ctx->buf) { ctx->buf = malloc(ctx->parms.frames * 2 * channels); if (!ctx->buf) return -ENOMEM; } memset (ctx->buf, 0, ctx->parms.frames * 2 * channels); if (!ctx->outbuf) { ctx->outbuf = malloc(ctx->parms.frames * 2 * channels); if (!ctx->outbuf) return -ENOMEM; } memset (ctx->outbuf, 0, ctx->parms.frames * 2 * channels); return 0;}static int fellowext_close(snd_pcm_extplug *ext){ snd_pcm_fellowext_t *ctx = (snd_pcm_fellowext_t *)ext; if (ctx->parms.enable_dump) fclose(ctx->parms.dump_fp); if (ctx->buf) { free(ctx->buf); ctx->buf = NULL; } if (ctx->outbuf) { free(ctx->outbuf); ctx->outbuf = NULL; } return 0;}static const snd_pcm_extplug_callback_t fellowext_callback = { .transfer = fellowext_transfer, .init = fellowext_init, .close = fellowext_close,};static int get_bool_parm(snd_config_t *n, const char *id, const char *str, int *val_ret){ int val; if (strcmp(id, str)) return 0; val = snd_config_get_bool(n); if (val < 0) { SNDERR("Invalid value for %s", id ); return val; } *val_ret = val; return 1;}static int get_int_parm(snd_config_t *n, const char *id, const char *str, int *val_ret){ long val; int err; if (strcmp(id, str)) return 0; err = snd_config_get_integer(n, &val); if (err < 0) { SNDERR("Invalid value for %s", id ); return err; } *val_ret = val; return 1;}static int get_float_parm(snd_config_t *n, const char *id, const char *str, float*val_ret){ double val; int err; if (strcmp(id, str)) return 0; err = snd_config_get_ireal(n, &val); if (err < 0) { SNDERR("Invalid value for %s", id ); return err; } *val_ret = val; return 1;}SND_PCM_PLUGIN_DEINE_FUNC(fellowext){ snd_config_iterator_t i, next; snd_pcm_fellowext_t *ctx; snd_config_t *sconf = NULL; int err; struct ctx_parms parms = { .frames = 512, .enable_dump = 0, .dump_fp = NULL, .gain = 0.5, }; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0) continue; if (strcmp(id, "slave") == 0) { sconf = n; continue; } err = get_int_parm(n, id, "frames", &parms.frames); if (err) goto ok; err = get_bool_parm(n, id, "enable_dump", &parms.enable_dump); if (err) goto ok; err = get_float_parm(n, id, "gain", &parms.gain); if (err) goto ok; SNDERR("Unknown field %s", id); return -EINVAL; ok: if (err < 0) return err; } if (!sconf) { SNDERR("No slave configuration for fellowext pcm"); } if (parms.enable_dump) parms.dump_fp = fopen("extplug.pcm", "wb"); ctx = calloc(1, sizeof(*ctx)) if (!ctx) return -ENOMEM; ctx->ext.version = SND_PCM_EXTPLUG_VERSION; ctx->ext.name = "Fellow Ext Plugin"; ctx->ext.callback = &fellowext_callback; ctx->ext.private_data = ctx; ctx->parms = parms; err = snd_pcm_extplug_create(&ctx->ext, name, root, conf, stream, mode); if (err < 0) { free(ctx); return err; } snd_pcm_explug_set_param(&ctx->ext, SND_PCM_EXTPLUG_HW_CHANNELS, 2); snd_pcm_explug_set_slave_param(&ctx->ext, SND_PCM_EXTPLUG_HW_CHANNELS, 2); snd_pcm_explug_set_param(&ctx->ext, SND_PCM_EXTPLUG_HW_FORMAT, SND_PCM_FORMAT_S16); snd_pcm_explug_set_slave_param(&ctx->ext, SND_PCM_EXTPLUG_HW_FORMAT, SND_PCM_FORMAT_S16); *pcmp = ctx->ext.pcm; return 0;}SND_PCM_PLUGIN_SYMBOL(fellowext);
External I/O plugin SDK
https://www.cnblogs.com/fellow1988/p/12375206.html
- I/O 类型插件，用作输入或输出的最终端点，即作为用户空间 PCM 程序。
通过代码设置alsa 音量
amixer -c2 cget name='playback_softvol’
#include <alsa/asoundlib.h>#include <iostream>int main() { snd_ctl_t *ctl; snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *control; snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); snd_ctl_elem_id_set_name(id, "playback_softvol"); if (snd_ctl_open(&ctl, "hw:2", 0) < 0) { std::cerr << "Cannot open control for card 2" << std::endl; return -1; } snd_ctl_elem_value_set_id(control, id); snd_ctl_elem_value_set_integer(control, 0, 50); // left channel snd_ctl_elem_value_set_integer(control, 1, 50); // right channel if (snd_ctl_elem_write(ctl, control) < 0) { std::cerr << "Cannot set playback_softvol volume" << std::endl; return -1; } snd_ctl_close(ctl); return 0;}
ALSA 异步IO
// AlsaDevice.h#ifndef ALSADEVICE_H#define ALSADEVICE_H#include <vector>#include "alsa/pcm.h"#include "AlsaDeviceParam.h"#include <sys/poll.h>#include <string>class AlsaDevice {private: snd_pcm_t *capture_handle; snd_pcm_t *playback_handle; std::vector<struct pollfd> ufds; int in_count; int out_count;public: AlsaDeviceParam alsaDeviceParam; AlsaDevice(const AlsaDeviceParam ¶m); ~AlsaDevice(); bool read(short *pcm, int len); bool write(short *pcm, int len); bool captureReady(); bool playbackReady(); void waitForPoll(); void start();};#endif
// AlsaDevice.cpp#include "AlsaDevice.h"#include "macrologger.h"#include <alsa/pcm.h>#include <iostream>#include <assert.h>#include <unistd.h>using namespace std;int xrun_recovery(snd_pcm_t *handle, int err) { if (err == -EPIPE) {/* under-run */ err = snd_pcm_prepare(handle); if (err < 0) LOG_ERROR("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) usleep(1000);/* wait until the suspend flag is released */if (err < 0) { err = snd_pcm_prepare(handle); if (err < 0) LOG_ERROR("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err)); } return 0; } return err;}AlsaDevice::AlsaDevice(const AlsaDeviceParam ¶m) { this->alsaDeviceParam = param; int dir; int err; snd_pcm_hw_params_t *hw_params; snd_pcm_sw_params_t *sw_params; snd_pcm_uframes_t buffer_size = 2048; static snd_output_t *jcd_out; err = snd_output_stdio_attach(&jcd_out, stderr, 0); if ((err = snd_pcm_open(&capture_handle, alsaDeviceParam.capture_dev.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) { LOG_ERROR("cannot open audio device %s (%s)", alsaDeviceParam.capture_dev.c_str(), snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) { LOG_ERROR("cannot allocate hardware parameter structure (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) { LOG_ERROR("cannot initialize hardware parameter structure (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { LOG_ERROR("cannot set access type (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params, alsaDeviceParam.capt_format)) < 0) { LOG_ERROR("cannot set sample format (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &alsaDeviceParam.capt_rate, 0)) < 0) { LOG_ERROR("cannot set sample alsaDeviceParam.capt_rate (%s)", snd_strerror(err)); assert(0); } LOG_INFO("capt_rate = %d", alsaDeviceParam.capt_rate); if ((err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, alsaDeviceParam.capt_nb_chan)) < 0) { LOG_ERROR("cannot set capt_nb_chan count (%s)", snd_strerror(err)); assert(0); } dir = 0; if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params, &alsaDeviceParam.capt_period_size, &dir)) < 0) { LOG_ERROR("cannot set period size (%s)", snd_strerror(err)); assert(0); } buffer_size = alsaDeviceParam.capt_period_size * alsaDeviceParam.capt_nb_periods; dir = 0; if ((err = snd_pcm_hw_params_set_buffer_size_near(capture_handle, hw_params, &buffer_size)) < 0) { LOG_ERROR("cannot set buffer time (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) { LOG_ERROR("cannot set capture parameters (%s)", snd_strerror(err)); assert(0); } snd_pcm_dump_setup(capture_handle, jcd_out); snd_pcm_hw_params_free(hw_params); if ((err = snd_pcm_sw_params_malloc(&sw_params)) < 0) { LOG_ERROR("cannot allocate software parameters structure (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_sw_params_current(capture_handle, sw_params)) < 0) { LOG_ERROR("cannot initialize software parameters structure (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_sw_params_set_avail_min(capture_handle, sw_params, alsaDeviceParam.capt_period_size)) < 0) { LOG_ERROR("cannot set minimum available count (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0) { LOG_ERROR("cannot set software parameters (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_prepare(capture_handle)) < 0) { LOG_ERROR("cannot prepare audio interface for use (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_open(&playback_handle, alsaDeviceParam.playback_dev.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) { LOG_ERROR("cannot open audio device %s (%s)", alsaDeviceParam.playback_dev.c_str(), snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) { LOG_ERROR("cannot allocate hardware parameter structure (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_any(playback_handle, hw_params)) < 0) { LOG_ERROR("cannot initialize hardware parameter structure (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { LOG_ERROR("cannot set access type (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_set_format(playback_handle, hw_params, alsaDeviceParam.play_format)) < 0) { LOG_ERROR("cannot set sample format (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &alsaDeviceParam.play_rate, 0)) < 0) { LOG_ERROR("cannot set sample alsaDeviceParam.play_rate (%s)", snd_strerror(err)); assert(0); } LOG_INFO("play_rate = %d", alsaDeviceParam.play_rate); if ((err = snd_pcm_hw_params_set_channels(playback_handle, hw_params, alsaDeviceParam.play_nb_chan)) < 0) { LOG_ERROR("cannot set alsaDeviceParam.play_channel count (%s)", snd_strerror(err)); assert(0); } dir = 0; if ((err = snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &alsaDeviceParam.play_period_size, &dir)) < 0) { LOG_ERROR("cannot set alsaDeviceParam.play_period size (%s)", snd_strerror(err)); assert(0); } buffer_size = alsaDeviceParam.play_period_size * alsaDeviceParam.play_nb_periods; dir = 0; if ((err = snd_pcm_hw_params_set_buffer_size_near(playback_handle, hw_params, &buffer_size)) < 0) { LOG_ERROR("cannot set buffer time (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_hw_params(playback_handle, hw_params)) < 0) { LOG_ERROR("cannot set playback parameters (%s)", snd_strerror(err)); assert(0); } snd_pcm_dump_setup(playback_handle, jcd_out); snd_pcm_hw_params_free(hw_params); if ((err = snd_pcm_sw_params_malloc(&sw_params)) < 0) { LOG_ERROR("cannot allocate software parameters structure (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_sw_params_current(playback_handle, sw_params)) < 0) { LOG_ERROR("cannot initialize software parameters structure (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_sw_params_set_avail_min(playback_handle, sw_params, alsaDeviceParam.play_period_size)) < 0) { LOG_ERROR("cannot set minimum available count (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_sw_params_set_start_threshold(playback_handle, sw_params, alsaDeviceParam.play_period_size)) < 0) { LOG_ERROR("cannot set start mode (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_sw_params(playback_handle, sw_params)) < 0) { LOG_ERROR("cannot set software parameters (%s)", snd_strerror(err)); assert(0); } if ((err = snd_pcm_prepare(playback_handle)) < 0) { LOG_ERROR("cannot prepare audio interface for use (%s)", snd_strerror(err)); assert(0); } in_count = snd_pcm_poll_descriptors_count(capture_handle); out_count = snd_pcm_poll_descriptors_count(playback_handle); int tot_count = in_count + out_count; ufds.resize(tot_count); if (snd_pcm_poll_descriptors(capture_handle, &ufds[0], in_count) != in_count) { LOG_ERROR("snd_pcm_poll_descriptors() doesn't match for capture"); } if (snd_pcm_poll_descriptors(playback_handle, &ufds[in_count], out_count) != out_count) { LOG_ERROR("snd_pcm_poll_descriptors() doesn't match for playback"); } LOG_INFO("devices are %d and %d, total is :%d", ufds[0].fd, ufds[1].fd, tot_count);}AlsaDevice::~AlsaDevice() {}bool AlsaDevice::read(short *pcm, int len) { int err; if ((err = snd_pcm_readi(capture_handle, pcm, len)) != len) { if (err < 0) {// LOG_ERROR("error %d, EPIPE = %d", err, EPIPE);if (err == -EPIPE) { LOG_ERROR("An overrun has occured, reseting capture"); } else { LOG_ERROR("read from audio interface failed (%s)", snd_strerror(err)); } if ((err = snd_pcm_prepare(capture_handle)) < 0) { LOG_ERROR("cannot prepare audio interface for use (%s)", snd_strerror(err)); } if ((err = snd_pcm_start(capture_handle)) < 0) { LOG_ERROR("cannot prepare audio interface for use (%s)", snd_strerror(err)); } } else { LOG_ERROR("Couldn't read as many samples as I wanted (%d instead of %d)", err, len); } return true; }// LOG_INFO("read: %d", err);return false;}bool AlsaDevice::write(short *pcm, int len) { int err; if ((err = snd_pcm_writei(playback_handle, pcm, len)) != len) { if (err < 0) { if (err == -EPIPE) { LOG_ERROR("An underrun has occured, reseting playback, len=%d", len); if (xrun_recovery(playback_handle, err) < 0) { LOG_ERROR("Write error: %s\n", snd_strerror(err)); } snd_pcm_writei(playback_handle, pcm, len); } else { LOG_ERROR("write to audio interface failed (%s)", snd_strerror(err)); } } else { LOG_ERROR("Couldn't write as many samples as I wanted (%d instead of %d)", err, len); } return true; }// LOG_INFO("wrote %d", err);return false;}bool AlsaDevice::captureReady() { unsigned short revents = 0; int err; if ((err = snd_pcm_poll_descriptors_revents(capture_handle, &ufds[0], in_count, &revents)) < 0) { LOG_ERROR("error in snd_pcm_poll_descriptors_revents for capture: %s", snd_strerror(err)); if (revents & POLLERR) { return false; } } return revents & POLLIN;}bool AlsaDevice::playbackReady() { unsigned short revents = 0; int err; if ((err = snd_pcm_poll_descriptors_revents(playback_handle, &ufds[in_count], out_count, &revents)) < 0) { LOG_ERROR("error in snd_pcm_poll_descriptors_revents for playback: %s", snd_strerror(err)); if (revents & POLLERR) { return false; } } return revents & POLLOUT;}void AlsaDevice::waitForPoll() { poll(&ufds[0], ufds.size(), -1);}void AlsaDevice::start() { int16_t lead_in = alsaDeviceParam.play_period_size; int16_t channels = alsaDeviceParam.play_nb_chan; snd_pcm_start(playback_handle); snd_pcm_start(capture_handle);}
linux & alsa info
https://magodo.github.io/alsa-pcm/
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