Hi there,
when using the newly added feature for adjusting the output volume of
ALSA devices (see https://trac.pjsip.org/repos/ticket/1991), I noticed
a couple of problems:
a) The very first mixer is used always. Even if it is the wrong card!
The master mixer must be managed in a card specific manner.
b) The output_vol parameter is not used when calling open_playback()
and the capability is set.
c) The output volume capability is not reported to via the
alsa_factory_get_dev_info() function.
The attached patch fixes all of the mentioned problems and includes
some small other improvements (e.g. parameter checks, removal of some
unused macros).
Please integrate. Thank a lot in advance.
Regrads,
Michael
Index: pjmedia/src/pjmedia-audiodev/alsa_dev.c
===================================================================
--- pjmedia/src/pjmedia-audiodev/alsa_dev.c (revision 5883)
+++ pjmedia/src/pjmedia-audiodev/alsa_dev.c (working copy)
@@ -39,12 +39,10 @@
#define THIS_FILE "alsa_dev.c"
#define ALSA_DEVICE_NAME "plughw:%d,%d"
-#define ALSASOUND_PLAYBACK 1
-#define ALSASOUND_CAPTURE 2
#define MAX_SOUND_CARDS 5
-#define MAX_SOUND_DEVICES_PER_CARD 5
#define MAX_DEVICES 32
-#define MAX_MIX_NAME_LEN 64
+#define MAX_MIX_NAME_LEN 64
+#define MAX_CARD_HCTL_LEN 8
/* Set to 1 to enable tracing */
#if 0
@@ -89,6 +87,15 @@
static pj_status_t alsa_stream_destroy(pjmedia_aud_stream *strm);
+typedef struct alsa_card_info
+{
+ unsigned id;
+ char name[PJMEDIA_AUD_DEV_INFO_NAME_LEN];
+ char hctl[MAX_CARD_HCTL_LEN];
+ char pb_mixer_name[MAX_MIX_NAME_LEN];
+} alsa_card_info;
+
+
struct alsa_factory
{
pjmedia_aud_dev_factory base;
@@ -96,9 +103,11 @@
pj_pool_t *pool;
pj_pool_t *base_pool;
+ unsigned card_cnt;
+ alsa_card_info cards[MAX_SOUND_CARDS];
+
unsigned dev_cnt;
pjmedia_aud_dev_info devs[MAX_DEVICES];
- char pb_mixer_name[MAX_MIX_NAME_LEN];
};
struct alsa_stream
@@ -110,7 +119,6 @@
struct alsa_factory *af;
void *user_data;
pjmedia_aud_param param; /* Running parameter */
- int rec_id; /* Capture device id */
int quit;
/* Playback */
@@ -151,6 +159,8 @@
&alsa_stream_destroy
};
+static void get_mixer_name(alsa_card_info *card);
+
static void null_alsa_error_handler (const char *file,
int line,
const char *function,
@@ -210,6 +220,38 @@
}
+static pj_status_t add_card (struct alsa_factory *af, unsigned id)
+{
+ alsa_card_info *card;
+ char *name;
+
+ if (af->card_cnt >= PJ_ARRAY_SIZE(af->cards))
+ return PJ_ETOOMANY;
+
+ card = &af->cards[af->card_cnt];
+
+ if (snd_card_get_name(id, &name) == 0) {
+ card->id = id;
+
+ strncpy(card->name, name, sizeof(card->name) - 1);
+ free(name);
+
+ snprintf(af->cards[af->card_cnt].hctl,
+ sizeof(af->cards[af->card_cnt].hctl), "hw:%d", id);
+
+ get_mixer_name(card);
+
+ ++af->card_cnt;
+
+ PJ_LOG (5,(THIS_FILE, "Added sound card %s", card->name));
+
+ return PJ_SUCCESS;
+ }
+
+ return PJMEDIA_EAUD_INVDEV;
+}
+
+
static pj_status_t add_dev (struct alsa_factory *af, const char *dev_name)
{
pjmedia_aud_dev_info *adi;
@@ -275,7 +317,8 @@
return PJ_SUCCESS;
}
-static void get_mixer_name(struct alsa_factory *af)
+
+static void get_mixer_name(alsa_card_info *card)
{
snd_mixer_t *handle;
snd_mixer_elem_t *elem;
@@ -283,7 +326,7 @@
if (snd_mixer_open(&handle, 0) < 0)
return;
- if (snd_mixer_attach(handle, "default") < 0) {
+ if (snd_mixer_attach(handle, card->hctl) < 0) {
snd_mixer_close(handle);
return;
}
@@ -304,9 +347,10 @@
if (snd_mixer_selem_is_active(elem) &&
snd_mixer_selem_has_playback_volume(elem))
{
- pj_ansi_strncpy(af->pb_mixer_name, snd_mixer_selem_get_name(elem),
- sizeof(af->pb_mixer_name));
- TRACE_((THIS_FILE, "Playback mixer name: %s", af->pb_mixer_name));
+ pj_ansi_strncpy(card->pb_mixer_name, snd_mixer_selem_get_name(elem),
+ sizeof(card->pb_mixer_name));
+ TRACE_((THIS_FILE, "Playback mixer name for card %d: %s", card->id,
+ card->mixer_name));
break;
}
}
@@ -367,6 +411,7 @@
static pj_status_t alsa_factory_refresh(pjmedia_aud_dev_factory *f)
{
struct alsa_factory *af = (struct alsa_factory*)f;
+ int card = -1;
char **hints, **n;
int err;
@@ -378,8 +423,22 @@
}
af->pool = pj_pool_create(af->pf, "alsa_aud", 256, 256, NULL);
+ af->card_cnt = 0;
af->dev_cnt = 0;
+ /* Enumerate sound cards */
+ err = snd_card_next(&card);
+
+ while ((err == 0) && (card != -1) ) {
+ if (add_card(af, card) == PJ_SUCCESS)
+ err = snd_card_next(&card);
+ else
+ err = -1;
+ }
+
+ if (err != 0)
+ return PJMEDIA_EAUD_SYSERR;
+
/* Enumerate sound devices */
err = snd_device_name_hint(-1, "pcm", (void***)&hints);
if (err != 0)
@@ -399,9 +458,6 @@
n++;
}
- /* Get the mixer name */
- get_mixer_name(af);
-
/* Install error handler after enumeration, otherwise we'll get many
* error messages about invalid card/device ID.
*/
@@ -434,10 +490,13 @@
pj_memcpy(info, &af->devs[index], sizeof(*info));
info->caps = PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY |
- PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY;
+ PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY |
+ PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING;
+
return PJ_SUCCESS;
}
+
/* API: create default parameter */
static pj_status_t alsa_factory_default_param(pjmedia_aud_dev_factory *f,
unsigned index,
@@ -531,7 +590,6 @@
}
-
static int ca_thread_func (void *arg)
{
struct alsa_stream* stream = (struct alsa_stream*) arg;
@@ -664,8 +722,15 @@
/* Set number of channels */
TRACE_((THIS_FILE, "open_playback: set channels: %d",
param->channel_count));
- snd_pcm_hw_params_set_channels (stream->pb_pcm, params,
- param->channel_count);
+ result = snd_pcm_hw_params_set_channels (stream->pb_pcm, params,
+ param->channel_count);
+ if (result < 0) {
+ PJ_LOG (3,(THIS_FILE, "Unable to set a channel count of %d for "
+ "playback device '%s'", param->channel_count,
+ stream->af->devs[param->play_id].name));
+ snd_pcm_close (stream->pb_pcm);
+ return PJMEDIA_EAUD_SYSERR;
+ }
/* Set clock rate */
rate = param->clock_rate;
@@ -710,6 +775,18 @@
return PJMEDIA_EAUD_SYSERR;
}
+ /* Apply output volume adjustment if provided */
+ if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING) {
+ pj_status_t status = alsa_stream_set_cap(stream,
+ PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING,
+ ¶m->output_vol);
+
+ if (status != PJ_SUCCESS)
+ PJ_LOG (4,(THIS_FILE, "Unable to set the output volume %d for "
+ "playback device '%s'", param->output_vol,
+ stream->af->devs[param->play_id].name));
+ }
+
PJ_LOG (5,(THIS_FILE, "Opened device alsa(%s) for playing, sample rate=%d"
", ch=%d, bits=%d, period size=%d frames, latency=%d ms",
stream->af->devs[param->play_id].name,
@@ -782,8 +859,15 @@
/* Set number of channels */
TRACE_((THIS_FILE, "open_capture: set channels: %d",
param->channel_count));
- snd_pcm_hw_params_set_channels (stream->ca_pcm, params,
- param->channel_count);
+ result = snd_pcm_hw_params_set_channels (stream->ca_pcm, params,
+ param->channel_count);
+ if (result < 0) {
+ PJ_LOG (3,(THIS_FILE, "Unable to set a channel count of %d for "
+ "capture device '%s'", param->channel_count,
+ stream->af->devs[param->rec_id].name));
+ snd_pcm_close (stream->ca_pcm);
+ return PJMEDIA_EAUD_SYSERR;
+ }
/* Set clock rate */
rate = param->clock_rate;
@@ -915,18 +999,21 @@
PJ_ASSERT_RETURN(s && pval, PJ_EINVAL);
- if (cap==PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY &&
- (stream->param.dir & PJMEDIA_DIR_CAPTURE))
- {
+ if ((cap == PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY) &&
+ (stream->param.dir & PJMEDIA_DIR_CAPTURE)) {
/* Recording latency */
*(unsigned*)pval = stream->param.input_latency_ms;
return PJ_SUCCESS;
- } else if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY &&
- (stream->param.dir & PJMEDIA_DIR_PLAYBACK))
- {
+ } else if ((cap == PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY) &&
+ (stream->param.dir & PJMEDIA_DIR_PLAYBACK)) {
/* Playback latency */
*(unsigned*)pval = stream->param.output_latency_ms;
return PJ_SUCCESS;
+ } else if ((cap == PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING) &&
+ (stream->param.dir & PJMEDIA_DIR_PLAYBACK)) {
+ /* Playback volume */
+ *(unsigned*)pval = stream->param.output_vol;
+ return PJ_SUCCESS;
} else {
return PJMEDIA_EAUD_INVCAP;
}
@@ -934,46 +1021,75 @@
/* API: set capability */
-static pj_status_t alsa_stream_set_cap(pjmedia_aud_stream *strm,
+static pj_status_t alsa_stream_set_cap(pjmedia_aud_stream *s,
pjmedia_aud_dev_cap cap,
const void *value)
{
- struct alsa_factory *af = ((struct alsa_stream*)strm)->af;
+ struct alsa_stream *stream = (struct alsa_stream*)s;
- if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING &&
- pj_ansi_strlen(af->pb_mixer_name))
+ PJ_ASSERT_RETURN(s && value, PJ_EINVAL);
+
+ /* only the output volume can be adjusted during runtime */
+ if (cap == PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING)
{
- pj_ssize_t min, max;
- snd_mixer_t *handle;
- snd_mixer_selem_id_t *sid;
- snd_mixer_elem_t* elem;
- unsigned vol = *(unsigned*)value;
+ struct alsa_factory *af = stream->af;
+ snd_pcm_info_t *info = NULL;
+ int card = -1;
- if (snd_mixer_open(&handle, 0) < 0)
- return PJMEDIA_EAUD_SYSERR;
+ /* get the corresponding card */
+ if (snd_pcm_info_malloc(&info) == 0) {
+ if (snd_pcm_info(stream->pb_pcm, info) == 0)
+ card = snd_pcm_info_get_card(info);
- if (snd_mixer_attach(handle, "default") < 0)
- return PJMEDIA_EAUD_SYSERR;
+ snd_pcm_info_free(info);
+ }
- if (snd_mixer_selem_register(handle, NULL, NULL) < 0)
+ /* card in range? */
+ if ((card < 0) || (card >= af->card_cnt))
return PJMEDIA_EAUD_SYSERR;
- if (snd_mixer_load(handle) < 0)
- return PJMEDIA_EAUD_SYSERR;
+ /* does this card have a mixer? */
+ if (pj_ansi_strlen(af->cards[card].pb_mixer_name))
+ {
+ pj_ssize_t min, max;
+ snd_mixer_t *handle;
+ snd_mixer_selem_id_t *sid;
+ snd_mixer_elem_t* elem;
+ unsigned vol = *(unsigned*)value;
- snd_mixer_selem_id_alloca(&sid);
- snd_mixer_selem_id_set_index(sid, 0);
- snd_mixer_selem_id_set_name(sid, af->pb_mixer_name);
- elem = snd_mixer_find_selem(handle, sid);
- if (!elem)
- return PJMEDIA_EAUD_SYSERR;
+ if (snd_mixer_open(&handle, 0) < 0)
+ return PJMEDIA_EAUD_SYSERR;
- snd_mixer_selem_get_playback_volume_range(elem, &min, &max);
- if (snd_mixer_selem_set_playback_volume_all(elem, vol * max / 100) < 0)
- return PJMEDIA_EAUD_SYSERR;
+ if (snd_mixer_attach(handle, af->cards[card].hctl) < 0)
+ return PJMEDIA_EAUD_SYSERR;
- snd_mixer_close(handle);
- return PJ_SUCCESS;
+ if (snd_mixer_selem_register(handle, NULL, NULL) < 0)
+ return PJMEDIA_EAUD_SYSERR;
+
+ if (snd_mixer_load(handle) < 0)
+ return PJMEDIA_EAUD_SYSERR;
+
+ snd_mixer_selem_id_alloca(&sid);
+ snd_mixer_selem_id_set_index(sid, 0);
+ snd_mixer_selem_id_set_name(sid, af->cards[card].pb_mixer_name);
+ elem = snd_mixer_find_selem(handle, sid);
+ if (!elem)
+ return PJMEDIA_EAUD_SYSERR;
+
+ snd_mixer_selem_get_playback_volume_range(elem, &min, &max);
+ if (snd_mixer_selem_set_playback_volume_all(elem,
+ vol * max / 100) < 0)
+ return PJMEDIA_EAUD_SYSERR;
+
+ /* store currently used value */
+ stream->param.output_vol = vol;
+
+ snd_mixer_close(handle);
+ return PJ_SUCCESS;
+ }
+ else
+ /* no mixer */
+ return PJMEDIA_EAUD_INVOP;
}
return PJMEDIA_EAUD_INVCAP;
@@ -986,6 +1102,8 @@
struct alsa_stream *stream = (struct alsa_stream*)s;
pj_status_t status = PJ_SUCCESS;
+ PJ_ASSERT_RETURN(s, PJ_EINVAL);
+
stream->quit = 0;
if (stream->param.dir & PJMEDIA_DIR_PLAYBACK) {
status = pj_thread_create (stream->pool,
@@ -1024,6 +1142,8 @@
{
struct alsa_stream *stream = (struct alsa_stream*)s;
+ PJ_ASSERT_RETURN(s, PJ_EINVAL);
+
stream->quit = 1;
if (stream->pb_thread) {
@@ -1054,11 +1174,13 @@
}
-
+/* API: destroy stream */
static pj_status_t alsa_stream_destroy (pjmedia_aud_stream *s)
{
struct alsa_stream *stream = (struct alsa_stream*)s;
+ PJ_ASSERT_RETURN(s, PJ_EINVAL);
+
alsa_stream_stop (s);
if (stream->param.dir & PJMEDIA_DIR_PLAYBACK) {
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