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[PATCH 4/5] audio/alsa, audio/oss - Assign all log messages a log level
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From: |
Luke Yelavich |
|
Subject: |
[PATCH 4/5] audio/alsa, audio/oss - Assign all log messages a log level |
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Date: |
Tue, 23 Jun 2009 16:54:39 +1000 |
From: Luke Yelavich <address@hidden>
To: address@hidden
All log messages that are logged using the MSG macro now have a log level
associated with them. The log levels that are currently assigned may not be
appropriate, however that can be tweaked at a later date. The documentation
also needs updating to indicate that the log level extends to audio output
activity logging.
---
ChangeLog | 14 ++++++---
src/audio/alsa.c | 82 +++++++++++++++++++++++++++---------------------------
src/audio/oss.c | 34 +++++++++++-----------
3 files changed, 67 insertions(+), 63 deletions(-)
diff --git a/ChangeLog b/ChangeLog
index 83955c1..bfa4180 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -15,11 +15,15 @@
* src/modules/module_main.c (main): Add a call to do_loglevel to set
the log level for the module.
- * src/audio/alsa.c: Adjust the MSG macro to take the log level setting
- into account before logging a message.
-
- * src/audio/oss.c: Adjust the MSG macro to take the log level setting
- into account before logging a message.
+ * src/audio/alsa.c:
+ - Adjust the MSG macro to take the log level setting into account
+ before logging a message.
+ - Assign all log messages to a log level.
+
+ * src/audio/oss.c:
+ - Adjust the MSG macro to take the log level setting into account
+ before logging a message.
+ - Assign all log messages to a log level.
2009-06-22 Luke Yelavich <luke.yelavich at canonical.com>
diff --git a/src/audio/alsa.c b/src/audio/alsa.c
index a25cc8b..5091f68 100644
--- a/src/audio/alsa.c
+++ b/src/audio/alsa.c
@@ -99,7 +99,7 @@ xrun(AudioID *id)
if (id == NULL) return -1;
- MSG("WARNING: Entering XRUN handler");
+ MSG(1, "WARNING: Entering XRUN handler");
snd_pcm_status_alloca(&status);
if ((res = snd_pcm_status(id->alsa_pcm, status))<0) {
@@ -112,7 +112,7 @@ xrun(AudioID *id)
gettimeofday(&now, 0);
snd_pcm_status_get_trigger_tstamp(status, &tstamp);
timersub(&now, &tstamp, &diff);
- MSG("underrun!!! (at least %.3f ms long)",
+ MSG(1, "underrun!!! (at least %.3f ms long)",
diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
if ((res = snd_pcm_prepare(id->alsa_pcm)) < 0) {
ERR("xrun: prepare error: %s", snd_strerror(res));
@@ -134,7 +134,7 @@ suspend(AudioID *id)
{
int res;
- MSG("WARNING: Entering SUSPEND handler.");
+ MSG(1, "WARNING: Entering SUSPEND handler.");
if (id == NULL) return -1;
@@ -160,7 +160,7 @@ _alsa_open(AudioID *id)
{
int err;
- MSG("Opening ALSA device");
+ MSG(1, "Opening ALSA device");
fflush(stderr);
/* Open the device */
@@ -172,14 +172,14 @@ _alsa_open(AudioID *id)
/* Allocate space for hw_params (description of the sound parameters) */
/* Allocate space for sw_params (description of the sound parameters) */
- MSG("Allocating new sw_params structure");
+ MSG(2, "Allocating new sw_params structure");
if ((err = snd_pcm_sw_params_malloc (&id->alsa_sw_params)) < 0) {
ERR("Cannot allocate hardware parameter structure (%s)",
snd_strerror(err));
return -1;
}
- MSG("Opening ALSA device ... success");
+ MSG(1, "Opening ALSA device ... success");
return 0;
}
@@ -193,7 +193,7 @@ _alsa_close(AudioID *id)
{
int err;
- MSG("Closing ALSA device");
+ MSG(1, "Closing ALSA device");
if (id->alsa_opened == 0) return 0;
@@ -201,7 +201,7 @@ _alsa_close(AudioID *id)
id->alsa_opened = 0;
if ((err = snd_pcm_close (id->alsa_pcm)) < 0) {
- MSG("Cannot close ALSA device (%s)", snd_strerror (err));
+ MSG(2, "Cannot close ALSA device (%s)", snd_strerror (err));
return -1;
}
@@ -210,7 +210,7 @@ _alsa_close(AudioID *id)
free(id->alsa_poll_fds);
pthread_mutex_unlock(&id->alsa_pipe_mutex);
- MSG("Closing ALSA device ... success");
+ MSG(1, "Closing ALSA device ... success");
return 0;
}
@@ -243,7 +243,7 @@ alsa_open(AudioID *id, void **pars)
return -1;
}
- MSG("Opening ALSA sound output");
+ MSG(1, "Opening ALSA sound output");
id->alsa_device_name = strdup(pars[0]);
@@ -253,7 +253,7 @@ alsa_open(AudioID *id, void **pars)
return -1;
}
- MSG("Device '%s' initialized succesfully.", (char*) pars[0]);
+ MSG(1, "Device '%s' initialized succesfully.", (char*) pars[0]);
return 0;
}
@@ -269,7 +269,7 @@ alsa_close(AudioID *id)
ERR("Cannot close audio device");
return -1;
}
- MSG("ALSA closed.");
+ MSG(1, "ALSA closed.");
id = NULL;
@@ -299,7 +299,7 @@ int wait_for_poll(AudioID *id, struct pollfd *alsa_poll_fds,
descriptors*/
if (revents = id->alsa_poll_fds[count-1].revents){
if (revents & POLLIN){
- MSG("wait_for_poll: stop requested");
+ MSG(4, "wait_for_poll: stop requested");
return 1;
}
}
@@ -313,24 +313,24 @@ int wait_for_poll(AudioID *id, struct pollfd
*alsa_poll_fds,
if (SND_PCM_STATE_XRUN == state){
if (!draining){
- MSG("WARNING: Buffer underrun detected!");
+ MSG(1, "WARNING: Buffer underrun detected!");
if (xrun(id) != 0) return -1;
return 0;
}else{
- MSG("Poll: Playback terminated");
+ MSG(4, "Poll: Playback terminated");
return 0;
}
}
if (SND_PCM_STATE_SUSPENDED == state){
- MSG("WARNING: Suspend detected!");
+ MSG(1, "WARNING: Suspend detected!");
if (suspend(id) != 0) return -1;
return 0;
}
/* Check for errors */
if (revents & POLLERR) {
- MSG("wait_for_poll: poll revents says POLLERR");
+ MSG(4, "wait_for_poll: poll revents says POLLERR");
return -EIO;
}
@@ -395,7 +395,7 @@ alsa_play(AudioID *id, AudioTrack track)
pthread_mutex_lock(&id->alsa_pipe_mutex);
- MSG("Start of playback on ALSA");
+ MSG(2, "Start of playback on ALSA");
/* Is it not an empty track? */
/* Passing an empty track is not an error */
@@ -404,7 +404,7 @@ alsa_play(AudioID *id, AudioTrack track)
return 0;
}
/* Allocate space for hw_params (description of the sound parameters) */
- MSG("Allocating new hw_params structure");
+ MSG(2, "Allocating new hw_params structure");
if ((err = snd_pcm_hw_params_malloc (&id->alsa_hw_params)) < 0) {
ERR("Cannot allocate hardware parameter structure (%s)",
snd_strerror(err));
@@ -459,7 +459,7 @@ alsa_play(AudioID *id, AudioTrack track)
/* Report current state */
state = snd_pcm_state(id->alsa_pcm);
- MSG("PCM state before setting audio parameters: %s",
+ MSG(4, "PCM state before setting audio parameters: %s",
snd_pcm_state_name(state));
/* Choose the correct format */
@@ -482,7 +482,7 @@ alsa_play(AudioID *id, AudioTrack track)
}
/* Set access mode, bitrate, sample rate and channels */
- MSG("Setting access type to INTERLEAVED");
+ MSG(4, "Setting access type to INTERLEAVED");
if ((err = snd_pcm_hw_params_set_access (id->alsa_pcm,
id->alsa_hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED)
@@ -492,14 +492,14 @@ alsa_play(AudioID *id, AudioTrack track)
return -1;
}
- MSG("Setting sample format to %s", snd_pcm_format_name(format));
+ MSG(4, "Setting sample format to %s", snd_pcm_format_name(format));
if ((err = snd_pcm_hw_params_set_format (id->alsa_pcm, id->alsa_hw_params,
format)) < 0) {
ERR("Cannot set sample format (%s)",
snd_strerror (err));
return -1;
}
- MSG("Setting sample rate to %i", track.sample_rate);
+ MSG(4, "Setting sample rate to %i", track.sample_rate);
sr = track.sample_rate;
if ((err = snd_pcm_hw_params_set_rate_near (id->alsa_pcm,
id->alsa_hw_params,
&sr, 0)) < 0) {
@@ -509,17 +509,17 @@ alsa_play(AudioID *id, AudioTrack track)
return -1;
}
- MSG("Setting channel count to %i", track.num_channels);
+ MSG(4, "Setting channel count to %i", track.num_channels);
if ((err = snd_pcm_hw_params_set_channels (id->alsa_pcm,
id->alsa_hw_params,
track.num_channels)) < 0) {
- MSG("cannot set channel count (%s)",
+ MSG(4, "cannot set channel count (%s)",
snd_strerror (err));
return -1;
}
- MSG("Setting hardware parameters on the ALSA device");
+ MSG(4, "Setting hardware parameters on the ALSA device");
if ((err = snd_pcm_hw_params (id->alsa_pcm, id->alsa_hw_params)) < 0) {
- MSG("cannot set parameters (%s) state=%s",
+ MSG(4, "cannot set parameters (%s) state=%s",
snd_strerror (err),
snd_pcm_state_name(snd_pcm_state(id->alsa_pcm)));
return -1;
}
@@ -536,7 +536,7 @@ alsa_play(AudioID *id, AudioTrack track)
ERR("Unable to get buffer size for playback: %s\n", snd_strerror(err));
return -1;
}
- MSG("Buffer size on ALSA device is %d bytes", (int) id->alsa_buffer_size);
+ MSG(4, "Buffer size on ALSA device is %d bytes", (int)
id->alsa_buffer_size);
/* This is probably better left for the device driver to decide */
/* allow the transfer when at least period_size samples can be processed */
@@ -557,7 +557,7 @@ alsa_play(AudioID *id, AudioTrack track)
// MSG("silent samples = %i", silent_samples);
- MSG("Preparing device for playback");
+ MSG(4, "Preparing device for playback");
if ((err = snd_pcm_prepare (id->alsa_pcm)) < 0) {
ERR("Cannot prepare audio interface for playback (%s)",
snd_strerror (err));
@@ -567,10 +567,10 @@ alsa_play(AudioID *id, AudioTrack track)
/* Calculate space needed to round up to nearest period size. */
volume_size = bytes_per_sample*(track.num_samples + silent_samples);
- MSG("volume size = %i", (int) volume_size);
+ MSG(4, "volume size = %i", (int) volume_size);
/* Create a copy of track with adjusted volume. */
- MSG("Making copy of track and adjusting volume");
+ MSG(4, "Making copy of track and adjusting volume");
track_volume = track;
track_volume.samples = (short*) malloc(volume_size);
real_volume = ((float) id->volume + 100)/(float)200;
@@ -582,7 +582,7 @@ alsa_play(AudioID *id, AudioTrack track)
u_int8_t silent8;
/* Fill remaining space with silence */
- MSG("Filling with silence up to the period size, silent_samples=%d",
(int) silent_samples);
+ MSG(4, "Filling with silence up to the period size,
silent_samples=%d", (int) silent_samples);
/* TODO: This hangs. Why?
snd_pcm_format_set_silence(format,
track_volume.samples + (track.num_samples * bytes_per_sample),
silent_samples);
@@ -621,14 +621,14 @@ alsa_play(AudioID *id, AudioTrack track)
// MSG("Sent %d of %d remaining bytes", ret*bytes_per_sample,
num_bytes);
if (ret == -EAGAIN) {
- MSG("Warning: Forced wait!");
+ MSG(4, "Warning: Forced wait!");
snd_pcm_wait(id->alsa_pcm, 100);
} else if (ret == -EPIPE) {
if (xrun(id) != 0) ERROR_EXIT();
} else if (ret == -ESTRPIPE) {
if (suspend(id) != 0) ERROR_EXIT();
} else if (ret == -EBUSY){
- MSG("WARNING: sleeping while PCM BUSY");
+ MSG(4, "WARNING: sleeping while PCM BUSY");
usleep(100);
continue;
} else if (ret < 0) {
@@ -654,7 +654,7 @@ alsa_play(AudioID *id, AudioTrack track)
ERROR_EXIT();
}
else if (err == 1){
- MSG("Playback stopped");
+ MSG(4, "Playback stopped");
/* Drop the playback on the sound device (probably
still in progress up till now) */
@@ -673,7 +673,7 @@ alsa_play(AudioID *id, AudioTrack track)
/* Stop requests can be issued again */
}
- MSG("Draining...");
+ MSG(4, "Draining...");
/* We want to next "device ready" notification only after the buffer is
already empty */
@@ -694,7 +694,7 @@ alsa_play(AudioID *id, AudioTrack track)
ERR("Wait for poll() failed\n");
return -1;
} else if (err == 1){
- MSG("Playback stopped while draining");
+ MSG(4, "Playback stopped while draining");
/* Drop the playback on the sound device (probably
still in progress up till now) */
@@ -704,7 +704,7 @@ alsa_play(AudioID *id, AudioTrack track)
return -1;
}
}
- MSG("Draining terminated");
+ MSG(4, "Draining terminated");
terminate:
/* Terminating (successfully or after a stop) */
@@ -718,7 +718,7 @@ alsa_play(AudioID *id, AudioTrack track)
}
- MSG("Freeing HW parameters");
+ MSG(2, "Freeing HW parameters");
snd_pcm_hw_params_free(id->alsa_hw_params);
pthread_mutex_lock(&id->alsa_pipe_mutex);
@@ -729,7 +729,7 @@ alsa_play(AudioID *id, AudioTrack track)
xfree(id->alsa_poll_fds);
pthread_mutex_unlock(&id->alsa_pipe_mutex);
- MSG("End of playback on ALSA");
+ MSG(1, "End of playback on ALSA");
return 0;
}
@@ -745,7 +745,7 @@ alsa_stop(AudioID *id)
char buf;
int ret;
- MSG("STOP!");
+ MSG(1, "STOP!");
pthread_mutex_lock(&id->alsa_pipe_mutex);
if (id->alsa_opened){
diff --git a/src/audio/oss.c b/src/audio/oss.c
index a65f12e..b868557 100644
--- a/src/audio/oss.c
+++ b/src/audio/oss.c
@@ -81,7 +81,7 @@ xfree(void* p)
int
_oss_open(AudioID *id)
{
- MSG("_oss_open()")
+ MSG(1, "_oss_open()")
pthread_mutex_lock(&id->fd_mutex);
id->fd = open(id->device_name, O_WRONLY, 0);
@@ -100,7 +100,7 @@ _oss_open(AudioID *id)
int
_oss_close(AudioID *id)
{
- MSG("_oss_close()")
+ MSG(1, "_oss_close()")
if (id == NULL) return 0;
if (id->fd == 0) return 0;
@@ -228,7 +228,7 @@ oss_play(AudioID *id, AudioTrack track)
return -3;
}
if (channels != track.num_channels){
- MSG("Device doesn't support stereo sound.\n");
+ MSG(1, "Device doesn't support stereo sound.\n");
_oss_close(id);
return -4;
}
@@ -254,10 +254,10 @@ oss_play(AudioID *id, AudioTrack track)
/* Loop until all samples are played on the device.
In the meantime, wait in pthread_cond_timedwait for more data
or for interruption. */
- MSG("Starting playback");
+ MSG(4, "Starting playback");
output_samples = track_volume.samples;
num_bytes = track.num_samples*bytes_per_sample;
- MSG("bytes to play: %d, (%f secs)", num_bytes, (((float) (num_bytes)/2) /
(float) track.sample_rate));
+ MSG(4, "bytes to play: %d, (%f secs)", num_bytes, (((float) (num_bytes)/2)
/ (float) track.sample_rate));
while(num_bytes > 0) {
/* OSS doesn't support non-blocking write, so lets check how much data
@@ -272,12 +272,12 @@ oss_play(AudioID *id, AudioTrack track)
/* If there is not enough space for a single fragment, try later.
(This shouldn't happen, it has very bad effect on synchronization!)
*/
if (info.fragments == 0){
- MSG("WARNING: There is not enough space for a single fragment,
looping");
+ MSG(4, "WARNING: There is not enough space for a single fragment,
looping");
usleep (100);
continue;
}
- MSG("There is space for %d more fragments, fragment size is %d bytes",
+ MSG(4, "There is space for %d more fragments, fragment size is %d
bytes",
info.fragments, info.fragsize);
bytes = info.fragments * info.fragsize;
@@ -293,7 +293,7 @@ oss_play(AudioID *id, AudioTrack track)
num_bytes -= ret;
output_samples += ret/2;
- MSG("%d bytes written to OSS, %d remaining", ret, num_bytes);
+ MSG(4, "%d bytes written to OSS, %d remaining", ret, num_bytes);
/* If there is some more data that is less than a
full fragment, we need to write it immediatelly so
@@ -302,7 +302,7 @@ oss_play(AudioID *id, AudioTrack track)
&& (num_bytes < info.fragsize)
&& (bytes+num_bytes < info.bytes)){
- MSG("Writing the rest of the data (%d bytes) to OSS, not a full
fragment", num_bytes);
+ MSG(4, "Writing the rest of the data (%d bytes) to OSS, not a full
fragment", num_bytes);
ret2 = write(id->fd, output_samples, num_bytes);
num_bytes -= ret2;
@@ -325,14 +325,14 @@ oss_play(AudioID *id, AudioTrack track)
in the first pass through the while() loop. Then our timer
will be DELAY nsecs backwards.
*/
- MSG("Now we will try to wait");
+ MSG(4, "Now we will try to wait");
pthread_mutex_lock(&id->pt_mutex);
lenght = (((float) (ret)/2) / (float) track.sample_rate);
if (!delay){
delay = lenght>DELAY ? DELAY : lenght;
lenght -= delay;
}
- MSG("Wait for %f secs (begin: %f, delay: %f)", lenght, lenght+delay,
delay)
+ MSG(4, "Wait for %f secs (begin: %f, delay: %f)", lenght, lenght+delay,
delay)
gettimeofday(&now, NULL);
timeout.tv_sec = now.tv_sec + (int) lenght;
timeout.tv_nsec = now.tv_usec * 1000 + (lenght - (int) lenght) *
1000000000;
@@ -345,12 +345,12 @@ oss_play(AudioID *id, AudioTrack track)
// now.tv_sec, now.tv_usec*1000, timeout.tv_sec - now.tv_sec,
timeout.tv_nsec-now.tv_usec*1000);
r = pthread_cond_timedwait(&id->pt_cond, &id->pt_mutex, &timeout);
pthread_mutex_unlock(&id->pt_mutex);
- MSG("End of wait");
+ MSG(4, "End of wait");
/* The pthread_cond_timedwait was interrupted by change in the
condition variable? if so, terminate.*/
if (r != ETIMEDOUT){
- MSG("Playback stopped, %d", r);
+ MSG(4, "Playback stopped, %d", r);
break;
}
}
@@ -361,7 +361,7 @@ oss_play(AudioID *id, AudioTrack track)
/* Ugly hack: correct for the time we spend outside timing segments */
delay -= 0.05;
- MSG("Wait for the resting delay = %f secs", delay)
+ MSG(4, "Wait for the resting delay = %f secs", delay)
if ((delay > 0) && (r == ETIMEDOUT)){
pthread_mutex_lock(&id->pt_mutex);
gettimeofday(&now, NULL);
@@ -376,7 +376,7 @@ oss_play(AudioID *id, AudioTrack track)
r = pthread_cond_timedwait(&id->pt_cond, &id->pt_mutex, &timeout);
pthread_mutex_unlock(&id->pt_mutex);
}
- MSG("End of wait");
+ MSG(4, "End of wait");
if (track_volume.samples!=NULL) free(track_volume.samples);
@@ -387,7 +387,7 @@ oss_play(AudioID *id, AudioTrack track)
access the device. */
_oss_close(id);
- MSG("Device closed");
+ MSG(4, "Device closed");
return 0;
}
@@ -400,7 +400,7 @@ oss_stop(AudioID *id)
if (id == NULL) return 0;
- MSG("stop() called");
+ MSG(4, "stop() called");
/* Stop the playback on /dev/dsp */
pthread_mutex_lock(&id->fd_mutex);
--
1.6.3.1