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Audio, ALSA: colorized warnings

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This commit is contained in:
Martin Pulec
2016-02-01 10:14:07 +01:00
parent 4fa90b6ac1
commit 07fa608501
2 changed files with 25 additions and 26 deletions
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8
src/audio/audio.cpp
View File

@@ -664,11 +664,11 @@ echo_play(s->echo_state, &pbuf_data.buffer);
if(audio_reconfigure(s, curr_desc.bps * 8,
curr_desc.ch_count,
curr_desc.sample_rate) != TRUE) {
fprintf(stderr, "Audio reconfiguration failed!");
log_msg(LOG_LEVEL_ERROR, "Audio reconfiguration failed!");
failed = true;
}
else {
fprintf(stderr, "Audio reconfiguration succeeded.");
log_msg(LOG_LEVEL_INFO, "Audio reconfiguration succeeded.");
device_desc = curr_desc;
rtp_flush_recv_buf(s->audio_network_device);
}
@@ -705,11 +705,11 @@ echo_play(s->echo_state, &pbuf_data.buffer);
if(audio_reconfigure(s, curr_desc.bps * 8,
curr_desc.ch_count,
curr_desc.sample_rate) != TRUE) {
fprintf(stderr, "Audio reconfiguration failed!");
log_msg(LOG_LEVEL_ERROR, "Audio reconfiguration failed!");
failed = true;
}
else {
fprintf(stderr, "Audio reconfiguration succeeded.");
log_msg(LOG_LEVEL_INFO, "Audio reconfiguration succeeded.");
device_desc = curr_desc;
rtp_flush_recv_buf(s->audio_network_device);
}

43
src/audio/playback/alsa.c
View File

@@ -3,7 +3,7 @@
* @author Martin Pulec <pulec@cesnet.cz>
*/
/*
* Copyright (c) 2011-2014 CESNET, z. s. p. o.
* Copyright (c) 2011-2016 CESNET, z. s. p. o.
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
@@ -63,7 +63,7 @@
#define BUFFER_MIN 41
#define BUFFER_MAX 200
#define MOD_NAME "[ALSA cap.] "
#define MOD_NAME "[ALSA play.] "
struct state_alsa_playback {
snd_pcm_t *handle;
@@ -220,7 +220,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
/* Fill it in with default values. */
rc = snd_pcm_hw_params_any(s->handle, params);
if (rc < 0) {
fprintf(stderr, "cannot obtain default hw parameters: %s\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot obtain default hw parameters: %s\n",
snd_strerror(rc));
return FALSE;
}
@@ -231,12 +231,12 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
rc = snd_pcm_hw_params_set_access(s->handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (rc < 0) {
fprintf(stderr, "cannot set interleaved hw access: %s\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set interleaved hw access: %s\n",
snd_strerror(rc));
rc = snd_pcm_hw_params_set_access(s->handle, params,
SND_PCM_ACCESS_RW_NONINTERLEAVED);
if (rc < 0) {
fprintf(stderr, "cannot set non-interleaved hw access: %s\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set non-interleaved hw access: %s\n",
snd_strerror(rc));
return FALSE;
}
@@ -246,7 +246,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
}
if (desc.bps > 4 || desc.bps < 1) {
log_msg(LOG_LEVEL_ERROR, "[ALSA playback] Unsupported BPS for audio (%d).\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "Unsupported BPS for audio (%d).\n",
desc.bps * 8);
return FALSE;
@@ -257,7 +257,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
rc = snd_pcm_hw_params_set_format(s->handle, params,
format);
if (rc < 0) {
fprintf(stderr, "cannot set format: %s\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set format: %s\n",
snd_strerror(rc));
return FALSE;
}
@@ -265,7 +265,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
/* Two channels (stereo) */
rc = snd_pcm_hw_params_set_channels(s->handle, params, desc.ch_count);
if (rc < 0) {
fprintf(stderr, "cannot set requested channel count: %s\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set requested channel count: %s\n",
snd_strerror(rc));
return FALSE;
}
@@ -275,7 +275,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
rc = snd_pcm_hw_params_set_rate_resample(s->handle,
params, val);
if(rc < 0) {
fprintf(stderr, "[ALSA play.] Warning: Unable to set resampling: %s\n",
log_msg(LOG_LEVEL_WARNING, MOD_NAME "Warning: Unable to set resampling: %s\n",
snd_strerror(rc));
}
@@ -286,7 +286,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
rc = snd_pcm_hw_params_set_rate_near(s->handle, params,
&val, &dir);
if (rc < 0) {
fprintf(stderr, "cannot set requested sample rate: %s\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set requested sample rate: %s\n",
snd_strerror(rc));
return FALSE;
}
@@ -300,7 +300,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
rc = snd_pcm_hw_params_set_period_time_min(s->handle,
params, &frames, &dir);
if (rc < 0) {
fprintf(stderr, "[ALSA play.] Warning: cannot set period time: %s\n",
log_msg(LOG_LEVEL_WARNING, MOD_NAME "Warning: cannot set period time: %s\n",
snd_strerror(rc));
}
@@ -309,7 +309,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
rc = snd_pcm_hw_params_set_buffer_time_min(s->handle, params,
&val, &dir);
if (rc < 0) {
fprintf(stderr, "Warining - unable to set minimal buffer size: %s\n",
log_msg(LOG_LEVEL_WARNING, MOD_NAME "Warning - unable to set minimal buffer size: %s\n",
snd_strerror(rc));
}
@@ -318,15 +318,14 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
rc = snd_pcm_hw_params_set_buffer_time_max(s->handle, params,
&val, &dir);
if (rc < 0) {
fprintf(stderr, "Warining - unable to set maximal buffer size: %s\n",
log_msg(LOG_LEVEL_WARNING, MOD_NAME "Warning - unable to set maximal buffer size: %s\n",
snd_strerror(rc));
}
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(s->handle, params);
if (rc < 0) {
fprintf(stderr,
"unable to set hw parameters: %s\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "unable to set hw parameters: %s\n",
snd_strerror(rc));
return FALSE;
}
@@ -494,14 +493,14 @@ static void * audio_play_alsa_init(const char *cfg)
}
if (rc < 0) {
fprintf(stderr, "unable to open pcm device: %s\n",
log_msg(LOG_LEVEL_ERROR, MOD_NAME "unable to open pcm device: %s\n",
snd_strerror(rc));
goto error;
}
rc = snd_pcm_nonblock(s->handle, 1);
if(rc < 0) {
fprintf(stderr, "ALSA Warning: Unable to set nonblock mode.\n");
log_msg(LOG_LEVEL_WARNING, MOD_NAME "Warning: Unable to set nonblock mode.\n");
}
return s;
@@ -546,7 +545,7 @@ static void audio_play_alsa_put_frame(void *state, struct audio_frame *frame)
rc = write_samples(s->handle, frame, frames, s->non_interleaved);
if (rc == -EPIPE) {
/* EPIPE means underrun */
fprintf(stderr, "underrun occurred\n");
log_msg(LOG_LEVEL_WARNING, MOD_NAME "underrun occurred\n");
snd_pcm_prepare(s->handle);
/* fill the stream with some sasmples */
for (double sec = 0.0; sec < BUFFER_MAX / 1000.0; sec += (double) frames / frame->sample_rate) {
@@ -556,16 +555,16 @@ static void audio_play_alsa_put_frame(void *state, struct audio_frame *frame)
}
int rc = write_samples(s->handle, frame, frames_to_write, s->non_interleaved);
if(rc < 0) {
fprintf(stderr, "error from writei: %s\n",
log_msg(LOG_LEVEL_WARNING, MOD_NAME "error from writei: %s\n",
snd_strerror(rc));
break;
}
}
} else if (rc < 0) {
fprintf(stderr, "error from writei: %s\n",
log_msg(LOG_LEVEL_WARNING, MOD_NAME "error from writei: %s\n",
snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr, "short write, written %d frames (overrun)\n", rc);
log_msg(LOG_LEVEL_WARNING, MOD_NAME "short write, written %d frames (overrun)\n", rc);
}
}
@@ -576,7 +575,7 @@ static void audio_play_alsa_done(void *state)
struct timeval t;
gettimeofday(&t, NULL);
printf("[ALSA play.] Played %lld samples in %f seconds (%f samples per second).\n",
log_msg(LOG_LEVEL_INFO, MOD_NAME "Played %lld samples in %f seconds (%f samples per second).\n",
s->played_samples, tv_diff(t, s->start_time),
s->played_samples / tv_diff(t, s->start_time));