diff options
author | Mike Brady <mikebrady@eircom.net> | 2018-11-08 14:43:52 +0000 |
---|---|---|
committer | Mike Brady <mikebrady@eircom.net> | 2018-11-08 14:43:52 +0000 |
commit | e7f70fba328bdcc88994b1963c1e85ffaba29af7 (patch) | |
tree | 80a410aceeaa0d23a5aefe1d4f539f7e21934b5d /audio_alsa.c | |
parent | cfd10647aad65314cbbde1ef5d959a57221d3090 (diff) |
Tidy up with clang-format
Diffstat (limited to 'audio_alsa.c')
-rw-r--r-- | audio_alsa.c | 89 |
1 files changed, 46 insertions, 43 deletions
diff --git a/audio_alsa.c b/audio_alsa.c index 815f342..0a03957 100644 --- a/audio_alsa.c +++ b/audio_alsa.c @@ -198,7 +198,6 @@ void do_snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t *mix_elem, double v static int init(int argc, char **argv) { snd_output_stdio_attach(&output, stdout, 0); - // debug(2,"audio_alsa init called."); int response = 0; // this will be what we return to the caller. const char *str; @@ -507,7 +506,6 @@ static int init(int argc, char **argv) { static void deinit(void) { // debug(2,"audio_alsa deinit called."); stop(); - } int open_alsa_device(void) { @@ -543,7 +541,7 @@ int open_alsa_device(void) { snd_pcm_hw_params_alloca(&alsa_params); snd_pcm_sw_params_alloca(&alsa_swparams); - + ret = snd_pcm_hw_params_any(alsa_handle, alsa_params); if (ret < 0) { warn("audio_alsa: Broken configuration for device \"%s\": no configurations " @@ -698,7 +696,7 @@ int open_alsa_device(void) { snd_strerror(ret)); return -9; } - + ret = snd_pcm_sw_params_current(alsa_handle, alsa_swparams); if (ret < 0) { warn("audio_alsa: Unable to get current sw parameters for device \"%s\": %s.", alsa_out_dev, @@ -712,15 +710,15 @@ int open_alsa_device(void) { snd_strerror(ret)); return -11; } - + /* write the sw parameters */ ret = snd_pcm_sw_params(alsa_handle, alsa_swparams); if (ret < 0) { warn("audio_alsa: Unable to set software parameters of device: \"%s\": %s.", alsa_out_dev, - snd_strerror(ret)); + snd_strerror(ret)); return -12; } - + if (actual_buffer_length < config.audio_backend_buffer_desired_length + minimal_buffer_headroom) { /* // the dac buffer is too small, so let's try to set it @@ -748,7 +746,7 @@ int open_alsa_device(void) { "length (%ld) you have chosen.", actual_buffer_length, config.audio_backend_buffer_desired_length); } - + if (alsa_characteristics_already_listed == 0) { alsa_characteristics_already_listed = 1; int log_level = 2; // the level at which debug information should be output @@ -896,38 +894,40 @@ static void start(int i_sample_rate, int i_sample_format) { measurement_data_is_valid = 0; } -int my_snd_pcm_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) { - int ret; +int my_snd_pcm_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) { + int ret; snd_pcm_status_t *alsa_snd_pcm_status; snd_pcm_status_alloca(&alsa_snd_pcm_status); - struct timespec tn; // time now - snd_htimestamp_t update_timestamp; //actually a struct timespec + struct timespec tn; // time now + snd_htimestamp_t update_timestamp; // actually a struct timespec ret = snd_pcm_status(pcm, alsa_snd_pcm_status); if (ret) { *delayp = 0; - return ret; + return ret; } - + snd_pcm_state_t state = snd_pcm_status_get_state(alsa_snd_pcm_status); if (state != SND_PCM_STATE_RUNNING) { *delayp = 0; - return -EIO; // might be a better code than this... + return -EIO; // might be a better code than this... } - + clock_gettime(CLOCK_MONOTONIC, &tn); snd_pcm_status_get_htstamp(alsa_snd_pcm_status, &update_timestamp); - - uint64_t t1 = tn.tv_sec * (uint64_t)1000000000 + tn.tv_nsec; - uint64_t t2 = update_timestamp.tv_sec * (uint64_t)1000000000 + update_timestamp.tv_nsec; + + uint64_t t1 = tn.tv_sec * (uint64_t)1000000000 + tn.tv_nsec; + uint64_t t2 = update_timestamp.tv_sec * (uint64_t)1000000000 + update_timestamp.tv_nsec; uint64_t delta = t1 - t2; - - uint64_t frames_played_since_last_interrupt = ((uint64_t)desired_sample_rate * delta)/1000000000; + + uint64_t frames_played_since_last_interrupt = + ((uint64_t)desired_sample_rate * delta) / 1000000000; snd_pcm_sframes_t frames_played_since_last_interrupt_sized = frames_played_since_last_interrupt; - - *delayp = snd_pcm_status_get_delay(alsa_snd_pcm_status) - frames_played_since_last_interrupt_sized; - return 0; + + *delayp = + snd_pcm_status_get_delay(alsa_snd_pcm_status) - frames_played_since_last_interrupt_sized; + return 0; } int delay(long *the_delay) { @@ -947,9 +947,9 @@ int delay(long *the_delay) { debug(1, "Error %d in delay(): \"%s\". Delay reported is %d frames.", reply, snd_strerror(reply), *the_delay); derr = snd_pcm_recover(alsa_handle, reply, 1); - if (derr < 0) - warn("Error %d -- could not clear an error after attempting delay(): \"%s\".", derr, - snd_strerror(derr)); + if (derr < 0) + warn("Error %d -- could not clear an error after attempting delay(): \"%s\".", derr, + snd_strerror(derr)); frame_index = 0; measurement_data_is_valid = 0; @@ -977,9 +977,10 @@ int delay(long *the_delay) { if ((derr = snd_pcm_prepare(alsa_handle))) { debug(1, "Error preparing after delay error: \"%s\".", snd_strerror(derr)); derr = snd_pcm_recover(alsa_handle, derr, 1); - if (derr < 0) - warn("Error %d -- could not clear an error after attempting to recover following a delay(): \"%s\".", derr, - snd_strerror(derr)); + if (derr < 0) + warn("Error %d -- could not clear an error after attempting to recover following a " + "delay(): \"%s\".", + derr, snd_strerror(derr)); } } } @@ -1028,10 +1029,10 @@ static int play(void *buf, int samples) { if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_XRUN) { if ((err = snd_pcm_prepare(alsa_handle))) { debug(1, "Error preparing after underrun: \"%s\".", snd_strerror(err)); - err = snd_pcm_recover(alsa_handle, err, 1); - if (err < 0) - warn("Error %d -- could not clear an error after detecting underrun in play(): \"%s\".", err, - snd_strerror(err)); + err = snd_pcm_recover(alsa_handle, err, 1); + if (err < 0) + warn("Error %d -- could not clear an error after detecting underrun in play(): \"%s\".", + err, snd_strerror(err)); } frame_index = 0; // we'll be starting over measurement_data_is_valid = 0; @@ -1049,9 +1050,10 @@ static int play(void *buf, int samples) { debug(1, "Error %d writing %d samples in play(): \"%s\".", err, samples, snd_strerror(err)); err = snd_pcm_recover(alsa_handle, err, 1); - if (err < 0) - warn("Error %d -- could not clear an error after attempting to write %d samples in play(): \"%s\".", err, samples, - snd_strerror(err)); + if (err < 0) + warn("Error %d -- could not clear an error after attempting to write %d samples in " + "play(): \"%s\".", + err, samples, snd_strerror(err)); } if (frame_index == 0) { frames_sent_for_playing = samples; @@ -1068,9 +1070,9 @@ static int play(void *buf, int samples) { debug(1, "Error %d in delay in play(): \"%s\". Delay reported is %d frames.", err2, snd_strerror(err2), fl); err2 = snd_pcm_recover(alsa_handle, err2, 1); - if (err2 < 0) - warn("Error %d -- could not clear an error after checking delay in play(): \"%s\".", err2, - snd_strerror(err2)); + if (err2 < 0) + warn("Error %d -- could not clear an error after checking delay in play(): \"%s\".", + err2, snd_strerror(err2)); frame_index = 0; measurement_data_is_valid = 0; } else { @@ -1097,9 +1099,10 @@ static int play(void *buf, int samples) { if ((err = snd_pcm_prepare(alsa_handle))) { debug(1, "Error preparing after play error: \"%s\".", snd_strerror(err)); err2 = snd_pcm_recover(alsa_handle, err, 1); - if (err2 < 0) - warn("Error %d -- could not clear an error after reporting ALSA device in incorrect state for play: \"%s\".", err2, - snd_strerror(err2)); + if (err2 < 0) + warn("Error %d -- could not clear an error after reporting ALSA device in incorrect " + "state for play: \"%s\".", + err2, snd_strerror(err2)); } frame_index = 0; measurement_data_is_valid = 0; |