diff options
author | Johns <johns98@gmx.net> | 2012-03-12 17:58:19 +0100 |
---|---|---|
committer | Johns <johns98@gmx.net> | 2012-03-12 17:58:19 +0100 |
commit | 7b570c507cf0634e1249f60e7d008ff43f0d264a (patch) | |
tree | 57ab7a357157e07a452c4e49f97b6d896171a4c3 /audio.c | |
parent | 09ba3e299321f49b1f330feb6778171a06968d73 (diff) | |
download | vdr-plugin-softhddevice-7b570c507cf0634e1249f60e7d008ff43f0d264a.tar.gz vdr-plugin-softhddevice-7b570c507cf0634e1249f60e7d008ff43f0d264a.tar.bz2 |
Cleanups.
Diffstat (limited to 'audio.c')
-rw-r--r-- | audio.c | 19 |
1 files changed, 9 insertions, 10 deletions
@@ -435,8 +435,7 @@ static void AlsaFlushBuffers(void) RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer)); state = snd_pcm_state(AlsaPCMHandle); - Debug(3, "audio/alsa: flush state %d - %s\n", state, - snd_pcm_state_name(state)); + Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state)); if (state != SND_PCM_STATE_OPEN) { if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) { Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err)); @@ -943,7 +942,7 @@ static int64_t AlsaGetDelay(void) pts = ((int64_t) delay * 90 * 1000) / AudioSampleRate; pts += ((int64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90 * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); - Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 " ms\n", + Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(AlsaRingBuffer), pts / 90); return pts; @@ -1147,7 +1146,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3) // update buffer snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size); - Info(_("audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n"), + Debug(3, "audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n", buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle, buffer_size) * 1000 / (AudioSampleRate * AudioChannels * AudioBytesProSample), period_size, @@ -1172,7 +1171,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3) if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) { AlsaStartThreshold = RingBufferFreeBytes(AlsaRingBuffer); } - Info(_("audio/alsa: delay %u ms\n"), (AlsaStartThreshold * 1000) + Info(_("audio/alsa: delay %ums\n"), (AlsaStartThreshold * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); return ret; @@ -1450,7 +1449,7 @@ static void OssEnqueue(const void *samples, int count) uint32_t tick; tick = GetMsTicks(); - Debug(4, "audio/oss: %4d %d ms\n", count, tick - last_tick); + Debug(4, "audio/oss: %4d %dms\n", count, tick - last_tick); last_tick = tick; #endif @@ -1756,7 +1755,7 @@ static int64_t OssGetDelay(void) pts = ((int64_t) (delay + RingBufferUsedBytes(OssRingBuffer)) * 90 * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); - Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 " ms\n", + Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(OssRingBuffer), pts / 90); return pts; @@ -1865,7 +1864,7 @@ static int OssSetup(int *freq, int *channels, int use_ac3) OssFragmentTime = (bi.fragsize * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); - Info(_("audio/oss: buffer size %d %dms, fragment size %d %dms\n"), + Debug(3, "audio/oss: buffer size %d %dms, fragment size %d %dms\n", bi.fragsize * bi.fragstotal, (bi.fragsize * bi.fragstotal * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample), bi.fragsize, OssFragmentTime); @@ -1890,7 +1889,7 @@ static int OssSetup(int *freq, int *channels, int use_ac3) OssStartThreshold = RingBufferFreeBytes(OssRingBuffer); } - Info(_("audio/oss: delay %u ms\n"), (OssStartThreshold * 1000) + Info(_("audio/oss: delay %ums\n"), (OssStartThreshold * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); return ret; @@ -2080,7 +2079,7 @@ static void *AudioPlayHandlerThread(void *dummy) // cond_wait can return, without signal! } while (!AudioRunning); - Debug(3, "audio: ----> %d ms start\n", (AudioUsedBytes() * 1000) + Debug(3, "audio: ----> %dms start\n", (AudioUsedBytes() * 1000) / (!AudioSampleRate + !AudioChannels + AudioSampleRate * AudioChannels * AudioBytesProSample)); |