From edd36a3afce4ca3778461e61df64f6a79ba94079 Mon Sep 17 00:00:00 2001 From: wm4 Date: Thu, 22 Aug 2013 23:12:35 +0200 Subject: audio/out: do some mp_msg conversions Use the new MP_ macros for some AOs instead of mp_msg. Not all AOs are converted, and some only partially. In some cases, some additional cosmetic changes are made. --- audio/out/ao_alsa.c | 89 +++++++++++++++++++----------------------------- audio/out/ao_jack.c | 10 +++--- audio/out/ao_lavc.c | 74 ++++++++++++++++------------------------ audio/out/ao_openal.c | 6 ++-- audio/out/ao_pcm.c | 16 ++++----- audio/out/ao_portaudio.c | 11 +++--- audio/out/ao_pulse.c | 45 +++++++++++------------- audio/out/ao_sdl.c | 30 ++++++++-------- 8 files changed, 118 insertions(+), 163 deletions(-) (limited to 'audio') diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c index 63309c3dbd..2db5041b95 100644 --- a/audio/out/ao_alsa.c +++ b/audio/out/ao_alsa.c @@ -73,8 +73,7 @@ struct priv { #define CHECK_ALSA_ERROR(message) \ do { \ if (err < 0) { \ - mp_msg(MSGT_VO, MSGL_ERR, "[AO_ALSA] %s: %s\n", \ - (message), snd_strerror(err)); \ + MP_ERR(ao, "%s: %s\n", (message), snd_strerror(err)); \ goto alsa_error; \ } \ } while (0) @@ -94,10 +93,10 @@ static void alsa_error_handler(const char *file, int line, const char *function, va_end(va); if (err) { - mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n", + mp_msg(MSGT_AO, MSGL_ERR, "alsa-lib: %s:%i:(%s) %s: %s\n", file, line, function, tmp, snd_strerror(err)); } else { - mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n", + mp_msg(MSGT_AO, MSGL_ERR, "alsa-lib: %s:%i:(%s) %s\n", file, line, function, tmp); } } @@ -145,10 +144,9 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg) elem = snd_mixer_find_selem(handle, sid); if (!elem) { - mp_tmsg(MSGT_AO, MSGL_ERR, - "[AO_ALSA] Unable to find simple control '%s',%i.\n", - snd_mixer_selem_id_get_name(sid), - snd_mixer_selem_id_get_index(sid)); + MP_VERBOSE(ao, "Unable to find simple control '%s',%i.\n", + snd_mixer_selem_id_get_name(sid), + snd_mixer_selem_id_get_index(sid)); goto alsa_error; } @@ -164,15 +162,14 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg) err = snd_mixer_selem_set_playback_volume (elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol); CHECK_ALSA_ERROR("Error setting left channel"); - mp_msg(MSGT_AO, MSGL_DBG2, "left=%li, ", set_vol); + MP_DBG(ao, "left=%li, ", set_vol); set_vol = vol->right / f_multi + pmin + 0.5; err = snd_mixer_selem_set_playback_volume (elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol); CHECK_ALSA_ERROR("Error setting right channel"); - mp_msg(MSGT_AO, MSGL_DBG2, - "right=%li, pmin=%li, pmax=%li, mult=%f\n", + MP_DBG(ao, "right=%li, pmin=%li, pmax=%li, mult=%f\n", set_vol, pmin, pmax, f_multi); break; @@ -185,8 +182,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg) snd_mixer_selem_get_playback_volume (elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol); vol->right = (get_vol - pmin) * f_multi; - mp_msg(MSGT_AO, MSGL_DBG2, "left=%f, right=%f\n", vol->left, - vol->right); + MP_DBG(ao, "left=%f, right=%f\n", vol->left, vol->right); break; } case AOCONTROL_SET_MUTE: { @@ -303,9 +299,8 @@ static const char *select_chmap(struct ao *ao) } char *name = mp_chmap_to_str(&ao->channels); - mp_tmsg(MSGT_AO, MSGL_ERR, - "[AO_ALSA] channel layout %s (%d ch) not supported.\n", - name, ao->channels.num); + MP_ERR(ao, "channel layout %s (%d ch) not supported.\n", + name, ao->channels.num); talloc_free(name); return "default"; } @@ -370,9 +365,8 @@ static int init(struct ao *ao) struct priv *p = ao->priv; - mp_msg(MSGT_AO, MSGL_V, - "alsa-init: requested format: %d Hz, %d channels, %x\n", - ao->samplerate, ao->channels.num, ao->format); + MP_VERBOSE(ao, "requested format: %d Hz, %d channels, %x\n", + ao->samplerate, ao->channels.num, ao->format); p->prepause_frames = 0; p->delay_before_pause = 0; @@ -386,9 +380,8 @@ static int init(struct ao *ao) const char *device; if (AF_FORMAT_IS_IEC61937(ao->format)) { device = "iec958"; - mp_msg(MSGT_AO, MSGL_V, - "alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", - ao->channels.num); + MP_VERBOSE(ao, "playing AC3/iec61937/iec958, %i channels\n", + ao->channels.num); } else { device = select_chmap(ao); if (strcmp(device, "default") != 0 && ao->format == AF_FORMAT_FLOAT_NE) @@ -400,11 +393,11 @@ static int init(struct ao *ao) if (p->cfg_device && p->cfg_device[0]) device = p->cfg_device; - mp_msg(MSGT_AO, MSGL_V, "alsa-init: using device %s\n", device); + MP_VERBOSE(ao, "using device: %s\n", device); p->can_pause = 1; - mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version()); + MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version()); snd_lib_error_set_handler(alsa_error_handler); int open_mode = p->cfg_block ? 0 : SND_PCM_NONBLOCK; @@ -413,7 +406,7 @@ static int init(struct ao *ao) err = try_open_device(ao, device, open_mode, isac3); if (err < 0) { if (err != -EBUSY && !p->cfg_block) { - mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Open in nonblock-mode " + MP_WARN(ao, "Open in nonblock-mode " "failed, trying to open in block-mode.\n"); err = try_open_device(ao, device, 0, isac3); } @@ -422,11 +415,9 @@ static int init(struct ao *ao) err = snd_pcm_nonblock(p->alsa, 0); if (err < 0) { - mp_tmsg(MSGT_AO, MSGL_ERR, - "[AL_ALSA] Error setting block-mode %s.\n", - snd_strerror(err)); + MP_ERR(ao, "Error setting block-mode: %s.\n", snd_strerror(err)); } else { - mp_msg(MSGT_AO, MSGL_V, "alsa-init: pcm opened in blocking mode\n"); + MP_VERBOSE(ao, "pcm opened in blocking mode\n"); } snd_pcm_hw_params_t *alsa_hwparams; @@ -451,8 +442,8 @@ static int init(struct ao *ao) err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt); if (err < 0) { - mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Format %s is not supported " - "by hardware, trying default.\n", af_fmt2str_short(ao->format)); + MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n", + af_fmt2str_short(ao->format)); p->alsa_fmt = SND_PCM_FORMAT_S16_LE; if (AF_FORMAT_IS_AC3(ao->format)) ao->format = AF_FORMAT_AC3_LE; @@ -471,8 +462,7 @@ static int init(struct ao *ao) CHECK_ALSA_ERROR("Unable to set channels"); if (num_channels != ao->channels.num) { - mp_tmsg(MSGT_AO, MSGL_ERR, - "[AO_ALSA] Couldn't get requested number of channels.\n"); + MP_ERR(ao, "Couldn't get requested number of channels.\n"); mp_chmap_from_channels_alsa(&ao->channels, num_channels); } @@ -508,13 +498,12 @@ static int init(struct ao *ao) CHECK_ALSA_ERROR("Unable to get buffersize"); p->buffersize = bufsize * p->bytes_per_sample; - mp_msg(MSGT_AO, MSGL_V, "alsa-init: got buffersize=%i\n", - p->buffersize); + MP_VERBOSE(ao, "got buffersize=%i\n", p->buffersize); err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL); CHECK_ALSA_ERROR("Unable to get period size"); - mp_msg(MSGT_AO, MSGL_V, "alsa-init: got period size %li\n", chunk_size); + MP_VERBOSE(ao, "got period size %li\n", chunk_size); p->outburst = chunk_size * p->bytes_per_sample; /* setting software parameters */ @@ -546,10 +535,9 @@ static int init(struct ao *ao) p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams); - mp_msg(MSGT_AO, MSGL_V, - "alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n", - ao->samplerate, ao->channels.num, (int)p->bytes_per_sample, - p->buffersize, snd_pcm_format_description(p->alsa_fmt)); + MP_VERBOSE(ao, "opened: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n", + ao->samplerate, ao->channels.num, (int)p->bytes_per_sample, + p->buffersize, snd_pcm_format_description(p->alsa_fmt)); return 0; @@ -573,7 +561,7 @@ static void uninit(struct ao *ao, bool immed) err = snd_pcm_close(p->alsa); CHECK_ALSA_ERROR("pcm close error"); - mp_msg(MSGT_AO, MSGL_V, "alsa-uninit: pcm closed\n"); + MP_VERBOSE(ao, "uninit: pcm closed\n"); } alsa_error: @@ -590,8 +578,8 @@ static void audio_pause(struct ao *ao) p->delay_before_pause = get_delay(ao); err = snd_pcm_pause(p->alsa, 1); CHECK_ALSA_ERROR("pcm pause error"); - mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n"); } else { + MP_VERBOSE(ao, "pause not supported by hardware\n"); if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0 || p->prepause_frames < 0) p->prepause_frames = 0; @@ -610,16 +598,15 @@ static void audio_resume(struct ao *ao) int err; if (snd_pcm_state(p->alsa) == SND_PCM_STATE_SUSPENDED) { - mp_tmsg(MSGT_AO, MSGL_INFO, - "[AO_ALSA] Pcm in suspend mode, trying to resume.\n"); + MP_INFO(ao, "PCM in suspend mode, trying to resume.\n"); while ((err = snd_pcm_resume(p->alsa)) == -EAGAIN) sleep(1); } if (p->can_pause) { err = snd_pcm_pause(p->alsa, 0); CHECK_ALSA_ERROR("pcm resume error"); - mp_msg(MSGT_AO, MSGL_V, "alsa-resume: resume supported by hardware\n"); } else { + MP_VERBOSE(ao, "resume not supported by hardware\n"); err = snd_pcm_prepare(p->alsa); CHECK_ALSA_ERROR("pcm prepare error"); if (p->prepause_frames) { @@ -664,10 +651,8 @@ static int play(struct ao *ao, void *data, int len, int flags) len = len / p->outburst * p->outburst; num_frames = len / p->bytes_per_sample; - //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len); - if (!p->alsa) { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Device configuration error."); + MP_ERR(ao, "Device configuration error."); return 0; } @@ -681,16 +666,12 @@ static int play(struct ao *ao, void *data, int len, int flags) /* nothing to do */ res = 0; } else if (res == -ESTRPIPE) { /* suspend */ - mp_tmsg(MSGT_AO, MSGL_INFO, - "[AO_ALSA] Pcm in suspend mode, trying to resume.\n"); + MP_INFO(ao, "PCM in suspend mode, trying to resume.\n"); while ((res = snd_pcm_resume(p->alsa)) == -EAGAIN) sleep(1); } if (res < 0) { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Write error: %s\n", - snd_strerror(res)); - mp_tmsg(MSGT_AO, MSGL_INFO, - "[AO_ALSA] Trying to reset soundcard.\n"); + MP_ERR(ao, "Write error: %s\n", snd_strerror(res)); res = snd_pcm_prepare(p->alsa); int err = res; CHECK_ALSA_ERROR("pcm prepare error"); diff --git a/audio/out/ao_jack.c b/audio/out/ao_jack.c index 579dee2631..b1f105248f 100644 --- a/audio/out/ao_jack.c +++ b/audio/out/ao_jack.c @@ -190,7 +190,7 @@ static int init(struct ao *ao) open_options |= JackNoStartServer; p->client = jack_client_open(p->cfg_client_name, open_options, NULL); if (!p->client) { - mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] cannot open server\n"); + MP_FATAL(ao, "cannot open server\n"); goto err_out; } jack_set_process_callback(p->client, outputaudio, ao); @@ -201,7 +201,7 @@ static int init(struct ao *ao) port_flags |= JackPortIsPhysical; matching_ports = jack_get_ports(p->client, port_name, NULL, port_flags); if (!matching_ports || !matching_ports[0]) { - mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] no physical ports available\n"); + MP_FATAL(ao, "no physical ports available\n"); goto err_out; } i = 1; @@ -220,19 +220,19 @@ static int init(struct ao *ao) jack_port_register(p->client, pname, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); if (!p->ports[i]) { - mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] not enough ports available\n"); + MP_FATAL(ao, "not enough ports available\n"); goto err_out; } } if (jack_activate(p->client)) { - mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] activate failed\n"); + MP_FATAL(ao, "activate failed\n"); goto err_out; } for (i = 0; i < p->num_ports; i++) { if (jack_connect(p->client, jack_port_name(p->ports[i]), matching_ports[i])) { - mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] connecting failed\n"); + MP_FATAL(ao, "connecting failed\n"); goto err_out; } } diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c index 0631d54aea..bf42a75ca3 100644 --- a/audio/out/ao_lavc.c +++ b/audio/out/ao_lavc.c @@ -71,13 +71,7 @@ static int init(struct ao *ao) AVCodec *codec; if (!encode_lavc_available(ao->encode_lavc_ctx)) { - mp_msg(MSGT_ENCODE, MSGL_ERR, - "ao-lavc: the option --o (output file) must be specified\n"); - return -1; - } - - if (ac->stream) { - mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: rejecting reinitialization\n"); + MP_ERR(ao, "the option --o (output file) must be specified\n"); return -1; } @@ -85,7 +79,7 @@ static int init(struct ao *ao) AVMEDIA_TYPE_AUDIO); if (!ac->stream) { - mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: could not get a new audio stream\n"); + MP_ERR(ao, "could not get a new audio stream\n"); return -1; } @@ -243,8 +237,7 @@ out_takefirst: } if (!found_format && !found_planar_format) { // shouldn't happen - mp_msg(MSGT_ENCODE, MSGL_ERR, - "ao-lavc: sample format not found\n"); + MP_ERR(ao, "sample format not found\n"); } } @@ -285,8 +278,7 @@ out_takefirst: ao->priv = ac; if (ac->planarize) - mp_msg(MSGT_ENCODE, MSGL_WARN, - "ao-lavc: need to planarize audio data\n"); + MP_WARN(ao, "need to planarize audio data\n"); return 0; } @@ -311,8 +303,7 @@ static void uninit(struct ao *ao, bool cut_audio) struct encode_lavc_context *ectx = ao->encode_lavc_ctx; if (!encode_lavc_start(ectx)) { - mp_msg(MSGT_ENCODE, MSGL_WARN, - "ao-lavc: not even ready to encode audio at end -> dropped"); + MP_WARN(ao, "not even ready to encode audio at end -> dropped"); return; } @@ -329,8 +320,7 @@ static void uninit(struct ao *ao, bool cut_audio) ac->sample_size, ac->sample_padding); int written = play(ao, paddingbuf, ao->buffer.len + extralen, 0); if (written < ao->buffer.len) { - mp_msg(MSGT_ENCODE, MSGL_ERR, - "ao-lavc: did not write enough data at the end\n"); + MP_ERR(ao, "did not write enough data at the end\n"); } talloc_free(paddingbuf); ao->buffer.len = 0; @@ -392,7 +382,7 @@ static int encode(struct ao *ao, double apts, void *data) ac->stream->codec->sample_fmt, data, audiolen, 1)) { - mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: error filling\n"); + MP_ERR(ao, "error filling\n"); return -1; } @@ -408,9 +398,8 @@ static int encode(struct ao *ao, double apts, void *data) if (ac->lastpts != MP_NOPTS_VALUE && frame_pts <= ac->lastpts) { // this indicates broken video // (video pts failing to increase fast enough to match audio) - mp_msg(MSGT_ENCODE, MSGL_WARN, "ao-lavc: audio frame pts went backwards " - "(%d <- %d), autofixed\n", (int)frame->pts, - (int)ac->lastpts); + MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n", + (int)frame->pts, (int)ac->lastpts); frame_pts = ac->lastpts + 1; frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, ac->stream->codec->time_base); } @@ -436,17 +425,15 @@ static int encode(struct ao *ao, double apts, void *data) status = avcodec_encode_audio2(ac->stream->codec, &packet, NULL, &gotpacket); } - if(status) - { - mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: error encoding\n"); + if(status) { + MP_ERR(ao, "error encoding\n"); return -1; } if(!gotpacket) return 0; - mp_msg(MSGT_ENCODE, MSGL_DBG2, - "ao-lavc: got pts %f (playback time: %f); out size: %d\n", + MP_DBG(ao, "got pts %f (playback time: %f); out size: %d\n", apts, realapts, packet.size); encode_lavc_write_stats(ao->encode_lavc_ctx, ac->stream); @@ -455,7 +442,7 @@ static int encode(struct ao *ao, double apts, void *data) // Do we need this at all? Better be safe than sorry... if (packet.pts == AV_NOPTS_VALUE) { - mp_msg(MSGT_ENCODE, MSGL_WARN, "ao-lavc: encoder lost pts, why?\n"); + MP_WARN(ao, "encoder lost pts, why?\n"); if (ac->savepts != MP_NOPTS_VALUE) packet.pts = ac->savepts; } @@ -475,7 +462,7 @@ static int encode(struct ao *ao, double apts, void *data) ac->savepts = MP_NOPTS_VALUE; if (encode_lavc_write_frame(ao->encode_lavc_ctx, &packet) < 0) { - mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: error writing at %f %f/%f\n", + MP_ERR(ao, "error writing at %f %f/%f\n", realapts, (double) ac->stream->time_base.num, (double) ac->stream->time_base.den); return -1; @@ -501,13 +488,11 @@ static int play(struct ao *ao, void *data, int len, int flags) len /= ac->sample_size * ao->channels.num; if (!encode_lavc_start(ectx)) { - mp_msg(MSGT_ENCODE, MSGL_WARN, - "ao-lavc: not ready yet for encoding audio\n"); + MP_WARN(ao, "not ready yet for encoding audio\n"); return 0; } if (pts == MP_NOPTS_VALUE) { - mp_msg(MSGT_ENCODE, MSGL_WARN, - "ao-lavc: frame without pts, please report; synthesizing pts instead\n"); + MP_WARN(ao, "frame without pts, please report; synthesizing pts instead\n"); // synthesize pts from previous expected next pts pts = ac->expected_next_pts; } @@ -516,19 +501,21 @@ static int play(struct ao *ao, void *data, int len, int flags) //if (ac->stream->codec->time_base.num / ac->stream->codec->time_base.den >= ac->stream->time_base.num / ac->stream->time_base.den) if (ac->stream->codec->time_base.num * (double) ac->stream->time_base.den >= ac->stream->time_base.num * (double) ac->stream->codec->time_base.den) { - mp_msg(MSGT_ENCODE, MSGL_V, "ao-lavc: NOTE: using codec time base " - "(%d/%d) for pts adjustment; the stream base (%d/%d) is " - "not worse.\n", (int)ac->stream->codec->time_base.num, - (int)ac->stream->codec->time_base.den, (int)ac->stream->time_base.num, - (int)ac->stream->time_base.den); + MP_VERBOSE(ao, "NOTE: using codec time base (%d/%d) for pts " + "adjustment; the stream base (%d/%d) is not worse.\n", + (int)ac->stream->codec->time_base.num, + (int)ac->stream->codec->time_base.den, + (int)ac->stream->time_base.num, + (int)ac->stream->time_base.den); ac->worst_time_base = ac->stream->codec->time_base; ac->worst_time_base_is_stream = 0; } else { - mp_msg(MSGT_ENCODE, MSGL_WARN, "ao-lavc: NOTE: not using codec time " - "base (%d/%d) for pts adjustment; the stream base (%d/%d) " - "is worse.\n", (int)ac->stream->codec->time_base.num, - (int)ac->stream->codec->time_base.den, (int)ac->stream->time_base.num, - (int)ac->stream->time_base.den); + MP_WARN(ao, "NOTE: not using codec time base (%d/%d) for pts " + "adjustment; the stream base (%d/%d) is worse.\n", + (int)ac->stream->codec->time_base.num, + (int)ac->stream->codec->time_base.den, + (int)ac->stream->time_base.num, + (int)ac->stream->time_base.den); ac->worst_time_base = ac->stream->time_base; ac->worst_time_base_is_stream = 1; } @@ -596,7 +583,7 @@ static int play(struct ao *ao, void *data, int len, int flags) */ int finalbufpos = len - (len - bufpos) % ac->aframesize; if (finalbufpos < 0) { - mp_msg(MSGT_ENCODE, MSGL_WARN, "ao-lavc: cannot attain the " + MP_WARN(ao, "cannot attain the " "exact requested audio sync; shifting by %d frames\n", -finalbufpos); bufpos -= finalbufpos; @@ -611,8 +598,7 @@ static int play(struct ao *ao, void *data, int len, int flags) ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts; } else if (fabs(nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts) > 30) { - mp_msg(MSGT_ENCODE, MSGL_WARN, - "ao-lavc: detected an unexpected discontinuity (pts jumped by " + MP_WARN(ao, "detected an unexpected discontinuity (pts jumped by " "%f seconds)\n", nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts); ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts; diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c index 1f9115a471..599672658d 100644 --- a/audio/out/ao_openal.c +++ b/audio/out/ao_openal.c @@ -127,7 +127,7 @@ static int init(struct ao *ao) int i; struct priv *p = ao->priv; if (ao_data) { - mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Not reentrant!\n"); + MP_FATAL(ao, "Not reentrant!\n"); return -1; } ao_data = ao; @@ -145,13 +145,13 @@ static int init(struct ao *ao) speakers[i] = speaker_pos[n]; } if (speakers[i].id < 0) { - mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Unknown channel layout\n"); + MP_FATAL(ao, "Unknown channel layout\n"); goto err_out; } } dev = alcOpenDevice(p->cfg_device && p->cfg_device[0] ? p->cfg_device : NULL); if (!dev) { - mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] could not open device\n"); + MP_FATAL(ao, "could not open device\n"); goto err_out; } ctx = alcCreateContext(dev, attribs); diff --git a/audio/out/ao_pcm.c b/audio/out/ao_pcm.c index 207b75fddd..b1e3a79708 100644 --- a/audio/out/ao_pcm.c +++ b/audio/out/ao_pcm.c @@ -141,19 +141,16 @@ static int init(struct ao *ao) ao->bps = ao->channels.num * ao->samplerate * (af_fmt2bits(ao->format) / 8); - mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\n" - "PCM: Samplerate: %d Hz Channels: %d Format: %s\n", + MP_INFO(ao, "File: %s (%s)\nPCM: Samplerate: %d Hz Channels: %d Format: %s\n", priv->outputfilename, priv->waveheader ? "WAVE" : "RAW PCM", ao->samplerate, ao->channels.num, af_fmt2str_short(ao->format)); - mp_tmsg(MSGT_AO, MSGL_INFO, - "[AO PCM] Info: Faster dumping is achieved with -no-video\n" - "[AO PCM] Info: To write WAVE files use -ao pcm:waveheader (default).\n"); + MP_INFO(ao, "Info: Faster dumping is achieved with -no-video\n"); + MP_INFO(ao, "Info: To write WAVE files use -ao pcm:waveheader (default).\n"); priv->fp = fopen(priv->outputfilename, "wb"); if (!priv->fp) { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n", - priv->outputfilename); + MP_ERR(ao, "Failed to open %s for writing!\n", priv->outputfilename); return -1; } if (priv->waveheader) // Reserve space for wave header @@ -177,11 +174,10 @@ static void uninit(struct ao *ao, bool cut_audio) GetFileType((HANDLE)_get_osfhandle(_fileno(priv->fp))); #endif if (broken_seek || fseek(priv->fp, 0, SEEK_SET) != 0) - mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, " - "WAV size headers not updated!\n"); + MP_ERR(ao, "Could not seek to start, WAV size headers not updated!\n"); else { if (priv->data_length > 0xfffff000) { - mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for " + MP_ERR(ao, "File larger than allowed for " "WAV files, may play truncated!\n"); priv->data_length = 0xfffff000; } diff --git a/audio/out/ao_portaudio.c b/audio/out/ao_portaudio.c index b80549398f..c9f55646a3 100644 --- a/audio/out/ao_portaudio.c +++ b/audio/out/ao_portaudio.c @@ -68,11 +68,11 @@ static const struct format_map format_maps[] = { static bool check_pa_ret(int ret) { if (ret < 0) { - mp_msg(MSGT_AO, MSGL_ERR, "[portaudio] %s\n", + mp_msg(MSGT_AO, MSGL_ERR, "[ao/portaudio] %s\n", Pa_GetErrorText(ret)); if (ret == paUnanticipatedHostError) { const PaHostErrorInfo* hosterr = Pa_GetLastHostErrorInfo(); - mp_msg(MSGT_AO, MSGL_ERR, "[portaudio] Host error: %s\n", + mp_msg(MSGT_AO, MSGL_ERR, "[ao/portaudio] Host error: %s\n", hosterr->errorText); } return false; @@ -121,7 +121,7 @@ static int find_device(const char *name) } } if (found == paNoDevice && !help) - mp_msg(MSGT_AO, MSGL_WARN, "[portaudio] Device '%s' not found!\n", + mp_msg(MSGT_AO, MSGL_WARN, "[ao/portaudio] Device '%s' not found!\n", name); return found; } @@ -183,7 +183,7 @@ static int stream_callback(const void *input, res = paComplete; priv->play_remaining = false; } else { - mp_msg(MSGT_AO, MSGL_ERR, "[portaudio] Buffer underflow!\n"); + MP_ERR(ao, "Buffer underflow!\n"); } fill_silence(output, len_bytes); } @@ -253,8 +253,7 @@ static int init(struct ao *ao) fmt++; } if (!fmt->pa_format) { - mp_msg(MSGT_AO, MSGL_V, - "[portaudio] Unsupported format, using default.\n"); + MP_VERBOSE(ao, "Unsupported format, using default.\n"); fmt = format_maps; } diff --git a/audio/out/ao_pulse.c b/audio/out/ao_pulse.c index 0a7d5347ba..7124ea34b4 100644 --- a/audio/out/ao_pulse.c +++ b/audio/out/ao_pulse.c @@ -57,9 +57,9 @@ struct priv { char *cfg_sink; }; -#define GENERIC_ERR_MSG(ctx, str) \ - mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] "str": %s\n", \ - pa_strerror(pa_context_errno(ctx))) +#define GENERIC_ERR_MSG(str) \ + MP_ERR(ao, str": %s\n", \ + pa_strerror(pa_context_errno(((struct priv *)ao->priv)->context))) static void context_state_cb(pa_context *c, void *userdata) { @@ -247,20 +247,19 @@ static int init(struct ao *ao) * hangs somewhen. */ if (strncmp(version, "0.9.1", 5) == 0 && version[5] >= '1' && version[5] <= '4') { - mp_msg(MSGT_AO, MSGL_WARN, - "[pulse] working around probably broken pause functionality,\n" - " see http://www.pulseaudio.org/ticket/440\n"); + MP_WARN(ao, "working around probably broken pause functionality,\n" + " see http://www.pulseaudio.org/ticket/440\n"); priv->broken_pause = true; } if (!(priv->mainloop = pa_threaded_mainloop_new())) { - mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate main loop\n"); + MP_ERR(ao, "Failed to allocate main loop\n"); goto fail; } if (!(priv->context = pa_context_new(pa_threaded_mainloop_get_api( priv->mainloop), PULSE_CLIENT_NAME))) { - mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate context\n"); + MP_ERR(ao, "Failed to allocate context\n"); goto fail; } @@ -286,8 +285,7 @@ static int init(struct ao *ao) const struct format_map *fmt_map = format_maps; while (fmt_map->mp_format != ao->format) { if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) { - mp_msg(MSGT_AO, MSGL_V, - "AO: [pulse] Unsupported format, using default\n"); + MP_VERBOSE(ao, "Unsupported format, using default\n"); fmt_map = format_maps; break; } @@ -297,7 +295,7 @@ static int init(struct ao *ao) ss.format = fmt_map->pa_format; if (!pa_sample_spec_valid(&ss)) { - mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Invalid sample spec\n"); + MP_ERR(ao, "Invalid sample spec\n"); goto fail; } @@ -342,7 +340,7 @@ fail: if (priv->context) { if (!(pa_context_errno(priv->context) == PA_ERR_CONNECTIONREFUSED && ao->probing)) - GENERIC_ERR_MSG(priv->context, "Init failed"); + GENERIC_ERR_MSG("Init failed"); } uninit(ao, true); return -1; @@ -355,7 +353,7 @@ static void cork(struct ao *ao, bool pause) priv->retval = 0; if (!waitop(priv, pa_stream_cork(priv->stream, pause, success_cb, ao)) || !priv->retval) - GENERIC_ERR_MSG(priv->context, "pa_stream_cork() failed"); + GENERIC_ERR_MSG("pa_stream_cork() failed"); } // Play the specified data to the pulseaudio server @@ -365,7 +363,7 @@ static int play(struct ao *ao, void *data, int len, int flags) pa_threaded_mainloop_lock(priv->mainloop); if (pa_stream_write(priv->stream, data, len, NULL, 0, PA_SEEK_RELATIVE) < 0) { - GENERIC_ERR_MSG(priv->context, "pa_stream_write() failed"); + GENERIC_ERR_MSG("pa_stream_write() failed"); len = -1; } if (flags & AOPLAY_FINAL_CHUNK) { @@ -387,7 +385,7 @@ static void reset(struct ao *ao) priv->retval = 0; if (!waitop(priv, pa_stream_flush(priv->stream, success_cb, ao)) || !priv->retval) - GENERIC_ERR_MSG(priv->context, "pa_stream_flush() failed"); + GENERIC_ERR_MSG("pa_stream_flush() failed"); cork(ao, false); } @@ -438,20 +436,20 @@ static float get_delay(struct ao *ao) struct priv *priv = ao->priv; pa_threaded_mainloop_lock(priv->mainloop); if (!waitop(priv, pa_stream_update_timing_info(priv->stream, NULL, NULL))) { - GENERIC_ERR_MSG(priv->context, "pa_stream_update_timing_info() failed"); + GENERIC_ERR_MSG("pa_stream_update_timing_info() failed"); return 0; } pa_threaded_mainloop_lock(priv->mainloop); const pa_timing_info *ti = pa_stream_get_timing_info(priv->stream); if (!ti) { pa_threaded_mainloop_unlock(priv->mainloop); - GENERIC_ERR_MSG(priv->context, "pa_stream_get_timing_info() failed"); + GENERIC_ERR_MSG("pa_stream_get_timing_info() failed"); return 0; } const struct pa_sample_spec *ss = pa_stream_get_sample_spec(priv->stream); if (!ss) { pa_threaded_mainloop_unlock(priv->mainloop); - GENERIC_ERR_MSG(priv->context, "pa_stream_get_sample_spec() failed"); + GENERIC_ERR_MSG("pa_stream_get_sample_spec() failed"); return 0; } // data left in PulseAudio's main buffers (not written to sink yet) @@ -485,7 +483,7 @@ static void info_func(struct pa_context *c, const struct pa_sink_input_info *i, struct ao *ao = userdata; struct priv *priv = ao->priv; if (is_last < 0) { - GENERIC_ERR_MSG(priv->context, "Failed to get sink input info"); + GENERIC_ERR_MSG("Failed to get sink input info"); return; } if (!i) @@ -504,8 +502,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg) pa_threaded_mainloop_lock(priv->mainloop); if (!waitop(priv, pa_context_get_sink_input_info(priv->context, devidx, info_func, ao))) { - GENERIC_ERR_MSG(priv->context, - "pa_stream_get_sink_input_info() failed"); + GENERIC_ERR_MSG("pa_stream_get_sink_input_info() failed"); return CONTROL_ERROR; } // Warning: some information in pi might be unaccessible, because @@ -548,8 +545,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg) &volume, NULL, NULL); if (!o) { pa_threaded_mainloop_unlock(priv->mainloop); - GENERIC_ERR_MSG(priv->context, - "pa_context_set_sink_input_volume() failed"); + GENERIC_ERR_MSG("pa_context_set_sink_input_volume() failed"); return CONTROL_ERROR; } } else if (cmd == AOCONTROL_SET_MUTE) { @@ -558,8 +554,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg) *mute, NULL, NULL); if (!o) { pa_threaded_mainloop_unlock(priv->mainloop); - GENERIC_ERR_MSG(priv->context, - "pa_context_set_sink_input_mute() failed"); + GENERIC_ERR_MSG("pa_context_set_sink_input_mute() failed"); return CONTROL_ERROR; } } else diff --git a/audio/out/ao_sdl.c b/audio/out/ao_sdl.c index 231b40681a..4ca153b4dd 100644 --- a/audio/out/ao_sdl.c +++ b/audio/out/ao_sdl.c @@ -124,7 +124,7 @@ static unsigned int ceil_power_of_two(unsigned int x) static int init(struct ao *ao) { if (SDL_WasInit(SDL_INIT_AUDIO)) { - mp_msg(MSGT_AO, MSGL_ERR, "[sdl] already initialized\n"); + MP_ERR(ao, "already initialized\n"); return -1; } @@ -132,7 +132,7 @@ static int init(struct ao *ao) if (SDL_InitSubSystem(SDL_INIT_AUDIO)) { if (!ao->probing) - mp_msg(MSGT_AO, MSGL_ERR, "[sdl] SDL_Init failed\n"); + MP_ERR(ao, "SDL_Init failed\n"); uninit(ao, true); return -1; } @@ -174,24 +174,23 @@ static int init(struct ao *ao) desired.callback = audio_callback; desired.userdata = ao; - mp_msg(MSGT_AO, MSGL_V, "[sdl] requested format: %d Hz, %d channels, %x, " - "buffer size: %d samples\n", - (int) desired.freq, (int) desired.channels, - (int) desired.format, (int) desired.samples); + MP_VERBOSE(ao, "requested format: %d Hz, %d channels, %x, " + "buffer size: %d samples\n", + (int) desired.freq, (int) desired.channels, + (int) desired.format, (int) desired.samples); obtained = desired; if (SDL_OpenAudio(&desired, &obtained)) { if (!ao->probing) - mp_msg(MSGT_AO, MSGL_ERR, "[sdl] could not open audio: %s\n", - SDL_GetError()); + MP_ERR(ao, "could not open audio: %s\n", SDL_GetError()); uninit(ao, true); return -1; } - mp_msg(MSGT_AO, MSGL_V, "[sdl] obtained format: %d Hz, %d channels, %x, " - "buffer size: %d samples\n", - (int) obtained.freq, (int) obtained.channels, - (int) obtained.format, (int) obtained.samples); + MP_VERBOSE(ao, "obtained format: %d Hz, %d channels, %x, " + "buffer size: %d samples\n", + (int) obtained.freq, (int) obtained.channels, + (int) obtained.format, (int) obtained.samples); switch (obtained.format) { case AUDIO_U8: ao->format = AF_FORMAT_U8; break; @@ -214,8 +213,7 @@ static int init(struct ao *ao) #endif default: if (!ao->probing) - mp_msg(MSGT_AO, MSGL_ERR, - "[sdl] could not find matching format\n"); + MP_ERR(ao, "could not find matching format\n"); uninit(ao, true); return -1; } @@ -229,13 +227,13 @@ static int init(struct ao *ao) priv->buffer = av_fifo_alloc(obtained.size * priv->bufcnt); priv->buffer_mutex = SDL_CreateMutex(); if (!priv->buffer_mutex) { - mp_msg(MSGT_AO, MSGL_ERR, "[sdl] SDL_CreateMutex failed\n"); + MP_ERR(ao, "SDL_CreateMutex failed\n"); uninit(ao, true); return -1; } priv->underrun_cond = SDL_CreateCond(); if (!priv->underrun_cond) { - mp_msg(MSGT_AO, MSGL_ERR, "[sdl] SDL_CreateCond failed\n"); + MP_ERR(ao, "SDL_CreateCond failed\n"); uninit(ao, true); return -1; } -- cgit v1.2.3