From 39b40e1ffb1e3fcf81ec4a4afe88c974adb2efcd Mon Sep 17 00:00:00 2001 From: wm4 Date: Fri, 24 Jan 2014 21:30:15 +0100 Subject: audio/filter: remove redundant log message prefixes These are now appended automatically, so you'd get them twice before this commit. --- audio/filter/af_bs2b.c | 6 +++--- audio/filter/af_channels.c | 8 ++++---- audio/filter/af_delay.c | 8 ++++---- audio/filter/af_dummy.c | 2 +- audio/filter/af_equalizer.c | 2 +- audio/filter/af_export.c | 12 ++++++------ audio/filter/af_hrtf.c | 18 +++++++++--------- audio/filter/af_lavcac3enc.c | 4 ++-- audio/filter/af_lavrresample.c | 4 ++-- audio/filter/af_pan.c | 4 ++-- audio/filter/af_scaletempo.c | 12 ++++++------ audio/filter/af_sinesuppress.c | 2 +- audio/filter/af_surround.c | 6 +++--- 13 files changed, 44 insertions(+), 44 deletions(-) (limited to 'audio/filter') diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c index 1ea31b260a..08d037bf4e 100644 --- a/audio/filter/af_bs2b.c +++ b/audio/filter/af_bs2b.c @@ -142,12 +142,12 @@ static int control(struct af_instance *af, int cmd, void *arg) // bs2b have srate limits, try to resample if needed if (af->data->rate > BS2B_MAXSRATE || af->data->rate < BS2B_MINSRATE) { af->data->rate = BS2B_DEFAULT_SRATE; - MP_WARN(af, "[bs2b] Requested sample rate %d Hz is out of bounds [%d..%d] Hz.\n" - "[bs2b] Trying to resample to %d Hz.\n", + MP_WARN(af, "Requested sample rate %d Hz is out of bounds [%d..%d] Hz.\n" + "Trying to resample to %d Hz.\n", af->data->rate, BS2B_MINSRATE, BS2B_MAXSRATE, BS2B_DEFAULT_SRATE); } bs2b_set_srate(s->filter, (long)af->data->rate); - MP_VERBOSE(af, "[bs2b] using format %s\n", + MP_VERBOSE(af, "using format %s\n", af_fmt_to_str(af->data->format)); return af_test_output(af,(struct mp_audio*)arg); diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c index 74a0de0fd6..6db44ba024 100644 --- a/audio/filter/af_channels.c +++ b/audio/filter/af_channels.c @@ -124,14 +124,14 @@ static int check_routes(struct af_instance *af, int nin, int nout) af_channels_t* s = af->priv; int i; if((s->nr < 1) || (s->nr > AF_NCH)){ - MP_ERR(af, "[channels] The number of routing pairs must be" + MP_ERR(af, "The number of routing pairs must be" " between 1 and %i. Current value is %i\n",AF_NCH,s->nr); return AF_ERROR; } for(i=0;inr;i++){ if((s->route[i][FR] >= nin) || (s->route[i][TO] >= nout)){ - MP_ERR(af, "[channels] Invalid routing in pair nr. %i.\n", i); + MP_ERR(af, "Invalid routing in pair nr. %i.\n", i); return AF_ERROR; } } @@ -220,11 +220,11 @@ static int af_open(struct af_instance* af){ do { int n = 0; if (ch >= AF_NCH) { - MP_FATAL(af, "[channels] Can't have more than %d routes.\n", AF_NCH); + MP_FATAL(af, "Can't have more than %d routes.\n", AF_NCH); return AF_ERROR; } sscanf(cp, "%i-%i%n" ,&s->route[ch][FR], &s->route[ch][TO], &n); - MP_VERBOSE(af, "[channels] Routing from channel %i to" + MP_VERBOSE(af, "Routing from channel %i to" " channel %i\n",s->route[ch][FR],s->route[ch][TO]); cp = &cp[n]; ch++; diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c index 51c453f6e5..a8cd79f117 100644 --- a/audio/filter/af_delay.c +++ b/audio/filter/af_delay.c @@ -54,7 +54,7 @@ static int control(struct af_instance* af, int cmd, void* arg) struct mp_audio *in = arg; if (in->bps != 1 && in->bps != 2 && in->bps != 4) { - MP_FATAL(af, "[delay] Sample format not supported\n"); + MP_FATAL(af, "Sample format not supported\n"); return AF_ERROR; } @@ -69,16 +69,16 @@ static int control(struct af_instance* af, int cmd, void* arg) for(i=0;idata->nch;i++){ s->q[i] = calloc(L,af->data->bps); if(NULL == s->q[i]) - MP_FATAL(af, "[delay] Out of memory\n"); + MP_FATAL(af, "Out of memory\n"); } if(AF_OK != af_from_ms(AF_NCH, s->d, s->wi, af->data->rate, 0.0, 1000.0)) return AF_ERROR; s->ri = 0; for(i=0;id[i],0.0,1000.0)); - MP_TRACE(af, "[delay] Channel %i delayed by %i samples\n", + MP_TRACE(af, "Channel %i delayed by %i samples\n", i,s->wi[i]); } return AF_OK; diff --git a/audio/filter/af_dummy.c b/audio/filter/af_dummy.c index 6822523efe..c13c32b968 100644 --- a/audio/filter/af_dummy.c +++ b/audio/filter/af_dummy.c @@ -33,7 +33,7 @@ static int control(struct af_instance* af, int cmd, void* arg) switch(cmd){ case AF_CONTROL_REINIT: ; *af->data = *(struct mp_audio*)arg; - MP_VERBOSE(af, "[dummy] Was reinitialized: %iHz/%ich/%s\n", + MP_VERBOSE(af, "Was reinitialized: %iHz/%ich/%s\n", af->data->rate,af->data->nch,af_fmt_to_str(af->data->format)); return AF_OK; } diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c index 49aedeb106..4f5a29706e 100644 --- a/audio/filter/af_equalizer.c +++ b/audio/filter/af_equalizer.c @@ -107,7 +107,7 @@ static int control(struct af_instance* af, int cmd, void* arg) s->K--; if(s->K != KM) - MP_INFO(af, "[equalizer] Limiting the number of filters to" + MP_INFO(af, "Limiting the number of filters to" " %i due to low sample rate.\n",s->K); // Generate filter taps diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c index a5a8257c39..542b2a1f9f 100644 --- a/audio/filter/af_export.c +++ b/audio/filter/af_export.c @@ -95,20 +95,20 @@ static int control(struct af_instance* af, int cmd, void* arg) // Allocate new buffers (as one continuous block) s->buf[0] = calloc(s->sz*af->data->nch, af->data->bps); if(NULL == s->buf[0]) - MP_FATAL(af, "[export] Out of memory\n"); + MP_FATAL(af, "Out of memory\n"); for(i = 1; i < af->data->nch; i++) s->buf[i] = (uint8_t *)s->buf[0] + i*s->sz*af->data->bps; if (!s->filename) { - MP_FATAL(af, "[export] No filename set.\n"); + MP_FATAL(af, "No filename set.\n"); return AF_ERROR; } // Init memory mapping s->fd = open(s->filename, O_RDWR | O_CREAT | O_TRUNC | O_CLOEXEC, 0640); - MP_INFO(af, "[export] Exporting to file: %s\n", s->filename); + MP_INFO(af, "Exporting to file: %s\n", s->filename); if(s->fd < 0) { - MP_FATAL(af, "[export] Could not open/create file: %s\n", + MP_FATAL(af, "Could not open/create file: %s\n", s->filename); return AF_ERROR; } @@ -125,8 +125,8 @@ static int control(struct af_instance* af, int cmd, void* arg) // mmap size s->mmap_area = mmap(0, mapsize, PROT_READ|PROT_WRITE,MAP_SHARED, s->fd, 0); if(s->mmap_area == NULL) - MP_FATAL(af, "[export] Could not mmap file %s\n", s->filename); - MP_INFO(af, "[export] Memory mapped to file: %s (%p)\n", + MP_FATAL(af, "Could not mmap file %s\n", s->filename); + MP_INFO(af, "Memory mapped to file: %s (%p)\n", s->filename, s->mmap_area); // Initialize header diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c index e3fecb8ad7..e329b4e558 100644 --- a/audio/filter/af_hrtf.c +++ b/audio/filter/af_hrtf.c @@ -296,7 +296,7 @@ static int control(struct af_instance *af, int cmd, void* arg) if(af->data->rate != 48000) { // automatic samplerate adjustment in the filter chain // is not yet supported. - MP_ERR(af, "[hrtf] ERROR: Sampling rate is not 48000 Hz (%d)!\n", + MP_ERR(af, "ERROR: Sampling rate is not 48000 Hz (%d)!\n", af->data->rate); return AF_ERROR; } @@ -367,25 +367,25 @@ static int filter(struct af_instance *af, struct mp_audio *data, int flags) s->print_flag = 0; switch (s->decode_mode) { case HRTF_MIX_51: - MP_INFO(af, "[hrtf] Using HRTF to mix %s discrete surround into " + MP_INFO(af, "Using HRTF to mix %s discrete surround into " "L, R channels\n", s->matrix_mode ? "5+1" : "5"); break; case HRTF_MIX_STEREO: - MP_INFO(af, "[hrtf] Using HRTF to mix stereo into " + MP_INFO(af, "Using HRTF to mix stereo into " "L, R channels\n"); break; case HRTF_MIX_MATRIX2CH: - MP_INFO(af, "[hrtf] Using active matrix to decode 2 channel " + MP_INFO(af, "Using active matrix to decode 2 channel " "input, HRTF to mix %s matrix surround into " "L, R channels\n", "3/2"); break; default: - MP_WARN(af, "[hrtf] bogus decode_mode: %d\n", s->decode_mode); + MP_WARN(af, "bogus decode_mode: %d\n", s->decode_mode); break; } if(s->matrix_mode) - MP_INFO(af, "[hrtf] Using active matrix to decode rear center " + MP_INFO(af, "Using active matrix to decode rear center " "channel\n"); } @@ -595,7 +595,7 @@ static int af_open(struct af_instance* af) s->print_flag = 1; if (allocate(s) != 0) { - MP_ERR(af, "[hrtf] Memory allocation error.\n"); + MP_ERR(af, "Memory allocation error.\n"); return AF_ERROR; } @@ -614,13 +614,13 @@ static int af_open(struct af_instance* af) s->cr_ir = cr_filt + (s->cr_o = pulse_detect(cr_filt)); if((s->ba_ir = malloc(s->basslen * sizeof(float))) == NULL) { - MP_ERR(af, "[hrtf] Memory allocation error.\n"); + MP_ERR(af, "Memory allocation error.\n"); return AF_ERROR; } fc = 2.0 * BASSFILTFREQ / (float)af->data->rate; if(af_filter_design_fir(s->basslen, s->ba_ir, &fc, LP | KAISER, 4 * M_PI) == -1) { - MP_ERR(af, "[hrtf] Unable to design low-pass " + MP_ERR(af, "Unable to design low-pass " "filter.\n"); return AF_ERROR; } diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c index 6df3820e75..34de6adb74 100644 --- a/audio/filter/af_lavcac3enc.c +++ b/audio/filter/af_lavcac3enc.c @@ -187,7 +187,7 @@ static int filter(struct af_instance* af, struct mp_audio* audio, int flags) AVFrame *frame = avcodec_alloc_frame(); if (!frame) { - MP_FATAL(af, "[libaf] Could not allocate memory \n"); + MP_FATAL(af, "Could not allocate memory \n"); return -1; } frame->nb_samples = s->in_samples; @@ -201,7 +201,7 @@ static int filter(struct af_instance* af, struct mp_audio* audio, int flags) int ok; ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok); if (ret < 0 || !ok) { - MP_FATAL(af, "[lavac3enc] Encode failed.\n"); + MP_FATAL(af, "Encode failed.\n"); return -1; } diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c index 4bd95459c0..f6537f1db0 100644 --- a/audio/filter/af_lavrresample.c +++ b/audio/filter/af_lavrresample.c @@ -221,7 +221,7 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in, if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) { - MP_ERR(af, "[lavrresample] Cannot open " + MP_ERR(af, "Cannot open " "Libavresample Context. \n"); return AF_ERROR; } @@ -397,7 +397,7 @@ static int af_open(struct af_instance *af) if (s->avrctx && s->avrctx_out) { return AF_OK; } else { - MP_ERR(af, "[lavrresample] Cannot initialize " + MP_ERR(af, "Cannot initialize " "Libavresample Context. \n"); uninit(af); return AF_ERROR; diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c index 081a8f72d0..01b0575ae3 100644 --- a/audio/filter/af_pan.c +++ b/audio/filter/af_pan.c @@ -80,7 +80,7 @@ static int control(struct af_instance* af, int cmd, void* arg) // Sanity check if(((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH){ - MP_ERR(af, "[pan] The number of output channels must be" + MP_ERR(af, "The number of output channels must be" " between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]); return AF_ERROR; } @@ -161,7 +161,7 @@ static int af_open(struct af_instance* af){ j = 0; k = 0; while(cp && k < AF_NCH){ sscanf(cp, "%f%n" , &s->level[j][k], &n); - MP_VERBOSE(af, "[pan] Pan level from channel %i to" + MP_VERBOSE(af, "Pan level from channel %i to" " channel %i = %f\n",k,j,s->level[j][k]); cp =&cp[n]; j++; diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c index eb5fc66a12..cd288e9d12 100644 --- a/audio/filter/af_scaletempo.c +++ b/audio/filter/af_scaletempo.c @@ -275,7 +275,7 @@ static int control(struct af_instance *af, int cmd, void *arg) int nch = data->nch; int use_int = 0; - MP_VERBOSE(af, "[scaletempo] %.3f speed * %.3f scale_nominal = %.3f\n", + MP_VERBOSE(af, "%.3f speed * %.3f scale_nominal = %.3f\n", s->speed, s->scale_nominal, s->scale); mp_audio_force_interleaved_format(data); @@ -317,7 +317,7 @@ static int control(struct af_instance *af, int cmd, void *arg) s->buf_overlap = realloc(s->buf_overlap, s->bytes_overlap); s->table_blend = realloc(s->table_blend, s->bytes_overlap * 4); if (!s->buf_overlap || !s->table_blend) { - MP_FATAL(af, "[scaletempo] Out of memory\n"); + MP_FATAL(af, "Out of memory\n"); return AF_ERROR; } memset(s->buf_overlap, 0, s->bytes_overlap); @@ -354,7 +354,7 @@ static int control(struct af_instance *af, int cmd, void *arg) s->table_window = realloc(s->table_window, s->bytes_overlap * 2 - nch * bps * 2); if (!s->buf_pre_corr || !s->table_window) { - MP_FATAL(af, "[scaletempo] Out of memory\n"); + MP_FATAL(af, "Out of memory\n"); return AF_ERROR; } memset((char *)s->buf_pre_corr + s->bytes_overlap * 2, 0, @@ -371,7 +371,7 @@ static int control(struct af_instance *af, int cmd, void *arg) s->table_window = realloc(s->table_window, s->bytes_overlap - nch * bps); if (!s->buf_pre_corr || !s->table_window) { - MP_FATAL(af, "[scaletempo] Out of memory\n"); + MP_FATAL(af, "Out of memory\n"); return AF_ERROR; } float *pw = s->table_window; @@ -391,14 +391,14 @@ static int control(struct af_instance *af, int cmd, void *arg) * bps * nch; s->buf_queue = realloc(s->buf_queue, s->bytes_queue + UNROLL_PADDING); if (!s->buf_queue) { - MP_FATAL(af, "[scaletempo] Out of memory\n"); + MP_FATAL(af, "Out of memory\n"); return AF_ERROR; } s->bytes_queued = 0; s->bytes_to_slide = 0; - MP_DBG(af, "[scaletempo] " + MP_DBG(af, "" "%.2f stride_in, %i stride_out, %i standing, " "%i overlap, %i search, %i queue, %s mode\n", s->frames_stride_scaled, diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c index 37952ab0be..9719d46828 100644 --- a/audio/filter/af_sinesuppress.c +++ b/audio/filter/af_sinesuppress.c @@ -99,7 +99,7 @@ static int play_s16(struct af_instance* af, struct mp_audio* data, int f) s->pos += 2 * M_PI * s->freq / data->rate; } - MP_VERBOSE(af, "[sinesuppress] f:%8.2f: amp:%8.2f\n", s->freq, sqrt(s->real*s->real + s->imag*s->imag) / s->ref); + MP_VERBOSE(af, "f:%8.2f: amp:%8.2f\n", s->freq, sqrt(s->real*s->real + s->imag*s->imag) / s->ref); return 0; } diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c index daa1f25ef1..9cabeedf9d 100644 --- a/audio/filter/af_surround.c +++ b/audio/filter/af_surround.c @@ -94,7 +94,7 @@ static int control(struct af_instance* af, int cmd, void* arg) struct mp_audio *in = arg; float fc; if (!mp_chmap_is_stereo(&in->channels)) { - MP_ERR(af, "[surround] Only stereo input is supported.\n"); + MP_ERR(af, "Only stereo input is supported.\n"); return AF_DETACH; } @@ -105,7 +105,7 @@ static int control(struct af_instance* af, int cmd, void* arg) // Surround filer coefficients fc = 2.0 * 7000.0/(float)af->data->rate; if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){ - MP_ERR(af, "[surround] Unable to design low-pass filter.\n"); + MP_ERR(af, "Unable to design low-pass filter.\n"); return AF_ERROR; } @@ -116,7 +116,7 @@ static int control(struct af_instance* af, int cmd, void* arg) s->dl = calloc(LD,af->data->bps); s->dr = calloc(LD,af->data->bps); if((NULL == s->dl) || (NULL == s->dr)) - MP_FATAL(af, "[delay] Out of memory\n"); + MP_FATAL(af, "Out of memory\n"); // Initialize delay queue index if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0)) -- cgit v1.2.3