diff options
Diffstat (limited to 'audio/filter')
-rw-r--r-- | audio/filter/af.c | 46 | ||||
-rw-r--r-- | audio/filter/af_bs2b.c | 2 | ||||
-rw-r--r-- | audio/filter/af_center.c | 4 | ||||
-rw-r--r-- | audio/filter/af_channels.c | 16 | ||||
-rw-r--r-- | audio/filter/af_convert24.c | 15 | ||||
-rw-r--r-- | audio/filter/af_convertsignendian.c | 9 | ||||
-rw-r--r-- | audio/filter/af_delay.c | 9 | ||||
-rw-r--r-- | audio/filter/af_drc.c | 17 | ||||
-rw-r--r-- | audio/filter/af_equalizer.c | 6 | ||||
-rw-r--r-- | audio/filter/af_export.c | 4 | ||||
-rw-r--r-- | audio/filter/af_extrastereo.c | 9 | ||||
-rw-r--r-- | audio/filter/af_format.c | 5 | ||||
-rw-r--r-- | audio/filter/af_hrtf.c | 11 | ||||
-rw-r--r-- | audio/filter/af_karaoke.c | 4 | ||||
-rw-r--r-- | audio/filter/af_ladspa.c | 12 | ||||
-rw-r--r-- | audio/filter/af_lavcac3enc.c | 30 | ||||
-rw-r--r-- | audio/filter/af_lavfi.c | 17 | ||||
-rw-r--r-- | audio/filter/af_lavrresample.c | 32 | ||||
-rw-r--r-- | audio/filter/af_pan.c | 11 | ||||
-rw-r--r-- | audio/filter/af_scaletempo.c | 27 | ||||
-rw-r--r-- | audio/filter/af_sinesuppress.c | 4 | ||||
-rw-r--r-- | audio/filter/af_sub.c | 4 | ||||
-rw-r--r-- | audio/filter/af_surround.c | 11 | ||||
-rw-r--r-- | audio/filter/af_sweep.c | 4 | ||||
-rw-r--r-- | audio/filter/af_volume.c | 9 |
25 files changed, 163 insertions, 155 deletions
diff --git a/audio/filter/af.c b/audio/filter/af.c index edee4bef65..95d0e43673 100644 --- a/audio/filter/af.c +++ b/audio/filter/af.c @@ -523,8 +523,7 @@ static int af_reinit(struct af_stream *s) // Check if this is the first filter struct mp_audio in = *af->prev->data; // Reset just in case... - in.audio = NULL; - in.len = 0; + mp_audio_set_null_data(&in); int rv = af->control(af, AF_CONTROL_REINIT, &in); if (rv == AF_OK && !mp_audio_config_equals(&in, af->prev->data)) @@ -640,8 +639,8 @@ int af_init(struct af_stream *s) return -1; // Precaution in case caller is misbehaving - s->input.audio = s->output.audio = NULL; - s->input.len = s->output.len = 0; + mp_audio_set_null_data(&s->input); + mp_audio_set_null_data(&s->output); // Check if this is the first call if (s->first->next == s->last) { @@ -731,36 +730,39 @@ double af_calc_delay(struct af_stream *s) return delay; } -/* Calculate the minimum output buffer size for given input data d - * when using the af_resize_local_buffer function. The +t+1 part ensures the - * value is >= len*mul rounded upwards to whole samples even if the - * double 'mul' is inexact. */ -static int af_lencalc(double mul, struct mp_audio *d) -{ - int t = d->bps * d->nch; - return d->len * mul + t + 1; -} - /* I a local buffer is used (i.e. if the filter doesn't operate on the incoming * buffer), this macro must be called to ensure the buffer is big enough. */ int af_resize_local_buffer(struct af_instance *af, struct mp_audio *data) { - if (af->data->len >= af_lencalc(af->mul, data)) + assert(data->format); + + if (!af->data->format && !af->data->planes[0]) { + // Dummy initialization + mp_audio_set_format(af->data, AF_FORMAT_U8); + } + + int oldlen = af->data->samples * af->data->sstride; + + /* Calculate the minimum output buffer size for given input data d + * when using the af_resize_local_buffer function. The +x part ensures + * the value is >= len*mul rounded upwards to whole samples even if the + * double 'mul' is inexact. */ + int newlen = data->samples * data->sstride * af->mul + data->sstride + 1; + + if (oldlen >= newlen) return AF_OK; - // Calculate new length - register int len = af_lencalc(af->mul, data); mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Reallocating memory in module %s, " - "old len = %i, new len = %i\n", af->info->name, af->data->len, len); + "old len = %i, new len = %i\n", af->info->name, oldlen, newlen); // If there is a buffer free it - free(af->data->audio); + free(af->data->planes[0]); // Create new buffer and check that it is OK - af->data->audio = malloc(len); - if (!af->data->audio) { + af->data->planes[0] = malloc(newlen); + if (!af->data->planes[0]) { mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n"); return AF_ERROR; } - af->data->len = len; + af->data->samples = newlen / af->data->sstride; return AF_OK; } diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c index 0e77b3e4eb..5e0caf28af 100644 --- a/audio/filter/af_bs2b.c +++ b/audio/filter/af_bs2b.c @@ -42,7 +42,7 @@ static struct mp_audio *play_##name(struct af_instance *af, struct mp_audio *dat { \ /* filter is called for all pairs of samples available in the buffer */ \ bs2b_cross_feed_##name(((struct af_bs2b*)(af->priv))->filter, \ - (type*)(data->audio), data->len/data->bps/2); \ + (type*)(data->planes[0]), data->samples); \ \ return data; \ } diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c index 0cfdbc3b0e..c64c551f1c 100644 --- a/audio/filter/af_center.c +++ b/audio/filter/af_center.c @@ -87,9 +87,9 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) { struct mp_audio* c = data; // Current working data af_center_t* s = af->setup; // Setup for this instance - float* a = c->audio; // Audio data - int len = c->len/4; // Number of samples in current audio block + float* a = c->planes[0]; // Audio data int nch = c->nch; // Number of channels + int len = c->samples*c->nch; // Number of samples in current audio block int ch = s->ch; // Channel in which to insert the center audio register int i; diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c index 27445aafe2..fd3b8262f5 100644 --- a/audio/filter/af_channels.c +++ b/audio/filter/af_channels.c @@ -169,6 +169,10 @@ static int control(struct af_instance* af, int cmd, void* arg) af->data->rate = ((struct mp_audio*)arg)->rate; mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format); af->mul = (double)af->data->nch / ((struct mp_audio*)arg)->nch; + mp_audio_force_interleaved_format(af->data); + int r = af_test_output(af,(struct mp_audio*)arg); + if (r != AF_OK) + return r; return check_routes(s,((struct mp_audio*)arg)->nch,af->data->nch); case AF_CONTROL_COMMAND_LINE:{ int nch = 0; @@ -219,7 +223,7 @@ static void uninit(struct af_instance* af) { free(af->setup); if (af->data) - free(af->data->audio); + free(af->data->planes[0]); free(af->data); } @@ -235,16 +239,16 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) return NULL; // Reset unused channels - memset(l->audio,0,c->len / c->nch * l->nch); + memset(l->planes[0],0,mp_audio_psize(c) / c->nch * l->nch); if(AF_OK == check_routes(s,c->nch,l->nch)) for(i=0;i<s->nr;i++) - copy(c->audio,l->audio,c->nch,s->route[i][FR], - l->nch,s->route[i][TO],c->len,c->bps); + copy(c->planes[0],l->planes[0],c->nch,s->route[i][FR], + l->nch,s->route[i][TO],mp_audio_psize(c),c->bps); // Set output data - c->audio = l->audio; - c->len = c->len / c->nch * l->nch; + c->planes[0] = l->planes[0]; + c->samples = c->samples / c->nch * l->nch; mp_audio_set_channels(c, &l->channels); return c; diff --git a/audio/filter/af_convert24.c b/audio/filter/af_convert24.c index 18ce156467..96924de344 100644 --- a/audio/filter/af_convert24.c +++ b/audio/filter/af_convert24.c @@ -78,12 +78,12 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) return NULL; struct mp_audio *out = af->data; - size_t len = data->len / data->bps; + size_t len = mp_audio_psize(data) / data->bps; if (data->bps == 4) { for (int s = 0; s < len; s++) { - uint32_t val = *((uint32_t *)data->audio + s); - uint8_t *ptr = (uint8_t *)out->audio + s * 3; + uint32_t val = *((uint32_t *)data->planes[0] + s); + uint8_t *ptr = (uint8_t *)out->planes[0] + s * 3; ptr[0] = val >> SHIFT(0); ptr[1] = val >> SHIFT(1); ptr[2] = val >> SHIFT(2); @@ -91,24 +91,23 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) mp_audio_set_format(data, af_fmt_change_bits(data->format, 24)); } else { for (int s = 0; s < len; s++) { - uint8_t *ptr = (uint8_t *)data->audio + s * 3; + uint8_t *ptr = (uint8_t *)data->planes[0] + s * 3; uint32_t val = ptr[0] << SHIFT(0) | ptr[1] << SHIFT(1) | ptr[2] << SHIFT(2); - *((uint32_t *)out->audio + s) = val; + *((uint32_t *)out->planes[0] + s) = val; } mp_audio_set_format(data, af_fmt_change_bits(data->format, 32)); } - data->audio = out->audio; - data->len = len * data->bps; + data->planes[0] = out->planes[0]; return data; } static void uninit(struct af_instance* af) { if (af->data) - free(af->data->audio); + free(af->data->planes[0]); } static int af_open(struct af_instance *af) diff --git a/audio/filter/af_convertsignendian.c b/audio/filter/af_convertsignendian.c index bfea004bb2..5565438aad 100644 --- a/audio/filter/af_convertsignendian.c +++ b/audio/filter/af_convertsignendian.c @@ -24,6 +24,9 @@ static bool test_conversion(int src_format, int dst_format) { + if ((src_format & AF_FORMAT_PLANAR) || + (dst_format & AF_FORMAT_PLANAR)) + return false; int src_noend = src_format & ~AF_FORMAT_END_MASK; int dst_noend = dst_format & ~AF_FORMAT_END_MASK; // We can swap endian for all formats, but sign only for integer formats. @@ -100,13 +103,13 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) { int infmt = data->format; int outfmt = af->data->format; - size_t len = data->len / data->bps; + size_t len = data->samples * data->nch; if ((infmt & AF_FORMAT_END_MASK) != (outfmt & AF_FORMAT_END_MASK)) - endian(data->audio, len, data->bps); + endian(data->planes[0], len, data->bps); if ((infmt & AF_FORMAT_SIGN_MASK) != (outfmt & AF_FORMAT_SIGN_MASK)) - si2us(data->audio, len, data->bps, + si2us(data->planes[0], len, data->bps, (outfmt & AF_FORMAT_END_MASK) == AF_FORMAT_LE); mp_audio_set_format(data, outfmt); diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c index a6515f84cf..c979060fe3 100644 --- a/audio/filter/af_delay.c +++ b/audio/filter/af_delay.c @@ -56,6 +56,7 @@ static int control(struct af_instance* af, int cmd, void* arg) free(s->q[i]); mp_audio_copy_config(af->data, (struct mp_audio*)arg); + mp_audio_force_interleaved_format(af->data); // Allocate new delay queues for(i=0;i<af->data->nch;i++){ @@ -123,13 +124,13 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) struct mp_audio* c = data; // Current working data af_delay_t* s = af->setup; // Setup for this instance int nch = c->nch; // Number of channels - int len = c->len/c->bps; // Number of sample in data chunk + int len = mp_audio_psize(c)/c->bps; // Number of sample in data chunk int ri = 0; int ch,i; for(ch=0;ch<nch;ch++){ switch(c->bps){ case 1:{ - int8_t* a = c->audio; + int8_t* a = c->planes[0]; int8_t* q = s->q[ch]; int wi = s->wi[ch]; ri = s->ri; @@ -143,7 +144,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) break; } case 2:{ - int16_t* a = c->audio; + int16_t* a = c->planes[0]; int16_t* q = s->q[ch]; int wi = s->wi[ch]; ri = s->ri; @@ -157,7 +158,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) break; } case 4:{ - int32_t* a = c->audio; + int32_t* a = c->planes[0]; int32_t* q = s->q[ch]; int wi = s->wi[ch]; ri = s->ri; diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c index 9bbbde4831..589844d89b 100644 --- a/audio/filter/af_drc.c +++ b/audio/filter/af_drc.c @@ -88,6 +88,7 @@ static int control(struct af_instance* af, int cmd, void* arg) // Sanity check if(!arg) return AF_ERROR; + mp_audio_force_interleaved_format((struct mp_audio*)arg); mp_audio_copy_config(af->data, (struct mp_audio*)arg); if(((struct mp_audio*)arg)->format != (AF_FORMAT_S16_NE)){ @@ -119,8 +120,8 @@ static void uninit(struct af_instance* af) static void method1_int16(af_drc_t *s, struct mp_audio *c) { register int i = 0; - int16_t *data = (int16_t*)c->audio; // Audio data - int len = c->len/2; // Number of samples + int16_t *data = (int16_t*)c->planes[0]; // Audio data + int len = c->samples*c->nch; // Number of samples float curavg = 0.0, newavg, neededmul; int tmp; @@ -161,8 +162,8 @@ static void method1_int16(af_drc_t *s, struct mp_audio *c) static void method1_float(af_drc_t *s, struct mp_audio *c) { register int i = 0; - float *data = (float*)c->audio; // Audio data - int len = c->len/4; // Number of samples + float *data = (float*)c->planes[0]; // Audio data + int len = c->samples*c->nch; // Number of samples float curavg = 0.0, newavg, neededmul, tmp; for (i = 0; i < len; i++) @@ -198,8 +199,8 @@ static void method1_float(af_drc_t *s, struct mp_audio *c) static void method2_int16(af_drc_t *s, struct mp_audio *c) { register int i = 0; - int16_t *data = (int16_t*)c->audio; // Audio data - int len = c->len/2; // Number of samples + int16_t *data = (int16_t*)c->planes[0]; // Audio data + int len = c->samples*c->nch; // Number of samples float curavg = 0.0, newavg, avg = 0.0; int tmp, totallen = 0; @@ -248,8 +249,8 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c) static void method2_float(af_drc_t *s, struct mp_audio *c) { register int i = 0; - float *data = (float*)c->audio; // Audio data - int len = c->len/4; // Number of samples + float *data = (float*)c->planes[0]; // Audio data + int len = c->samples*c->nch; // Number of samples float curavg = 0.0, newavg, avg = 0.0, tmp; int totallen = 0; diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c index cbdcd3f84a..718445c001 100644 --- a/audio/filter/af_equalizer.c +++ b/audio/filter/af_equalizer.c @@ -170,9 +170,9 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) while(ci--){ float* g = s->g[ci]; // Gain factor - float* in = ((float*)c->audio)+ci; - float* out = ((float*)c->audio)+ci; - float* end = in + c->len/4; // Block loop end + float* in = ((float*)c->planes[0])+ci; + float* out = ((float*)c->planes[0])+ci; + float* end = in + c->samples*c->nch; // Block loop end while(in < end){ register int k = 0; // Frequency band index diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c index 45143075a1..a6ebdc322e 100644 --- a/audio/filter/af_export.c +++ b/audio/filter/af_export.c @@ -213,9 +213,9 @@ static struct mp_audio* play( struct af_instance* af, struct mp_audio* data ) { struct mp_audio* c = data; // Current working data af_export_t* s = af->setup; // Setup for this instance - int16_t* a = c->audio; // Incomming sound + int16_t* a = c->planes[0]; // Incomming sound int nch = c->nch; // Number of channels - int len = c->len/c->bps; // Number of sample in data chunk + int len = c->samples*c->nch; // Number of sample in data chunk int sz = s->sz; // buffer size (in samples) int flag = 0; // Set to 1 if buffer is filled diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c index 4cf27f2724..5b3792763b 100644 --- a/audio/filter/af_extrastereo.c +++ b/audio/filter/af_extrastereo.c @@ -49,6 +49,7 @@ static int control(struct af_instance* af, int cmd, void* arg) if(!arg) return AF_ERROR; mp_audio_copy_config(af->data, (struct mp_audio*)arg); + mp_audio_force_interleaved_format(af->data); mp_audio_set_num_channels(af->data, 2); if (af->data->format == AF_FORMAT_FLOAT_NE) { @@ -83,8 +84,8 @@ static struct mp_audio* play_s16(struct af_instance* af, struct mp_audio* data) { af_extrastereo_t *s = af->setup; register int i = 0; - int16_t *a = (int16_t*)data->audio; // Audio data - int len = data->len/2; // Number of samples + int16_t *a = (int16_t*)data->planes[0]; // Audio data + int len = data->samples*data->nch; // Number of samples int avg, l, r; for (i = 0; i < len; i+=2) @@ -105,8 +106,8 @@ static struct mp_audio* play_float(struct af_instance* af, struct mp_audio* data { af_extrastereo_t *s = af->setup; register int i = 0; - float *a = (float*)data->audio; // Audio data - int len = data->len/4; // Number of samples + float *a = (float*)data->planes[0]; // Audio data + int len = data->samples * data->nch; // Number of samples float avg, l, r; for (i = 0; i < len; i+=2) diff --git a/audio/filter/af_format.c b/audio/filter/af_format.c index 642ef927bb..b551ddba42 100644 --- a/audio/filter/af_format.c +++ b/audio/filter/af_format.c @@ -105,8 +105,9 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) struct mp_audio *r = &priv->temp; *r = *af->data; - r->audio = data->audio; - r->len = data->len; + for (int n = 0; n < r->nch; n++) + r->planes[n] = data->planes[n]; + r->samples = data->samples; return r; } diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c index 5b80bf0eec..01148dc6a6 100644 --- a/audio/filter/af_hrtf.c +++ b/audio/filter/af_hrtf.c @@ -367,7 +367,7 @@ static void uninit(struct af_instance *af) free(af->setup); } if(af->data) - free(af->data->audio); + free(af->data->planes[0]); free(af->data); } @@ -385,9 +385,9 @@ damped (without any real 3D acoustical image, however). static struct mp_audio* play(struct af_instance *af, struct mp_audio *data) { af_hrtf_t *s = af->setup; - short *in = data->audio; // Input audio data + short *in = data->planes[0]; // Input audio data short *out = NULL; // Output audio data - short *end = in + data->len / sizeof(short); // Loop end + short *end = in + data->samples * data->nch; // Loop end float common, left, right, diff, left_b, right_b; const int dblen = s->dlbuflen, hlen = s->hrflen, blen = s->basslen; @@ -425,7 +425,7 @@ static struct mp_audio* play(struct af_instance *af, struct mp_audio *data) "channel\n"); } - out = af->data->audio; + out = af->data->planes[0]; /* MPlayer's 5 channel layout (notation for the variable): * @@ -565,8 +565,7 @@ static struct mp_audio* play(struct af_instance *af, struct mp_audio *data) } /* Set output data */ - data->audio = af->data->audio; - data->len = data->len / data->nch * 2; + data->planes[0] = af->data->planes[0]; mp_audio_set_num_channels(data, 2); return data; diff --git a/audio/filter/af_karaoke.c b/audio/filter/af_karaoke.c index b24ba0d877..d9c62aa85c 100644 --- a/audio/filter/af_karaoke.c +++ b/audio/filter/af_karaoke.c @@ -51,9 +51,9 @@ static void uninit(struct af_instance* af) static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) { struct mp_audio* c = data; // Current working data - float* a = c->audio; // Audio data - int len = c->len/4; // Number of samples in current audio block + float* a = c->planes[0]; // Audio data int nch = c->nch; // Number of channels + int len = c->samples*nch; // Number of samples in current audio block register int i; /* diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c index 73b7430201..6acdc150d7 100644 --- a/audio/filter/af_ladspa.c +++ b/audio/filter/af_ladspa.c @@ -497,10 +497,6 @@ static int control(struct af_instance *af, int cmd, void *arg) { mp_audio_copy_config(af->data, (struct mp_audio*)arg); mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); - /* arg->len is not set here yet, so init of buffers and connecting the - * filter, has to be done in play() :-/ - */ - return af_test_output(af, (struct mp_audio*)arg); case AF_CONTROL_COMMAND_LINE: { char *buf; @@ -710,8 +706,8 @@ static void uninit(struct af_instance *af) { static struct mp_audio* play(struct af_instance *af, struct mp_audio *data) { af_ladspa_t *setup = af->setup; const LADSPA_Descriptor *pdes = setup->plugin_descriptor; - float *audio = (float*)data->audio; - int nsamples = data->len/4; /* /4 because it's 32-bit float */ + float *audio = (float*)data->planes[0]; + int nsamples = data->samples*data->nch; int nch = data->nch; int rate = data->rate; int i, p; @@ -723,10 +719,6 @@ static struct mp_audio* play(struct af_instance *af, struct mp_audio *data) { * plugin, connect ports and activate plugin */ - /* 2004-12-07: Also check if the buffersize has to be changed! - * data->len is not constant per se! re-init buffers. - */ - if ( (setup->bufsize != nsamples/nch) || (setup->nch != nch) ) { /* if setup->nch==0, it's the first call, if not, something has diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c index 9ca193017a..d4abf2e942 100644 --- a/audio/filter/af_lavcac3enc.c +++ b/audio/filter/af_lavcac3enc.c @@ -161,7 +161,7 @@ static void uninit(struct af_instance* af) af_ac3enc_t *s = af->setup; if (af->data) - free(af->data->audio); + free(af->data->planes[0]); free(af->data); if (s) { av_free_packet(&s->pkt); @@ -183,7 +183,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* audio) int left, outsize = 0; char *buf, *src; int max_output_len; - int frame_num = (audio->len + s->pending_len) / s->expect_len; + int frame_num = (mp_audio_psize(audio) + s->pending_len) / s->expect_len; int samplesize = af_fmt2bits(s->in_sampleformat) / 8; if (s->add_iec61937_header) @@ -191,23 +191,23 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* audio) else max_output_len = AC3_MAX_CODED_FRAME_SIZE * frame_num; - if (af->data->len < max_output_len) { + if (mp_audio_psize(af->data) < max_output_len) { mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Reallocating memory in module %s, " - "old len = %i, new len = %i\n", af->info->name, af->data->len, - max_output_len); - free(af->data->audio); - af->data->audio = malloc(max_output_len); - if (!af->data->audio) { + "old len = %i, new len = %i\n", af->info->name, + mp_audio_psize(af->data), max_output_len); + free(af->data->planes[0]); + af->data->planes[0] = malloc(max_output_len); + if (!af->data->planes[0]) { mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n"); return NULL; } - af->data->len = max_output_len; + af->data->samples = max_output_len / af->data->sstride; } l = af->data; // Local data - buf = l->audio; - src = c->audio; - left = c->len; + buf = l->planes[0]; + src = c->planes[0]; + left = mp_audio_psize(c); while (left > 0) { @@ -296,7 +296,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* audio) len = AC3_FRAME_SIZE * 2 * 2; } - assert(buf + len <= (char *)af->data->audio + af->data->len); + assert(buf + len <= (char *)af->data->planes[0] + mp_audio_psize(af->data)); assert(s->pkt.size <= len - header_len); memcpy(buf + header_len, s->pkt.data, s->pkt.size); @@ -304,10 +304,10 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* audio) outsize += len; buf += len; } - c->audio = l->audio; + c->planes[0] = l->planes[0]; mp_audio_set_num_channels(c, 2); mp_audio_set_format(c, af->data->format); - c->len = outsize; + c->samples = outsize / c->sstride; mp_msg(MSGT_AFILTER, MSGL_DBG2, "play return size %d, pending %d\n", outsize, s->pending_len); return c; diff --git a/audio/filter/af_lavfi.c b/audio/filter/af_lavfi.c index 3c6d981f1d..f08e7ff85c 100644 --- a/audio/filter/af_lavfi.c +++ b/audio/filter/af_lavfi.c @@ -181,6 +181,7 @@ static int control(struct af_instance *af, int cmd, void *arg) if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE) mp_audio_set_format(in, AF_FORMAT_FLOAT_NE); + mp_audio_force_interleaved_format(in); if (!mp_chmap_is_lavc(&in->channels)) mp_chmap_reorder_to_lavc(&in->channels); // will always work @@ -193,6 +194,7 @@ static int control(struct af_instance *af, int cmd, void *arg) out->rate = l_out->sample_rate; mp_audio_set_format(out, af_from_avformat(l_out->format)); + mp_audio_force_interleaved_format(out); struct mp_chmap out_cm; mp_chmap_from_lavc(&out_cm, l_out->channel_layout); @@ -228,7 +230,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) int out_frame_size = r->bps * r->channels.num; AVFrame *frame = av_frame_alloc(); - frame->nb_samples = data->len / in_frame_size; + frame->nb_samples = data->samples; frame->format = l_in->format; // Timebase is 1/sample_rate @@ -238,7 +240,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) av_frame_set_channel_layout(frame, l_in->channel_layout); av_frame_set_sample_rate(frame, l_in->sample_rate); - frame->data[0] = data->audio; + frame->data[0] = data->planes[0]; frame->extended_data = frame->data; if (av_buffersrc_add_frame(p->in, frame) < 0) { @@ -268,11 +270,11 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) av_frame_free(&frame); } - r->audio = p->out_buffer; - r->len = out_len; + r->planes[0] = p->out_buffer; + r->samples = out_len / r->sstride; - p->bytes_in += data->len; - p->bytes_out += r->len; + p->bytes_in += data->samples * data->sstride; + p->bytes_out += r->samples * r->sstride; if (out_pts != AV_NOPTS_VALUE) { int64_t num_in_frames = p->bytes_in / in_frame_size; @@ -280,8 +282,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) double out_time = out_pts * av_q2d(p->timebase_out); // Need pts past the last output sample. - int out_frames = r->len / out_frame_size; - out_time += out_frames / (double)r->rate; + out_time += r->samples / (double)r->rate; af->delay = (in_time - out_time) * r->rate * out_frame_size; } diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c index 860e5a52d3..a6abd32aed 100644 --- a/audio/filter/af_lavrresample.c +++ b/audio/filter/af_lavrresample.c @@ -244,8 +244,10 @@ static int control(struct af_instance *af, int cmd, void *arg) if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE) mp_audio_set_format(in, AF_FORMAT_FLOAT_NE); + mp_audio_force_interleaved_format(in); if (af_to_avformat(out->format) == AV_SAMPLE_FMT_NONE) mp_audio_set_format(out, in->format); + mp_audio_force_interleaved_format(out); af->mul = (double) (out->rate * out->nch) / (in->rate * in->nch); af->delay = out->nch * s->opts.filter_size / FFMIN(af->mul, 1); @@ -306,28 +308,30 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) struct mp_audio *out = af->data; - int in_size = data->len; - int in_samples = |