mirror of
https://github.com/mpv-player/mpv
synced 2024-12-21 22:30:22 +00:00
caaa1189ba
Just reimplement it in some way, as mp_audio is GPL-only. Actually I wanted to get rid of audio_buffer.c completely (and instead have a list of mp_aframes), but to do so would require rewriting some more player core audio code. So to get this LGPL relicensing over quickly, just do some extra work.
158 lines
4.9 KiB
C
158 lines
4.9 KiB
C
/*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stddef.h>
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#include <limits.h>
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#include <assert.h>
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#include "common/common.h"
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#include "chmap.h"
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#include "audio_buffer.h"
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#include "format.h"
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struct mp_audio_buffer {
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int format;
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struct mp_chmap channels;
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int srate;
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int sstride;
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int num_planes;
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uint8_t *data[MP_NUM_CHANNELS];
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int allocated;
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int num_samples;
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};
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struct mp_audio_buffer *mp_audio_buffer_create(void *talloc_ctx)
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{
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return talloc_zero(talloc_ctx, struct mp_audio_buffer);
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}
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// Reinitialize the buffer, set a new format, drop old data.
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// The audio data in fmt is not used, only the format.
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void mp_audio_buffer_reinit_fmt(struct mp_audio_buffer *ab, int format,
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const struct mp_chmap *channels, int srate)
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{
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for (int n = 0; n < MP_NUM_CHANNELS; n++)
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TA_FREEP(&ab->data[n]);
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ab->format = format;
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ab->channels = *channels;
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ab->srate = srate;
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ab->allocated = 0;
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ab->num_samples = 0;
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ab->sstride = af_fmt_to_bytes(ab->format);
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ab->num_planes = 1;
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if (af_fmt_is_planar(ab->format)) {
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ab->num_planes = ab->channels.num;
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} else {
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ab->sstride *= ab->channels.num;
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}
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}
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// Make the total size of the internal buffer at least this number of samples.
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void mp_audio_buffer_preallocate_min(struct mp_audio_buffer *ab, int samples)
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{
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if (samples > ab->allocated) {
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for (int n = 0; n < ab->num_planes; n++) {
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ab->data[n] = talloc_realloc(ab, ab->data[n], char,
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ab->sstride * samples);
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}
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ab->allocated = samples;
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}
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}
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// Get number of samples that can be written without forcing a resize of the
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// internal buffer.
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int mp_audio_buffer_get_write_available(struct mp_audio_buffer *ab)
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{
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return ab->allocated - ab->num_samples;
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}
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// All integer parameters are in samples.
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// dst and src can overlap.
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static void copy_planes(struct mp_audio_buffer *ab,
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uint8_t **dst, int dst_offset,
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uint8_t **src, int src_offset, int length)
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{
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for (int n = 0; n < ab->num_planes; n++) {
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memmove((char *)dst[n] + dst_offset * ab->sstride,
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(char *)src[n] + src_offset * ab->sstride,
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length * ab->sstride);
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}
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}
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// Append data to the end of the buffer.
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// If the buffer is not large enough, it is transparently resized.
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void mp_audio_buffer_append(struct mp_audio_buffer *ab, void **ptr, int samples)
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{
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mp_audio_buffer_preallocate_min(ab, ab->num_samples + samples);
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copy_planes(ab, ab->data, ab->num_samples, (uint8_t **)ptr, 0, samples);
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ab->num_samples += samples;
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}
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// Prepend silence to the start of the buffer.
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void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples)
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{
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assert(samples >= 0);
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mp_audio_buffer_preallocate_min(ab, ab->num_samples + samples);
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copy_planes(ab, ab->data, samples, ab->data, 0, ab->num_samples);
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ab->num_samples += samples;
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for (int n = 0; n < ab->num_planes; n++)
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af_fill_silence(ab->data[n], samples * ab->sstride, ab->format);
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}
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void mp_audio_buffer_duplicate(struct mp_audio_buffer *ab, int samples)
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{
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assert(samples >= 0 && samples <= ab->num_samples);
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mp_audio_buffer_preallocate_min(ab, ab->num_samples + samples);
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copy_planes(ab, ab->data, ab->num_samples,
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ab->data, ab->num_samples - samples, samples);
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ab->num_samples += samples;
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}
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// Get the start of the current readable buffer.
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void mp_audio_buffer_peek(struct mp_audio_buffer *ab, uint8_t ***ptr,
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int *samples)
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{
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*ptr = ab->data;
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*samples = ab->num_samples;
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}
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// Skip leading samples. (Used with mp_audio_buffer_peek() to read data.)
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void mp_audio_buffer_skip(struct mp_audio_buffer *ab, int samples)
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{
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assert(samples >= 0 && samples <= ab->num_samples);
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copy_planes(ab, ab->data, 0, ab->data, samples, ab->num_samples - samples);
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ab->num_samples -= samples;
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}
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void mp_audio_buffer_clear(struct mp_audio_buffer *ab)
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{
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ab->num_samples = 0;
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}
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// Return number of buffered audio samples
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int mp_audio_buffer_samples(struct mp_audio_buffer *ab)
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{
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return ab->num_samples;
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}
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// Return amount of buffered audio in seconds.
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double mp_audio_buffer_seconds(struct mp_audio_buffer *ab)
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{
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return ab->num_samples / (double)ab->srate;
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}
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