/* * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include <stdint.h> #include <string.h> #include "libavutil/mem.h" #include "audio_data.h" static const AVClass audio_data_class = { .class_name = "AudioData", .item_name = av_default_item_name, .version = LIBAVUTIL_VERSION_INT, }; /* * Calculate alignment for data pointers. */ static void calc_ptr_alignment(AudioData *a) { int p; int min_align = 128; for (p = 0; p < a->planes; p++) { int cur_align = 128; while ((intptr_t)a->data[p] % cur_align) cur_align >>= 1; if (cur_align < min_align) min_align = cur_align; } a->ptr_align = min_align; } int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels) { if (channels == 1) return 1; else return av_sample_fmt_is_planar(sample_fmt); } int ff_audio_data_set_channels(AudioData *a, int channels) { if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS || channels > a->allocated_channels) return AVERROR(EINVAL); a->channels = channels; a->planes = a->is_planar ? channels : 1; calc_ptr_alignment(a); return 0; } int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels, int nb_samples, enum AVSampleFormat sample_fmt, int read_only, const char *name) { int p; memset(a, 0, sizeof(*a)); a->class = &audio_data_class; if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) { av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels); return AVERROR(EINVAL); } a->sample_size = av_get_bytes_per_sample(sample_fmt); if (!a->sample_size) { av_log(a, AV_LOG_ERROR, "invalid sample format\n"); return AVERROR(EINVAL); } a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels); a->planes = a->is_planar ? channels : 1; a->stride = a->sample_size * (a->is_planar ? 1 : channels); for (p = 0; p < (a->is_planar ? channels : 1); p++) { if (!src[p]) { av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p); return AVERROR(EINVAL); } a->data[p] = src[p]; } a->allocated_samples = nb_samples * !read_only; a->nb_samples = nb_samples; a->sample_fmt = sample_fmt; a->channels = channels; a->allocated_channels = channels; a->read_only = read_only; a->allow_realloc = 0; a->name = name ? name : "{no name}"; calc_ptr_alignment(a); a->samples_align = plane_size / a->stride; return 0; } AudioData *ff_audio_data_alloc(int channels, int nb_samples, enum AVSampleFormat sample_fmt, const char *name) { AudioData *a; int ret; if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) return NULL; a = av_mallocz(sizeof(*a)); if (!a) return NULL; a->sample_size = av_get_bytes_per_sample(sample_fmt); if (!a->sample_size) { av_free(a); return NULL; } a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels); a->planes = a->is_planar ? channels : 1; a->stride = a->sample_size * (a->is_planar ? 1 : channels); a->class = &audio_data_class; a->sample_fmt = sample_fmt; a->channels = channels; a->allocated_channels = channels; a->read_only = 0; a->allow_realloc = 1; a->name = name ? name : "{no name}"; if (nb_samples > 0) { ret = ff_audio_data_realloc(a, nb_samples); if (ret < 0) { av_free(a); return NULL; } return a; } else { calc_ptr_alignment(a); return a; } } int ff_audio_data_realloc(AudioData *a, int nb_samples) { int ret, new_buf_size, plane_size, p; /* check if buffer is already large enough */ if (a->allocated_samples >= nb_samples) return 0; /* validate that the output is not read-only and realloc is allowed */ if (a->read_only || !a->allow_realloc) return AVERROR(EINVAL); new_buf_size = av_samples_get_buffer_size(&plane_size, a->allocated_channels, nb_samples, a->sample_fmt, 0); if (new_buf_size < 0) return new_buf_size; /* if there is already data in the buffer and the sample format is planar, allocate a new buffer and copy the data, otherwise just realloc the internal buffer and set new data pointers */ if (a->nb_samples > 0 && a->is_planar) { uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL }; ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels, nb_samples, a->sample_fmt, 0); if (ret < 0) return ret; for (p = 0; p < a->planes; p++) memcpy(new_data[p], a->data[p], a->nb_samples * a->stride); av_freep(&a->buffer); memcpy(a->data, new_data, sizeof(new_data)); a->buffer = a->data[0]; } else { av_freep(&a->buffer); a->buffer = av_malloc(new_buf_size); if (!a->buffer) return AVERROR(ENOMEM); ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer, a->allocated_channels, nb_samples, a->sample_fmt, 0); if (ret < 0) return ret; } a->buffer_size = new_buf_size; a->allocated_samples = nb_samples; calc_ptr_alignment(a); a->samples_align = plane_size / a->stride; return 0; } void ff_audio_data_free(AudioData **a) { if (!*a) return; av_free((*a)->buffer); av_freep(a); } int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map) { int ret, p; /* validate input/output compatibility */ if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels) return AVERROR(EINVAL); if (map && !src->is_planar) { av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n"); return AVERROR(EINVAL); } /* if the input is empty, just empty the output */ if (!src->nb_samples) { dst->nb_samples = 0; return 0; } /* reallocate output if necessary */ ret = ff_audio_data_realloc(dst, src->nb_samples); if (ret < 0) return ret; /* copy data */ if (map) { if (map->do_remap) { for (p = 0; p < src->planes; p++) { if (map->channel_map[p] >= 0) memcpy(dst->data[p], src->data[map->channel_map[p]], src->nb_samples * src->stride); } } if (map->do_copy || map->do_zero) { for (p = 0; p < src->planes; p++) { if (map->channel_copy[p]) memcpy(dst->data[p], dst->data[map->channel_copy[p]], src->nb_samples * src->stride); else if (map->channel_zero[p]) av_samples_set_silence(&dst->data[p], 0, src->nb_samples, 1, dst->sample_fmt); } } } else { for (p = 0; p < src->planes; p++) memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride); } dst->nb_samples = src->nb_samples; return 0; } int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, int src_offset, int nb_samples) { int ret, p, dst_offset2, dst_move_size; /* validate input/output compatibility */ if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) { av_log(src, AV_LOG_ERROR, "sample format mismatch\n"); return AVERROR(EINVAL); } /* validate offsets are within the buffer bounds */ if (dst_offset < 0 || dst_offset > dst->nb_samples || src_offset < 0 || src_offset > src->nb_samples) { av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n", src_offset, dst_offset); return AVERROR(EINVAL); } /* check offsets and sizes to see if we can just do nothing and return */ if (nb_samples > src->nb_samples - src_offset) nb_samples = src->nb_samples - src_offset; if (nb_samples <= 0) return 0; /* validate that the output is not read-only */ if (dst->read_only) { av_log(dst, AV_LOG_ERROR, "dst is read-only\n"); return AVERROR(EINVAL); } /* reallocate output if necessary */ ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples); if (ret < 0) { av_log(dst, AV_LOG_ERROR, "error reallocating dst\n"); return ret; } dst_offset2 = dst_offset + nb_samples; dst_move_size = dst->nb_samples - dst_offset; for (p = 0; p < src->planes; p++) { if (dst_move_size > 0) { memmove(dst->data[p] + dst_offset2 * dst->stride, dst->data[p] + dst_offset * dst->stride, dst_move_size * dst->stride); } memcpy(dst->data[p] + dst_offset * dst->stride, src->data[p] + src_offset * src->stride, nb_samples * src->stride); } dst->nb_samples += nb_samples; return 0; } void ff_audio_data_drain(AudioData *a, int nb_samples) { if (a->nb_samples <= nb_samples) { /* drain the whole buffer */ a->nb_samples = 0; } else { int p; int move_offset = a->stride * nb_samples; int move_size = a->stride * (a->nb_samples - nb_samples); for (p = 0; p < a->planes; p++) memmove(a->data[p], a->data[p] + move_offset, move_size); a->nb_samples -= nb_samples; } } int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, int nb_samples) { uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS]; int offset_size, p; if (offset >= a->nb_samples) return 0; offset_size = offset * a->stride; for (p = 0; p < a->planes; p++) offset_data[p] = a->data[p] + offset_size; return av_audio_fifo_write(af, (void **)offset_data, nb_samples); } int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples) { int ret; if (a->read_only) return AVERROR(EINVAL); ret = ff_audio_data_realloc(a, nb_samples); if (ret < 0) return ret; ret = av_audio_fifo_read(af, (void **)a->data, nb_samples); if (ret >= 0) a->nb_samples = ret; return ret; }