diff options
-rw-r--r-- | libavfilter/avfilter.h | 9 | ||||
-rw-r--r-- | libavfilter/fifo.c | 159 |
2 files changed, 160 insertions, 8 deletions
diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h index c92f7e1..f09f086 100644 --- a/libavfilter/avfilter.h +++ b/libavfilter/avfilter.h @@ -595,6 +595,15 @@ struct AVFilterLink { AVFilterFormats *out_samplerates; struct AVFilterChannelLayouts *in_channel_layouts; struct AVFilterChannelLayouts *out_channel_layouts; + + /** + * Audio only, the destination filter sets this to a non-zero value to + * request that buffers with the given number of samples should be sent to + * it. AVFilterPad.needs_fifo must also be set on the corresponding input + * pad. + * Last buffer before EOF will be padded with silence. + */ + int request_samples; }; /** diff --git a/libavfilter/fifo.c b/libavfilter/fifo.c index 3fa4faa..6d28757 100644 --- a/libavfilter/fifo.c +++ b/libavfilter/fifo.c @@ -23,6 +23,11 @@ * FIFO buffering filter */ +#include "libavutil/avassert.h" +#include "libavutil/audioconvert.h" +#include "libavutil/mathematics.h" +#include "libavutil/samplefmt.h" + #include "audio.h" #include "avfilter.h" #include "internal.h" @@ -36,6 +41,13 @@ typedef struct Buf { typedef struct { Buf root; Buf *last; ///< last buffered frame + + /** + * When a specific number of output samples is requested, the partial + * buffer is stored here + */ + AVFilterBufferRef *buf_out; + int allocated_samples; ///< number of samples buf_out was allocated for } FifoContext; static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque) @@ -57,6 +69,8 @@ static av_cold void uninit(AVFilterContext *ctx) avfilter_unref_buffer(buf->buf); av_free(buf); } + + avfilter_unref_buffer(fifo->buf_out); } static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf) @@ -68,14 +82,143 @@ static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf) fifo->last->buf = buf; } +static void queue_pop(FifoContext *s) +{ + Buf *tmp = s->root.next->next; + if (s->last == s->root.next) + s->last = &s->root; + av_freep(&s->root.next); + s->root.next = tmp; +} + static void end_frame(AVFilterLink *inlink) { } static void draw_slice(AVFilterLink *inlink, int y, int h, int slice_dir) { } +/** + * Move data pointers and pts offset samples forward. + */ +static void buffer_offset(AVFilterLink *link, AVFilterBufferRef *buf, + int offset) +{ + int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout); + int planar = av_sample_fmt_is_planar(link->format); + int planes = planar ? nb_channels : 1; + int block_align = av_get_bytes_per_sample(link->format) * (planar ? 1 : nb_channels); + int i; + + av_assert0(buf->audio->nb_samples > offset); + + for (i = 0; i < planes; i++) + buf->extended_data[i] += block_align*offset; + if (buf->data != buf->extended_data) + memcpy(buf->data, buf->extended_data, + FFMIN(planes, FF_ARRAY_ELEMS(buf->data)) * sizeof(*buf->data)); + buf->linesize[0] -= block_align*offset; + buf->audio->nb_samples -= offset; + + if (buf->pts != AV_NOPTS_VALUE) { + buf->pts += av_rescale_q(offset, (AVRational){1, link->sample_rate}, + link->time_base); + } +} + +static int calc_ptr_alignment(AVFilterBufferRef *buf) +{ + int planes = av_sample_fmt_is_planar(buf->format) ? + av_get_channel_layout_nb_channels(buf->audio->channel_layout) : 1; + int min_align = 128; + int p; + + for (p = 0; p < planes; p++) { + int cur_align = 128; + while ((intptr_t)buf->extended_data[p] % cur_align) + cur_align >>= 1; + if (cur_align < min_align) + min_align = cur_align; + } + return min_align; +} + +static int return_audio_frame(AVFilterContext *ctx) +{ + AVFilterLink *link = ctx->outputs[0]; + FifoContext *s = ctx->priv; + AVFilterBufferRef *head = s->root.next->buf; + AVFilterBufferRef *buf_out; + int ret; + + if (!s->buf_out && + head->audio->nb_samples >= link->request_samples && + calc_ptr_alignment(head) >= 32) { + if (head->audio->nb_samples == link->request_samples) { + buf_out = head; + queue_pop(s); + } else { + buf_out = avfilter_ref_buffer(head, AV_PERM_READ); + buf_out->audio->nb_samples = link->request_samples; + buffer_offset(link, head, link->request_samples); + } + } else { + int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout); + + if (!s->buf_out) { + s->buf_out = ff_get_audio_buffer(link, AV_PERM_WRITE, + link->request_samples); + if (!s->buf_out) + return AVERROR(ENOMEM); + + s->buf_out->audio->nb_samples = 0; + s->buf_out->pts = head->pts; + s->allocated_samples = link->request_samples; + } else if (link->request_samples != s->allocated_samples) { + av_log(ctx, AV_LOG_ERROR, "request_samples changed before the " + "buffer was returned.\n"); + return AVERROR(EINVAL); + } + + while (s->buf_out->audio->nb_samples < s->allocated_samples) { + int len = FFMIN(s->allocated_samples - s->buf_out->audio->nb_samples, + head->audio->nb_samples); + + av_samples_copy(s->buf_out->extended_data, head->extended_data, + s->buf_out->audio->nb_samples, 0, len, nb_channels, + link->format); + s->buf_out->audio->nb_samples += len; + + if (len == head->audio->nb_samples) { + avfilter_unref_buffer(head); + queue_pop(s); + + if (!s->root.next && + (ret = ff_request_frame(ctx->inputs[0])) < 0) { + if (ret == AVERROR_EOF) { + av_samples_set_silence(s->buf_out->extended_data, + s->buf_out->audio->nb_samples, + s->allocated_samples - + s->buf_out->audio->nb_samples, + nb_channels, link->format); + s->buf_out->audio->nb_samples = s->allocated_samples; + break; + } + return ret; + } + head = s->root.next->buf; + } else { + buffer_offset(link, head, len); + } + } + buf_out = s->buf_out; + s->buf_out = NULL; + } + ff_filter_samples(link, buf_out); + + return 0; +} + static int request_frame(AVFilterLink *outlink) { FifoContext *fifo = outlink->src->priv; - Buf *tmp; int ret; if (!fifo->root.next) { @@ -90,20 +233,20 @@ static int request_frame(AVFilterLink *outlink) ff_start_frame(outlink, fifo->root.next->buf); ff_draw_slice (outlink, 0, outlink->h, 1); ff_end_frame (outlink); + queue_pop(fifo); break; case AVMEDIA_TYPE_AUDIO: - ff_filter_samples(outlink, fifo->root.next->buf); + if (outlink->request_samples) { + return return_audio_frame(outlink->src); + } else { + ff_filter_samples(outlink, fifo->root.next->buf); + queue_pop(fifo); + } break; default: return AVERROR(EINVAL); } - if (fifo->last == fifo->root.next) - fifo->last = &fifo->root; - tmp = fifo->root.next->next; - av_free(fifo->root.next); - fifo->root.next = tmp; - return 0; } |