diff options
author | Nicolas George <nicolas.george@normalesup.org> | 2013-03-10 16:44:46 +0100 |
---|---|---|
committer | Nicolas George <nicolas.george@normalesup.org> | 2013-03-10 19:39:25 +0100 |
commit | de54a96aa8f5959ebd87516d514c77055b369fed (patch) | |
tree | 491fb7fae2bd8178c0ff42e9777596716ff3d509 /libavfilter | |
parent | b71db3f38eafeb05e068804e002984b0d232e5f6 (diff) | |
download | ffmpeg-de54a96aa8f5959ebd87516d514c77055b369fed.tar.gz |
lavfi/buffersink: implement av_buffersink_get_samples().
Note: the implementation could be more efficient, but at
the cost of more diff.
Most of the code from the following commit:
commit a2cd9be212fca02dd3d6ee65cb6ab9f84c5e28e5
Author: Anton Khirnov <anton@khirnov.net>
Date: Fri May 4 19:22:38 2012 +0200
lavfi: add an audio buffer sink.
Adapted to call av_buffersink_get_frame_flags() instead of
accessing the frame directly.
Diffstat (limited to 'libavfilter')
-rw-r--r-- | libavfilter/buffersink.c | 72 |
1 files changed, 70 insertions, 2 deletions
diff --git a/libavfilter/buffersink.c b/libavfilter/buffersink.c index 9f92051c2b..35607850f0 100644 --- a/libavfilter/buffersink.c +++ b/libavfilter/buffersink.c @@ -23,7 +23,7 @@ * buffer sink */ -#include "libavutil/fifo.h" +#include "libavutil/audio_fifo.h" #include "libavutil/avassert.h" #include "libavutil/channel_layout.h" #include "libavutil/common.h" @@ -46,6 +46,10 @@ typedef struct { int64_t *channel_layouts; ///< list of accepted channel layouts, terminated by -1 int all_channel_counts; int *sample_rates; ///< list of accepted sample rates, terminated by -1 + + /* only used for compat API */ + AVAudioFifo *audio_fifo; ///< FIFO for audio samples + int64_t next_pts; ///< interpolating audio pts } BufferSinkContext; static av_cold void uninit(AVFilterContext *ctx) @@ -53,6 +57,9 @@ static av_cold void uninit(AVFilterContext *ctx) BufferSinkContext *sink = ctx->priv; AVFrame *frame; + if (sink->audio_fifo) + av_audio_fifo_free(sink->audio_fifo); + if (sink->fifo) { while (av_fifo_size(sink->fifo) >= sizeof(AVFilterBufferRef *)) { av_fifo_generic_read(sink->fifo, &frame, sizeof(frame), NULL); @@ -140,9 +147,70 @@ int av_buffersink_get_frame_flags(AVFilterContext *ctx, AVFrame *frame, int flag return 0; } +static int read_from_fifo(AVFilterContext *ctx, AVFrame *frame, + int nb_samples) +{ + BufferSinkContext *s = ctx->priv; + AVFilterLink *link = ctx->inputs[0]; + AVFrame *tmp; + + if (!(tmp = ff_get_audio_buffer(link, nb_samples))) + return AVERROR(ENOMEM); + av_audio_fifo_read(s->audio_fifo, (void**)tmp->extended_data, nb_samples); + + tmp->pts = s->next_pts; + s->next_pts += av_rescale_q(nb_samples, (AVRational){1, link->sample_rate}, + link->time_base); + + av_frame_move_ref(frame, tmp); + av_frame_free(&tmp); + + return 0; + +} + int av_buffersink_get_samples(AVFilterContext *ctx, AVFrame *frame, int nb_samples) { - av_assert0(!"TODO"); + BufferSinkContext *s = ctx->priv; + AVFilterLink *link = ctx->inputs[0]; + AVFrame *cur_frame; + int ret = 0; + + if (!s->audio_fifo) { + int nb_channels = link->channels; + if (!(s->audio_fifo = av_audio_fifo_alloc(link->format, nb_channels, nb_samples))) + return AVERROR(ENOMEM); + } + + while (ret >= 0) { + if (av_audio_fifo_size(s->audio_fifo) >= nb_samples) + return read_from_fifo(ctx, frame, nb_samples); + + if (!(cur_frame = av_frame_alloc())) + return AVERROR(ENOMEM); + ret = av_buffersink_get_frame_flags(ctx, cur_frame, 0); + if (ret == AVERROR_EOF && av_audio_fifo_size(s->audio_fifo)) { + av_frame_free(&cur_frame); + return read_from_fifo(ctx, frame, av_audio_fifo_size(s->audio_fifo)); + } else if (ret < 0) { + av_frame_free(&cur_frame); + return ret; + } + + if (cur_frame->pts != AV_NOPTS_VALUE) { + s->next_pts = cur_frame->pts - + av_rescale_q(av_audio_fifo_size(s->audio_fifo), + (AVRational){ 1, link->sample_rate }, + link->time_base); + } + + ret = av_audio_fifo_write(s->audio_fifo, (void**)cur_frame->extended_data, + cur_frame->nb_samples); + av_frame_free(&cur_frame); + } + + return ret; + } AVBufferSinkParams *av_buffersink_params_alloc(void) |