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authorMina Nagy Zaki <mnzaki@gmail.com>2011-08-01 11:33:26 +0300
committerStefano Sabatini <stefano.sabatini-lala@poste.it>2011-08-21 11:37:57 +0200
commit587c8ab9128455ccf2580c5350992e4a402dc8fd (patch)
tree90e8b2389003a3fde082987cbfdd3cb6e3bcdd56 /libavfilter/asrc_abuffer.c
parentf138c7f993e1aaf5223c546da5292993a467ee8d (diff)
lavfi: add asrc_abuffer - audio buffer source
Originally based on code by Stefano Sabatini and S. N. Hemanth. Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Diffstat (limited to 'libavfilter/asrc_abuffer.c')
-rw-r--r--libavfilter/asrc_abuffer.c366
1 files changed, 366 insertions, 0 deletions
diff --git a/libavfilter/asrc_abuffer.c b/libavfilter/asrc_abuffer.c
new file mode 100644
index 0000000000..badc2d8adf
--- /dev/null
+++ b/libavfilter/asrc_abuffer.c
@@ -0,0 +1,366 @@
+/*
+ * Copyright (c) 2010 S.N. Hemanth Meenakshisundaram
+ * Copyright (c) 2011 Mina Nagy Zaki
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * memory buffer source for audio
+ */
+
+#include "libavutil/audioconvert.h"
+#include "libavutil/fifo.h"
+#include "asrc_abuffer.h"
+#include "internal.h"
+
+typedef struct {
+ // Audio format of incoming buffers
+ int sample_rate;
+ unsigned int sample_format;
+ int64_t channel_layout;
+ int packing_format;
+
+ // FIFO buffer of audio buffer ref pointers
+ AVFifoBuffer *fifo;
+
+ // Normalization filters
+ AVFilterContext *aconvert;
+ AVFilterContext *aresample;
+} ABufferSourceContext;
+
+#define FIFO_SIZE 8
+
+static void buf_free(AVFilterBuffer *ptr)
+{
+ av_free(ptr);
+ return;
+}
+
+static void set_link_source(AVFilterContext *src, AVFilterLink *link)
+{
+ link->src = src;
+ link->srcpad = &(src->output_pads[0]);
+ src->outputs[0] = link;
+}
+
+static int reconfigure_filter(ABufferSourceContext *abuffer, AVFilterContext *filt_ctx)
+{
+ int ret;
+ AVFilterLink * const inlink = filt_ctx->inputs[0];
+ AVFilterLink * const outlink = filt_ctx->outputs[0];
+
+ inlink->format = abuffer->sample_format;
+ inlink->channel_layout = abuffer->channel_layout;
+ inlink->planar = abuffer->packing_format;
+ inlink->sample_rate = abuffer->sample_rate;
+
+ filt_ctx->filter->uninit(filt_ctx);
+ memset(filt_ctx->priv, 0, filt_ctx->filter->priv_size);
+ if ((ret = filt_ctx->filter->init(filt_ctx, NULL , NULL)) < 0)
+ return ret;
+ if ((ret = inlink->srcpad->config_props(inlink)) < 0)
+ return ret;
+ return outlink->srcpad->config_props(outlink);
+}
+
+static int insert_filter(ABufferSourceContext *abuffer,
+ AVFilterLink *link, AVFilterContext **filt_ctx,
+ const char *filt_name)
+{
+ int ret;
+
+ if ((ret = avfilter_open(filt_ctx, avfilter_get_by_name(filt_name), NULL)) < 0)
+ return ret;
+
+ link->src->outputs[0] = NULL;
+ if ((ret = avfilter_link(link->src, 0, *filt_ctx, 0)) < 0) {
+ link->src->outputs[0] = link;
+ return ret;
+ }
+
+ set_link_source(*filt_ctx, link);
+
+ if ((ret = reconfigure_filter(abuffer, *filt_ctx)) < 0) {
+ avfilter_free(*filt_ctx);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void remove_filter(AVFilterContext **filt_ctx)
+{
+ AVFilterLink *outlink = (*filt_ctx)->outputs[0];
+ AVFilterContext *src = (*filt_ctx)->inputs[0]->src;
+
+ (*filt_ctx)->outputs[0] = NULL;
+ avfilter_free(*filt_ctx);
+ *filt_ctx = NULL;
+
+ set_link_source(src, outlink);
+}
+
+static inline void log_input_change(void *ctx, AVFilterLink *link, AVFilterBufferRef *ref)
+{
+ char old_layout_str[16], new_layout_str[16];
+ av_get_channel_layout_string(old_layout_str, sizeof(old_layout_str),
+ -1, link->channel_layout);
+ av_get_channel_layout_string(new_layout_str, sizeof(new_layout_str),
+ -1, ref->audio->channel_layout);
+ av_log(ctx, AV_LOG_INFO,
+ "Audio input format changed: "
+ "%s:%s:%"PRId64" -> %s:%s:%u, normalizing\n",
+ av_get_sample_fmt_name(link->format),
+ old_layout_str, link->sample_rate,
+ av_get_sample_fmt_name(ref->format),
+ new_layout_str, ref->audio->sample_rate);
+}
+
+int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *ctx,
+ AVFilterBufferRef *samplesref,
+ int av_unused flags)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ AVFilterLink *link;
+ int ret, logged = 0;
+
+ if (av_fifo_space(abuffer->fifo) < sizeof(samplesref)) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Buffering limit reached. Please consume some available frames "
+ "before adding new ones.\n");
+ return AVERROR(EINVAL);
+ }
+
+ // Normalize input
+
+ link = ctx->outputs[0];
+ if (samplesref->audio->sample_rate != link->sample_rate) {
+
+ log_input_change(ctx, link, samplesref);
+ logged = 1;
+
+ abuffer->sample_rate = samplesref->audio->sample_rate;
+
+ if (!abuffer->aresample) {
+ ret = insert_filter(abuffer, link, &abuffer->aresample, "aresample");
+ if (ret < 0) return ret;
+ } else {
+ link = abuffer->aresample->outputs[0];
+ if (samplesref->audio->sample_rate == link->sample_rate)
+ remove_filter(&abuffer->aresample);
+ else
+ if ((ret = reconfigure_filter(abuffer, abuffer->aresample)) < 0)
+ return ret;
+ }
+ }
+
+ link = ctx->outputs[0];
+ if (samplesref->format != link->format ||
+ samplesref->audio->channel_layout != link->channel_layout ||
+ samplesref->audio->planar != link->planar) {
+
+ if (!logged) log_input_change(ctx, link, samplesref);
+
+ abuffer->sample_format = samplesref->format;
+ abuffer->channel_layout = samplesref->audio->channel_layout;
+ abuffer->packing_format = samplesref->audio->planar;
+
+ if (!abuffer->aconvert) {
+ ret = insert_filter(abuffer, link, &abuffer->aconvert, "aconvert");
+ if (ret < 0) return ret;
+ } else {
+ link = abuffer->aconvert->outputs[0];
+ if (samplesref->format == link->format &&
+ samplesref->audio->channel_layout == link->channel_layout &&
+ samplesref->audio->planar == link->planar
+ )
+ remove_filter(&abuffer->aconvert);
+ else
+ if ((ret = reconfigure_filter(abuffer, abuffer->aconvert)) < 0)
+ return ret;
+ }
+ }
+
+ if (sizeof(samplesref) != av_fifo_generic_write(abuffer->fifo, &samplesref,
+ sizeof(samplesref), NULL)) {
+ av_log(ctx, AV_LOG_ERROR, "Error while writing to FIFO\n");
+ return AVERROR(EINVAL);
+ }
+
+ return 0;
+}
+
+int av_asrc_buffer_add_samples(AVFilterContext *ctx,
+ uint8_t *data[8], int linesize[8],
+ int nb_samples, int sample_rate,
+ int sample_fmt, int64_t channel_layout, int planar,
+ int64_t pts, int av_unused flags)
+{
+ AVFilterBufferRef *samplesref;
+
+ samplesref = avfilter_get_audio_buffer_ref_from_arrays(
+ data, linesize, AV_PERM_WRITE,
+ nb_samples,
+ sample_fmt, channel_layout, planar);
+ if (!samplesref)
+ return AVERROR(ENOMEM);
+
+ samplesref->buf->free = buf_free;
+ samplesref->pts = pts;
+ samplesref->audio->sample_rate = sample_rate;
+
+ return av_asrc_buffer_add_audio_buffer_ref(ctx, samplesref, 0);
+}
+
+int av_asrc_buffer_add_buffer(AVFilterContext *ctx,
+ uint8_t *buf, int buf_size, int sample_rate,
+ int sample_fmt, int64_t channel_layout, int planar,
+ int64_t pts, int av_unused flags)
+{
+ uint8_t *data[8];
+ int linesize[8];
+ int nb_channels = av_get_channel_layout_nb_channels(channel_layout),
+ nb_samples = buf_size / nb_channels / av_get_bytes_per_sample(sample_fmt);
+
+ av_samples_fill_arrays(data, linesize,
+ buf, nb_channels, nb_samples,
+ sample_fmt, planar, 16);
+
+ return av_asrc_buffer_add_samples(ctx,
+ data, linesize, nb_samples,
+ sample_rate,
+ sample_fmt, channel_layout, planar,
+ pts, flags);
+}
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ char *arg = NULL, *ptr, chlayout_str[16];
+ int ret;
+
+ arg = strtok_r(args, ":", &ptr);
+
+#define ADD_FORMAT(fmt_name) \
+ if (!arg) \
+ goto arg_fail; \
+ if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0) \
+ return ret; \
+ if (*args) \
+ arg = strtok_r(NULL, ":", &ptr)
+
+ ADD_FORMAT(sample_rate);
+ ADD_FORMAT(sample_format);
+ ADD_FORMAT(channel_layout);
+ ADD_FORMAT(packing_format);
+
+ abuffer->fifo = av_fifo_alloc(FIFO_SIZE*sizeof(AVFilterBufferRef*));
+ if (!abuffer->fifo) {
+ av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo, filter init failed.\n");
+ return AVERROR(ENOMEM);
+ }
+
+ av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str),
+ -1, abuffer->channel_layout);
+ av_log(ctx, AV_LOG_INFO, "format:%s layout:%s rate:%d\n",
+ av_get_sample_fmt_name(abuffer->sample_format), chlayout_str,
+ abuffer->sample_rate);
+
+ return 0;
+
+arg_fail:
+ av_log(ctx, AV_LOG_ERROR, "Invalid arguments, must be of the form "
+ "sample_rate:sample_fmt:channel_layout:packing\n");
+ return AVERROR(EINVAL);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ av_fifo_free(abuffer->fifo);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ AVFilterFormats *formats;
+
+ formats = NULL;
+ avfilter_add_format(&formats, abuffer->sample_format);
+ avfilter_set_common_sample_formats(ctx, formats);
+
+ formats = NULL;
+ avfilter_add_format(&formats, abuffer->channel_layout);
+ avfilter_set_common_channel_layouts(ctx, formats);
+
+ formats = NULL;
+ avfilter_add_format(&formats, abuffer->packing_format);
+ avfilter_set_common_packing_formats(ctx, formats);
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ ABufferSourceContext *abuffer = outlink->src->priv;
+ outlink->sample_rate = abuffer->sample_rate;
+ return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ ABufferSourceContext *abuffer = outlink->src->priv;
+ AVFilterBufferRef *samplesref;
+
+ if (!av_fifo_size(abuffer->fifo)) {
+ av_log(outlink->src, AV_LOG_ERROR,
+ "request_frame() called with no available frames!\n");
+ return AVERROR(EINVAL);
+ }
+
+ av_fifo_generic_read(abuffer->fifo, &samplesref, sizeof(samplesref), NULL);
+ avfilter_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0));
+ avfilter_unref_buffer(samplesref);
+
+ return 0;
+}
+
+static int poll_frame(AVFilterLink *outlink)
+{
+ ABufferSourceContext *abuffer = outlink->src->priv;
+ return av_fifo_size(abuffer->fifo)/sizeof(AVFilterBufferRef*);
+}
+
+AVFilter avfilter_asrc_abuffer = {
+ .name = "abuffer",
+ .description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them accessible to the filterchain."),
+ .priv_size = sizeof(ABufferSourceContext),
+ .query_formats = query_formats,
+
+ .init = init,
+ .uninit = uninit,
+
+ .inputs = (AVFilterPad[]) {{ .name = NULL }},
+ .outputs = (AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .poll_frame = poll_frame,
+ .config_props = config_output, },
+ { .name = NULL}},
+};