[FFmpeg-devel] [PATCH 5/5] amerge: accept multiple inputs.
Stefano Sabatini
stefasab at gmail.com
Tue Jun 5 17:01:44 CEST 2012
On date Sunday 2012-06-03 21:44:50 +0200, Nicolas George encoded:
>
> Signed-off-by: Nicolas George <nicolas.george at normalesup.org>
> ---
> doc/filters.texi | 22 +++---
> libavfilter/af_amerge.c | 178 ++++++++++++++++++++++++++++++++---------------
> 2 files changed, 132 insertions(+), 68 deletions(-)
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 324a154..3aecc5e 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -168,9 +168,16 @@ aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo
>
> @section amerge
>
> -Merge two audio streams into a single multi-channel stream.
> +Merge two audio streams or more into a single multi-channel stream.
Nit: two or more audio streams ...
sounds nicer to my Englian ears.
>
> -This filter does not need any argument.
> +The filter accepts the following named options:
> +
> + at table @option
> +
> + at item inputs
> +Number of inputs. Default is 2.
Nit: Set the number of input.
> +
> + at end table
>
> If the channel layouts of the inputs are disjoint, and therefore compatible,
> the channel layout of the output will be set accordingly and the channels
> @@ -189,7 +196,7 @@ On the other hand, if both input are in stereo, the output channels will be
> in the default order: a1, a2, b1, b2, and the channel layout will be
> arbitrarily set to 4.0, which may or may not be the expected value.
>
> -Both inputs must have the same sample rate, and format.
> +All inputs must have the same sample rate, and format.
>
> If inputs do not have the same duration, the output will stop with the
> shortest.
> @@ -199,8 +206,7 @@ Example: merge two mono files into a stereo stream:
> amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
> @end example
>
> -If you need to do multiple merges (for instance multiple mono audio streams in
> -a single video media), you can do:
> +Example: multiple merges:
> @example
> ffmpeg -f lavfi -i "
> amovie=input.mkv:si=0 [a0];
> @@ -209,11 +215,7 @@ amovie=input.mkv:si=2 [a2];
> amovie=input.mkv:si=3 [a3];
> amovie=input.mkv:si=4 [a4];
> amovie=input.mkv:si=5 [a5];
> -[a0][a1] amerge [x0];
> -[x0][a2] amerge [x1];
> -[x1][a3] amerge [x2];
> -[x2][a4] amerge [x3];
> -[x3][a5] amerge" -c:a pcm_s16le output.mkv
> +[a0][a1][a2][a3][a4][a5] amerge=inputs=6" -c:a pcm_s16le output.mkv
> @end example
>
> @section amix
> diff --git a/libavfilter/af_amerge.c b/libavfilter/af_amerge.c
> index 394067e..364534a 100644
> --- a/libavfilter/af_amerge.c
> +++ b/libavfilter/af_amerge.c
> @@ -23,6 +23,8 @@
> * Audio merging filter
> */
>
> +#include "libavutil/opt.h"
> +#include "libavutil/bprint.h"
Nit++: alphabetical order
> #include "libswresample/swresample.h" // only for SWR_CH_MAX
> #include "avfilter.h"
> #include "audio.h"
> @@ -30,6 +32,8 @@
> #include "internal.h"
>
> typedef struct {
> + const AVClass *class;
> + int nb_inputs;
> int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
> int bps;
> struct amerge_input {
> @@ -37,27 +41,46 @@ typedef struct {
> int nb_ch; /**< number of channels for the input */
> int nb_samples;
> int pos;
> - } in[2];
> + } *in;
> } AMergeContext;
>
> +#define OFFSET(x) offsetof(AMergeContext, x)
> +
> +static const AVOption amerge_options[] = {
> + { "inputs", "specify the number of inputs", OFFSET(nb_inputs),
> + AV_OPT_TYPE_INT, { .dbl = 2 }, 2, SWR_CH_MAX },
> +};
Also maybe a short alias ("n"?) may be useful.
> +
> +static const char *amerge_get_name(void *ctx)
> +{
> + return "amerge";
> +}
> +
> +static const AVClass amerge_class = {
> + .class_name = "AMergeContext",
> + .item_name = amerge_get_name,
av_default_item_name() should be enough (and class_name = "amerge").
> + .option = amerge_options,
> +};
> +
> static av_cold void uninit(AVFilterContext *ctx)
> {
> AMergeContext *am = ctx->priv;
> int i;
>
> - for (i = 0; i < 2; i++)
> + for (i = 0; i < am->nb_inputs; i++)
> ff_bufqueue_discard_all(&am->in[i].queue);
> + av_freep(&am->in);
> }
>
> static int query_formats(AVFilterContext *ctx)
> {
> AMergeContext *am = ctx->priv;
> - int64_t inlayout[2], outlayout;
> + int64_t inlayout[SWR_CH_MAX], outlayout = 0;
> AVFilterFormats *formats;
> AVFilterChannelLayouts *layouts;
> - int i;
> + int i, overlap = 0, nb_ch = 0;
>
> - for (i = 0; i < 2; i++) {
> + for (i = 0; i < am->nb_inputs; i++) {
> if (!ctx->inputs[i]->in_channel_layouts ||
> !ctx->inputs[i]->in_channel_layouts->nb_channel_layouts) {
> av_log(ctx, AV_LOG_ERROR,
> @@ -71,33 +94,38 @@ static int query_formats(AVFilterContext *ctx)
> av_log(ctx, AV_LOG_INFO, "Using \"%s\" for input %d\n", buf, i + 1);
> }
> am->in[i].nb_ch = av_get_channel_layout_nb_channels(inlayout[i]);
> + if (outlayout & inlayout[i])
> + overlap++;
> + outlayout |= inlayout[i];
> + nb_ch += am->in[i].nb_ch;
> }
> - if (am->in[0].nb_ch + am->in[1].nb_ch > SWR_CH_MAX) {
> + if (nb_ch > SWR_CH_MAX) {
> av_log(ctx, AV_LOG_ERROR, "Too many channels (max %d)\n", SWR_CH_MAX);
> return AVERROR(EINVAL);
> }
> - if (inlayout[0] & inlayout[1]) {
> + if (overlap) {
> av_log(ctx, AV_LOG_WARNING,
> "Inputs overlap: output layout will be meaningless\n");
> - for (i = 0; i < am->in[0].nb_ch + am->in[1].nb_ch; i++)
> + for (i = 0; i < nb_ch; i++)
> am->route[i] = i;
> - outlayout = av_get_default_channel_layout(am->in[0].nb_ch +
> - am->in[1].nb_ch);
> + outlayout = av_get_default_channel_layout(nb_ch);
> if (!outlayout)
> - outlayout = ((int64_t)1 << (am->in[0].nb_ch + am->in[1].nb_ch)) - 1;
> + outlayout = ((int64_t)1 << nb_ch) - 1;
> } else {
> - int *route[2] = { am->route, am->route + am->in[0].nb_ch };
> + int *route[SWR_CH_MAX];
> int c, out_ch_number = 0;
>
> - outlayout = inlayout[0] | inlayout[1];
> + route[0] = am->route;
> + for (i = 1; i < am->nb_inputs; i++)
> + route[i] = route[i - 1] + am->in[i - 1].nb_ch;
> for (c = 0; c < 64; c++)
> - for (i = 0; i < 2; i++)
> + for (i = 0; i < am->nb_inputs; i++)
> if ((inlayout[i] >> c) & 1)
> *(route[i]++) = out_ch_number++;
> }
> formats = avfilter_make_format_list(ff_packed_sample_fmts);
> avfilter_set_common_sample_formats(ctx, formats);
> - for (i = 0; i < 2; i++) {
> + for (i = 0; i < am->nb_inputs; i++) {
> layouts = NULL;
> ff_add_channel_layout(&layouts, inlayout[i]);
> ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts);
> @@ -113,26 +141,31 @@ static int config_output(AVFilterLink *outlink)
> {
> AVFilterContext *ctx = outlink->src;
> AMergeContext *am = ctx->priv;
> - int64_t layout;
> - char name[3][256];
> + AVBPrint bp;
> int i;
>
> - if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
> - av_log(ctx, AV_LOG_ERROR,
> - "Inputs must have the same sample rate "
> - "(%"PRIi64" vs %"PRIi64")\n",
> - ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
> - return AVERROR(EINVAL);
> + for (i = 1; i < am->nb_inputs; i++) {
> + if (ctx->inputs[i]->sample_rate != ctx->inputs[0]->sample_rate) {
> + av_log(ctx, AV_LOG_ERROR,
> + "Inputs must have the same sample rate "
> + "(%"PRIi64" for in%d vs %"PRIi64")\n",
> + ctx->inputs[i]->sample_rate, i, ctx->inputs[0]->sample_rate);
> + return AVERROR(EINVAL);
> + }
> }
> am->bps = av_get_bytes_per_sample(ctx->outputs[0]->format);
> outlink->sample_rate = ctx->inputs[0]->sample_rate;
> outlink->time_base = ctx->inputs[0]->time_base;
> - for (i = 0; i < 3; i++) {
> - layout = (i < 2 ? ctx->inputs[i] : ctx->outputs[0])->channel_layout;
> - av_get_channel_layout_string(name[i], sizeof(name[i]), -1, layout);
> +
> + av_bprint_init(&bp, 0, 1);
> + for (i = 0; i < am->nb_inputs; i++) {
> + av_bprintf(&bp, "%sin%d:", i ? " + " : "", i);
> + av_bprint_channel_layout(&bp, -1, ctx->inputs[i]->channel_layout);
> }
> - av_log(ctx, AV_LOG_INFO,
> - "in1:%s + in2:%s -> out:%s\n", name[0], name[1], name[2]);
> + av_bprintf(&bp, " -> out:");
> + av_bprint_channel_layout(&bp, -1, ctx->outputs[0]->channel_layout);
> + av_log(ctx, AV_LOG_INFO, "%s\n", bp.str);
> +
> return 0;
> }
>
> @@ -142,7 +175,7 @@ static int request_frame(AVFilterLink *outlink)
> AMergeContext *am = ctx->priv;
> int i, ret;
>
> - for (i = 0; i < 2; i++)
> + for (i = 0; i < am->nb_inputs; i++)
> if (!am->in[i].nb_samples)
> if ((ret = avfilter_request_frame(ctx->inputs[i])) < 0)
> return ret;
> @@ -150,7 +183,8 @@ static int request_frame(AVFilterLink *outlink)
> }
>
> /**
> - * Copy samples from two input streams to one output stream.
> + * Copy samples from several input streams to one output stream.
> + * @param nb_inputs number of inputs
> * @param in inputs; used only for the nb_ch field;
> * @param route routing values;
> * input channel i goes to output channel route[i];
> @@ -164,21 +198,24 @@ static int request_frame(AVFilterLink *outlink)
> * @param ns number of samples to copy
> * @param bps bytes per sample
> */
> -static inline void copy_samples(struct amerge_input in[2], int *route, uint8_t *ins[2],
> +static inline void copy_samples(int nb_inputs, struct amerge_input in[],
> + int *route, uint8_t *ins[],
> uint8_t **outs, int ns, int bps)
> {
> int *route_cur;
> - int i, c;
> + int i, c, nb_ch = 0;
>
> + for (i = 0; i < nb_inputs; i++)
> + nb_ch += in[i].nb_ch;
> while (ns--) {
> route_cur = route;
> - for (i = 0; i < 2; i++) {
> + for (i = 0; i < nb_inputs; i++) {
> for (c = 0; c < in[i].nb_ch; c++) {
> memcpy((*outs) + bps * *(route_cur++), ins[i], bps);
> ins[i] += bps;
> }
> }
> - *outs += (in[0].nb_ch + in[1].nb_ch) * bps;
> + *outs += nb_ch * bps;
> }
> }
>
> @@ -187,21 +224,26 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
> AVFilterContext *ctx = inlink->dst;
> AMergeContext *am = ctx->priv;
> AVFilterLink *const outlink = ctx->outputs[0];
> - int input_number = inlink == ctx->inputs[1];
> + int input_number;
> int nb_samples, ns, i;
> - AVFilterBufferRef *outbuf, *inbuf[2];
> - uint8_t *ins[2], *outs;
> + AVFilterBufferRef *outbuf, *inbuf[SWR_CH_MAX];
> + uint8_t *ins[SWR_CH_MAX], *outs;
>
> + for (input_number = 0; input_number < am->nb_inputs; input_number++)
> + if (inlink == ctx->inputs[input_number])
> + break;
> + av_assert1(input_number < am->nb_inputs);
> ff_bufqueue_add(ctx, &am->in[input_number].queue, insamples);
> am->in[input_number].nb_samples += insamples->audio->nb_samples;
> - if (!am->in[!input_number].nb_samples)
> + nb_samples = am->in[0].nb_samples;
> + for (i = 1; i < am->nb_inputs; i++)
> + nb_samples = FFMIN(nb_samples, am->in[i].nb_samples);
> + if (!nb_samples)
> return;
>
> - nb_samples = FFMIN(am->in[0].nb_samples,
> - am->in[1].nb_samples);
> outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples);
> outs = outbuf->data[0];
> - for (i = 0; i < 2; i++) {
> + for (i = 0; i < am->nb_inputs; i++) {
> inbuf[i] = ff_bufqueue_peek(&am->in[i].queue, 0);
> ins[i] = inbuf[i]->data[0] +
> am->in[i].pos * am->in[i].nb_ch * am->bps;
> @@ -218,27 +260,27 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
>
> while (nb_samples) {
> ns = nb_samples;
> - for (i = 0; i < 2; i++)
> + for (i = 0; i < am->nb_inputs; i++)
> ns = FFMIN(ns, inbuf[i]->audio->nb_samples - am->in[i].pos);
> /* Unroll the most common sample formats: speed +~350% for the loop,
> +~13% overall (including two common decoders) */
> switch (am->bps) {
> case 1:
> - copy_samples(am->in, am->route, ins, &outs, ns, 1);
> + copy_samples(am->nb_inputs, am->in, am->route, ins, &outs, ns, 1);
> break;
> case 2:
> - copy_samples(am->in, am->route, ins, &outs, ns, 2);
> + copy_samples(am->nb_inputs, am->in, am->route, ins, &outs, ns, 2);
> break;
> case 4:
> - copy_samples(am->in, am->route, ins, &outs, ns, 4);
> + copy_samples(am->nb_inputs, am->in, am->route, ins, &outs, ns, 4);
> break;
> default:
> - copy_samples(am->in, am->route, ins, &outs, ns, am->bps);
> + copy_samples(am->nb_inputs, am->in, am->route, ins, &outs, ns, am->bps);
> break;
> }
>
> nb_samples -= ns;
> - for (i = 0; i < 2; i++) {
> + for (i = 0; i < am->nb_inputs; i++) {
> am->in[i].nb_samples -= ns;
> am->in[i].pos += ns;
> if (am->in[i].pos == inbuf[i]->audio->nb_samples) {
> @@ -253,25 +295,45 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
> ff_filter_samples(ctx->outputs[0], outbuf);
> }
>
> +static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
> +{
> + AMergeContext *am = ctx->priv;
> + int ret, i;
> + char name[16];
> +
> + am->class = &amerge_class;
> + av_opt_set_defaults(am);
> + ret = av_set_options_string(am, args, "=", ":");
> + if (ret < 0) {
> + av_log(ctx, AV_LOG_ERROR, "Error parsing options: '%s'\n", args);
> + return ret;
> + }
> + am->in = av_calloc(am->nb_inputs, sizeof(*am->in));
> + if (!am->in)
> + return AVERROR(ENOMEM);
> + for (i = 0; i < am->nb_inputs; i++) {
> + AVFilterPad pad = {
> + .name = name,
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_samples = filter_samples,
> + .min_perms = AV_PERM_READ | AV_PERM_PRESERVE,
> + };
> + snprintf(name, sizeof(name), "in%d", i);
> + avfilter_insert_inpad(ctx, i, &pad);
Uhm... is this really working? Shouldn't you strdup the name?
[...]
--
FFmpeg = Fanciful & Fierce Magical Purposeless Everlasting Governor
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