From e763d528e2b9370521b1e28450f48e65af33ca7f Mon Sep 17 00:00:00 2001 From: wm4 Date: Sun, 10 Nov 2013 23:18:31 +0100 Subject: af_lavfi: add support for non-interleaved audio --- audio/filter/af_lavfi.c | 54 ++++++++++++++++++++++--------------------------- 1 file changed, 24 insertions(+), 30 deletions(-) (limited to 'audio') diff --git a/audio/filter/af_lavfi.c b/audio/filter/af_lavfi.c index f08e7ff85c..381e2d0583 100644 --- a/audio/filter/af_lavfi.c +++ b/audio/filter/af_lavfi.c @@ -61,11 +61,9 @@ struct priv { // Guarantee that the data stays valid until next filter call char *out_buffer; - struct mp_audio data; struct mp_audio temp; - int64_t bytes_in; - int64_t bytes_out; + int64_t samples_in; AVRational timebase_out; @@ -129,6 +127,8 @@ static bool recreate_graph(struct af_instance *af, struct mp_audio *config) static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL, + AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }; r = av_opt_set_int_list(out, "sample_fmts", sample_fmts, @@ -181,7 +181,6 @@ static int control(struct af_instance *af, int cmd, void *arg) if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE) mp_audio_set_format(in, AF_FORMAT_FLOAT_NE); - mp_audio_force_interleaved_format(in); if (!mp_chmap_is_lavc(&in->channels)) mp_chmap_reorder_to_lavc(&in->channels); // will always work @@ -194,7 +193,6 @@ static int control(struct af_instance *af, int cmd, void *arg) out->rate = l_out->sample_rate; mp_audio_set_format(out, af_from_avformat(l_out->format)); - mp_audio_force_interleaved_format(out); struct mp_chmap out_cm; mp_chmap_from_lavc(&out_cm, l_out->channel_layout); @@ -220,28 +218,25 @@ static int control(struct af_instance *af, int cmd, void *arg) static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) { struct priv *p = af->priv; + struct mp_audio *r = af->data; AVFilterLink *l_in = p->in->outputs[0]; - struct mp_audio *r = &p->temp; - *r = *af->data; - - int in_frame_size = data->bps * data->channels.num; - int out_frame_size = r->bps * r->channels.num; - AVFrame *frame = av_frame_alloc(); frame->nb_samples = data->samples; frame->format = l_in->format; // Timebase is 1/sample_rate - frame->pts = p->bytes_in / in_frame_size; + frame->pts = p->samples_in; av_frame_set_channels(frame, l_in->channels); av_frame_set_channel_layout(frame, l_in->channel_layout); av_frame_set_sample_rate(frame, l_in->sample_rate); - frame->data[0] = data->planes[0]; frame->extended_data = frame->data; + for (int n = 0; n < data->num_planes; n++) + frame->data[n] = data->planes[n]; + frame->linesize[0] = frame->nb_samples * data->sstride; if (av_buffersrc_add_frame(p->in, frame) < 0) { av_frame_free(&frame); @@ -250,7 +245,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) av_frame_free(&frame); int64_t out_pts = AV_NOPTS_VALUE; - size_t out_len = 0; + r->samples = 0; for (;;) { frame = av_frame_alloc(); if (av_buffersink_get_frame(p->out, frame) < 0) { @@ -259,35 +254,32 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) break; } - size_t new_len = out_len + frame->nb_samples * out_frame_size; - if (new_len > talloc_get_size(p->out_buffer)) - p->out_buffer = talloc_realloc(p, p->out_buffer, char, new_len); - memcpy(p->out_buffer + out_len, frame->data[0], new_len - out_len); - out_len = new_len; + mp_audio_realloc_min(r, r->samples + frame->nb_samples); + for (int n = 0; n < r->num_planes; n++) { + memcpy((char *)r->planes[n] + r->samples * r->sstride, + frame->extended_data[n], frame->nb_samples * r->sstride); + } + r->samples += frame->nb_samples; + if (out_pts == AV_NOPTS_VALUE) out_pts = frame->pts; av_frame_free(&frame); } - r->planes[0] = p->out_buffer; - r->samples = out_len / r->sstride; - - p->bytes_in += data->samples * data->sstride; - p->bytes_out += r->samples * r->sstride; + p->samples_in += data->samples; if (out_pts != AV_NOPTS_VALUE) { - int64_t num_in_frames = p->bytes_in / in_frame_size; - double in_time = num_in_frames / (double)data->rate; - + double in_time = p->samples_in / (double)data->rate; double out_time = out_pts * av_q2d(p->timebase_out); // Need pts past the last output sample. out_time += r->samples / (double)r->rate; - af->delay = (in_time - out_time) * r->rate * out_frame_size; + af->delay = (in_time - out_time) * r->rate * r->sstride; } - return r; + p->temp = *r; + return &p->temp; } static void uninit(struct af_instance *af) @@ -301,7 +293,9 @@ static int af_open(struct af_instance *af) af->play = play; af->mul = 1; struct priv *priv = af->priv; - af->data = &priv->data; + af->data = talloc_zero(priv, struct mp_audio), + // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data + assert(MP_NUM_CHANNELS <= AV_NUM_DATA_POINTERS); return AF_OK; } -- cgit v1.2.3