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mpv/audio/decode/ad_spdif.c
wm4 5fd8a1e04c audio: make decoders output refcounted frames
This rewrites the audio decode loop to some degree. Audio filters don't
do refcounted frames yet, so af.c contains a hacky "emulation".

Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of
estimating how much audio we need to filter, we always filter full
frames. Maybe this should be adjusted later: in case filtering increases
the volume of the audio data, we should try not to buffer too much
filter output by reducing the input that is fed at once.

For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it
doesn't seem worth the trouble.
2014-11-10 22:02:05 +01:00

258 lines
7.4 KiB
C

/*
* This file is part of MPlayer.
*
* Copyright (C) 2012 Naoya OYAMA
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <string.h>
#include <assert.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include "config.h"
#include "common/msg.h"
#include "common/av_common.h"
#include "options/options.h"
#include "ad.h"
#define OUTBUF_SIZE 65536
struct spdifContext {
struct mp_log *log;
AVFormatContext *lavf_ctx;
int iec61937_packet_size;
int out_buffer_len;
uint8_t out_buffer[OUTBUF_SIZE];
bool need_close;
struct mp_audio fmt;
};
static int write_packet(void *p, uint8_t *buf, int buf_size)
{
struct spdifContext *ctx = p;
int buffer_left = OUTBUF_SIZE - ctx->out_buffer_len;
if (buf_size > buffer_left) {
MP_ERR(ctx, "spdif packet too large.\n");
buf_size = buffer_left;
}
memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, buf_size);
ctx->out_buffer_len += buf_size;
return buf_size;
}
static void uninit(struct dec_audio *da)
{
struct spdifContext *spdif_ctx = da->priv;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
if (lavf_ctx) {
if (spdif_ctx->need_close)
av_write_trailer(lavf_ctx);
if (lavf_ctx->pb)
av_freep(&lavf_ctx->pb->buffer);
av_freep(&lavf_ctx->pb);
avformat_free_context(lavf_ctx);
}
}
static int init(struct dec_audio *da, const char *decoder)
{
struct spdifContext *spdif_ctx = talloc_zero(NULL, struct spdifContext);
da->priv = spdif_ctx;
spdif_ctx->log = da->log;
AVFormatContext *lavf_ctx = avformat_alloc_context();
if (!lavf_ctx)
goto fail;
lavf_ctx->oformat = av_guess_format("spdif", NULL, NULL);
if (!lavf_ctx->oformat)
goto fail;
spdif_ctx->lavf_ctx = lavf_ctx;
void *buffer = av_mallocz(OUTBUF_SIZE);
if (!buffer)
abort();
lavf_ctx->pb = avio_alloc_context(buffer, OUTBUF_SIZE, 1, spdif_ctx, NULL,
write_packet, NULL);
if (!lavf_ctx->pb) {
av_free(buffer);
goto fail;
}
// Request minimal buffering (not available on Libav)
#if LIBAVFORMAT_VERSION_MICRO >= 100
lavf_ctx->pb->direct = 1;
#endif
AVStream *stream = avformat_new_stream(lavf_ctx, 0);
if (!stream)
goto fail;
stream->codec->codec_id = mp_codec_to_av_codec_id(decoder);
AVDictionary *format_opts = NULL;
int num_channels = 0;
int sample_format = 0;
int samplerate = 0;
switch (stream->codec->codec_id) {
case AV_CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sample_format = AF_FORMAT_S_AAC;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sample_format = AF_FORMAT_S_AC3;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_DTS:
if (da->opts->dtshd) {
av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000
spdif_ctx->iec61937_packet_size = 32768;
sample_format = AF_FORMAT_S_DTSHD;
samplerate = 192000;
num_channels = 2*4;
} else {
spdif_ctx->iec61937_packet_size = 32768;
sample_format = AF_FORMAT_S_DTS;
samplerate = 48000;
num_channels = 2;
}
break;
case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sample_format = AF_FORMAT_S_EAC3;
samplerate = 192000;
num_channels = 2;
break;
case AV_CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sample_format = AF_FORMAT_S_MP3;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sample_format = AF_FORMAT_S_TRUEHD;
samplerate = 192000;
num_channels = 8;
break;
default:
abort();
}
mp_audio_set_num_channels(&spdif_ctx->fmt, num_channels);
mp_audio_set_format(&spdif_ctx->fmt, sample_format);
spdif_ctx->fmt.rate = samplerate;
if (avformat_write_header(lavf_ctx, &format_opts) < 0) {
MP_FATAL(da, "libavformat spdif initialization failed.\n");
av_dict_free(&format_opts);
goto fail;
}
av_dict_free(&format_opts);
spdif_ctx->need_close = true;
return 1;
fail:
uninit(da);
return 0;
}
static int decode_packet(struct dec_audio *da, struct mp_audio **out)
{
struct spdifContext *spdif_ctx = da->priv;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
spdif_ctx->out_buffer_len = 0;
struct demux_packet *mpkt;
if (demux_read_packet_async(da->header, &mpkt) == 0)
return AD_WAIT;
if (!mpkt)
return AD_EOF;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
pkt.pts = pkt.dts = 0;
MP_VERBOSE(da, "spdif packet, size=%d\n", pkt.size);
if (mpkt->pts != MP_NOPTS_VALUE) {
da->pts = mpkt->pts;
da->pts_offset = 0;
}
int ret = av_write_frame(lavf_ctx, &pkt);
talloc_free(mpkt);
avio_flush(lavf_ctx->pb);
if (ret < 0)
return AD_ERR;
int samples = spdif_ctx->out_buffer_len / spdif_ctx->fmt.sstride;
*out = mp_audio_pool_get(da->pool, &spdif_ctx->fmt, samples);
if (!*out)
return AD_ERR;
memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
return 0;
}
static int control(struct dec_audio *da, int cmd, void *arg)
{
return CONTROL_UNKNOWN;
}
static const int codecs[] = {
AV_CODEC_ID_AAC,
AV_CODEC_ID_AC3,
AV_CODEC_ID_DTS,
AV_CODEC_ID_EAC3,
AV_CODEC_ID_MP3,
AV_CODEC_ID_TRUEHD,
AV_CODEC_ID_NONE
};
static void add_decoders(struct mp_decoder_list *list)
{
for (int n = 0; codecs[n] != AV_CODEC_ID_NONE; n++) {
const char *format = mp_codec_from_av_codec_id(codecs[n]);
if (format) {
mp_add_decoder(list, "spdif", format, format,
"libavformat/spdifenc audio pass-through decoder");
}
}
}
const struct ad_functions ad_spdif = {
.name = "spdif",
.add_decoders = add_decoders,
.init = init,
.uninit = uninit,
.control = control,
.decode_packet = decode_packet,
};