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mpv/audio/decode/ad_spdif.c
wm4 81bf9a1963 audio: cleanup spdif format definitions
Before this commit, there was AF_FORMAT_AC3 (the original spdif format,
used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS
and DTS-HD), which was handled as some sort of superset for
AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used
IEC61937-framing, but still was handled as something "separate".

Technically, all of them are pretty similar, but may use different
bitrates. Since digital passthrough pretends to be PCM (just with
special headers that wrap digital packets), this is easily detectable by
the higher samplerate or higher number of channels, so I don't know why
you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs.
AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is
just a mess.

Simplify this by handling all these formats the same way.
AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3).
All AOs just accept all spdif formats now - whether that works or not is
not really clear (seems inconsistent due to earlier attempts to make
DTS-HD work). But on the other hand, enabling spdif requires manual user
interaction, so it doesn't matter much if initialization fails in
slightly less graceful ways if it can't work at all.

At a later point, we will support passthrough with ao_pulse. It seems
the PulseAudio API wants to know the codec type (or maybe not - feeding
it DTS while telling it it's AC3 works), add separate formats for each
codecs. While this reminds of the earlier chaos, it's stricter, and most
code just uses AF_FORMAT_IS_IEC61937().

Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to
include special formats, so that it always describes the fundamental
sample format type. This also ensures valid AF formats are never 0 (this
was probably broken in one of the earlier commits from today).
2014-09-23 23:11:54 +02:00

255 lines
7.2 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;
};
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(&da->decoded, num_channels);
mp_audio_set_format(&da->decoded, sample_format);
da->decoded.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 spdifContext *spdif_ctx = da->priv;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
mp_audio_set_null_data(&da->decoded);
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;
da->decoded.planes[0] = spdif_ctx->out_buffer;
da->decoded.samples = spdif_ctx->out_buffer_len / da->decoded.sstride;
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,
};