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fef8b7984b
mpv/audio/decode/ad_spdif.c
wm4 fef8b7984b audio: refactor: work towards unentangling audio decoding and filtering 
Similar to the video path. dec_audio.c now handles decoding only. It
also looks very similar to dec_video.c, and actually contains some of
the rewritten code from it. (A further goal might be unifying the
decoders, I guess.)

High potential for regressions.
2016-01-22 00:25:44 +01:00

318 lines
8.7 KiB
C
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/*
* Copyright (C) 2012 Naoya OYAMA
*
* This file is part of mpv.
*
* mpv 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.
*
* mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#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;
enum AVCodecID codec_id;
AVFormatContext *lavf_ctx;
int out_buffer_len;
uint8_t out_buffer[OUTBUF_SIZE];
bool need_close;
bool use_dts_hd;
struct mp_audio fmt;
struct mp_audio_pool *pool;
};
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;
spdif_ctx->use_dts_hd = da->opts->dtshd;
spdif_ctx->pool = mp_audio_pool_create(spdif_ctx);
if (strcmp(decoder, "dts-hd") == 0) {
decoder = "dts";
spdif_ctx->use_dts_hd = true;
}
spdif_ctx->codec_id = mp_codec_to_av_codec_id(decoder);
return spdif_ctx->codec_id != AV_CODEC_ID_NONE;
}
static int determine_codec_profile(struct dec_audio *da, AVPacket *pkt)
{
struct spdifContext *spdif_ctx = da->priv;
int profile = FF_PROFILE_UNKNOWN;
AVCodecContext *ctx = NULL;
AVFrame *frame = NULL;
AVCodec *codec = avcodec_find_decoder(spdif_ctx->codec_id);
if (!codec)
goto done;
frame = av_frame_alloc();
if (!frame)
goto done;
ctx = avcodec_alloc_context3(codec);
if (!ctx)
goto done;
if (avcodec_open2(ctx, codec, NULL) < 0) {
av_free(ctx); // don't attempt to avcodec_close() an unopened ctx
ctx = NULL;
goto done;
}
int got_frame = 0;
if (avcodec_decode_audio4(ctx, frame, &got_frame, pkt) < 1 || !got_frame)
goto done;
profile = ctx->profile;
done:
av_frame_free(&frame);
if (ctx)
avcodec_close(ctx);
avcodec_free_context(&ctx);
if (profile == FF_PROFILE_UNKNOWN)
MP_WARN(da, "Failed to parse codec profile.\n");
return profile;
}
static int init_filter(struct dec_audio *da, AVPacket *pkt)
{
struct spdifContext *spdif_ctx = da->priv;
int profile = FF_PROFILE_UNKNOWN;
if (spdif_ctx->codec_id == AV_CODEC_ID_DTS)
profile = determine_codec_profile(da, pkt);
AVFormatContext *lavf_ctx = avformat_alloc_context();
if (!lavf_ctx)
goto fail;
spdif_ctx->lavf_ctx = lavf_ctx;
lavf_ctx->oformat = av_guess_format("spdif", NULL, NULL);
if (!lavf_ctx->oformat)
goto fail;
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 = spdif_ctx->codec_id;
AVDictionary *format_opts = NULL;
int num_channels = 0;
int sample_format = 0;
int samplerate = 0;
switch (spdif_ctx->codec_id) {
case AV_CODEC_ID_AAC:
sample_format = AF_FORMAT_S_AAC;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_AC3:
sample_format = AF_FORMAT_S_AC3;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_DTS: {
bool is_hd = profile == FF_PROFILE_DTS_HD_HRA ||
profile == FF_PROFILE_DTS_HD_MA ||
profile == FF_PROFILE_UNKNOWN;
if (spdif_ctx->use_dts_hd && is_hd) {
av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000
sample_format = AF_FORMAT_S_DTSHD;
samplerate = 192000;
num_channels = 2*4;
} else {
sample_format = AF_FORMAT_S_DTS;
samplerate = 48000;
num_channels = 2;
}
break;
}
case AV_CODEC_ID_EAC3:
sample_format = AF_FORMAT_S_EAC3;
samplerate = 192000;
num_channels = 2;
break;
case AV_CODEC_ID_MP3:
sample_format = AF_FORMAT_S_MP3;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_TRUEHD:
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 0;
fail:
uninit(da);
return -1;
}
static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
struct mp_audio **out)
{
struct spdifContext *spdif_ctx = da->priv;
spdif_ctx->out_buffer_len = 0;
if (!mpkt)
return 0;
double pts = mpkt->pts;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
mpkt->len = 0; // will be fully consumed
pkt.pts = pkt.dts = 0;
if (!spdif_ctx->lavf_ctx) {
if (init_filter(da, &pkt) < 0)
return -1;
}
int ret = av_write_frame(spdif_ctx->lavf_ctx, &pkt);
avio_flush(spdif_ctx->lavf_ctx->pb);
if (ret < 0)
return -1;
int samples = spdif_ctx->out_buffer_len / spdif_ctx->fmt.sstride;
*out = mp_audio_pool_get(spdif_ctx->pool, &spdif_ctx->fmt, samples);
if (!*out)
return -1;
memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
(*out)->pts = pts;
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");
}
}
mp_add_decoder(list, "spdif", "dts", "dts-hd",
"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,
};
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