mirror of
https://github.com/mpv-player/mpv
synced 2024-12-21 22:30:22 +00:00
5fd8a1e04c
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.
258 lines
7.4 KiB
C
258 lines
7.4 KiB
C
/*
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* This file is part of MPlayer.
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*
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* Copyright (C) 2012 Naoya OYAMA
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <string.h>
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#include <assert.h>
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#include <libavformat/avformat.h>
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#include <libavcodec/avcodec.h>
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#include <libavutil/opt.h>
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#include "config.h"
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#include "common/msg.h"
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#include "common/av_common.h"
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#include "options/options.h"
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#include "ad.h"
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#define OUTBUF_SIZE 65536
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struct spdifContext {
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struct mp_log *log;
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AVFormatContext *lavf_ctx;
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int iec61937_packet_size;
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int out_buffer_len;
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uint8_t out_buffer[OUTBUF_SIZE];
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bool need_close;
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struct mp_audio fmt;
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};
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static int write_packet(void *p, uint8_t *buf, int buf_size)
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{
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struct spdifContext *ctx = p;
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int buffer_left = OUTBUF_SIZE - ctx->out_buffer_len;
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if (buf_size > buffer_left) {
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MP_ERR(ctx, "spdif packet too large.\n");
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buf_size = buffer_left;
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}
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memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, buf_size);
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ctx->out_buffer_len += buf_size;
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return buf_size;
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}
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static void uninit(struct dec_audio *da)
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{
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struct spdifContext *spdif_ctx = da->priv;
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AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
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if (lavf_ctx) {
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if (spdif_ctx->need_close)
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av_write_trailer(lavf_ctx);
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if (lavf_ctx->pb)
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av_freep(&lavf_ctx->pb->buffer);
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av_freep(&lavf_ctx->pb);
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avformat_free_context(lavf_ctx);
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}
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}
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static int init(struct dec_audio *da, const char *decoder)
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{
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struct spdifContext *spdif_ctx = talloc_zero(NULL, struct spdifContext);
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da->priv = spdif_ctx;
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spdif_ctx->log = da->log;
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AVFormatContext *lavf_ctx = avformat_alloc_context();
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if (!lavf_ctx)
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goto fail;
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lavf_ctx->oformat = av_guess_format("spdif", NULL, NULL);
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if (!lavf_ctx->oformat)
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goto fail;
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spdif_ctx->lavf_ctx = lavf_ctx;
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void *buffer = av_mallocz(OUTBUF_SIZE);
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if (!buffer)
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abort();
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lavf_ctx->pb = avio_alloc_context(buffer, OUTBUF_SIZE, 1, spdif_ctx, NULL,
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write_packet, NULL);
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if (!lavf_ctx->pb) {
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av_free(buffer);
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goto fail;
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}
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// Request minimal buffering (not available on Libav)
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#if LIBAVFORMAT_VERSION_MICRO >= 100
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lavf_ctx->pb->direct = 1;
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#endif
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AVStream *stream = avformat_new_stream(lavf_ctx, 0);
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if (!stream)
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goto fail;
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stream->codec->codec_id = mp_codec_to_av_codec_id(decoder);
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AVDictionary *format_opts = NULL;
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int num_channels = 0;
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int sample_format = 0;
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int samplerate = 0;
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switch (stream->codec->codec_id) {
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case AV_CODEC_ID_AAC:
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spdif_ctx->iec61937_packet_size = 16384;
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sample_format = AF_FORMAT_S_AAC;
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samplerate = 48000;
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num_channels = 2;
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break;
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case AV_CODEC_ID_AC3:
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spdif_ctx->iec61937_packet_size = 6144;
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sample_format = AF_FORMAT_S_AC3;
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samplerate = 48000;
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num_channels = 2;
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break;
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case AV_CODEC_ID_DTS:
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if (da->opts->dtshd) {
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av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000
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spdif_ctx->iec61937_packet_size = 32768;
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sample_format = AF_FORMAT_S_DTSHD;
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samplerate = 192000;
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num_channels = 2*4;
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} else {
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spdif_ctx->iec61937_packet_size = 32768;
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sample_format = AF_FORMAT_S_DTS;
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samplerate = 48000;
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num_channels = 2;
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}
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break;
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case AV_CODEC_ID_EAC3:
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spdif_ctx->iec61937_packet_size = 24576;
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sample_format = AF_FORMAT_S_EAC3;
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samplerate = 192000;
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num_channels = 2;
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break;
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case AV_CODEC_ID_MP3:
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spdif_ctx->iec61937_packet_size = 4608;
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sample_format = AF_FORMAT_S_MP3;
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samplerate = 48000;
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num_channels = 2;
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break;
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case AV_CODEC_ID_TRUEHD:
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spdif_ctx->iec61937_packet_size = 61440;
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sample_format = AF_FORMAT_S_TRUEHD;
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samplerate = 192000;
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num_channels = 8;
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break;
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default:
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abort();
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}
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mp_audio_set_num_channels(&spdif_ctx->fmt, num_channels);
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mp_audio_set_format(&spdif_ctx->fmt, sample_format);
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spdif_ctx->fmt.rate = samplerate;
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if (avformat_write_header(lavf_ctx, &format_opts) < 0) {
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MP_FATAL(da, "libavformat spdif initialization failed.\n");
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av_dict_free(&format_opts);
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goto fail;
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}
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av_dict_free(&format_opts);
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spdif_ctx->need_close = true;
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return 1;
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fail:
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uninit(da);
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return 0;
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}
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static int decode_packet(struct dec_audio *da, struct mp_audio **out)
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{
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struct spdifContext *spdif_ctx = da->priv;
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AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
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spdif_ctx->out_buffer_len = 0;
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struct demux_packet *mpkt;
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if (demux_read_packet_async(da->header, &mpkt) == 0)
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return AD_WAIT;
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if (!mpkt)
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return AD_EOF;
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AVPacket pkt;
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mp_set_av_packet(&pkt, mpkt, NULL);
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pkt.pts = pkt.dts = 0;
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MP_VERBOSE(da, "spdif packet, size=%d\n", pkt.size);
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if (mpkt->pts != MP_NOPTS_VALUE) {
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da->pts = mpkt->pts;
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da->pts_offset = 0;
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}
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int ret = av_write_frame(lavf_ctx, &pkt);
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talloc_free(mpkt);
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avio_flush(lavf_ctx->pb);
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if (ret < 0)
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return AD_ERR;
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int samples = spdif_ctx->out_buffer_len / spdif_ctx->fmt.sstride;
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*out = mp_audio_pool_get(da->pool, &spdif_ctx->fmt, samples);
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if (!*out)
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return AD_ERR;
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memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
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return 0;
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}
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static int control(struct dec_audio *da, int cmd, void *arg)
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{
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return CONTROL_UNKNOWN;
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}
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static const int codecs[] = {
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AV_CODEC_ID_AAC,
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AV_CODEC_ID_AC3,
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AV_CODEC_ID_DTS,
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AV_CODEC_ID_EAC3,
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AV_CODEC_ID_MP3,
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AV_CODEC_ID_TRUEHD,
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AV_CODEC_ID_NONE
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};
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static void add_decoders(struct mp_decoder_list *list)
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{
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for (int n = 0; codecs[n] != AV_CODEC_ID_NONE; n++) {
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const char *format = mp_codec_from_av_codec_id(codecs[n]);
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if (format) {
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mp_add_decoder(list, "spdif", format, format,
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"libavformat/spdifenc audio pass-through decoder");
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}
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}
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}
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const struct ad_functions ad_spdif = {
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.name = "spdif",
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.add_decoders = add_decoders,
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.init = init,
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.uninit = uninit,
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.control = control,
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.decode_packet = decode_packet,
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};
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