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mpv/audio/decode/ad_spdif.c
wm4 7d11eda72e Remove remains of Libav compatibility
Libav seems rather dead: no release for 2 years, no new git commits in
master for almost a year (with one exception ~6 months ago). From what I
can tell, some developers resigned themselves to the horrifying idea to
post patches to ffmpeg-devel instead, while the rest of the developers
went on to greener pastures.

Libav was a better project than FFmpeg. Unfortunately, FFmpeg won,
because it managed to keep the name and website. Libav was pushed more
and more into obscurity: while there was initially a big push for Libav,
FFmpeg just remained "in place" and visible for most people. FFmpeg was
slowly draining all manpower and energy from Libav. A big part of this
was that FFmpeg stole code from Libav (regular merges of the entire
Libav git tree), making it some sort of Frankenstein mirror of Libav,
think decaying zombie with additional legs ("features") nailed to it.
"Stealing" surely is the wrong word; I'm just aping the language that
some of the FFmpeg members used to use. All that is in the past now, I'm
probably the only person left who is annoyed by this, and with this
commit I'm putting this decade long problem finally to an end. I just
thought I'd express my annoyance about this fucking shitshow one last
time.

The most intrusive change in this commit is the resample filter, which
originally used libavresample. Since the FFmpeg developer refused to
enable libavresample by default for drama reasons, and the API was
slightly different, so the filter used some big preprocessor mess to
make it compatible to libswresample. All that falls away now. The
simplification to the build system is also significant.
2020-02-16 15:14:55 +01:00

433 lines
12 KiB
C

/*
* 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 Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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 "audio/aframe.h"
#include "audio/format.h"
#include "common/av_common.h"
#include "common/codecs.h"
#include "common/msg.h"
#include "demux/packet.h"
#include "demux/stheader.h"
#include "filters/f_decoder_wrapper.h"
#include "filters/filter_internal.h"
#include "options/options.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_aframe *fmt;
int sstride;
struct mp_aframe_pool *pool;
struct mp_decoder public;
};
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;
}
// (called on both filter destruction _and_ if lavf fails to init)
static void destroy(struct mp_filter *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);
spdif_ctx->lavf_ctx = NULL;
}
}
static void determine_codec_params(struct mp_filter *da, AVPacket *pkt,
int *out_profile, int *out_rate)
{
struct spdifContext *spdif_ctx = da->priv;
int profile = FF_PROFILE_UNKNOWN;
AVCodecContext *ctx = NULL;
AVFrame *frame = NULL;
AVCodecParserContext *parser = av_parser_init(spdif_ctx->codec_id);
if (parser) {
// Don't make it wait for the next frame.
parser->flags |= PARSER_FLAG_COMPLETE_FRAMES;
ctx = avcodec_alloc_context3(NULL);
if (!ctx) {
av_parser_close(parser);
goto done;
}
uint8_t *d = NULL;
int s = 0;
av_parser_parse2(parser, ctx, &d, &s, pkt->data, pkt->size, 0, 0, 0);
*out_profile = profile = ctx->profile;
*out_rate = ctx->sample_rate;
avcodec_free_context(&ctx);
av_parser_close(parser);
}
if (profile != FF_PROFILE_UNKNOWN || spdif_ctx->codec_id != AV_CODEC_ID_DTS)
return;
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)
goto done;
if (avcodec_send_packet(ctx, pkt) < 0)
goto done;
if (avcodec_receive_frame(ctx, frame) < 0)
goto done;
*out_profile = profile = ctx->profile;
*out_rate = ctx->sample_rate;
done:
av_frame_free(&frame);
avcodec_free_context(&ctx);
if (profile == FF_PROFILE_UNKNOWN)
MP_WARN(da, "Failed to parse codec profile.\n");
}
static int init_filter(struct mp_filter *da, AVPacket *pkt)
{
struct spdifContext *spdif_ctx = da->priv;
int profile = FF_PROFILE_UNKNOWN;
int c_rate = 0;
determine_codec_params(da, pkt, &profile, &c_rate);
MP_VERBOSE(da, "In: profile=%d samplerate=%d\n", profile, c_rate);
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
lavf_ctx->pb->direct = 1;
AVStream *stream = avformat_new_stream(lavf_ctx, 0);
if (!stream)
goto fail;
stream->codecpar->codec_id = spdif_ctx->codec_id;
AVDictionary *format_opts = NULL;
spdif_ctx->fmt = mp_aframe_create();
talloc_steal(spdif_ctx, spdif_ctx->fmt);
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 = c_rate > 0 ? c_rate : 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;
// Apparently, DTS-HD over SPDIF is specified to be 7.1 (8 channels)
// for DTS-HD MA, and stereo (2 channels) for DTS-HD HRA. The bit
// streaming rate as well as the signaled channel count are defined
// based on this value.
int dts_hd_spdif_channel_count = profile == FF_PROFILE_DTS_HD_HRA ?
2 : 8;
if (spdif_ctx->use_dts_hd && is_hd) {
av_dict_set_int(&format_opts, "dtshd_rate",
dts_hd_spdif_channel_count * 96000, 0);
sample_format = AF_FORMAT_S_DTSHD;
samplerate = 192000;
num_channels = dts_hd_spdif_channel_count;
} 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();
}
struct mp_chmap chmap;
mp_chmap_from_channels(&chmap, num_channels);
mp_aframe_set_chmap(spdif_ctx->fmt, &chmap);
mp_aframe_set_format(spdif_ctx->fmt, sample_format);
mp_aframe_set_rate(spdif_ctx->fmt, samplerate);
spdif_ctx->sstride = mp_aframe_get_sstride(spdif_ctx->fmt);
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:
destroy(da);
mp_filter_internal_mark_failed(da);
return -1;
}
static void process(struct mp_filter *da)
{
struct spdifContext *spdif_ctx = da->priv;
if (!mp_pin_can_transfer_data(da->ppins[1], da->ppins[0]))
return;
struct mp_frame inframe = mp_pin_out_read(da->ppins[0]);
if (inframe.type == MP_FRAME_EOF) {
mp_pin_in_write(da->ppins[1], inframe);
return;
} else if (inframe.type != MP_FRAME_PACKET) {
if (inframe.type) {
MP_ERR(da, "unknown frame type\n");
mp_filter_internal_mark_failed(da);
}
return;
}
struct demux_packet *mpkt = inframe.data;
struct mp_aframe *out = NULL;
double pts = mpkt->pts;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
pkt.pts = pkt.dts = 0;
if (!spdif_ctx->lavf_ctx) {
if (init_filter(da, &pkt) < 0)
goto done;
}
spdif_ctx->out_buffer_len = 0;
int ret = av_write_frame(spdif_ctx->lavf_ctx, &pkt);
avio_flush(spdif_ctx->lavf_ctx->pb);
if (ret < 0) {
MP_ERR(da, "spdif mux error: '%s'\n", mp_strerror(AVUNERROR(ret)));
goto done;
}
out = mp_aframe_new_ref(spdif_ctx->fmt);
int samples = spdif_ctx->out_buffer_len / spdif_ctx->sstride;
if (mp_aframe_pool_allocate(spdif_ctx->pool, out, samples) < 0) {
TA_FREEP(&out);
goto done;
}
uint8_t **data = mp_aframe_get_data_rw(out);
if (!data) {
TA_FREEP(&out);
goto done;
}
memcpy(data[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
mp_aframe_set_pts(out, pts);
done:
talloc_free(mpkt);
if (out) {
mp_pin_in_write(da->ppins[1], MAKE_FRAME(MP_FRAME_AUDIO, out));
} else {
mp_filter_internal_mark_failed(da);
}
}
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 bool find_codec(const char *name)
{
for (int n = 0; codecs[n] != AV_CODEC_ID_NONE; n++) {
const char *format = mp_codec_from_av_codec_id(codecs[n]);
if (format && name && strcmp(format, name) == 0)
return true;
}
return false;
}
// codec is the libavcodec name of the source audio codec.
// pref is a ","-separated list of names, some of them which do not match with
// libavcodec names (like dts-hd).
struct mp_decoder_list *select_spdif_codec(const char *codec, const char *pref)
{
struct mp_decoder_list *list = talloc_zero(NULL, struct mp_decoder_list);
if (!find_codec(codec))
return list;
bool spdif_allowed = false, dts_hd_allowed = false;
bstr sel = bstr0(pref);
while (sel.len) {
bstr decoder;
bstr_split_tok(sel, ",", &decoder, &sel);
if (decoder.len) {
if (bstr_equals0(decoder, codec))
spdif_allowed = true;
if (bstr_equals0(decoder, "dts-hd") && strcmp(codec, "dts") == 0)
spdif_allowed = dts_hd_allowed = true;
}
}
if (!spdif_allowed)
return list;
const char *suffix_name = dts_hd_allowed ? "dts_hd" : codec;
char name[80];
snprintf(name, sizeof(name), "spdif_%s", suffix_name);
mp_add_decoder(list, codec, name,
"libavformat/spdifenc audio pass-through decoder");
return list;
}
static const struct mp_filter_info ad_spdif_filter = {
.name = "ad_spdif",
.priv_size = sizeof(struct spdifContext),
.process = process,
.destroy = destroy,
};
static struct mp_decoder *create(struct mp_filter *parent,
struct mp_codec_params *codec,
const char *decoder)
{
struct mp_filter *da = mp_filter_create(parent, &ad_spdif_filter);
if (!da)
return NULL;
mp_filter_add_pin(da, MP_PIN_IN, "in");
mp_filter_add_pin(da, MP_PIN_OUT, "out");
da->log = mp_log_new(da, parent->log, NULL);
struct spdifContext *spdif_ctx = da->priv;
spdif_ctx->log = da->log;
spdif_ctx->pool = mp_aframe_pool_create(spdif_ctx);
spdif_ctx->public.f = da;
if (strcmp(decoder, "spdif_dts_hd") == 0)
spdif_ctx->use_dts_hd = true;
spdif_ctx->codec_id = mp_codec_to_av_codec_id(codec->codec);
if (spdif_ctx->codec_id == AV_CODEC_ID_NONE) {
talloc_free(da);
return NULL;
}
return &spdif_ctx->public;
}
const struct mp_decoder_fns ad_spdif = {
.create = create,
};