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mpv/audio/decode/ad_spdif.c
Philip Langdale 4574dd5dc6 ffmpeg: update to handle deprecation of av_init_packet
This has been a long standing annoyance - ffmpeg is removing
sizeof(AVPacket) from the API which means you cannot stack-allocate
AVPacket anymore. However, that is something we take advantage of
because we use short-lived AVPackets to bridge from native mpv packets
in our main decoding paths.

We don't think that switching these to `av_packet_alloc` is desirable,
given the cost of heap allocation, so this change takes a different
approach - allocating a single packet in the relevant context and
reusing it over and over.

That's fairly straight-forward, with the main caveat being that
re-initialising the packet is unintuitive. There is no function that
does exactly what we need (what `av_init_packet` did). The closest is
`av_packet_unref`, which additionally frees buffers and side-data.
However, we don't copy those things - we just assign them in from our
own packet, so we have to explicitly clear the pointers before calling
`av_packet_unref`. But at least we can make a wrapper function for
that.

The weirdest part of the change is the handling of the vtt subtitle
conversion. This requires two packets, so I had to pre-allocate two in
the context struct. That sounds excessive, but if allocating the
primary packet is too expensive, then allocating the secondary one for
vtt subtitles must also be too expensive.

This change is not conditional as heap allocated AVPackets were
available for years and years before the deprecation.
2022-12-03 14:44:18 -08:00

438 lines
13 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;
AVPacket *avpkt;
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;
}
mp_free_av_packet(&spdif_ctx->avpkt);
}
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;
const 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;
mp_set_av_packet(spdif_ctx->avpkt, mpkt, NULL);
spdif_ctx->avpkt->pts = spdif_ctx->avpkt->dts = 0;
if (!spdif_ctx->lavf_ctx) {
if (init_filter(da, spdif_ctx->avpkt) < 0)
goto done;
}
spdif_ctx->out_buffer_len = 0;
int ret = av_write_frame(spdif_ctx->lavf_ctx, spdif_ctx->avpkt);
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;
}
spdif_ctx->avpkt = av_packet_alloc();
MP_HANDLE_OOM(spdif_ctx->avpkt);
return &spdif_ctx->public;
}
const struct mp_decoder_fns ad_spdif = {
.create = create,
};