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mpv/audio/decode/dec_audio.c
wm4 7dd3822d09 audio: refactor some aspects of filter chain setup
There's no real reason why audio_init_filter() should exist. Just use
af_init or af_reinit directly. (We lose a useless message; the same
information is printed in a quite close place with more details.)

Requires less code, and the way the filter chain is marked as having
failed to initialize allows just switching off audio instead of
crashing if trying to insert a volume filter in mixer.c fails, and
recreating the old filter chain fails too.
2014-10-02 02:42:23 +02:00

319 lines
10 KiB
C

/*
* This file is part of MPlayer.
*
* 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 <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <assert.h>
#include <libavutil/mem.h>
#include "demux/codec_tags.h"
#include "config.h"
#include "common/codecs.h"
#include "common/msg.h"
#include "misc/bstr.h"
#include "stream/stream.h"
#include "demux/demux.h"
#include "demux/stheader.h"
#include "dec_audio.h"
#include "ad.h"
#include "audio/format.h"
#include "audio/audio.h"
#include "audio/audio_buffer.h"
#include "audio/filter/af.h"
extern const struct ad_functions ad_mpg123;
extern const struct ad_functions ad_lavc;
extern const struct ad_functions ad_spdif;
static const struct ad_functions * const ad_drivers[] = {
&ad_lavc,
#if HAVE_MPG123
&ad_mpg123,
#endif
&ad_spdif,
NULL
};
// Drop audio buffer and reinit it (after format change)
// Returns whether the format was valid at all.
static bool reinit_audio_buffer(struct dec_audio *da)
{
if (!mp_audio_config_valid(&da->decoded)) {
MP_ERR(da, "Audio decoder did not specify audio "
"format, or requested an unsupported configuration!\n");
return false;
}
mp_audio_buffer_reinit(da->decode_buffer, &da->decoded);
return true;
}
static void uninit_decoder(struct dec_audio *d_audio)
{
if (d_audio->ad_driver) {
MP_VERBOSE(d_audio, "Uninit audio decoder.\n");
d_audio->ad_driver->uninit(d_audio);
}
d_audio->ad_driver = NULL;
talloc_free(d_audio->priv);
d_audio->priv = NULL;
}
static int init_audio_codec(struct dec_audio *d_audio, const char *decoder)
{
if (!d_audio->ad_driver->init(d_audio, decoder)) {
MP_VERBOSE(d_audio, "Audio decoder init failed.\n");
d_audio->ad_driver = NULL;
uninit_decoder(d_audio);
return 0;
}
d_audio->decode_buffer = mp_audio_buffer_create(NULL);
return 1;
}
struct mp_decoder_list *audio_decoder_list(void)
{
struct mp_decoder_list *list = talloc_zero(NULL, struct mp_decoder_list);
for (int i = 0; ad_drivers[i] != NULL; i++)
ad_drivers[i]->add_decoders(list);
return list;
}
static struct mp_decoder_list *audio_select_decoders(const char *codec,
char *selection)
{
struct mp_decoder_list *list = audio_decoder_list();
struct mp_decoder_list *new = mp_select_decoders(list, codec, selection);
talloc_free(list);
return new;
}
static const struct ad_functions *find_driver(const char *name)
{
for (int i = 0; ad_drivers[i] != NULL; i++) {
if (strcmp(ad_drivers[i]->name, name) == 0)
return ad_drivers[i];
}
return NULL;
}
int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders)
{
assert(!d_audio->ad_driver);
audio_reset_decoding(d_audio);
struct mp_decoder_entry *decoder = NULL;
struct mp_decoder_list *list =
audio_select_decoders(d_audio->header->codec, audio_decoders);
mp_print_decoders(d_audio->log, MSGL_V, "Codec list:", list);
for (int n = 0; n < list->num_entries; n++) {
struct mp_decoder_entry *sel = &list->entries[n];
const struct ad_functions *driver = find_driver(sel->family);
if (!driver)
continue;
MP_VERBOSE(d_audio, "Opening audio decoder %s:%s\n",
sel->family, sel->decoder);
d_audio->ad_driver = driver;
if (init_audio_codec(d_audio, sel->decoder)) {
decoder = sel;
break;
}
MP_WARN(d_audio, "Audio decoder init failed for "
"%s:%s\n", sel->family, sel->decoder);
}
if (d_audio->ad_driver) {
d_audio->decoder_desc =
talloc_asprintf(d_audio, "%s [%s:%s]", decoder->desc, decoder->family,
decoder->decoder);
MP_VERBOSE(d_audio, "Selected audio codec: %s\n", d_audio->decoder_desc);
} else {
MP_ERR(d_audio, "Failed to initialize an audio decoder for codec '%s'.\n",
d_audio->header->codec ? d_audio->header->codec : "<unknown>");
}
talloc_free(list);
return !!d_audio->ad_driver;
}
void audio_uninit(struct dec_audio *d_audio)
{
if (!d_audio)
return;
MP_VERBOSE(d_audio, "Uninit audio filters...\n");
af_destroy(d_audio->afilter);
uninit_decoder(d_audio);
talloc_free(d_audio->decode_buffer);
talloc_free(d_audio);
}
/* Decode packets until we know the audio format. Then reinit the buffer.
* Returns AD_OK on success, negative AD_* code otherwise.
* Also returns AD_OK if already initialized (and does nothing).
*/
int initial_audio_decode(struct dec_audio *da)
{
while (!mp_audio_config_valid(&da->decoded)) {
if (da->decoded.samples > 0)
return AD_ERR; // invalid format, rather than uninitialized
int ret = da->ad_driver->decode_packet(da);
if (ret < 0)
return ret;
}
if (mp_audio_buffer_samples(da->decode_buffer) > 0) // avoid accidental flush
return AD_OK;
return reinit_audio_buffer(da) ? AD_OK : AD_ERR;
}
// Filter len bytes of input, put result into outbuf.
static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
int len)
{
bool format_change = false;
int error = 0;
assert(len > 0); // would break EOF logic below
while (mp_audio_buffer_samples(da->decode_buffer) < len) {
// Check for a format change
struct mp_audio config;
mp_audio_buffer_get_format(da->decode_buffer, &config);
format_change = !mp_audio_config_equals(&da->decoded, &config);
if (format_change) {
error = AD_EOF; // drain remaining data left in the current buffer
break;
}
if (da->decoded.samples > 0) {
int copy = MPMIN(da->decoded.samples, len);
struct mp_audio append = da->decoded;
append.samples = copy;
mp_audio_buffer_append(da->decode_buffer, &append);
mp_audio_skip_samples(&da->decoded, copy);
da->pts_offset += copy;
continue;
}
error = da->ad_driver->decode_packet(da);
if (error < 0)
break;
}
if (error == AD_WAIT)
return error;
// Filter
struct mp_audio filter_data;
mp_audio_buffer_peek(da->decode_buffer, &filter_data);
filter_data.rate = da->afilter->input.rate; // due to playback speed change
len = MPMIN(filter_data.samples, len);
filter_data.samples = len;
bool eof = error == AD_EOF && filter_data.samples == 0;
if (af_filter(da->afilter, &filter_data, eof ? AF_FILTER_FLAG_EOF : 0) < 0)
return AD_ERR;
mp_audio_buffer_append(outbuf, &filter_data);
if (error == AD_EOF && filter_data.samples > 0)
error = 0; // don't end playback yet
// remove processed data from decoder buffer:
mp_audio_buffer_skip(da->decode_buffer, len);
// if format was changed, and all data was drained, execute the format change
if (format_change && eof) {
error = AD_NEW_FMT;
if (!reinit_audio_buffer(da))
error = AD_ERR; // switch to invalid format
}
return error;
}
/* Try to get at least minsamples decoded+filtered samples in outbuf
* (total length including possible existing data).
* Return 0 on success, or negative AD_* error code.
* In the former case outbuf has at least minsamples buffered on return.
* In case of EOF/error it might or might not be. */
int audio_decode(struct dec_audio *d_audio, struct mp_audio_buffer *outbuf,
int minsamples)
{
if (d_audio->afilter->initialized < 1)
return AD_ERR;
// Indicates that a filter seems to be buffering large amounts of data
int huge_filter_buffer = 0;
/* Filter output size will be about filter_multiplier times input size.
* If some filter buffers audio in big blocks this might only hold
* as average over time. */
double filter_multiplier = af_calc_filter_multiplier(d_audio->afilter);
int prev_buffered = -1;
int res = 0;
MP_STATS(d_audio, "start audio");
while (res >= 0 && minsamples >= 0) {
int buffered = mp_audio_buffer_samples(outbuf);
if (minsamples < buffered || buffered == prev_buffered)
break;
prev_buffered = buffered;
int decsamples = (minsamples - buffered) / filter_multiplier;
// + some extra for possible filter buffering, and avoid 0
decsamples += 512;
if (huge_filter_buffer) {
/* Some filter must be doing significant buffering if the estimated
* input length didn't produce enough output from filters.
* Feed the filters 250 samples at a time until we have enough
* output. Very small amounts could make filtering inefficient while
* large amounts can make mpv demux the file unnecessarily far ahead
* to get audio data and buffer video frames in memory while doing
* so. However the performance impact of either is probably not too
* significant as long as the value is not completely insane. */
decsamples = 250;
}
/* if this iteration does not fill buffer, we must have lots
* of buffering in filters */
huge_filter_buffer = 1;
res = filter_n_bytes(d_audio, outbuf, decsamples);
}
MP_STATS(d_audio, "end audio");
return res;
}
void audio_reset_decoding(struct dec_audio *d_audio)
{
if (d_audio->ad_driver)
d_audio->ad_driver->control(d_audio, ADCTRL_RESET, NULL);
af_control_all(d_audio->afilter, AF_CONTROL_RESET, NULL);
d_audio->pts = MP_NOPTS_VALUE;
d_audio->pts_offset = 0;
if (d_audio->decode_buffer)
mp_audio_buffer_clear(d_audio->decode_buffer);
}