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mpv/audio/out/ao_pcm.c
wm4 d27ad96542 audio: redo internal AO API
This affects "pull" AOs only: ao_alsa, ao_pulse, ao_openal, ao_pcm,
ao_lavc. There are changes to the other AOs too, but that's only about
renaming ao_driver.resume to ao_driver.start.

ao_openal is broken because I didn't manage to fix it, so it exits with
an error message. If you want it, why don't _you_ put effort into it? I
see no reason to waste my own precious lifetime over this (I realize the
irony).

ao_alsa loses the poll() mechanism, but it was mostly broken and didn't
really do what it was supposed to. There doesn't seem to be anything in
the ALSA API to watch the playback status without polling (unless you
want to use raw UNIX signals).

No idea if ao_pulse is correct, or whether it's subtly broken now. There
is no documentation, so I can't tell what is correct, without reverse
engineering the whole project. I recommend using ALSA.

This was supposed to be just a simple fix, but somehow it expanded scope
like a train wreck. Very high chance of regressions, but probably only
for the AOs listed above. The rest you can figure out from reading the
diff.
2020-06-01 01:08:16 +02:00

251 lines
6.8 KiB
C

/*
* PCM audio output driver
*
* Original author: Atmosfear
*
* 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 "config.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavutil/common.h>
#include "mpv_talloc.h"
#include "options/m_option.h"
#include "audio/format.h"
#include "ao.h"
#include "internal.h"
#include "common/msg.h"
#include "osdep/endian.h"
#ifdef __MINGW32__
// for GetFileType to detect pipes
#include <windows.h>
#include <io.h>
#endif
struct priv {
char *outputfilename;
int waveheader;
int append;
uint64_t data_length;
FILE *fp;
};
#define WAV_ID_RIFF 0x46464952 /* "RIFF" */
#define WAV_ID_WAVE 0x45564157 /* "WAVE" */
#define WAV_ID_FMT 0x20746d66 /* "fmt " */
#define WAV_ID_DATA 0x61746164 /* "data" */
#define WAV_ID_PCM 0x0001
#define WAV_ID_FLOAT_PCM 0x0003
#define WAV_ID_FORMAT_EXTENSIBLE 0xfffe
static void fput16le(uint16_t val, FILE *fp)
{
uint8_t bytes[2] = {val, val >> 8};
fwrite(bytes, 1, 2, fp);
}
static void fput32le(uint32_t val, FILE *fp)
{
uint8_t bytes[4] = {val, val >> 8, val >> 16, val >> 24};
fwrite(bytes, 1, 4, fp);
}
static void write_wave_header(struct ao *ao, FILE *fp, uint64_t data_length)
{
uint16_t fmt = ao->format == AF_FORMAT_FLOAT ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
int bits = af_fmt_to_bytes(ao->format) * 8;
// Master RIFF chunk
fput32le(WAV_ID_RIFF, fp);
// RIFF chunk size: 'WAVE' + 'fmt ' + 4 + 40 +
// data chunk hdr (8) + data length
fput32le(12 + 40 + 8 + data_length, fp);
fput32le(WAV_ID_WAVE, fp);
// Format chunk
fput32le(WAV_ID_FMT, fp);
fput32le(40, fp);
fput16le(WAV_ID_FORMAT_EXTENSIBLE, fp);
fput16le(ao->channels.num, fp);
fput32le(ao->samplerate, fp);
fput32le(ao->bps, fp);
fput16le(ao->channels.num * (bits / 8), fp);
fput16le(bits, fp);
// Extension chunk
fput16le(22, fp);
fput16le(bits, fp);
fput32le(mp_chmap_to_waveext(&ao->channels), fp);
// 2 bytes format + 14 bytes guid
fput32le(fmt, fp);
fput32le(0x00100000, fp);
fput32le(0xAA000080, fp);
fput32le(0x719B3800, fp);
// Data chunk
fput32le(WAV_ID_DATA, fp);
fput32le(data_length, fp);
}
static int init(struct ao *ao)
{
struct priv *priv = ao->priv;
char *outputfilename = priv->outputfilename;
if (!outputfilename) {
outputfilename = talloc_strdup(priv, priv->waveheader ? "audiodump.wav"
: "audiodump.pcm");
}
ao->format = af_fmt_from_planar(ao->format);
if (priv->waveheader) {
// WAV files must have one of the following formats
// And they don't work in big endian; fixing it would be simple, but
// nobody cares.
if (BYTE_ORDER == BIG_ENDIAN) {
MP_FATAL(ao, "Not supported on big endian.\n");
return -1;
}
switch (ao->format) {
case AF_FORMAT_U8:
case AF_FORMAT_S16:
case AF_FORMAT_S32:
case AF_FORMAT_FLOAT:
break;
default:
if (!af_fmt_is_spdif(ao->format))
ao->format = AF_FORMAT_S16;
break;
}
}
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_waveext(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
return -1;
ao->bps = ao->channels.num * ao->samplerate * af_fmt_to_bytes(ao->format);
MP_INFO(ao, "File: %s (%s)\nPCM: Samplerate: %d Hz Channels: %d Format: %s\n",
outputfilename,
priv->waveheader ? "WAVE" : "RAW PCM", ao->samplerate,
ao->channels.num, af_fmt_to_str(ao->format));
priv->fp = fopen(outputfilename, priv->append ? "ab" : "wb");
if (!priv->fp) {
MP_ERR(ao, "Failed to open %s for writing!\n", outputfilename);
return -1;
}
if (priv->waveheader) // Reserve space for wave header
write_wave_header(ao, priv->fp, 0x7ffff000);
ao->untimed = true;
ao->device_buffer = 1 << 16;
return 0;
}
// close audio device
static void uninit(struct ao *ao)
{
struct priv *priv = ao->priv;
if (priv->waveheader) { // Rewrite wave header
bool broken_seek = false;
#ifdef __MINGW32__
// Windows, in its usual idiocy "emulates" seeks on pipes so it always
// looks like they work. So we have to detect them brute-force.
broken_seek = FILE_TYPE_DISK !=
GetFileType((HANDLE)_get_osfhandle(_fileno(priv->fp)));
#endif
if (broken_seek || fseek(priv->fp, 0, SEEK_SET) != 0)
MP_ERR(ao, "Could not seek to start, WAV size headers not updated!\n");
else {
if (priv->data_length > 0xfffff000) {
MP_ERR(ao, "File larger than allowed for "
"WAV files, may play truncated!\n");
priv->data_length = 0xfffff000;
}
write_wave_header(ao, priv->fp, priv->data_length);
}
}
fclose(priv->fp);
}
static bool audio_write(struct ao *ao, void **data, int samples)
{
struct priv *priv = ao->priv;
int len = samples * ao->sstride;
fwrite(data[0], len, 1, priv->fp);
priv->data_length += len;
return true;
}
static void get_state(struct ao *ao, struct mp_pcm_state *state)
{
state->free_samples = ao->device_buffer;
state->queued_samples = 0;
state->delay = 0;
}
static bool set_pause(struct ao *ao, bool paused)
{
return true; // signal support so common code doesn't write silence
}
static void start(struct ao *ao)
{
// we use data immediately
}
static void reset(struct ao *ao)
{
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_pcm = {
.description = "RAW PCM/WAVE file writer audio output",
.name = "pcm",
.init = init,
.uninit = uninit,
.get_state = get_state,
.set_pause = set_pause,
.write = audio_write,
.start = start,
.reset = reset,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) { .waveheader = 1 },
.options = (const struct m_option[]) {
{"file", OPT_STRING(outputfilename), .flags = M_OPT_FILE},
{"waveheader", OPT_FLAG(waveheader)},
{"append", OPT_FLAG(append)},
{0}
},
.options_prefix = "ao-pcm",
};