mpv/audio/out/ao_pcm.c

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/*
* 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 General Public License as published by
* the Free Software Foundation; either version 2 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 General Public License for more details.
*
* You should have received a copy of the GNU 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)
{
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struct priv *priv = ao->priv;
if (!priv->outputfilename)
priv->outputfilename =
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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_S24:
case AF_FORMAT_S32:
case AF_FORMAT_FLOAT:
break;
default:
if (!af_fmt_is_spdif(ao->format))
audio: cleanup spdif format definitions Before this commit, there was AF_FORMAT_AC3 (the original spdif format, used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS and DTS-HD), which was handled as some sort of superset for AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used IEC61937-framing, but still was handled as something "separate". Technically, all of them are pretty similar, but may use different bitrates. Since digital passthrough pretends to be PCM (just with special headers that wrap digital packets), this is easily detectable by the higher samplerate or higher number of channels, so I don't know why you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs. AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is just a mess. Simplify this by handling all these formats the same way. AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3). All AOs just accept all spdif formats now - whether that works or not is not really clear (seems inconsistent due to earlier attempts to make DTS-HD work). But on the other hand, enabling spdif requires manual user interaction, so it doesn't matter much if initialization fails in slightly less graceful ways if it can't work at all. At a later point, we will support passthrough with ao_pulse. It seems the PulseAudio API wants to know the codec type (or maybe not - feeding it DTS while telling it it's AC3 works), add separate formats for each codecs. While this reminds of the earlier chaos, it's stricter, and most code just uses AF_FORMAT_IS_IEC61937(). Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to include special formats, so that it always describes the fundamental sample format type. This also ensures valid AF formats are never 0 (this was probably broken in one of the earlier commits from today).
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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",
priv->outputfilename,
priv->waveheader ? "WAVE" : "RAW PCM", ao->samplerate,
ao->channels.num, af_fmt_to_str(ao->format));
MP_INFO(ao, "Info: Faster dumping is achieved with --no-video\n");
MP_INFO(ao, "Info: To write WAVE files use --ao=pcm:waveheader (default).\n");
priv->fp = fopen(priv->outputfilename, priv->append ? "ab" : "wb");
if (!priv->fp) {
MP_ERR(ao, "Failed to open %s for writing!\n", priv->outputfilename);
return -1;
}
if (priv->waveheader) // Reserve space for wave header
write_wave_header(ao, priv->fp, 0x7ffff000);
ao->untimed = true;
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 int get_space(struct ao *ao)
{
return 65536;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *priv = ao->priv;
int len = samples * ao->sstride;
fwrite(data[0], len, 1, priv->fp);
priv->data_length += len;
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return samples;
}
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#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_space = get_space,
.play = play,
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.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) { .waveheader = 1 },
.options = (const struct m_option[]) {
OPT_STRING("file", outputfilename, 0),
OPT_FLAG("waveheader", waveheader, 0),
OPT_FLAG("append", append, 0),
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{0}
},
};