mirror of
https://github.com/mpv-player/mpv
synced 2024-11-18 21:16:10 +01:00
380fc765e4
This comes with two internal AO API changes: 1. ao_driver.play now can take non-interleaved audio. For this purpose, the data pointer is changed to void **data, where data[0] corresponds to the pointer in the old API. Also, the len argument as well as the return value are now in samples, not bytes. "Sample" in this context means the unit of the smallest possible audio frame, i.e. sample_size * channels. 2. ao_driver.get_space now returns samples instead of bytes. (Similar to the play function.) Change all AOs to use the new API. The AO API as exposed to the rest of the player still uses the old API. It's emulated in ao.c. This is purely to split the commits changing all AOs and the commits adding actual support for outputting N-I audio.
226 lines
6.4 KiB
C
226 lines
6.4 KiB
C
/*
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* PCM audio output driver
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*
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* This file is part of MPlayer.
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*
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* Original author: Atmosfear
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <libavutil/common.h>
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#include "talloc.h"
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#include "mpvcore/m_option.h"
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#include "audio/format.h"
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#include "audio/reorder_ch.h"
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#include "ao.h"
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#include "mpvcore/mp_msg.h"
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#ifdef __MINGW32__
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// for GetFileType to detect pipes
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#include <windows.h>
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#include <io.h>
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#endif
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struct priv {
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char *outputfilename;
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int waveheader;
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uint64_t data_length;
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FILE *fp;
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};
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#define WAV_ID_RIFF 0x46464952 /* "RIFF" */
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#define WAV_ID_WAVE 0x45564157 /* "WAVE" */
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#define WAV_ID_FMT 0x20746d66 /* "fmt " */
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#define WAV_ID_DATA 0x61746164 /* "data" */
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#define WAV_ID_PCM 0x0001
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#define WAV_ID_FLOAT_PCM 0x0003
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#define WAV_ID_FORMAT_EXTENSIBLE 0xfffe
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static void fput16le(uint16_t val, FILE *fp)
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{
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uint8_t bytes[2] = {val, val >> 8};
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fwrite(bytes, 1, 2, fp);
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}
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static void fput32le(uint32_t val, FILE *fp)
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{
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uint8_t bytes[4] = {val, val >> 8, val >> 16, val >> 24};
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fwrite(bytes, 1, 4, fp);
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}
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static void write_wave_header(struct ao *ao, FILE *fp, uint64_t data_length)
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{
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bool use_waveex = true;
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uint16_t fmt = ao->format == AF_FORMAT_FLOAT_LE ?
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WAV_ID_FLOAT_PCM : WAV_ID_PCM;
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uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
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int bits = af_fmt2bits(ao->format);
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// Master RIFF chunk
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fput32le(WAV_ID_RIFF, fp);
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// RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size +
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// data chunk hdr (8) + data length
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fput32le(12 + fmt_chunk_size + 8 + data_length, fp);
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fput32le(WAV_ID_WAVE, fp);
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// Format chunk
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fput32le(WAV_ID_FMT, fp);
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fput32le(fmt_chunk_size, fp);
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fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp);
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fput16le(ao->channels.num, fp);
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fput32le(ao->samplerate, fp);
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fput32le(ao->bps, fp);
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fput16le(ao->channels.num * (bits / 8), fp);
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fput16le(bits, fp);
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if (use_waveex) {
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// Extension chunk
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fput16le(22, fp);
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fput16le(bits, fp);
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fput32le(mp_chmap_to_waveext(&ao->channels), fp);
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// 2 bytes format + 14 bytes guid
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fput32le(fmt, fp);
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fput32le(0x00100000, fp);
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fput32le(0xAA000080, fp);
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fput32le(0x719B3800, fp);
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}
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// Data chunk
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fput32le(WAV_ID_DATA, fp);
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fput32le(data_length, fp);
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}
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static int init(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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if (!priv->outputfilename)
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priv->outputfilename =
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talloc_strdup(priv, priv->waveheader ? "audiodump.wav" : "audiodump.pcm");
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ao->format = af_fmt_from_planar(ao->format);
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if (priv->waveheader) {
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// WAV files must have one of the following formats
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switch (ao->format) {
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case AF_FORMAT_U8:
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case AF_FORMAT_S16_LE:
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case AF_FORMAT_S24_LE:
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case AF_FORMAT_S32_LE:
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case AF_FORMAT_FLOAT_LE:
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case AF_FORMAT_AC3_LE:
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break;
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default:
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ao->format = AF_FORMAT_S16_LE;
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break;
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}
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}
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struct mp_chmap_sel sel = {0};
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mp_chmap_sel_add_waveext(&sel);
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
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return -1;
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ao->bps = ao->channels.num * ao->samplerate * (af_fmt2bits(ao->format) / 8);
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MP_INFO(ao, "File: %s (%s)\nPCM: Samplerate: %d Hz Channels: %d Format: %s\n",
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priv->outputfilename,
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priv->waveheader ? "WAVE" : "RAW PCM", ao->samplerate,
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ao->channels.num, af_fmt_to_str(ao->format));
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MP_INFO(ao, "Info: Faster dumping is achieved with -no-video\n");
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MP_INFO(ao, "Info: To write WAVE files use -ao pcm:waveheader (default).\n");
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priv->fp = fopen(priv->outputfilename, "wb");
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if (!priv->fp) {
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MP_ERR(ao, "Failed to open %s for writing!\n", priv->outputfilename);
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return -1;
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}
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if (priv->waveheader) // Reserve space for wave header
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write_wave_header(ao, priv->fp, 0x7ffff000);
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ao->untimed = true;
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return 0;
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}
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// close audio device
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static void uninit(struct ao *ao, bool cut_audio)
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{
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struct priv *priv = ao->priv;
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if (priv->waveheader) { // Rewrite wave header
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bool broken_seek = false;
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#ifdef __MINGW32__
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// Windows, in its usual idiocy "emulates" seeks on pipes so it always
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// looks like they work. So we have to detect them brute-force.
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broken_seek = FILE_TYPE_DISK !=
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GetFileType((HANDLE)_get_osfhandle(_fileno(priv->fp)));
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#endif
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if (broken_seek || fseek(priv->fp, 0, SEEK_SET) != 0)
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MP_ERR(ao, "Could not seek to start, WAV size headers not updated!\n");
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else {
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if (priv->data_length > 0xfffff000) {
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MP_ERR(ao, "File larger than allowed for "
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"WAV files, may play truncated!\n");
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priv->data_length = 0xfffff000;
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}
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write_wave_header(ao, priv->fp, priv->data_length);
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}
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}
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fclose(priv->fp);
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}
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static int get_space(struct ao *ao)
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{
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return 65536;
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}
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static int play(struct ao *ao, void **data, int samples, int flags)
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{
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struct priv *priv = ao->priv;
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int len = samples * ao->sstride;
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fwrite(data[0], len, 1, priv->fp);
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priv->data_length += len;
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return len / ao->sstride;
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}
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#define OPT_BASE_STRUCT struct priv
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const struct ao_driver audio_out_pcm = {
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.description = "RAW PCM/WAVE file writer audio output",
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.name = "pcm",
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.init = init,
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.uninit = uninit,
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.get_space = get_space,
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.play = play,
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.priv_size = sizeof(struct priv),
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.priv_defaults = &(const struct priv) { .waveheader = 1 },
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.options = (const struct m_option[]) {
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OPT_STRING("file", outputfilename, 0),
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OPT_FLAG("waveheader", waveheader, 0),
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{0}
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},
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};
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