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-rw-r--r--libao2/ao_pcm.c249
1 files changed, 114 insertions, 135 deletions
diff --git a/libao2/ao_pcm.c b/libao2/ao_pcm.c
index 64eda888f7..20399152c5 100644
--- a/libao2/ao_pcm.c
+++ b/libao2/ao_pcm.c
@@ -24,13 +24,14 @@
#include <stdlib.h>
#include <string.h>
-#include "libavutil/common.h"
-#include "mpbswap.h"
+#include <libavutil/common.h>
+
+#include "talloc.h"
+
#include "subopt-helper.h"
#include "libaf/af_format.h"
#include "libaf/reorder_ch.h"
#include "audio_out.h"
-#include "audio_out_internal.h"
#include "mp_msg.h"
#ifdef __MINGW32__
@@ -38,22 +39,13 @@
#include <windows.h>
#endif
-static const ao_info_t info =
-{
- "RAW PCM/WAVE file writer audio output",
- "pcm",
- "Atmosfear",
- ""
+struct priv {
+ char *outputfilename;
+ int waveheader;
+ uint64_t data_length;
+ FILE *fp;
};
-LIBAO_EXTERN(pcm)
-
-extern int vo_pts;
-
-static char *ao_outputfilename = NULL;
-static int ao_pcm_waveheader = 1;
-static int fast = 0;
-
#define WAV_ID_RIFF 0x46464952 /* "RIFF" */
#define WAV_ID_WAVE 0x45564157 /* "WAVE" */
#define WAV_ID_FMT 0x20746d66 /* "fmt " */
@@ -62,30 +54,30 @@ static int fast = 0;
#define WAV_ID_FLOAT_PCM 0x0003
#define WAV_ID_FORMAT_EXTENSIBLE 0xfffe
-/* init with default values */
-static uint64_t data_length;
-static FILE *fp = NULL;
-
-
-static void fput16le(uint16_t val, FILE *fp) {
+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) {
+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(FILE *fp, uint64_t data_length) {
- int use_waveex = (ao_data.channels >= 5 && ao_data.channels <= 8);
- uint16_t fmt = (ao_data.format == AF_FORMAT_FLOAT_LE) ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
+static void write_wave_header(struct ao *ao, FILE *fp, uint64_t data_length)
+{
+ bool use_waveex = ao->channels >= 5 && ao->channels <= 8;
+ uint16_t fmt = ao->format == AF_FORMAT_FLOAT_LE ?
+ WAV_ID_FLOAT_PCM : WAV_ID_PCM;
uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
- int bits = af_fmt2bits(ao_data.format);
+ int bits = af_fmt2bits(ao->format);
// Master RIFF chunk
fput32le(WAV_ID_RIFF, fp);
- // RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size + data chunk hdr (8) + data length
+ // RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size +
+ // data chunk hdr (8) + data length
fput32le(12 + fmt_chunk_size + 8 + data_length, fp);
fput32le(WAV_ID_WAVE, fp);
@@ -93,17 +85,17 @@ static void write_wave_header(FILE *fp, uint64_t data_length) {
fput32le(WAV_ID_FMT, fp);
fput32le(fmt_chunk_size, fp);
fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp);
- fput16le(ao_data.channels, fp);
- fput32le(ao_data.samplerate, fp);
- fput32le(ao_data.bps, fp);
- fput16le(ao_data.channels * (bits / 8), fp);
+ fput16le(ao->channels, fp);
+ fput32le(ao->samplerate, fp);
+ fput32le(ao->bps, fp);
+ fput16le(ao->channels * (bits / 8), fp);
fput16le(bits, fp);
if (use_waveex) {
// Extension chunk
fput16le(22, fp);
fput16le(bits, fp);
- switch (ao_data.channels) {
+ switch (ao->channels) {
case 5:
fput32le(0x0607, fp); // L R C Lb Rb
break;
@@ -129,36 +121,36 @@ static void write_wave_header(FILE *fp, uint64_t data_length) {
fput32le(data_length, fp);
}
-// to set/get/query special features/parameters
-static int control(int cmd,void *arg){
- return -1;
-}
+static int init(struct ao *ao, char *params)
+{
+ struct priv *priv = talloc_zero(ao, struct priv);
+ ao->priv = priv;
-// open & setup audio device
-// return: 1=success 0=fail
-static int init(int rate,int channels,int format,int flags){
+ int fast;
const opt_t subopts[] = {
- {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
- {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
- {"fast", OPT_ARG_BOOL, &fast, NULL},
+ {"waveheader", OPT_ARG_BOOL, &priv->waveheader, NULL},
+ {"file", OPT_ARG_MSTRZ, &priv->outputfilename, NULL},
+ {"fast", OPT_ARG_BOOL, &fast, NULL},
{NULL}
};
// set defaults
- ao_pcm_waveheader = 1;
-
- if (subopt_parse(ao_subdevice, subopts) != 0) {
- return 0;
- }
- if (!ao_outputfilename){
- ao_outputfilename =
- strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
- }
-
- if (ao_pcm_waveheader)
- {
+ priv->waveheader = 1;
+
+ if (subopt_parse(params, subopts) != 0)
+ return -1;
+
+ if (fast)
+ mp_msg(MSGT_AO, MSGL_WARN,
+ "[AO PCM] Suboption \"fast\" is deprecated.\n"
+ "[AO PCM] Use -novideo, or -benchmark if you want "
+ "faster playback with video.\n");
+ if (!priv->outputfilename)
+ priv->outputfilename =
+ strdup(priv->waveheader ? "audiodump.wav" : "audiodump.pcm");
+ if (priv->waveheader) {
// WAV files must have one of the following formats
- switch(format){
+ switch (ao->format) {
case AF_FORMAT_U8:
case AF_FORMAT_S16_LE:
case AF_FORMAT_S24_LE:
@@ -168,110 +160,97 @@ static int init(int rate,int channels,int format,int flags){
case AF_FORMAT_AC3_LE:
break;
default:
- format = AF_FORMAT_S16_LE;
+ ao->format = AF_FORMAT_S16_LE;
break;
}
}
- ao_data.outburst = 65536;
- ao_data.buffersize= 2*65536;
- ao_data.channels=channels;
- ao_data.samplerate=rate;
- ao_data.format=format;
- ao_data.bps=channels*rate*(af_fmt2bits(format)/8);
-
- mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\nPCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename,
- (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
- (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
- mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] Info: Faster dumping is achieved with -novideo -ao pcm:fast\n[AO PCM] Info: To write WAVE files use -ao pcm:waveheader (default).\n");
-
- fp = fopen(ao_outputfilename, "wb");
- if(fp) {
- if(ao_pcm_waveheader){ /* Reserve space for wave header */
- write_wave_header(fp, 0x7ffff000);
- }
- return 1;
+ ao->outburst = 65536;
+ ao->buffersize = 2 * 65536;
+ ao->bps = ao->channels * ao->samplerate * (af_fmt2bits(ao->format) / 8);
+
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\n"
+ "PCM: Samplerate: %d Hz Channels: %d Format: %s\n",
+ priv->outputfilename,
+ priv->waveheader ? "WAVE" : "RAW PCM", ao->samplerate,
+ ao->channels, af_fmt2str_short(ao->format));
+ mp_tmsg(MSGT_AO, MSGL_INFO,
+ "[AO PCM] Info: Faster dumping is achieved with -novideo\n"
+ "[AO PCM] Info: To write WAVE files use -ao pcm:waveheader (default).\n");
+
+ priv->fp = fopen(priv->outputfilename, "wb");
+ if (!priv->fp) {
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n",
+ priv->outputfilename);
+ return -1;
}
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n",
- ao_outputfilename);
+ 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(int immed){
+static void uninit(struct ao *ao, bool cut_audio)
+{
+ struct priv *priv = ao->priv;
- if(ao_pcm_waveheader){ /* Rewrite wave header */
- int broken_seek = 0;
+ 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 = GetFileType((HANDLE)_get_osfhandle(_fileno(fp))) != FILE_TYPE_DISK;
+ // 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(fp, 0, SEEK_SET) != 0)
- mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
+ if (broken_seek || fseek(priv->fp, 0, SEEK_SET) != 0)
+ mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, "
+ "WAV size headers not updated!\n");
else {
- if (data_length > 0xfffff000) {
- mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
- data_length = 0xfffff000;
+ if (priv->data_length > 0xfffff000) {
+ mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for "
+ "WAV files, may play truncated!\n");
+ priv->data_length = 0xfffff000;
}
- write_wave_header(fp, data_length);
+ write_wave_header(ao, priv->fp, priv->data_length);
}
}
- fclose(fp);
- free(ao_outputfilename);
- ao_outputfilename = NULL;
+ fclose(priv->fp);
+ free(priv->outputfilename);
}
-// stop playing and empty buffers (for seeking/pause)
-static void reset(void){
-
-}
-
-// stop playing, keep buffers (for pause)
-static void audio_pause(void)
+static int get_space(struct ao *ao)
{
- // for now, just call reset();
- reset();
+ return ao->outburst;
}
-// resume playing, after audio_pause()
-static void audio_resume(void)
+static int play(struct ao *ao, void *data, int len, int flags)
{
-}
-
-// return: how many bytes can be played without blocking
-static int get_space(void){
+ struct priv *priv = ao->priv;
- if(vo_pts)
- return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
- return ao_data.outburst;
-}
-
-// plays 'len' bytes of 'data'
-// it should round it down to outburst*n
-// return: number of bytes played
-static int play(void* data,int len,int flags){
-
- if (ao_data.channels == 5 || ao_data.channels == 6 || ao_data.channels == 8) {
- int frame_size = af_fmt2bits(ao_data.format) / 8;
- len -= len % (frame_size * ao_data.channels);
+ if (ao->channels == 5 || ao->channels == 6 || ao->channels == 8) {
+ int frame_size = af_fmt2bits(ao->format) / 8;
+ len -= len % (frame_size * ao->channels);
reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
- ao_data.channels,
- len / frame_size, frame_size);
+ ao->channels, len / frame_size, frame_size);
}
-
- //printf("PCM: Writing chunk!\n");
- fwrite(data,len,1,fp);
-
- if(ao_pcm_waveheader)
- data_length += len;
-
+ fwrite(data, len, 1, priv->fp);
+ priv->data_length += len;
return len;
}
-// return: delay in seconds between first and last sample in buffer
-static float get_delay(void){
-
- return 0.0;
-}
+const struct ao_driver audio_out_pcm = {
+ .is_new = true,
+ .info = &(const struct ao_info) {
+ "RAW PCM/WAVE file writer audio output",
+ "pcm",
+ "Atmosfear",
+ "",
+ },
+ .init = init,
+ .uninit = uninit,
+ .get_space = get_space,
+ .play = play,
+};