summaryrefslogtreecommitdiffstats
path: root/audio/decode
diff options
context:
space:
mode:
Diffstat (limited to 'audio/decode')
-rw-r--r--audio/decode/ad.c2
-rw-r--r--audio/decode/ad_lavc.c99
-rw-r--r--audio/decode/ad_pcm.c220
3 files changed, 95 insertions, 226 deletions
diff --git a/audio/decode/ad.c b/audio/decode/ad.c
index ac344636e6..f091914244 100644
--- a/audio/decode/ad.c
+++ b/audio/decode/ad.c
@@ -33,7 +33,6 @@
extern const ad_functions_t mpcodecs_ad_mpg123;
extern const ad_functions_t mpcodecs_ad_ffmpeg;
-extern const ad_functions_t mpcodecs_ad_pcm;
extern const ad_functions_t mpcodecs_ad_dvdpcm;
extern const ad_functions_t mpcodecs_ad_spdif;
@@ -43,7 +42,6 @@ const ad_functions_t * const mpcodecs_ad_drivers[] =
&mpcodecs_ad_mpg123,
#endif
&mpcodecs_ad_ffmpeg,
- &mpcodecs_ad_pcm,
&mpcodecs_ad_dvdpcm,
&mpcodecs_ad_spdif,
NULL
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 8553e93cf1..e0e39e1e7d 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -59,6 +59,86 @@ struct priv {
int previous_data_left; // input demuxer packet data
};
+struct pcm_map
+{
+ int tag;
+ const char *codecs[5]; // {any, 1byte, 2bytes, 3bytes, 4bytes}
+};
+
+// NOTE: some of these are needed to make rawaudio with demux_mkv and others
+// work. ffmpeg does similar mapping internally, not part of the public
+// API. Some of these might be dead leftovers for demux_mov support.
+static const struct pcm_map tag_map[] = {
+ // Microsoft PCM
+ {0x0, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
+ {0x1, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
+ // MS PCM, Extended
+ {0xfffe, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
+ // IEEE float
+ {0x3, {"pcm_f32le"}},
+ // 'raw '
+ {0x20776172, {"pcm_s16be", [1] = "pcm_u8"}},
+ // 'twos'/'sowt'
+ {0x736F7774, {"pcm_s16be", [1] = "pcm_s8"}},
+ {0x74776F73, {"pcm_s16be", [1] = "pcm_s8"}},
+ // 'fl32'/'FL32'
+ {0x32336c66, {"pcm_f32be"}},
+ {0x32334C46, {"pcm_f32be"}},
+ // '23lf'/'lpcm'
+ {0x666c3332, {"pcm_f32le"}},
+ {0x6D63706C, {"pcm_f32le"}},
+ // 'in24', bigendian int24
+ {0x34326e69, {"pcm_s24be"}},
+ // '42ni', little endian int24, MPlayer internal fourCC
+ {0x696e3234, {"pcm_s24le"}},
+ // 'in32', bigendian int32
+ {0x32336e69, {"pcm_s32be"}},
+ // '23ni', little endian int32, MPlayer internal fourCC
+ {0x696e3332, {"pcm_s32le"}},
+ {-1},
+};
+
+// For demux_rawaudio.c; needed because ffmpeg doesn't have these sample
+// formats natively.
+static const struct pcm_map af_map[] = {
+ {AF_FORMAT_U8, {"pcm_u8"}},
+ {AF_FORMAT_S8, {"pcm_u8"}},
+ {AF_FORMAT_U16_LE, {"pcm_u16le"}},
+ {AF_FORMAT_U16_BE, {"pcm_u16be"}},
+ {AF_FORMAT_S16_LE, {"pcm_s16le"}},
+ {AF_FORMAT_S16_BE, {"pcm_s16be"}},
+ {AF_FORMAT_U24_LE, {"pcm_u24le"}},
+ {AF_FORMAT_U24_BE, {"pcm_u24be"}},
+ {AF_FORMAT_S24_LE, {"pcm_s24le"}},
+ {AF_FORMAT_S24_BE, {"pcm_s24be"}},
+ {AF_FORMAT_U32_LE, {"pcm_u32le"}},
+ {AF_FORMAT_U32_BE, {"pcm_u32be"}},
+ {AF_FORMAT_S32_LE, {"pcm_s32le"}},
+ {AF_FORMAT_S32_BE, {"pcm_s32be"}},
+ {AF_FORMAT_FLOAT_LE, {"pcm_f32le"}},
+ {AF_FORMAT_FLOAT_BE, {"pcm_f32be"}},
+ {-1},
+};
+
+static const char *find_pcm_decoder(const struct pcm_map *map, int format,
+ int bits_per_sample)
+{
+ int bytes = (bits_per_sample + 7) / 8;
+ for (int n = 0; map[n].tag != -1; n++) {
+ const struct pcm_map *entry = &map[n];
+ if (entry->tag == format) {
+ const char *dec = NULL;
+ if (bytes >= 1 && bytes <= 4)
+ dec = entry->codecs[bytes];
+ if (!dec)
+ dec = entry->codecs[0];
+ if (dec)
+ return dec;
+ }
+ }
+ return NULL;
+}
+
static int preinit(sh_audio_t *sh)
{
return 1;
@@ -115,12 +195,23 @@ static int init(sh_audio_t *sh_audio)
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
- if (sh_audio->codec->dll) {
- lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
+ const char *dll = sh_audio->codec->dll;
+
+ if (sh_audio->wf && dll && strcmp(dll, "pcm") == 0) {
+ if (sh_audio->format == MKTAG('M', 'P', 'a', 'f')) {
+ // demuxer_rawaudio convenience (abuses wFormatTag)
+ dll = find_pcm_decoder(af_map, sh_audio->wf->wFormatTag, 0);
+ } else {
+ dll = find_pcm_decoder(tag_map, sh_audio->format,
+ sh_audio->wf->wBitsPerSample);
+ }
+ }
+
+ if (dll) {
+ lavc_codec = avcodec_find_decoder_by_name(dll);
if (!lavc_codec) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
- "Cannot find codec '%s' in libavcodec...\n",
- sh_audio->codec->dll);
+ "Cannot find codec '%s' in libavcodec...\n", dll);
return 0;
}
} else if (!sh_audio->libav_codec_id) {
diff --git a/audio/decode/ad_pcm.c b/audio/decode/ad_pcm.c
deleted file mode 100644
index 01500b1274..0000000000
--- a/audio/decode/ad_pcm.c
+++ /dev/null
@@ -1,220 +0,0 @@
-/*
- * 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 <stdbool.h>
-
-#include <libavutil/common.h>
-
-#include "talloc.h"
-#include "config.h"
-#include "ad_internal.h"
-#include "audio/format.h"
-#include "audio/reorder_ch.h"
-
-static const ad_info_t info = {
- "Uncompressed PCM audio decoder",
- "pcm",
- "Nick Kurshev",
- "A'rpi",
- ""
-};
-
-struct ad_pcm_context {
- unsigned char *buffer;
- int buffer_pos;
- int buffer_len;
- int buffer_size;
-};
-
-LIBAD_EXTERN(pcm)
-
-static int init(sh_audio_t * sh_audio)
-{
- WAVEFORMATEX *h = sh_audio->wf;
- if (!h)
- return 0;
- sh_audio->i_bps = h->nAvgBytesPerSec;
- sh_audio->channels = h->nChannels;
- sh_audio->samplerate = h->nSamplesPerSec;
- sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
- sh_audio->sample_format = AF_FORMAT_S16_LE; // default
- switch (sh_audio->format) { /* hardware formats: */
- case 0x0:
- case 0x1: // Microsoft PCM
- case 0xfffe: // Extended
- switch (sh_audio->samplesize) {
- case 1: sh_audio->sample_format = AF_FORMAT_U8; break;
- case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
- case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
- case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
- }
- break;
- case 0x3: // IEEE float
- sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
- break;
- case 0x6: sh_audio->sample_format = AF_FORMAT_A_LAW; break;
- case 0x7: sh_audio->sample_format = AF_FORMAT_MU_LAW; break;
- case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM; break;
- case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2; break;
-/* case 0x2000: sh_audio->sample_format=AFMT_AC3; */
- case 0x20776172: // 'raw '
- sh_audio->sample_format = AF_FORMAT_S16_BE;
- if (sh_audio->samplesize == 1)
- sh_audio->sample_format = AF_FORMAT_U8;
- break;
- case 0x736F7774: // 'twos'
- sh_audio->sample_format = AF_FORMAT_S16_BE;
- // intended fall-through
- case 0x74776F73: // 'sowt'
- if (sh_audio->samplesize == 1)
- sh_audio->sample_format = AF_FORMAT_S8;
- break;
- case 0x32336c66: // 'fl32', bigendian float32
- case 0x32334C46: // 'FL32', bigendian float32 in aiff
- sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
- sh_audio->samplesize = 4;
- break;
- case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
- case 0x6D63706C: // 'lpcm'
- sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
- sh_audio->samplesize = 4;
- break;
-/* case 0x34366c66: // 'fl64', bigendian float64
- sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
- sh_audio->samplesize=8;
- break;
- case 0x666c3634: // '46lf', little endian float64, MPlayer internal fourCC
- sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
- sh_audio->samplesize=8;
- break;*/
- case 0x34326e69: // 'in24', bigendian int24
- sh_audio->sample_format = AF_FORMAT_S24_BE;
- sh_audio->samplesize = 3;
- break;
- case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
- sh_audio->sample_format = AF_FORMAT_S24_LE;
- sh_audio->samplesize = 3;
- break;
- case 0x32336e69: // 'in32', bigendian int32
- sh_audio->sample_format = AF_FORMAT_S32_BE;
- sh_audio->samplesize = 4;
- break;
- case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
- sh_audio->sample_format = AF_FORMAT_S32_LE;
- sh_audio->samplesize = 4;
- break;
- case MKTAG('M', 'P', 'a', 'f'):
- sh_audio->sample_format = h->wFormatTag;
- sh_audio->samplesize = (af_fmt2bits(sh_audio->sample_format) + 7) / 8;
- break;
- default:
- if (sh_audio->samplesize != 2)
- sh_audio->sample_format = AF_FORMAT_U8;
- }
- if (!sh_audio->samplesize) // this would cause MPlayer to hang later
- sh_audio->samplesize = 2;
- sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
- return 1;
-}
-
-static int preinit(sh_audio_t *sh)
-{
- sh->audio_out_minsize = 2048;
- return 1;
-}
-
-static void uninit(sh_audio_t *sh)
-{
- talloc_free(sh->context);
-}
-
-static int control(sh_audio_t *sh, int cmd, void *arg, ...)
-{
- struct ad_pcm_context *ctx = sh->context;
- int skip;
- switch (cmd) {
- case ADCTRL_RESYNC_STREAM:
- ctx->buffer_len = 0;
- return true;
- case ADCTRL_SKIP_FRAME:
- skip = sh->i_bps / 16;
- skip = skip & (~3);
- demux_read_data(sh->ds, NULL, skip);
- return CONTROL_TRUE;
- }
- return CONTROL_UNKNOWN;
-}
-
-static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
- int maxlen)
-{
- int unitsize = sh_audio->channels * sh_audio->samplesize;
- minlen = (minlen + unitsize - 1) / unitsize * unitsize;
- if (minlen > maxlen)
- // if someone needs hundreds of channels adjust audio_out_minsize
- // based on channels in preinit()
- return -1;
-
- int len = 0;
- struct ad_pcm_context *ctx = sh_audio->context;
- while (len < minlen) {
- if (ctx->buffer_len - ctx->buffer_pos <= 0) {
- double pts;
- unsigned char *ptr;
- int plen = ds_get_packet_pts(sh_audio->ds, &ptr, &pts);
- if (plen < 0)
- break;
- if (ctx->buffer_size < plen) {
- talloc_free(ctx->buffer);
- ctx->buffer = talloc_size(ctx, plen);
- ctx->buffer_size = plen;
- }
- memcpy(ctx->buffer, ptr, plen);
- ctx->buffer_len = plen;
- ctx->buffer_pos = 0;
- if (pts != MP_NOPTS_VALUE) {
- sh_audio->pts = pts;
- sh_audio->pts_bytes = 0;
- }
- }
- int from_stored = ctx->buffer_len - ctx->buffer_pos;
- if (from_stored > minlen - len)
- from_stored = minlen - len;
- memcpy(buf + len, ctx->buffer + ctx->buffer_pos, from_stored);
- ctx->buffer_pos += from_stored;
- sh_audio->pts_bytes += from_stored;
- len += from_stored;
- }
- if (len % unitsize) {
- mp_msg(MSGT_DECAUDIO, MSGL_WARN, "[ad_pcm] discarding partial sample "
- "at end\n");
- len -= len % unitsize;
- }
- if (len == 0)
- len = -1; // The loop above only exits at error/EOF
- if (len > 0 && sh_audio->channels >= 5) {
- reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
- AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
- sh_audio->channels, len / sh_audio->samplesize,
- sh_audio->samplesize);
- }
- return len;
-}