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authorwm4 <wm4@nowhere>2012-09-02 19:30:34 +0200
committerwm4 <wm4@nowhere>2012-09-18 21:04:47 +0200
commitee65b39cbe8a1df6bf4b2eaeef413c881ae16456 (patch)
tree6877c6130a51a67ae12f08272a18c8548dc05ba5 /libmpcodecs/ad_pcm.c
parent53bfaecd40aa7b34dcc9c90064e36f2ffe227fc0 (diff)
downloadmpv-ee65b39cbe8a1df6bf4b2eaeef413c881ae16456.tar.bz2
mpv-ee65b39cbe8a1df6bf4b2eaeef413c881ae16456.tar.xz
ad_pcm: add back raw decoder
This was removed in commit 6a26b4a66504. Add it back, because it was needed by demuxer_rawaudio and for PCM audio with demuxers other than demux_lavf. (In practice, this broke rawaudio and PCM-in-Matroska only.) Unlike with raw video, there is no single raw audio "decoder" in libavcodec. Instead of trying to mess raw audio input into ad_ffmpeg using a table to map audio formats to the respective libavcodec decoders, it seems advantageous to simply add back ad_pcm.
Diffstat (limited to 'libmpcodecs/ad_pcm.c')
-rw-r--r--libmpcodecs/ad_pcm.c214
1 files changed, 214 insertions, 0 deletions
diff --git a/libmpcodecs/ad_pcm.c b/libmpcodecs/ad_pcm.c
new file mode 100644
index 0000000000..6ddae6afeb
--- /dev/null
+++ b/libmpcodecs/ad_pcm.c
@@ -0,0 +1,214 @@
+/*
+ * 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 "talloc.h"
+#include "config.h"
+#include "ad_internal.h"
+#include "libaf/format.h"
+#include "libaf/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;
+ 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;
+}