[FFmpeg-cvslog] westwood: split the AUD and VQA demuxers into separate files.
Justin Ruggles
git at videolan.org
Tue Jan 24 03:04:43 CET 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Fri Jan 20 13:56:51 2012 -0500| [e924a4ba715da13260ed83262469a8a56aad427b] | committer: Justin Ruggles
westwood: split the AUD and VQA demuxers into separate files.
They have no code in common.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e924a4ba715da13260ed83262469a8a56aad427b
---
libavformat/Makefile | 4 +-
libavformat/westwood_aud.c | 173 ++++++++++++++++++++++++++++
libavformat/{westwood.c => westwood_vqa.c} | 147 +-----------------------
3 files changed, 177 insertions(+), 147 deletions(-)
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 2a2a946..a19ff3c 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -308,8 +308,8 @@ OBJS-$(CONFIG_WC3_DEMUXER) += wc3movie.o
OBJS-$(CONFIG_WEBM_MUXER) += matroskaenc.o matroska.o \
riff.o isom.o avc.o \
flacenc_header.o avlanguage.o
-OBJS-$(CONFIG_WSAUD_DEMUXER) += westwood.o
-OBJS-$(CONFIG_WSVQA_DEMUXER) += westwood.o
+OBJS-$(CONFIG_WSAUD_DEMUXER) += westwood_aud.o
+OBJS-$(CONFIG_WSVQA_DEMUXER) += westwood_vqa.o
OBJS-$(CONFIG_WTV_DEMUXER) += wtv.o asfdec.o asf.o asfcrypt.o \
avlanguage.o mpegts.o isom.o riff.o
OBJS-$(CONFIG_WV_DEMUXER) += wv.o apetag.o
diff --git a/libavformat/westwood_aud.c b/libavformat/westwood_aud.c
new file mode 100644
index 0000000..b1eb768
--- /dev/null
+++ b/libavformat/westwood_aud.c
@@ -0,0 +1,173 @@
+/*
+ * Westwood Studios AUD Format Demuxer
+ * Copyright (c) 2003 The ffmpeg Project
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Westwood Studios AUD file demuxer
+ * by Mike Melanson (melanson at pcisys.net)
+ * for more information on the Westwood file formats, visit:
+ * http://www.pcisys.net/~melanson/codecs/
+ * http://www.geocities.com/SiliconValley/8682/aud3.txt
+ *
+ * Implementation note: There is no definite file signature for AUD files.
+ * The demuxer uses a probabilistic strategy for content detection. This
+ * entails performing sanity checks on certain header values in order to
+ * qualify a file. Refer to wsaud_probe() for the precise parameters.
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "avformat.h"
+#include "internal.h"
+
+#define AUD_HEADER_SIZE 12
+#define AUD_CHUNK_PREAMBLE_SIZE 8
+#define AUD_CHUNK_SIGNATURE 0x0000DEAF
+
+typedef struct WsAudDemuxContext {
+ int audio_samplerate;
+ int audio_channels;
+ int audio_bits;
+ enum CodecID audio_type;
+ int audio_stream_index;
+ int64_t audio_frame_counter;
+} WsAudDemuxContext;
+
+static int wsaud_probe(AVProbeData *p)
+{
+ int field;
+
+ /* Probabilistic content detection strategy: There is no file signature
+ * so perform sanity checks on various header parameters:
+ * 8000 <= sample rate (16 bits) <= 48000 ==> 40001 acceptable numbers
+ * flags <= 0x03 (2 LSBs are used) ==> 4 acceptable numbers
+ * compression type (8 bits) = 1 or 99 ==> 2 acceptable numbers
+ * first audio chunk signature (32 bits) ==> 1 acceptable number
+ * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
+ * 320008 acceptable number combinations.
+ */
+
+ if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
+ return 0;
+
+ /* check sample rate */
+ field = AV_RL16(&p->buf[0]);
+ if ((field < 8000) || (field > 48000))
+ return 0;
+
+ /* enforce the rule that the top 6 bits of this flags field are reserved (0);
+ * this might not be true, but enforce it until deemed unnecessary */
+ if (p->buf[10] & 0xFC)
+ return 0;
+
+ /* note: only check for WS IMA (type 99) right now since there is no
+ * support for type 1 */
+ if (p->buf[11] != 99)
+ return 0;
+
+ /* read ahead to the first audio chunk and validate the first header signature */
+ if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
+ return 0;
+
+ /* return 1/2 certainty since this file check is a little sketchy */
+ return AVPROBE_SCORE_MAX / 2;
+}
+
+static int wsaud_read_header(AVFormatContext *s,
+ AVFormatParameters *ap)
+{
+ WsAudDemuxContext *wsaud = s->priv_data;
+ AVIOContext *pb = s->pb;
+ AVStream *st;
+ unsigned char header[AUD_HEADER_SIZE];
+
+ if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
+ return AVERROR(EIO);
+ wsaud->audio_samplerate = AV_RL16(&header[0]);
+ if (header[11] == 99)
+ wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
+ else
+ return AVERROR_INVALIDDATA;
+
+ /* flag 0 indicates stereo */
+ wsaud->audio_channels = (header[10] & 0x1) + 1;
+ /* flag 1 indicates 16 bit audio */
+ wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
+
+ /* initialize the audio decoder stream */
+ st = avformat_new_stream(s, NULL);
+ if (!st)
+ return AVERROR(ENOMEM);
+ avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
+ st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codec->codec_id = wsaud->audio_type;
+ st->codec->codec_tag = 0; /* no tag */
+ st->codec->channels = wsaud->audio_channels;
+ st->codec->sample_rate = wsaud->audio_samplerate;
+ st->codec->bits_per_coded_sample = wsaud->audio_bits;
+ st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
+ st->codec->bits_per_coded_sample / 4;
+ st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
+
+ wsaud->audio_stream_index = st->index;
+ wsaud->audio_frame_counter = 0;
+
+ return 0;
+}
+
+static int wsaud_read_packet(AVFormatContext *s,
+ AVPacket *pkt)
+{
+ WsAudDemuxContext *wsaud = s->priv_data;
+ AVIOContext *pb = s->pb;
+ unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
+ unsigned int chunk_size;
+ int ret = 0;
+
+ if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
+ AUD_CHUNK_PREAMBLE_SIZE)
+ return AVERROR(EIO);
+
+ /* validate the chunk */
+ if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
+ return AVERROR_INVALIDDATA;
+
+ chunk_size = AV_RL16(&preamble[0]);
+ ret= av_get_packet(pb, pkt, chunk_size);
+ if (ret != chunk_size)
+ return AVERROR(EIO);
+ pkt->stream_index = wsaud->audio_stream_index;
+ pkt->pts = wsaud->audio_frame_counter;
+ pkt->pts /= wsaud->audio_samplerate;
+
+ /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
+ wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
+
+ return ret;
+}
+
+AVInputFormat ff_wsaud_demuxer = {
+ .name = "wsaud",
+ .long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
+ .priv_data_size = sizeof(WsAudDemuxContext),
+ .read_probe = wsaud_probe,
+ .read_header = wsaud_read_header,
+ .read_packet = wsaud_read_packet,
+};
diff --git a/libavformat/westwood.c b/libavformat/westwood_vqa.c
similarity index 63%
rename from libavformat/westwood.c
rename to libavformat/westwood_vqa.c
index 82b7e94..2eff4cc 100644
--- a/libavformat/westwood.c
+++ b/libavformat/westwood_vqa.c
@@ -1,5 +1,5 @@
/*
- * Westwood Studios Multimedia Formats Demuxer (VQA, AUD)
+ * Westwood Studios VQA Format Demuxer
* Copyright (c) 2003 The ffmpeg Project
*
* This file is part of Libav.
@@ -21,26 +21,17 @@
/**
* @file
- * Westwood Studios VQA & AUD file demuxers
+ * Westwood Studios VQA file demuxer
* by Mike Melanson (melanson at pcisys.net)
* for more information on the Westwood file formats, visit:
* http://www.pcisys.net/~melanson/codecs/
* http://www.geocities.com/SiliconValley/8682/aud3.txt
- *
- * Implementation note: There is no definite file signature for AUD files.
- * The demuxer uses a probabilistic strategy for content detection. This
- * entails performing sanity checks on certain header values in order to
- * qualify a file. Refer to wsaud_probe() for the precise parameters.
*/
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
-#define AUD_HEADER_SIZE 12
-#define AUD_CHUNK_PREAMBLE_SIZE 8
-#define AUD_CHUNK_SIGNATURE 0x0000DEAF
-
#define FORM_TAG MKBETAG('F', 'O', 'R', 'M')
#define WVQA_TAG MKBETAG('W', 'V', 'Q', 'A')
#define VQHD_TAG MKBETAG('V', 'Q', 'H', 'D')
@@ -63,15 +54,6 @@
#define VQA_FRAMERATE 15
#define VQA_PREAMBLE_SIZE 8
-typedef struct WsAudDemuxContext {
- int audio_samplerate;
- int audio_channels;
- int audio_bits;
- enum CodecID audio_type;
- int audio_stream_index;
- int64_t audio_frame_counter;
-} WsAudDemuxContext;
-
typedef struct WsVqaDemuxContext {
int audio_samplerate;
int audio_channels;
@@ -83,119 +65,6 @@ typedef struct WsVqaDemuxContext {
int64_t audio_frame_counter;
} WsVqaDemuxContext;
-static int wsaud_probe(AVProbeData *p)
-{
- int field;
-
- /* Probabilistic content detection strategy: There is no file signature
- * so perform sanity checks on various header parameters:
- * 8000 <= sample rate (16 bits) <= 48000 ==> 40001 acceptable numbers
- * flags <= 0x03 (2 LSBs are used) ==> 4 acceptable numbers
- * compression type (8 bits) = 1 or 99 ==> 2 acceptable numbers
- * first audio chunk signature (32 bits) ==> 1 acceptable number
- * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
- * 320008 acceptable number combinations.
- */
-
- if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
- return 0;
-
- /* check sample rate */
- field = AV_RL16(&p->buf[0]);
- if ((field < 8000) || (field > 48000))
- return 0;
-
- /* enforce the rule that the top 6 bits of this flags field are reserved (0);
- * this might not be true, but enforce it until deemed unnecessary */
- if (p->buf[10] & 0xFC)
- return 0;
-
- /* note: only check for WS IMA (type 99) right now since there is no
- * support for type 1 */
- if (p->buf[11] != 99)
- return 0;
-
- /* read ahead to the first audio chunk and validate the first header signature */
- if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
- return 0;
-
- /* return 1/2 certainty since this file check is a little sketchy */
- return AVPROBE_SCORE_MAX / 2;
-}
-
-static int wsaud_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
-{
- WsAudDemuxContext *wsaud = s->priv_data;
- AVIOContext *pb = s->pb;
- AVStream *st;
- unsigned char header[AUD_HEADER_SIZE];
-
- if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
- return AVERROR(EIO);
- wsaud->audio_samplerate = AV_RL16(&header[0]);
- if (header[11] == 99)
- wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
- else
- return AVERROR_INVALIDDATA;
-
- /* flag 0 indicates stereo */
- wsaud->audio_channels = (header[10] & 0x1) + 1;
- /* flag 1 indicates 16 bit audio */
- wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
-
- /* initialize the audio decoder stream */
- st = avformat_new_stream(s, NULL);
- if (!st)
- return AVERROR(ENOMEM);
- avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
- st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
- st->codec->codec_id = wsaud->audio_type;
- st->codec->codec_tag = 0; /* no tag */
- st->codec->channels = wsaud->audio_channels;
- st->codec->sample_rate = wsaud->audio_samplerate;
- st->codec->bits_per_coded_sample = wsaud->audio_bits;
- st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
- st->codec->bits_per_coded_sample / 4;
- st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
-
- wsaud->audio_stream_index = st->index;
- wsaud->audio_frame_counter = 0;
-
- return 0;
-}
-
-static int wsaud_read_packet(AVFormatContext *s,
- AVPacket *pkt)
-{
- WsAudDemuxContext *wsaud = s->priv_data;
- AVIOContext *pb = s->pb;
- unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
- unsigned int chunk_size;
- int ret = 0;
-
- if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
- AUD_CHUNK_PREAMBLE_SIZE)
- return AVERROR(EIO);
-
- /* validate the chunk */
- if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
- return AVERROR_INVALIDDATA;
-
- chunk_size = AV_RL16(&preamble[0]);
- ret= av_get_packet(pb, pkt, chunk_size);
- if (ret != chunk_size)
- return AVERROR(EIO);
- pkt->stream_index = wsaud->audio_stream_index;
- pkt->pts = wsaud->audio_frame_counter;
- pkt->pts /= wsaud->audio_samplerate;
-
- /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
- wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
-
- return ret;
-}
-
static int wsvqa_probe(AVProbeData *p)
{
/* need 12 bytes to qualify */
@@ -372,17 +241,6 @@ static int wsvqa_read_packet(AVFormatContext *s,
return ret;
}
-#if CONFIG_WSAUD_DEMUXER
-AVInputFormat ff_wsaud_demuxer = {
- .name = "wsaud",
- .long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
- .priv_data_size = sizeof(WsAudDemuxContext),
- .read_probe = wsaud_probe,
- .read_header = wsaud_read_header,
- .read_packet = wsaud_read_packet,
-};
-#endif
-#if CONFIG_WSVQA_DEMUXER
AVInputFormat ff_wsvqa_demuxer = {
.name = "wsvqa",
.long_name = NULL_IF_CONFIG_SMALL("Westwood Studios VQA format"),
@@ -391,4 +249,3 @@ AVInputFormat ff_wsvqa_demuxer = {
.read_header = wsvqa_read_header,
.read_packet = wsvqa_read_packet,
};
-#endif
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