[FFmpeg-cvslog] vmd: Split audio and video decoder
Nidhi Makhijani
git at videolan.org
Mon Jul 7 18:47:18 CEST 2014
ffmpeg | branch: master | Nidhi Makhijani <nidhimj22 at gmail.com> | Mon Jul 7 09:59:04 2014 +0530| [246f869590b8c7313d26e1c2ef56db01f6fd2503] | committer: Diego Biurrun
vmd: Split audio and video decoder
Signed-off-by: Diego Biurrun <diego at biurrun.de>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=246f869590b8c7313d26e1c2ef56db01f6fd2503
---
libavcodec/Makefile | 4 +-
libavcodec/vmdaudio.c | 233 ++++++++++++++++++++++++++++++++++++
libavcodec/{vmdav.c => vmdvideo.c} | 220 +---------------------------------
3 files changed, 238 insertions(+), 219 deletions(-)
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index d3d531e..fcd35cd 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -390,8 +390,8 @@ OBJS-$(CONFIG_VBLE_DECODER) += vble.o
OBJS-$(CONFIG_VC1_DECODER) += vc1dec.o vc1.o vc1data.o vc1dsp.o \
msmpeg4dec.o msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_VCR1_DECODER) += vcr1.o
-OBJS-$(CONFIG_VMDAUDIO_DECODER) += vmdav.o
-OBJS-$(CONFIG_VMDVIDEO_DECODER) += vmdav.o
+OBJS-$(CONFIG_VMDAUDIO_DECODER) += vmdaudio.o
+OBJS-$(CONFIG_VMDVIDEO_DECODER) += vmdvideo.o
OBJS-$(CONFIG_VMNC_DECODER) += vmnc.o
OBJS-$(CONFIG_VORBIS_DECODER) += vorbisdec.o vorbisdsp.o vorbis.o \
vorbis_data.o xiph.o
diff --git a/libavcodec/vmdaudio.c b/libavcodec/vmdaudio.c
new file mode 100644
index 0000000..66c5865
--- /dev/null
+++ b/libavcodec/vmdaudio.c
@@ -0,0 +1,233 @@
+/*
+ * Sierra VMD audio decoder
+ *
+ * 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
+ * Sierra VMD audio decoder
+ * by Vladimir "VAG" Gneushev (vagsoft at mail.ru)
+ * for more information on the Sierra VMD format, visit:
+ * http://www.pcisys.net/~melanson/codecs/
+ *
+ * The audio decoder, expects each encoded data
+ * chunk to be prepended with the appropriate 16-byte frame information
+ * record from the VMD file. It does not require the 0x330-byte VMD file
+ * header, but it does need the audio setup parameters passed in through
+ * normal libavcodec API means.
+ */
+
+#include <string.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/intreadwrite.h"
+
+#include "avcodec.h"
+#include "internal.h"
+
+#define BLOCK_TYPE_AUDIO 1
+#define BLOCK_TYPE_INITIAL 2
+#define BLOCK_TYPE_SILENCE 3
+
+typedef struct VmdAudioContext {
+ int out_bps;
+ int chunk_size;
+} VmdAudioContext;
+
+static const uint16_t vmdaudio_table[128] = {
+ 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
+ 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
+ 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
+ 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
+ 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
+ 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
+ 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
+ 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
+ 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
+ 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
+ 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
+ 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
+ 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
+};
+
+static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
+{
+ VmdAudioContext *s = avctx->priv_data;
+
+ if (avctx->channels < 1 || avctx->channels > 2) {
+ av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
+ return AVERROR(EINVAL);
+ }
+ if (avctx->block_align < 1) {
+ av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
+ return AVERROR(EINVAL);
+ }
+
+ avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
+ AV_CH_LAYOUT_STEREO;
+
+ if (avctx->bits_per_coded_sample == 16)
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ else
+ avctx->sample_fmt = AV_SAMPLE_FMT_U8;
+ s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
+
+ s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
+
+ av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
+ "block align = %d, sample rate = %d\n",
+ avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
+ avctx->sample_rate);
+
+ return 0;
+}
+
+static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
+ int channels)
+{
+ int ch;
+ const uint8_t *buf_end = buf + buf_size;
+ int predictor[2];
+ int st = channels - 1;
+
+ /* decode initial raw sample */
+ for (ch = 0; ch < channels; ch++) {
+ predictor[ch] = (int16_t)AV_RL16(buf);
+ buf += 2;
+ *out++ = predictor[ch];
+ }
+
+ /* decode DPCM samples */
+ ch = 0;
+ while (buf < buf_end) {
+ uint8_t b = *buf++;
+ if (b & 0x80)
+ predictor[ch] -= vmdaudio_table[b & 0x7F];
+ else
+ predictor[ch] += vmdaudio_table[b];
+ predictor[ch] = av_clip_int16(predictor[ch]);
+ *out++ = predictor[ch];
+ ch ^= st;
+ }
+}
+
+static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ AVFrame *frame = data;
+ const uint8_t *buf = avpkt->data;
+ const uint8_t *buf_end;
+ int buf_size = avpkt->size;
+ VmdAudioContext *s = avctx->priv_data;
+ int block_type, silent_chunks, audio_chunks;
+ int ret;
+ uint8_t *output_samples_u8;
+ int16_t *output_samples_s16;
+
+ if (buf_size < 16) {
+ av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
+ *got_frame_ptr = 0;
+ return buf_size;
+ }
+
+ block_type = buf[6];
+ if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {
+ av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);
+ return AVERROR(EINVAL);
+ }
+ buf += 16;
+ buf_size -= 16;
+
+ /* get number of silent chunks */
+ silent_chunks = 0;
+ if (block_type == BLOCK_TYPE_INITIAL) {
+ uint32_t flags;
+ if (buf_size < 4) {
+ av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
+ return AVERROR(EINVAL);
+ }
+ flags = AV_RB32(buf);
+ silent_chunks = av_popcount(flags);
+ buf += 4;
+ buf_size -= 4;
+ } else if (block_type == BLOCK_TYPE_SILENCE) {
+ silent_chunks = 1;
+ buf_size = 0; // should already be zero but set it just to be sure
+ }
+
+ /* ensure output buffer is large enough */
+ audio_chunks = buf_size / s->chunk_size;
+
+ /* drop incomplete chunks */
+ buf_size = audio_chunks * s->chunk_size;
+
+ /* get output buffer */
+ frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
+ avctx->channels;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ output_samples_u8 = frame->data[0];
+ output_samples_s16 = (int16_t *)frame->data[0];
+
+ /* decode silent chunks */
+ if (silent_chunks > 0) {
+ int silent_size = FFMIN(avctx->block_align * silent_chunks,
+ frame->nb_samples * avctx->channels);
+ if (s->out_bps == 2) {
+ memset(output_samples_s16, 0x00, silent_size * 2);
+ output_samples_s16 += silent_size;
+ } else {
+ memset(output_samples_u8, 0x80, silent_size);
+ output_samples_u8 += silent_size;
+ }
+ }
+
+ /* decode audio chunks */
+ if (audio_chunks > 0) {
+ buf_end = buf + (buf_size & ~(avctx->channels > 1));
+ while (buf + s->chunk_size <= buf_end) {
+ if (s->out_bps == 2) {
+ decode_audio_s16(output_samples_s16, buf, s->chunk_size,
+ avctx->channels);
+ output_samples_s16 += avctx->block_align;
+ } else {
+ memcpy(output_samples_u8, buf, s->chunk_size);
+ output_samples_u8 += avctx->block_align;
+ }
+ buf += s->chunk_size;
+ }
+ }
+
+ *got_frame_ptr = 1;
+
+ return avpkt->size;
+}
+
+AVCodec ff_vmdaudio_decoder = {
+ .name = "vmdaudio",
+ .long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_VMDAUDIO,
+ .priv_data_size = sizeof(VmdAudioContext),
+ .init = vmdaudio_decode_init,
+ .decode = vmdaudio_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
+};
diff --git a/libavcodec/vmdav.c b/libavcodec/vmdvideo.c
similarity index 67%
rename from libavcodec/vmdav.c
rename to libavcodec/vmdvideo.c
index a1e39c0..aaeff43 100644
--- a/libavcodec/vmdav.c
+++ b/libavcodec/vmdvideo.c
@@ -1,6 +1,5 @@
/*
- * Sierra VMD Audio & Video Decoders
- * Copyright (C) 2004 the ffmpeg project
+ * Sierra VMD video decoder
*
* This file is part of Libav.
*
@@ -21,7 +20,7 @@
/**
* @file
- * Sierra VMD audio & video decoders
+ * Sierra VMD video decoder
* by Vladimir "VAG" Gneushev (vagsoft at mail.ru)
* for more information on the Sierra VMD format, visit:
* http://www.pcisys.net/~melanson/codecs/
@@ -31,21 +30,13 @@
* codec initialization. Each encoded frame that is sent to this decoder
* is expected to be prepended with the appropriate 16-byte frame
* information record from the VMD file.
- *
- * The audio decoder, like the video decoder, expects each encoded data
- * chunk to be prepended with the appropriate 16-byte frame information
- * record from the VMD file. It does not require the 0x330-byte VMD file
- * header, but it does need the audio setup parameters passed in through
- * normal libavcodec API means.
*/
-#include <stdio.h>
-#include <stdlib.h>
#include <string.h>
-#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"
+
#include "avcodec.h"
#include "internal.h"
#include "bytestream.h"
@@ -53,10 +44,6 @@
#define VMD_HEADER_SIZE 0x330
#define PALETTE_COUNT 256
-/*
- * Video Decoder
- */
-
typedef struct VmdVideoContext {
AVCodecContext *avctx;
@@ -467,196 +454,6 @@ static int vmdvideo_decode_frame(AVCodecContext *avctx,
return buf_size;
}
-
-/*
- * Audio Decoder
- */
-
-#define BLOCK_TYPE_AUDIO 1
-#define BLOCK_TYPE_INITIAL 2
-#define BLOCK_TYPE_SILENCE 3
-
-typedef struct VmdAudioContext {
- int out_bps;
- int chunk_size;
-} VmdAudioContext;
-
-static const uint16_t vmdaudio_table[128] = {
- 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
- 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
- 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
- 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
- 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
- 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
- 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
- 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
- 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
- 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
- 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
- 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
- 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
-};
-
-static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
-{
- VmdAudioContext *s = avctx->priv_data;
-
- if (avctx->channels < 1 || avctx->channels > 2) {
- av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
- return AVERROR(EINVAL);
- }
- if (avctx->block_align < 1) {
- av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
- return AVERROR(EINVAL);
- }
-
- avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
- AV_CH_LAYOUT_STEREO;
-
- if (avctx->bits_per_coded_sample == 16)
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- else
- avctx->sample_fmt = AV_SAMPLE_FMT_U8;
- s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
-
- s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
-
- av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
- "block align = %d, sample rate = %d\n",
- avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
- avctx->sample_rate);
-
- return 0;
-}
-
-static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
- int channels)
-{
- int ch;
- const uint8_t *buf_end = buf + buf_size;
- int predictor[2];
- int st = channels - 1;
-
- /* decode initial raw sample */
- for (ch = 0; ch < channels; ch++) {
- predictor[ch] = (int16_t)AV_RL16(buf);
- buf += 2;
- *out++ = predictor[ch];
- }
-
- /* decode DPCM samples */
- ch = 0;
- while (buf < buf_end) {
- uint8_t b = *buf++;
- if (b & 0x80)
- predictor[ch] -= vmdaudio_table[b & 0x7F];
- else
- predictor[ch] += vmdaudio_table[b];
- predictor[ch] = av_clip_int16(predictor[ch]);
- *out++ = predictor[ch];
- ch ^= st;
- }
-}
-
-static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
- int *got_frame_ptr, AVPacket *avpkt)
-{
- AVFrame *frame = data;
- const uint8_t *buf = avpkt->data;
- const uint8_t *buf_end;
- int buf_size = avpkt->size;
- VmdAudioContext *s = avctx->priv_data;
- int block_type, silent_chunks, audio_chunks;
- int ret;
- uint8_t *output_samples_u8;
- int16_t *output_samples_s16;
-
- if (buf_size < 16) {
- av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
- *got_frame_ptr = 0;
- return buf_size;
- }
-
- block_type = buf[6];
- if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {
- av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);
- return AVERROR(EINVAL);
- }
- buf += 16;
- buf_size -= 16;
-
- /* get number of silent chunks */
- silent_chunks = 0;
- if (block_type == BLOCK_TYPE_INITIAL) {
- uint32_t flags;
- if (buf_size < 4) {
- av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
- return AVERROR(EINVAL);
- }
- flags = AV_RB32(buf);
- silent_chunks = av_popcount(flags);
- buf += 4;
- buf_size -= 4;
- } else if (block_type == BLOCK_TYPE_SILENCE) {
- silent_chunks = 1;
- buf_size = 0; // should already be zero but set it just to be sure
- }
-
- /* ensure output buffer is large enough */
- audio_chunks = buf_size / s->chunk_size;
-
- /* drop incomplete chunks */
- buf_size = audio_chunks * s->chunk_size;
-
- /* get output buffer */
- frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
- avctx->channels;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
- return ret;
- }
- output_samples_u8 = frame->data[0];
- output_samples_s16 = (int16_t *)frame->data[0];
-
- /* decode silent chunks */
- if (silent_chunks > 0) {
- int silent_size = FFMIN(avctx->block_align * silent_chunks,
- frame->nb_samples * avctx->channels);
- if (s->out_bps == 2) {
- memset(output_samples_s16, 0x00, silent_size * 2);
- output_samples_s16 += silent_size;
- } else {
- memset(output_samples_u8, 0x80, silent_size);
- output_samples_u8 += silent_size;
- }
- }
-
- /* decode audio chunks */
- if (audio_chunks > 0) {
- buf_end = buf + (buf_size & ~(avctx->channels > 1));
- while (buf + s->chunk_size <= buf_end) {
- if (s->out_bps == 2) {
- decode_audio_s16(output_samples_s16, buf, s->chunk_size,
- avctx->channels);
- output_samples_s16 += avctx->block_align;
- } else {
- memcpy(output_samples_u8, buf, s->chunk_size);
- output_samples_u8 += avctx->block_align;
- }
- buf += s->chunk_size;
- }
- }
-
- *got_frame_ptr = 1;
-
- return avpkt->size;
-}
-
-
-/*
- * Public Data Structures
- */
-
AVCodec ff_vmdvideo_decoder = {
.name = "vmdvideo",
.long_name = NULL_IF_CONFIG_SMALL("Sierra VMD video"),
@@ -668,14 +465,3 @@ AVCodec ff_vmdvideo_decoder = {
.decode = vmdvideo_decode_frame,
.capabilities = CODEC_CAP_DR1,
};
-
-AVCodec ff_vmdaudio_decoder = {
- .name = "vmdaudio",
- .long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_VMDAUDIO,
- .priv_data_size = sizeof(VmdAudioContext),
- .init = vmdaudio_decode_init,
- .decode = vmdaudio_decode_frame,
- .capabilities = CODEC_CAP_DR1,
-};
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