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Code Issues Releases Activity

Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master: (23 commits)
  aacenc: Fix identification padding when the bitstream is already aligned.
  aacenc: Write correct length for long identification strings.
  aud: remove unneeded field, audio_stream_index from context
  aud: fix time stamp calculation for ADPCM IMA WS
  aud: simplify header parsing
  aud: set pts_wrap_bits to 64.
  cosmetics: indentation
  aud: support Westwood SND1 audio in AUD files.
  adpcm_ima_ws: fix stereo decoding
  avcodec: add a new codec_id for CRYO APC IMA ADPCM.
  vqa: remove unused context fields, audio_samplerate and audio_bits
  vqa: clean up audio header parsing
  vqa: set time base to frame rate as coded in the header.
  vqa: set packet duration.
  vqa: use 1/sample_rate as the audio stream time base
  vqa: set stream start_time to 0.
  lavc: postpone the removal of AVCodecContext.request_channels.
  lavf: postpone removing av_close_input_file().
  lavc: postpone removing old audio encoding and decoding API
  avplay: remove the -er option.
  ...

Conflicts:
	Changelog
	libavcodec/version.h
	libavdevice/v4l.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-01-24 22:53:59 +01:00
commit 1d9569f9e8
18 changed files with 151 additions and 91 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
Changelog
View File

@ -1,7 +1,6 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version next:
- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
- SBaGen (SBG) binaural beats script demuxer

3
ffplay.c
View File

@ -265,7 +265,6 @@ static int idct = FF_IDCT_AUTO;
static enum AVDiscard skip_frame = AVDISCARD_DEFAULT;
static enum AVDiscard skip_idct = AVDISCARD_DEFAULT;
static enum AVDiscard skip_loop_filter = AVDISCARD_DEFAULT;
static int error_recognition = FF_ER_CAREFUL;
static int error_concealment = 3;
static int decoder_reorder_pts = -1;
static int autoexit;
@ -2236,7 +2235,6 @@ static int stream_component_open(VideoState *is, int stream_index)
avctx->skip_frame = skip_frame;
avctx->skip_idct = skip_idct;
avctx->skip_loop_filter = skip_loop_filter;
avctx->error_recognition = error_recognition;
avctx->error_concealment = error_concealment;
if(avctx->lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
@ -3071,7 +3069,6 @@ static const OptionDef options[] = {
{ "skipframe", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&skip_frame }, "", "" },
{ "skipidct", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&skip_idct }, "", "" },
{ "idct", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&idct }, "set idct algo", "algo" },
{ "er", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&error_recognition }, "set error detection threshold (0-4)", "threshold" },
{ "ec", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&error_concealment }, "set error concealment options", "bit_mask" },
{ "sync", HAS_ARG | OPT_EXPERT, { (void*)opt_sync }, "set audio-video sync. type (type=audio/video/ext)", "type" },
{ "autoexit", OPT_BOOL | OPT_EXPERT, { (void*)&autoexit }, "exit at the end", "" },

1
libavcodec/Makefile
View File

@ -548,6 +548,7 @@ OBJS-$(CONFIG_ADPCM_G722_ENCODER) += g722.o g722enc.o
OBJS-$(CONFIG_ADPCM_G726_DECODER) += g726.o
OBJS-$(CONFIG_ADPCM_G726_ENCODER) += g726.o
OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_APC_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER) += adpcm.o adpcm_data.o

2
libavcodec/aacenc.c
View File

@ -464,7 +464,7 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
if (namelen >= 15)
put_bits(&s->pb, 8, namelen - 14);
put_bits(&s->pb, 4, 0); //extension type - filler
padbits = 8 - (put_bits_count(&s->pb) & 7);
padbits = -put_bits_count(&s->pb) & 7;
avpriv_align_put_bits(&s->pb);
for (i = 0; i < namelen - 2; i++)
put_bits(&s->pb, 8, name[i]);

37
libavcodec/adpcm.c
View File

@ -86,6 +86,7 @@ static const int swf_index_tables[4][16] = {
typedef struct ADPCMDecodeContext {
AVFrame frame;
ADPCMChannelStatus status[6];
int vqa_version; /**< VQA version. Used for ADPCM_IMA_WS */
} ADPCMDecodeContext;
static av_cold int adpcm_decode_init(AVCodecContext * avctx)
@ -120,12 +121,16 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
return -1;
}
break;
case CODEC_ID_ADPCM_IMA_WS:
if (avctx->extradata && avctx->extradata_size == 2 * 4) {
case CODEC_ID_ADPCM_IMA_APC:
if (avctx->extradata && avctx->extradata_size >= 8) {
c->status[0].predictor = AV_RL32(avctx->extradata);
c->status[1].predictor = AV_RL32(avctx->extradata + 4);
}
break;
case CODEC_ID_ADPCM_IMA_WS:
if (avctx->extradata && avctx->extradata_size >= 42)
c->vqa_version = AV_RL16(avctx->extradata);
break;
default:
break;
}
@ -362,6 +367,7 @@ static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf,
break;
/* simple 4-bit adpcm */
case CODEC_ID_ADPCM_CT:
case CODEC_ID_ADPCM_IMA_APC:
case CODEC_ID_ADPCM_IMA_EA_SEAD:
case CODEC_ID_ADPCM_IMA_WS:
case CODEC_ID_ADPCM_YAMAHA:
@ -776,13 +782,37 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v2, 3);
}
break;
case CODEC_ID_ADPCM_IMA_WS:
case CODEC_ID_ADPCM_IMA_APC:
while (src < buf + buf_size) {
uint8_t v = *src++;
*samples++ = adpcm_ima_expand_nibble(&c->status[0], v >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
}
break;
case CODEC_ID_ADPCM_IMA_WS:
for (channel = 0; channel < avctx->channels; channel++) {
const uint8_t *src0;
int src_stride;
int16_t *smp = samples + channel;
if (c->vqa_version == 3) {
src0 = src + channel * buf_size / 2;
src_stride = 1;
} else {
src0 = src + channel;
src_stride = avctx->channels;
}
for (n = nb_samples / 2; n > 0; n--) {
uint8_t v = *src0;
src0 += src_stride;
*smp = adpcm_ima_expand_nibble(&c->status[channel], v >> 4 , 3);
smp += avctx->channels;
*smp = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
smp += avctx->channels;
}
}
src = buf + buf_size;
break;
case CODEC_ID_ADPCM_XA:
while (buf_size >= 128) {
xa_decode(samples, src, &c->status[0], &c->status[1],
@ -1226,6 +1256,7 @@ ADPCM_DECODER(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2, "ADPCM Electronic Arts R2");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3, "ADPCM Electronic Arts R3");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_XAS, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv, "ADPCM IMA AMV");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_APC, adpcm_ima_apc, "ADPCM IMA CRYO APC");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4, "ADPCM IMA Duck DK4");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");

1
libavcodec/allcodecs.c
View File

@ -365,6 +365,7 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (ADPCM_G722, adpcm_g722);
REGISTER_ENCDEC (ADPCM_G726, adpcm_g726);
REGISTER_DECODER (ADPCM_IMA_AMV, adpcm_ima_amv);
REGISTER_DECODER (ADPCM_IMA_APC, adpcm_ima_apc);
REGISTER_DECODER (ADPCM_IMA_DK3, adpcm_ima_dk3);
REGISTER_DECODER (ADPCM_IMA_DK4, adpcm_ima_dk4);
REGISTER_DECODER (ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs);

1
libavcodec/avcodec.h
View File

@ -326,6 +326,7 @@ enum CodecID {
CODEC_ID_ADPCM_EA_MAXIS_XA,
CODEC_ID_ADPCM_IMA_ISS,
CODEC_ID_ADPCM_G722,
CODEC_ID_ADPCM_IMA_APC,
/* AMR */
CODEC_ID_AMR_NB = 0x12000,

9
libavcodec/rv10.c
View File

@ -666,9 +666,12 @@ static int rv10_decode_frame(AVCodecContext *avctx,
slice_count = avctx->slice_count;
for(i=0; i<slice_count; i++){
int offset= get_slice_offset(avctx, slices_hdr, i);
unsigned offset = get_slice_offset(avctx, slices_hdr, i);
int size, size2;
if (offset >= buf_size)
return AVERROR_INVALIDDATA;
if(i+1 == slice_count)
size= buf_size - offset;
else
@ -679,6 +682,10 @@ static int rv10_decode_frame(AVCodecContext *avctx,
else
size2= get_slice_offset(avctx, slices_hdr, i+2) - offset;
if (size <= 0 || size2 <= 0 ||
offset + FFMAX(size, size2) > buf_size)
return AVERROR_INVALIDDATA;
if(rv10_decode_packet(avctx, buf+offset, size, size2) > 8*size)
i++;
}

1
libavcodec/utils.c
View File

@ -1780,6 +1780,7 @@ int av_get_bits_per_sample(enum CodecID codec_id){
return 3;
case CODEC_ID_ADPCM_SBPRO_4:
case CODEC_ID_ADPCM_CT:
case CODEC_ID_ADPCM_IMA_APC:
case CODEC_ID_ADPCM_IMA_WAV:
case CODEC_ID_ADPCM_IMA_QT:
case CODEC_ID_ADPCM_SWF:

8
libavcodec/version.h
View File

@ -21,7 +21,7 @@
#define AVCODEC_VERSION_H
#define LIBAVCODEC_VERSION_MAJOR 53
#define LIBAVCODEC_VERSION_MINOR 58
#define LIBAVCODEC_VERSION_MINOR 59
#define LIBAVCODEC_VERSION_MICRO 100
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
@ -51,7 +51,7 @@
#define FF_API_ANTIALIAS_ALGO (LIBAVCODEC_VERSION_MAJOR < 54)
#endif
#ifndef FF_API_REQUEST_CHANNELS
#define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 54)
#define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 55)
#endif
#ifndef FF_API_OPT_H
#define FF_API_OPT_H (LIBAVCODEC_VERSION_MAJOR < 54)
@ -114,7 +114,7 @@
#define FF_API_DATA_POINTERS (LIBAVCODEC_VERSION_MAJOR < 54)
#endif
#ifndef FF_API_OLD_DECODE_AUDIO
#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 54)
#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
#endif
#ifndef FF_API_OLD_TIMECODE
#define FF_API_OLD_TIMECODE (LIBAVCODEC_VERSION_MAJOR < 54)
@ -124,7 +124,7 @@
#define FF_API_AVFRAME_AGE (LIBAVCODEC_VERSION_MAJOR < 54)
#endif
#ifndef FF_API_OLD_ENCODE_AUDIO
#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 54)
#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
#endif
#endif /* AVCODEC_VERSION_H */

1
libavdevice/Makefile
View File

@ -29,7 +29,6 @@ OBJS-$(CONFIG_SDL_OUTDEV) += sdl.o
OBJS-$(CONFIG_SNDIO_INDEV) += sndio_common.o sndio_dec.o
OBJS-$(CONFIG_SNDIO_OUTDEV) += sndio_common.o sndio_enc.o
OBJS-$(CONFIG_V4L2_INDEV) += v4l2.o
OBJS-$(CONFIG_V4L_INDEV) += v4l.o
OBJS-$(CONFIG_VFWCAP_INDEV) += vfwcap.o
OBJS-$(CONFIG_X11_GRAB_DEVICE_INDEV) += x11grab.o

2
libavdevice/alldevices.c
View File

@ -51,9 +51,7 @@ void avdevice_register_all(void)
REGISTER_OUTDEV (SDL, sdl);
REGISTER_INOUTDEV (SNDIO, sndio);
REGISTER_INDEV (V4L2, v4l2);
#if FF_API_V4L
REGISTER_INDEV (V4L, v4l);
#endif
REGISTER_INDEV (VFWCAP, vfwcap);
REGISTER_INDEV (X11_GRAB_DEVICE, x11_grab_device);

4
libavdevice/avdevice.h
View File

@ -56,10 +56,6 @@
LIBAVDEVICE_VERSION_MICRO)
#define LIBAVDEVICE_BUILD LIBAVDEVICE_VERSION_INT
#ifndef FF_API_V4L
#define FF_API_V4L (LIBAVDEVICE_VERSION_MAJOR < 54)
#endif
/**
* Return the LIBAVDEVICE_VERSION_INT constant.
*/

2
libavformat/apc.c
View File

@ -44,7 +44,7 @@ static int apc_read_header(AVFormatContext *s, AVFormatParameters *ap)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
st->codec->codec_id = CODEC_ID_ADPCM_IMA_APC;
avio_rl32(pb); /* number of samples */
st->codec->sample_rate = avio_rl32(pb);

2
libavformat/version.h
View File

@ -123,7 +123,7 @@
#define FF_API_SET_PTS_INFO (LIBAVFORMAT_VERSION_MAJOR < 54)
#endif
#ifndef FF_API_CLOSE_INPUT_FILE
#define FF_API_CLOSE_INPUT_FILE (LIBAVFORMAT_VERSION_MAJOR < 54)
#define FF_API_CLOSE_INPUT_FILE (LIBAVFORMAT_VERSION_MAJOR < 55)
#endif
#endif /* AVFORMAT_VERSION_H */

92
libavformat/westwood_aud.c
View File

@ -41,15 +41,6 @@
#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;
@ -79,7 +70,7 @@ static int wsaud_probe(AVProbeData *p)
/* note: only check for WS IMA (type 99) right now since there is no
* support for type 1 */
if (p->buf[11] != 99)
if (p->buf[11] != 99 && p->buf[11] != 1)
return 0;
/* read ahead to the first audio chunk and validate the first header signature */
@ -93,41 +84,44 @@ static int wsaud_probe(AVProbeData *p)
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];
int sample_rate, channels, codec;
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;
sample_rate = AV_RL16(&header[0]);
channels = (header[10] & 0x1) + 1;
codec = header[11];
/* 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;
switch (codec) {
case 1:
if (channels != 1) {
av_log_ask_for_sample(s, "Stereo WS-SND1 is not supported.\n");
return AVERROR_PATCHWELCOME;
}
st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
break;
case 99:
st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
st->codec->bits_per_coded_sample = 4;
st->codec->bit_rate = channels * sample_rate * 4;
break;
default:
av_log_ask_for_sample(s, "Unknown codec: %d\n", codec);
return AVERROR_PATCHWELCOME;
}
avpriv_set_pts_info(st, 64, 1, sample_rate);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->channels = channels;
st->codec->sample_rate = sample_rate;
return 0;
}
@ -135,11 +129,11 @@ static int wsaud_read_header(AVFormatContext *s,
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;
AVStream *st = s->streams[0];
if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
AUD_CHUNK_PREAMBLE_SIZE)
@ -150,15 +144,30 @@ static int wsaud_read_packet(AVFormatContext *s,
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;
if (st->codec->codec_id == CODEC_ID_WESTWOOD_SND1) {
/* For Westwood SND1 audio we need to add the output size and input
size to the start of the packet to match what is in VQA.
Specifically, this is needed to signal when a packet should be
decoding as raw 8-bit pcm or variable-size ADPCM. */
int out_size = AV_RL16(&preamble[2]);
if ((ret = av_new_packet(pkt, chunk_size + 4)))
return ret;
if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
return ret < 0 ? ret : AVERROR(EIO);
AV_WL16(&pkt->data[0], out_size);
AV_WL16(&pkt->data[2], chunk_size);
pkt->duration = out_size;
} else {
ret = av_get_packet(pb, pkt, chunk_size);
if (ret != chunk_size)
return AVERROR(EIO);
/* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
pkt->duration = (chunk_size * 2) / st->codec->channels;
}
pkt->stream_index = st->index;
return ret;
}
@ -166,7 +175,6 @@ static int wsaud_read_packet(AVFormatContext *s,
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,

73
libavformat/westwood_vqa.c
View File

@ -51,18 +51,12 @@
#define CMDS_TAG MKBETAG('C', 'M', 'D', 'S')
#define VQA_HEADER_SIZE 0x2A
#define VQA_FRAMERATE 15
#define VQA_PREAMBLE_SIZE 8
typedef struct WsVqaDemuxContext {
int audio_samplerate;
int audio_channels;
int audio_bits;
int audio_stream_index;
int video_stream_index;
int64_t audio_frame_counter;
} WsVqaDemuxContext;
static int wsvqa_probe(AVProbeData *p)
@ -89,12 +83,13 @@ static int wsvqa_read_header(AVFormatContext *s,
unsigned char scratch[VQA_PREAMBLE_SIZE];
unsigned int chunk_tag;
unsigned int chunk_size;
int fps, version, flags, sample_rate, channels;
/* initialize the video decoder stream */
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avpriv_set_pts_info(st, 33, 1, VQA_FRAMERATE);
st->start_time = 0;
wsvqa->video_stream_index = st->index;
st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
st->codec->codec_id = CODEC_ID_WS_VQA;
@ -114,34 +109,59 @@ static int wsvqa_read_header(AVFormatContext *s,
}
st->codec->width = AV_RL16(&header[6]);
st->codec->height = AV_RL16(&header[8]);
fps = header[12];
if (fps < 1 || fps > 30) {
av_log(s, AV_LOG_ERROR, "invalid fps: %d\n", fps);
return AVERROR_INVALIDDATA;
}
avpriv_set_pts_info(st, 64, 1, fps);
/* initialize the audio decoder stream for VQA v1 or nonzero samplerate */
if (AV_RL16(&header[24]) || (AV_RL16(&header[0]) == 1 && AV_RL16(&header[2]) == 1)) {
version = AV_RL16(&header[ 0]);
flags = AV_RL16(&header[ 2]);
sample_rate = AV_RL16(&header[24]);
channels = header[26];
if (sample_rate || (version == 1 && flags == 1)) {
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avpriv_set_pts_info(st, 33, 1, VQA_FRAMERATE);
st->start_time = 0;
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
if (AV_RL16(&header[0]) == 1)
st->codec->extradata_size = VQA_HEADER_SIZE;
st->codec->extradata = av_mallocz(VQA_HEADER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
if (!st->codec->extradata)
return AVERROR(ENOMEM);
memcpy(st->codec->extradata, header, VQA_HEADER_SIZE);
if (!sample_rate)
sample_rate = 22050;
st->codec->sample_rate = sample_rate;
avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
if (!channels)
channels = 1;
st->codec->channels = channels;
switch (version) {
case 1:
st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
else
break;
case 2:
case 3:
st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
st->codec->codec_tag = 0; /* no tag */
st->codec->sample_rate = AV_RL16(&header[24]);
if (!st->codec->sample_rate)
st->codec->sample_rate = 22050;
st->codec->channels = header[26];
if (!st->codec->channels)
st->codec->channels = 1;
st->codec->bits_per_coded_sample = 16;
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;
st->codec->bits_per_coded_sample = 4;
st->codec->bit_rate = channels * sample_rate * 4;
break;
default:
/* NOTE: version 0 is supposedly raw pcm_u8 or pcm_s16le, but we do
not have any samples to validate this */
av_log_ask_for_sample(s, "VQA version %d audio\n", version);
return AVERROR_PATCHWELCOME;
}
wsvqa->audio_stream_index = st->index;
wsvqa->audio_samplerate = st->codec->sample_rate;
wsvqa->audio_channels = st->codec->channels;
wsvqa->audio_frame_counter = 0;
}
/* there are 0 or more chunks before the FINF chunk; iterate until
@ -208,13 +228,14 @@ static int wsvqa_read_packet(AVFormatContext *s,
if (chunk_type == SND2_TAG) {
pkt->stream_index = wsvqa->audio_stream_index;
/* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
wsvqa->audio_frame_counter += (chunk_size * 2) / wsvqa->audio_channels;
pkt->duration = (chunk_size * 2) / wsvqa->audio_channels;
} else if(chunk_type == SND1_TAG) {
pkt->stream_index = wsvqa->audio_stream_index;
/* unpacked size is stored in header */
wsvqa->audio_frame_counter += AV_RL16(pkt->data) / wsvqa->audio_channels;
pkt->duration = AV_RL16(pkt->data) / wsvqa->audio_channels;
} else {
pkt->stream_index = wsvqa->video_stream_index;
pkt->duration = 1;
}
/* stay on 16-bit alignment */
if (skip_byte)

2
libavformat/yop.c
View File

@ -72,7 +72,7 @@ static int yop_read_header(AVFormatContext *s, AVFormatParameters *ap)
// Audio
audio_dec = audio_stream->codec;
audio_dec->codec_type = AVMEDIA_TYPE_AUDIO;
audio_dec->codec_id = CODEC_ID_ADPCM_IMA_WS;
audio_dec->codec_id = CODEC_ID_ADPCM_IMA_APC;
audio_dec->channels = 1;
audio_dec->sample_rate = 22050;