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FFmpeg
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Merge commit '1cd432e167b1a80853760c89a33606e2b5f229c2' 

* commit '1cd432e167b1a80853760c89a33606e2b5f229c2':
  configure: fix libcdio check
  rtsp: Allow setting the reordering buffer size via an AVOption
  rtsp: Vertically align a constant definition
  rtp: Update the check for distinguishing between RTP and RTCP
  aac: fix build with hardcoded tables
  fate: dependencies for screen codec tests
  riff: Move functions around to be covered by appropriate #ifdefs

Conflicts:
	configure
	tests/fate/screen.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
tags/n1.1
Michael Niedermayer 13 years ago
parent
93f244e3ab 1cd432e167
commit
81ff0c24ef
9 changed files with 124 additions and 95 deletions
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  1. +4
    -3
      configure
  2. +0
    -1
      libavcodec/aac.h
  3. +0
    -1
      libavcodec/aac_tablegen.h
  4. +2
    -0
      libavcodec/aac_tablegen_decl.h
  5. +79
    -79
      libavformat/riff.c
  6. +13
    -2
      libavformat/rtp.h
  7. +14
    -2
      libavformat/rtsp.c
  8. +6
    -1
      libavformat/rtsp.h
  9. +6
    -6
      tests/fate/screen.mak

+ 4
- 3
configure View File

@@ -769,8 +769,8 @@ check_ld(){
log check_ld "$@"
type=$1
shift 1
flags=$(filter_out '-l*' $@)
libs=$(filter '-l*' $@)
flags=$(filter_out '-l*' "$@")
libs=$(filter '-l*' "$@")
check_$type $($cflags_filter $flags) || return
flags=$($ldflags_filter $flags)
libs=$($ldflags_filter $libs)
@@ -3649,7 +3649,7 @@ enabled libiec61883 && require libiec61883 libiec61883/iec61883.h iec61883_cmp_c
enabled libaacplus && require "libaacplus >= 2.0.0" aacplus.h aacplusEncOpen -laacplus
enabled libass && require_pkg_config libass ass/ass.h ass_library_init
enabled libbluray && require libbluray libbluray/bluray.h bd_open -lbluray
enabled libcdio && require2 libcdio "cdio/cdda.h cdio/paranoia.h" cdio_cddap_open "-lcdio_paranoia -lcdio_cdda -lcdio"
enabled libcdio && require2 libcdio "cdio/cdda.h cdio/paranoia.h" cdio_cddap_open -lcdio_paranoia -lcdio_cdda -lcdio
enabled libcelt && require libcelt celt/celt.h celt_decode -lcelt0 &&
{ check_lib celt/celt.h celt_decoder_create_custom -lcelt0 ||
die "ERROR: libcelt must be installed and version must be >= 0.11.0."; }
@@ -3782,6 +3782,7 @@ enabled jack_indev && check_lib2 jack/jack.h jack_client_open -ljack && check_fu

enabled_any sndio_indev sndio_outdev && check_lib2 sndio.h sio_open -lsndio


enabled x11grab &&
require X11 X11/Xlib.h XOpenDisplay -lX11 &&
require Xext X11/extensions/XShm.h XShmCreateImage -lXext &&


+ 0
- 1
libavcodec/aac.h View File

@@ -141,7 +141,6 @@ typedef struct PredictorState {
#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);

/**
* Long Term Prediction


+ 0
- 1
libavcodec/aac_tablegen.h View File

@@ -29,7 +29,6 @@
#include "libavcodec/aac_tables.h"
#else
#include "libavutil/mathematics.h"
#include "aac.h"
float ff_aac_pow2sf_tab[428];

void ff_aac_tableinit(void)


+ 2
- 0
libavcodec/aac_tablegen_decl.h View File

@@ -23,6 +23,8 @@
#ifndef AVCODEC_AAC_TABLEGEN_DECL_H
#define AVCODEC_AAC_TABLEGEN_DECL_H

#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);

#if CONFIG_HARDCODED_TABLES
#define ff_aac_tableinit()
extern const float ff_aac_pow2sf_tab[428];


+ 79
- 79
libavformat/riff.c View File

@@ -589,6 +589,84 @@ void ff_put_bmp_header(AVIOContext *pb, AVCodecContext *enc, const AVCodecTag *t
if (!for_asf && enc->extradata_size & 1)
avio_w8(pb, 0);
}

void ff_parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale)
{
int gcd;
int audio_frame_size;

/* We use the known constant frame size for the codec if known, otherwise
fallback to using AVCodecContext.frame_size, which is not as reliable
for indicating packet duration */
audio_frame_size = av_get_audio_frame_duration(stream, 0);
if (!audio_frame_size)
audio_frame_size = stream->frame_size;

*au_ssize= stream->block_align;
if (audio_frame_size && stream->sample_rate) {
*au_scale = audio_frame_size;
*au_rate= stream->sample_rate;
}else if(stream->codec_type == AVMEDIA_TYPE_VIDEO ||
stream->codec_type == AVMEDIA_TYPE_DATA ||
stream->codec_type == AVMEDIA_TYPE_SUBTITLE){
*au_scale= stream->time_base.num;
*au_rate = stream->time_base.den;
}else{
*au_scale= stream->block_align ? stream->block_align*8 : 8;
*au_rate = stream->bit_rate ? stream->bit_rate : 8*stream->sample_rate;
}
gcd= av_gcd(*au_scale, *au_rate);
*au_scale /= gcd;
*au_rate /= gcd;
}

void ff_riff_write_info_tag(AVIOContext *pb, const char *tag, const char *str)
{
int len = strlen(str);
if (len > 0) {
len++;
ffio_wfourcc(pb, tag);
avio_wl32(pb, len);
avio_put_str(pb, str);
if (len & 1)
avio_w8(pb, 0);
}
}

static int riff_has_valid_tags(AVFormatContext *s)
{
int i;
AVDictionaryEntry *t = NULL;

for (i = 0; *ff_riff_tags[i]; i++) {
if ((t = av_dict_get(s->metadata, ff_riff_tags[i], NULL, AV_DICT_MATCH_CASE)))
return 1;
}

return 0;
}

void ff_riff_write_info(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
int i;
int64_t list_pos;
AVDictionaryEntry *t = NULL;

ff_metadata_conv(&s->metadata, ff_riff_info_conv, NULL);

/* writing empty LIST is not nice and may cause problems */
if (!riff_has_valid_tags(s))
return;

list_pos = ff_start_tag(pb, "LIST");
ffio_wfourcc(pb, "INFO");
for (i = 0; *ff_riff_tags[i]; i++) {
if ((t = av_dict_get(s->metadata, ff_riff_tags[i], NULL, AV_DICT_MATCH_CASE)))
ff_riff_write_info_tag(s->pb, t->key, t->value);
}
ff_end_tag(pb, list_pos);
}
#endif //CONFIG_MUXERS

#if CONFIG_DEMUXERS
@@ -705,37 +783,6 @@ int ff_get_bmp_header(AVIOContext *pb, AVStream *st, unsigned *esize)
avio_rl32(pb); /* ClrImportant */
return tag1;
}
#endif // CONFIG_DEMUXERS

void ff_parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale)
{
int gcd;
int audio_frame_size;

/* We use the known constant frame size for the codec if known, otherwise
fallback to using AVCodecContext.frame_size, which is not as reliable
for indicating packet duration */
audio_frame_size = av_get_audio_frame_duration(stream, 0);
if (!audio_frame_size)
audio_frame_size = stream->frame_size;

*au_ssize= stream->block_align;
if (audio_frame_size && stream->sample_rate) {
*au_scale = audio_frame_size;
*au_rate= stream->sample_rate;
}else if(stream->codec_type == AVMEDIA_TYPE_VIDEO ||
stream->codec_type == AVMEDIA_TYPE_DATA ||
stream->codec_type == AVMEDIA_TYPE_SUBTITLE){
*au_scale= stream->time_base.num;
*au_rate = stream->time_base.den;
}else{
*au_scale= stream->block_align ? stream->block_align*8 : 8;
*au_rate = stream->bit_rate ? stream->bit_rate : 8*stream->sample_rate;
}
gcd= av_gcd(*au_scale, *au_rate);
*au_scale /= gcd;
*au_rate /= gcd;
}

void ff_get_guid(AVIOContext *s, ff_asf_guid *g)
{
@@ -799,51 +846,4 @@ int ff_read_riff_info(AVFormatContext *s, int64_t size)

return 0;
}

static int riff_has_valid_tags(AVFormatContext *s)
{
int i;
AVDictionaryEntry *t = NULL;

for (i = 0; *ff_riff_tags[i]; i++) {
if ((t = av_dict_get(s->metadata, ff_riff_tags[i], NULL, AV_DICT_MATCH_CASE)))
return 1;
}

return 0;
}

void ff_riff_write_info_tag(AVIOContext *pb, const char *tag, const char *str)
{
int len = strlen(str);
if (len > 0) {
len++;
ffio_wfourcc(pb, tag);
avio_wl32(pb, len);
avio_put_str(pb, str);
if (len & 1)
avio_w8(pb, 0);
}
}

void ff_riff_write_info(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
int i;
int64_t list_pos;
AVDictionaryEntry *t = NULL;

ff_metadata_conv(&s->metadata, ff_riff_info_conv, NULL);

/* writing empty LIST is not nice and may cause problems */
if (!riff_has_valid_tags(s))
return;

list_pos = ff_start_tag(pb, "LIST");
ffio_wfourcc(pb, "INFO");
for (i = 0; *ff_riff_tags[i]; i++) {
if ((t = av_dict_get(s->metadata, ff_riff_tags[i], NULL, AV_DICT_MATCH_CASE)))
ff_riff_write_info_tag(s->pb, t->key, t->value);
}
ff_end_tag(pb, list_pos);
}
#endif // CONFIG_DEMUXERS

+ 13
- 2
libavformat/rtp.h View File

@@ -84,13 +84,24 @@ enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type);

/* RTCP packet types */
enum RTCPType {
RTCP_FIR = 192,
RTCP_NACK, // 193
RTCP_SMPTETC,// 194
RTCP_IJ, // 195
RTCP_SR = 200,
RTCP_RR, // 201
RTCP_SDES, // 202
RTCP_BYE, // 203
RTCP_APP // 204
RTCP_APP, // 204
RTCP_RTPFB,// 205
RTCP_PSFB, // 206
RTCP_XR, // 207
RTCP_AVB, // 208
RTCP_RSI, // 209
RTCP_TOKEN,// 210
};

#define RTP_PT_IS_RTCP(x) ((x) >= RTCP_SR && (x) <= RTCP_APP)
#define RTP_PT_IS_RTCP(x) (((x) >= RTCP_FIR && (x) <= RTCP_IJ) || \
((x) >= RTCP_SR && (x) <= RTCP_TOKEN))

#endif /* AVFORMAT_RTP_H */

+ 14
- 2
libavformat/rtsp.c View File

@@ -75,6 +75,9 @@
{ "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
{ "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }

#define RTSP_REORDERING_OPTS() \
{ "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }

const AVOption ff_rtsp_options[] = {
{ "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
@@ -88,17 +91,20 @@ const AVOption ff_rtsp_options[] = {
{ "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
{ "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
{ "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
RTSP_REORDERING_OPTS(),
{ NULL },
};

static const AVOption sdp_options[] = {
RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
RTSP_REORDERING_OPTS(),
{ NULL },
};

static const AVOption rtp_options[] = {
RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
RTSP_REORDERING_OPTS(),
{ NULL },
};

@@ -608,6 +614,13 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
AVStream *st = NULL;
int reordering_queue_size = rt->reordering_queue_size;
if (reordering_queue_size < 0) {
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
reordering_queue_size = 0;
else
reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
}

/* open the RTP context */
if (rtsp_st->stream_index >= 0)
@@ -632,8 +645,7 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
else if (CONFIG_RTPDEC)
rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
rtsp_st->sdp_payload_type,
(rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
reordering_queue_size);

if (!rtsp_st->transport_priv) {
return AVERROR(ENOMEM);


+ 6
- 1
libavformat/rtsp.h View File

@@ -385,12 +385,17 @@ typedef struct RTSPState {
* Timeout to wait for incoming connections.
*/
int initial_timeout;

/**
* Size of RTP packet reordering queue.
*/
int reordering_queue_size;
} RTSPState;

#define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
receive packets only from the right
source address and port. */
#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */

/**
* Describe a single stream, as identified by a single m= line block in the


+ 6
- 6
tests/fate/screen.mak View File

@@ -1,8 +1,8 @@
# FIXME dropped frames in this test because of coarse timebase
FATE_SCREEN += fate-cscd
FATE_SCREEN-$(call DEMDEC, AVI, CSCD) += fate-cscd
fate-cscd: CMD = framecrc -i $(SAMPLES)/CSCD/sample_video.avi -an -pix_fmt rgb24

FATE_SCREEN += fate-dxtory
FATE_SCREEN-$(call DEMDEC, AVI, DXTORY) += fate-dxtory
fate-dxtory: CMD = framecrc -i $(SAMPLES)/dxtory/dxtory_mic.avi

FATE_FRAPS += fate-fraps-v0
@@ -23,7 +23,7 @@ fate-fraps-v4: CMD = framecrc -i $(SAMPLES)/fraps/WoW_2006-11-03_14-58-17-19-nos
FATE_FRAPS += fate-fraps-v5
fate-fraps-v5: CMD = framecrc -i $(SAMPLES)/fraps/fraps-v5-bouncing-balls-partial.avi

FATE_SCREEN += $(FATE_FRAPS)
FATE_SCREEN-$(call DEMDEC, AVI, FRAPS) += $(FATE_FRAPS)
fate-fraps: $(FATE_FRAPS)

FATE_TSCC += fate-tscc-15bit
@@ -32,7 +32,7 @@ fate-tscc-15bit: CMD = framecrc -i $(SAMPLES)/tscc/oneminute.avi -t 15 -pix_fmt
FATE_TSCC += fate-tscc-32bit
fate-tscc-32bit: CMD = framecrc -i $(SAMPLES)/tscc/2004-12-17-uebung9-partial.avi -pix_fmt rgb24 -an

FATE_SCREEN-$(CONFIG_ZLIB) += $(FATE_TSCC)
FATE_SCREEN-$(call DEMDEC, AVI, TSCC) += $(FATE_TSCC)
fate-tscc: $(FATE_TSCC)

FATE_VMNC += fate-vmnc-16bit
@@ -41,7 +41,7 @@ fate-vmnc-16bit: CMD = framecrc -i $(SAMPLES)/VMnc/test.avi -pix_fmt rgb24
FATE_VMNC += fate-vmnc-32bit
fate-vmnc-32bit: CMD = framecrc -i $(SAMPLES)/VMnc/VS2k5DebugDemo-01-partial.avi -pix_fmt rgb24

FATE_SCREEN += $(FATE_VMNC)
FATE_SCREEN-$(call DEMDEC, AVI, VMNC) += $(FATE_VMNC)
fate-vmnc: $(FATE_VMNC)

FATE_ZMBV += fate-zmbv-8bit
@@ -56,7 +56,7 @@ fate-zmbv-16bit: CMD = framecrc -i $(SAMPLES)/zmbv/zmbv_16bit.avi -pix_fmt rgb24
FATE_ZMBV += fate-zmbv-32bit
fate-zmbv-32bit: CMD = framecrc -i $(SAMPLES)/zmbv/zmbv_32bit.avi -pix_fmt rgb24 -t 25

FATE_SCREEN-$(CONFIG_ZLIB) += $(FATE_ZMBV)
FATE_SCREEN-$(call DEMDEC, AVI, ZMBV) += $(FATE_ZMBV)
fate-zmbv: $(FATE_ZMBV)

FATE_SCREEN += $(FATE_SCREEN-yes)


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