May 2010 - FFmpeg-soc - lists.ffmpeg.org

shrink rtsp sdp buffer
by Josh Allmann 14 Jun '10

14 Jun '10

Hi, The following reverts changes made in r18508 and shrinks the sdp buffer in rtsp.c. I only do this because 16k on the stack is a lot. This cleans up the last remnants of Vorbis in rtsp.c -- its sdp/fmtp processing is now done in rtpdec_xiph.c, so this codepath is not touched by Vorbis over RTP anymore except in the case of an error in parsing sdp/fmtp. Even then, it should still fall-through rather cleanly. Josh

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GSoC start
by Ronald S. Bultje 10 Jun '10

10 Jun '10

Hi guys, last week, GSoC started. It's time we start our gears and get in full coding mode. Who needs good weather anyway. If you haven't yet emailed your mentor with your first progress report, update or other form of regular email exchange (IRC is fine also, but please send out a weekly summary by email), then now would be a great moment to start. For all students, please CC me on your weekly emails just so I can see that all projects are moving fine, and I can make pretty graphs (a la Mike) of how everyone is progressing overtime. All students, please do the following: 1) make sure you're subscribed to ffmpeg-soc(a)mplayerhq.hu (if something doesn't work, ask for help) 2) reply to this email with what you're currently working on (1-2 lines is enough), so that we know you're ready and have started coding 3) if you don't yet have a SVN account for our summer-of-code repository, please reply to this email so we can fix that 4) if anything else needs to be done or I did something wrong, blame me and I'll fix it Good luck! Ronald

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[RFC] rename vsrc movie to vsrc file
by Baptiste Coudurier 09 Jun '10

09 Jun '10

Guys, What do you think about it ? -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org

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[PATCH] factor out writing rtsp headers
by Josh Allmann 31 May '10

31 May '10

$subj, this may not be needed for RTSP over HTTP (depending on how I do it), but I think this patch cleans up the code a bit regardless. Messy diff though; all I did was factor out a hunk of code to a separate function, but old and new lines are interleaved here. My git-fu is weak; if there's a command to make a cleaner diff (I used git diff -b), please let me know. Tested with feng and the cathedral mp4, works OK. Josh

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[soc] soc g723.1
by naufal11＠gmail.com 29 May '10

29 May '10

- Log ----------------------------------------------------------------- commit 590ea9c2cd3c7c73307356f22b86a6e68c49c4d7 Author: Naufal <naufal11(a)gmail.com> Date: Sat May 29 23:20:23 2010 +0530 Break long line diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c index 7a62b91..bc8e0cf 100755 --- a/libavcodec/g723_1.c +++ b/libavcodec/g723_1.c @@ -112,7 +112,8 @@ static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf, } p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS); if (p->subframe[i].ad_cb_gain < ad_cb_len) { - p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain * GAIN_LEVELS; + p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain * + GAIN_LEVELS; } else { return -1; } ----------------------------------------------------------------------- Summary of changes: libavcodec/g723_1.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) -- http://github.com/naufal/ffmpeg-soc

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[soc] soc g723.1
by naufal11＠gmail.com 29 May '10

29 May '10

- Log ----------------------------------------------------------------- commit f88ce69634a9def332abda541439dfad031bd8b6 Author: Naufal <naufal11(a)gmail.com> Date: Sat May 29 21:13:59 2010 +0530 Add G723.1 raw bitstream read support diff --git a/libavformat/Makefile b/libavformat/Makefile index ac812cc..1cb5d00 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -79,6 +79,7 @@ OBJS-$(CONFIG_FLV_MUXER) += flvenc.o avc.o OBJS-$(CONFIG_FOURXM_DEMUXER) += 4xm.o OBJS-$(CONFIG_FRAMECRC_MUXER) += framecrcenc.o OBJS-$(CONFIG_FRAMEMD5_MUXER) += md5enc.o +OBJS-$(CONFIG_G723_1_DEMUXER) += raw.o OBJS-$(CONFIG_GIF_MUXER) += gif.o OBJS-$(CONFIG_GSM_DEMUXER) += raw.o id3v2.o OBJS-$(CONFIG_GXF_DEMUXER) += gxf.o diff --git a/libavformat/raw.c b/libavformat/raw.c index 7837de8..00c43f1 100644 --- a/libavformat/raw.c +++ b/libavformat/raw.c @@ -871,6 +871,20 @@ AVOutputFormat eac3_muxer = { }; #endif +#if CONFIG_G723_1_DEMUXER +AVInputFormat g723_1_demuxer = { + "g723_1", + NULL_IF_CONFIG_SMALL("raw G723.1"), + 0, + NULL, + audio_read_header, + ff_raw_read_partial_packet, + .flags = AVFMT_GENERIC_INDEX, + .extensions = "tco", + .value = CODEC_ID_G723_1, +}; +#endif + #if CONFIG_GSM_DEMUXER AVInputFormat gsm_demuxer = { "gsm", ----------------------------------------------------------------------- Summary of changes: libavformat/Makefile | 1 + libavformat/raw.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+), 0 deletions(-) -- http://github.com/naufal/ffmpeg-soc

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[soc] UNNAMED PROJECT
by naufal11＠gmail.com 29 May '10

29 May '10

- Log ----------------------------------------------------------------- commit 6e9317c15dc62a45a2325acc145c00e086f1026e Author: Naufal <naufal11(a)gmail.com> Date: Sat May 29 20:59:59 2010 +0530 Add G723.1 decoder wireframe diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 764555e..d3cd00b 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -129,6 +129,7 @@ OBJS-$(CONFIG_FLIC_DECODER) += flicvideo.o OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o huffman.o OBJS-$(CONFIG_FRWU_DECODER) += frwu.o +OBJS-$(CONFIG_G723_1_DECODER) += g723_1.o OBJS-$(CONFIG_GIF_DECODER) += gifdec.o lzw.o OBJS-$(CONFIG_GIF_ENCODER) += gif.o lzwenc.o OBJS-$(CONFIG_H261_DECODER) += h261dec.o h261.o \ diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index a5d8ce1..696ed3c 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -227,6 +227,7 @@ void avcodec_register_all(void) REGISTER_DECODER (DSICINAUDIO, dsicinaudio); REGISTER_DECODER (EAC3, eac3); REGISTER_ENCDEC (FLAC, flac); + REGISTER_DECODER (G723_1, g723_1); REGISTER_DECODER (IMC, imc); REGISTER_DECODER (MACE3, mace3); REGISTER_DECODER (MACE6, mace6); diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index 60aa876..3f2a2a9 100644 --- a/libavcodec/avcodec.h +++ b/libavcodec/avcodec.h @@ -336,6 +336,7 @@ enum CodecID { CODEC_ID_ATRAC1, CODEC_ID_BINKAUDIO_RDFT, CODEC_ID_BINKAUDIO_DCT, + CODEC_ID_G723_1, /* subtitle codecs */ CODEC_ID_DVD_SUBTITLE= 0x17000, diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c new file mode 100755 index 0000000..7a62b91 --- /dev/null +++ b/libavcodec/g723_1.c @@ -0,0 +1,191 @@ +#include "avcodec.h" +#define ALT_BITSTREAM_READER_LE +#include "get_bits.h" + +#define SUBFRAMES 4 +#define FRAME_LEN 240 +#define SUBFRAME_LEN (FRAME_LEN / SUBFRAMES) +#define PITCH_MIN 18 +#define GAIN_LEVELS 24 + +/* + * G723.1 frame types + */ +typedef enum { + Frame6k3 = 0, ///< 6.3 kbps frame rate + Frame5k3, ///< 5.3 kbps frame rate + FrameSID, ///< Silence Insertion Descriptor frame + FrameUntransmitted +} FrameType; + +static const uint8_t frame_size[4] = {24, 20, 4, 1}; + +typedef enum { + Rate6k3, + Rate5k3 +} Rate; + +/* + * G723.1 unpacked data subframe + */ +typedef struct { + uint16_t ad_cb_lag; ///< adaptive codebook lag + uint16_t ad_cb_gain; + uint16_t trans_gain; + uint16_t pulse_pos; + uint16_t pulse_sign; + uint16_t grid_index; + uint16_t amp_index; +} G723_1_Subframe; + +typedef struct g723_1_context { + uint32_t lsp_index; + uint16_t pitch_lag[2]; + G723_1_Subframe subframe[4]; + FrameType cur_frame_type; + Rate cur_rate; +} G723_1_Context; + +static av_cold int g723_1_decode_init(AVCodecContext *avctx) +{ + avctx->sample_fmt = SAMPLE_FMT_S16; + avctx->channels = 1; + avctx->sample_rate = 8000; + return 0; +} + +/* + * Unpack the frame into parameters. + * + * @param p the context + * @param buf pointer to the input buffer + * @param buf_size size of the input buffer + */ +static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf, + int buf_size) +{ + GetBitContext gb; + int ad_cb_len; + int temp, i; + + init_get_bits(&gb, buf, buf_size * 8); + + // Extract frame type and rate info + p->cur_frame_type = get_bits(&gb, 2); + + if (p->cur_frame_type == FrameUntransmitted) + return 0; + + p->lsp_index = get_bits(&gb, 24); + + if (p->cur_frame_type == FrameSID) { + p->subframe[0].amp_index = get_bits(&gb, 6); + return 0; + } + + // Extract the info common to both rates + p->cur_rate = p->cur_frame_type ? Rate5k3 : Rate6k3; + + p->pitch_lag[0] = get_bits(&gb, 7); + if (p->pitch_lag[0] > 123) // test if forbidden code + return -1; + p->pitch_lag[0] += PITCH_MIN; + p->subframe[1].ad_cb_lag = get_bits(&gb, 2); + + p->pitch_lag[1] = get_bits(&gb, 7); + if (p->pitch_lag[1] > 123) + return -1; + p->pitch_lag[1] += PITCH_MIN; + p->subframe[3].ad_cb_lag = get_bits(&gb, 2); + p->subframe[0].ad_cb_lag = 1; + p->subframe[2].ad_cb_lag = 1; + + for (i = 0; i < SUBFRAMES; i++) { + // Extract combined gain + temp = get_bits(&gb, 12); + ad_cb_len = 170; + p->subframe[i].trans_gain = 0; + if (p->cur_rate == Rate6k3 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) { + p->subframe[i].trans_gain = temp >> 11; + temp &= 0x7ff; + ad_cb_len = 85; + } + p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS); + if (p->subframe[i].ad_cb_gain < ad_cb_len) { + p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain * GAIN_LEVELS; + } else { + return -1; + } + } + + for (i = 0; i < SUBFRAMES; i++) + p->subframe[i].grid_index = get_bits(&gb, 1); + + if (p->cur_frame_type == Frame6k3) { + skip_bits(&gb, 1); // skip reserved bit + + // Compute pulse_pos index using the 13-bit combined position index + temp = get_bits(&gb, 13); + p->subframe[0].pulse_pos = temp / 810; + + temp -= p->subframe[0].pulse_pos * 810; + p->subframe[1].pulse_pos = FASTDIV(temp, 90); + + temp -= p->subframe[1].pulse_pos * 90; + p->subframe[2].pulse_pos = FASTDIV(temp, 9); + p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9; + + p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) + + get_bits(&gb, 16); + p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) + + get_bits(&gb, 14); + p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) + + get_bits(&gb, 16); + p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) + + get_bits(&gb, 14); + + p->subframe[0].pulse_sign = get_bits(&gb, 6); + p->subframe[1].pulse_sign = get_bits(&gb, 5); + p->subframe[2].pulse_sign = get_bits(&gb, 6); + p->subframe[3].pulse_sign = get_bits(&gb, 5); + } else { // Frame5k3 + for (i = 0; i < SUBFRAMES; i++) + p->subframe[i].pulse_pos = get_bits(&gb, 12); + + for (i = 0; i < SUBFRAMES; i++) + p->subframe[i].pulse_sign = get_bits(&gb, 4); + } + + return 0; +} + +static int g723_1_decode_frame(AVCodecContext *avctx, void *data, + int *data_size, AVPacket *avpkt) +{ + G723_1_Context *p = avctx->priv_data; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; + + if (!buf_size || buf_size < frame_size[buf[0] & 3]) { + *data_size = 0; + return buf_size; + } + + if (unpack_bitstream(p, buf, buf_size) < 0) { + av_log(avctx, AV_LOG_ERROR, "G723.1: Bad frame\n"); + return AVERROR_INVALIDDATA; + } + + return frame_size[p->cur_frame_type]; +} + +AVCodec g723_1_decoder = { + .name = "g723_1", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_G723_1, + .priv_data_size = sizeof(G723_1_Context), + .init = g723_1_decode_init, + .decode = g723_1_decode_frame, + .long_name = NULL_IF_CONFIG_SMALL("G.723.1"), + .capabilities = CODEC_CAP_SUBFRAMES, +}; diff --git a/libavformat/rtp.c b/libavformat/rtp.c index a8dcfd7..b7ab999 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -43,7 +43,7 @@ static const struct { {0, "PCMU", AVMEDIA_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, {3, "GSM", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, - {4, "G723", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {4, "G723", AVMEDIA_TYPE_AUDIO, CODEC_ID_G723_1, 8000, 1}, {5, "DVI4", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {6, "DVI4", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, {7, "LPC", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, ----------------------------------------------------------------------- Summary of changes: libavcodec/Makefile | 1 + libavcodec/allcodecs.c | 1 + libavcodec/avcodec.h | 1 + libavcodec/g723_1.c | 191 ++++++++++++++++++++++++++++++++++++++++++++++++ libavformat/rtp.c | 2 +- 5 files changed, 195 insertions(+), 1 deletions(-) create mode 100755 libavcodec/g723_1.c -- http://github.com/naufal/ffmpeg-soc

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[RFC] rtsp http tunneling implementation
by Josh Allmann 29 May '10

29 May '10

Since RTSP over HTTP basically involves a few http request-replies and base64-encoding RTSP into a HTTP POST, I have the following thoughts: -Add HTTP as a lower transport mechanism in rtsp.c To do this we need to signal to http.c to add the "x-sessioncookie" header when it is required. I was thinking of adding a new field to HTTPContext for this, and setting it via a flag passed to http_open (eg, URL_WRONLY | URL_RTSP ). bikeshed: There is some weirdness in the spec, eg it mandates HTTP/1.0 but uses 1.1-specific headers such as cache-control (albeit optionally). Adding configurable http/1.{0,1} to http.c seems more trouble than it's worth (we send out 1.1 requests at present) -- feng handles 1.1 requests fine. Haven't tried DSS though. Let me know what you think, especially if there is a better way. Josh

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[soc]: r5818 - libavfilter/README
by stefano 26 May '10

26 May '10

Author: stefano Date: Wed May 26 00:54:24 2010 New Revision: 5818 Log: Remove reference to the option -vfilters, replaced with -vf. Modified: libavfilter/README Modified: libavfilter/README ============================================================================== --- libavfilter/README Mon May 24 18:43:37 2010 (r5817) +++ libavfilter/README Wed May 26 00:54:24 2010 (r5818) @@ -56,12 +56,12 @@ RUNNING IT: The command line syntax for specifying filters is similar to that of mplayer. For example, to vertically flip a video, you would do: -./ffplay -vfilters vflip input_video.avi +./ffplay -vf vflip input_video.avi -The argument to -vfilters lists the video filters to load. A comma separates +The argument to -vf lists the video filters to load. A comma separates sequential filters. For example, this command: -./ffplay -vfilters vflip,vflip input_video.avi +./ffplay -vf vflip,vflip input_video.avi will run the video through two instances of the vflip filter, which should result in the original, unflipped video. @@ -70,7 +70,7 @@ Some filters accept parameters, which ca would be done for mplayer. For example, to scale a video to 640x480, you could use this command: -./ffplay -vfilters scale=640:480 input_video.avi +./ffplay -vf scale=640:480 input_video.avi Be aware that some filters, notably the crop filter, interpret the parameters differently than mplayer does. Documentation for each filter is provided in the

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[soc]: r5817 - mms/mmst.c
by spyfeng 24 May '10

24 May '10

Author: spyfeng Date: Mon May 24 18:43:37 2010 New Revision: 5817 Log: fix goto style in consistent. Modified: mms/mmst.c Modified: mms/mmst.c ============================================================================== --- mms/mmst.c Mon May 24 18:32:38 2010 (r5816) +++ mms/mmst.c Mon May 24 18:43:37 2010 (r5817) @@ -609,18 +609,16 @@ static int mms_open(URLContext *h, const if (err) goto fail; err = mms_safe_send_recv(mms, NULL, SC_PKT_ASF_HEADER); + if (err) + goto fail; + if((mms->incoming_flags != 0X08) && (mms->incoming_flags != 0X0C)) + goto fail; + err = asf_header_parser(mms); if (err) { + dprintf(NULL, "asf header parsed failed!\n"); goto fail; - } else { - if((mms->incoming_flags == 0X08) || (mms->incoming_flags == 0X0C)) { - err = asf_header_parser(mms); - if (err) { - dprintf(NULL, "asf header parsed failed!\n"); - goto fail; - } - mms->header_parsed = 1; - } } + mms->header_parsed = 1; if (!mms->asf_packet_len || !mms->stream_num) goto fail;

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