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

Re: [FFmpeg-soc] libavfilter audio work - qualification task
by S.N. Hemanth Meenakshisundaram 17 May '10

17 May '10

On 04/21/2010 03:35 PM, Stefano Sabatini wrote: > > [...] > _______________________________________________ > FFmpeg-soc mailing list > FFmpeg-soc(a)mplayerhq.hu > https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-soc > > Hi, Attached is the vf_drawtext.c and changes to libavfilter.texi (both are patches against libavfilter soc like you asked for) Following is the list of changes : 1. Using a single context structure now. Descriptive variable names. 2. Input parameters also are descriptive now, but sometimes different from the corresponding context variable where the context variable name is too long. 3. Using macro instead of function to parse libfreetype messages now. 4. Changed warning messages to AV_LOG_WARNINGS. 4. User now needs to provide only one of text or file, if both are given then file is used. 5. Made color string to YUV conversion separate function called for both fg and bg color. 6. Alignments, indents, description of filter and redundant '!='s fixed. 7. Reading file only once now in init. 8. Using re-entrant localtime_r now. 9. The documentation now uses $var{x} instead of ``x''. Made these changes throughout the libavfilter doc, not just drawtext. Is that ok? 10. Updated documentation with new parameter names, strftime behavior and the file/text requirement. As for the alpha channel support in drawbox function, do we need to update the alpha values? Should the caller be allowed to specify bgcolor or fgcolor in #RRGGBBAA format? Please review and let me know if there's anything else that needs to be fixed. Regards, Hemanth

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[PATCH] updated! vf_overlay alpha patch and watermarking using PNG with alpha
by Artur Bodera 15 May '10

15 May '10

For those that have given up on ffmpeg and +libavfilter -vhook debacle. Here's a "just add water" solution to all your trouble: transparent PNG's and watermarks in recent ffmpeg compilations. I'm attaching a working patch that will enable alpha-transparency in overlay filter. This, on the other hand, will let you use overlay filter to superimpose a video clip or transparent png watermark on another video. This has been tested with recent revisions of svn:// svn.ffmpeg.org/soc/libavfilter rev. 5387 and ffmpeg rev. 19868 (16 Sep 2009). Compiled and running under both x86 and x86_64 on linux. Example 1 - insert transparent PNG watermark in bottom left corner of the video: -vfilters "movie=0:png:logo.png [wm];[in][wm] overlay=10:mainH-overlayH-10:1 [out]" Notice the last parameter to overlay ":1" - this enables alpha blending. Example 2 - insert 2 different transparent PNG watermarks (second watermark on bottom right corner): -vfilters "movie=0:png:logo.png [wm];movie=0:png:logo2.png [awm];[in][wm] overlay=10:mainH-overlayH-10:1 [int];[int][awm] overlay=mainW-overlayW-10:mainH-overlayH-10:1 [out]" You could chain and add more overlays this way but the efficiency of such approach is yet to be tested. This should save you a lot of time and you can continue to use your favorite transcoder for web video publishing! With best regards. -- __ /.)\ +48 695 600 936 \(./ abodera(a)gmail.com

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libavfilter audio work - qualification task
by S.N. Hemanth Meenakshisundaram 10 May '10

10 May '10

Hi All, I would like to take up the libavfilter audio work for gsoc. What would be a good qualification task for this? The wiki says the mentor is yet to be determined, so am asking on this list. I was thinking of taking up the vf_imlib2 work or Cedric's blending routines from the interesting patches list. Is someone else already working on these or is there a more suitable qualification task I can take up? Thanks, Hemanth

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segmenter Error : 32bit rollover in MPEG
by 김동규 27 Apr '10

27 Apr '10

I use an ffmpeg base segmenter application: http://svn.assembla.com/svn/legend/segmenter/segmenter.c The segmenter working fine, but I have a problem with an ffmpeg feature The input stream (mpeg2ts) contains pts and dts information. These are OK, but after 26 hours running, the ffmpeg reaches the 2exp33 limitation, and the timestamp restart (33 bit rollover in MPEG). At this point the segmenter stopped working with the "non monotone timestamp" error messages. When I restart the segmenter, everything works fine again - up to 26 hours-. My question is, how can I solve this situation in the code of the segmenter? How can I do that? Regards..

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[soc]: r5776 - mms/mmst.c
by spyfeng 26 Apr '10

26 Apr '10

Author: spyfeng Date: Sun Apr 25 10:12:44 2010 New Revision: 5776 Log: add mms_safe_send_recv() function to check the expected packet type according to send packet type. Modified: mms/mmst.c Modified: mms/mmst.c ============================================================================== --- mms/mmst.c Sun Apr 25 08:10:48 2010 (r5775) +++ mms/mmst.c Sun Apr 25 10:12:44 2010 (r5776) @@ -353,6 +353,22 @@ static MMSSCPacketType get_tcp_server_re return packet_type; } +static int mms_safe_send_recv(MMSContext *mms, int (*send_fun)(), const MMSSCPacketType expect_type) +{ + MMSSCPacketType type; + if(send_fun) { + if (send_fun(mms) < 0) { + dprintf(NULL, "Send Packet error before expecting recv packet %d\n", expect_type); + return -1; + } + } + if((type = get_tcp_server_response(mms)) != expect_type) { + dprintf(NULL,"Unhandled packet type %d\n", type); + return -1; + } + return 0; +} + static int send_media_header_request(MMSContext *mms) { start_command_packet(mms, CS_PKT_MEDIA_HEADER_REQUEST); @@ -469,7 +485,6 @@ static int read_data(MMSContext *mms, ui static int read_mms_packet(MMSContext *mms, uint8_t *buf, int buf_size) { int result = 0; - MMSSCPacketType packet_type; int size_to_copy; do { @@ -488,8 +503,8 @@ static int read_mms_packet(MMSContext *m result = read_data(mms, buf, buf_size); } else { /* Read from network */ - packet_type= get_tcp_server_response(mms); - if (packet_type == SC_PKT_ASF_MEDIA) { + int err = mms_safe_send_recv(mms, NULL, SC_PKT_ASF_MEDIA); + if (!err) { if(mms->remaining_in_len>mms->asf_packet_len) { dprintf(NULL, "Incoming packet" "larger than the asf packet size stated (%d>%d)\n", @@ -499,8 +514,6 @@ static int read_mms_packet(MMSContext *m // copy the data to the packet buffer. result = read_data(mms, buf, buf_size); } - } else { - dprintf(NULL, "Got a unkown Packet Type: 0x%x\n", packet_type); } } } while(!result); // only return one packet. @@ -532,40 +545,9 @@ static int mms_close(URLContext *h) return 0; } -static int handle_mms_msg_pkt(MMSContext *mms, const MMSSCPacketType packet_type) -{ - int ret = -1; - - switch(packet_type) { - case SC_PKT_CLIENT_ACCEPTED: - ret = send_protocol_select(mms); - break; - case SC_PKT_PROTOCOL_ACCEPTED: - ret = send_media_file_request(mms); - break; - case SC_PKT_MEDIA_FILE_DETAILS: - ret = send_media_header_request(mms); - break; - case SC_PKT_HEADER_REQUEST_ACCEPTED: - ret = 0; - break; - case SC_PKT_ASF_HEADER: - if((mms->incoming_flags == 0X08) || (mms->incoming_flags == 0X0C)) { - ret = asf_header_parser(mms); - mms->header_parsed = 1; - } - break; - default: - dprintf(NULL, "Unhandled packet type %d\n", packet_type); - break; - } - return ret; -} - static int mms_open(URLContext *h, const char *uri, int flags) { MMSContext *mms; - MMSSCPacketType packet_type; int port; char tcpname[256]; @@ -593,13 +575,26 @@ static int mms_open(URLContext *h, const mms->packet_id = 3; // default, initial value. mms->header_packet_id = 2; // default, initial value. - - send_startup_packet(mms); - while (!mms->header_parsed) { - packet_type = get_tcp_server_response(mms); - ret = handle_mms_msg_pkt(mms, packet_type); - if (ret < 0) - break; + err = mms_safe_send_recv(mms, send_startup_packet, SC_PKT_CLIENT_ACCEPTED); + if (err) + goto fail; + err = mms_safe_send_recv(mms, send_protocol_select, SC_PKT_PROTOCOL_ACCEPTED); + if (err) + goto fail; + err = mms_safe_send_recv(mms, send_media_file_request, SC_PKT_MEDIA_FILE_DETAILS); + if (err) + goto fail; + err = mms_safe_send_recv(mms, send_media_header_request, SC_PKT_HEADER_REQUEST_ACCEPTED); + if (err) + goto fail; + err = mms_safe_send_recv(mms, NULL, SC_PKT_ASF_HEADER); + if (err) + goto fail; + else { + if((mms->incoming_flags == 0X08) || (mms->incoming_flags == 0X0C)) { + ret = asf_header_parser(mms); + mms->header_parsed = 1; + } } if (!mms->asf_packet_len || !mms->stream_num) @@ -651,22 +646,15 @@ static int mms_read(URLContext *h, uint8 && !mms->is_playing) { dprintf(NULL, "mms_read() before play().\n"); clear_stream_buffers(mms); - result = send_stream_selection_request(mms); - if(result < 0) + result = mms_safe_send_recv(mms, send_stream_selection_request, SC_PKT_STREAM_ID_ACCEPTED); + if (result) return result; - if (get_tcp_server_response(mms) != SC_PKT_STREAM_ID_ACCEPTED) { - dprintf(NULL, "Can't get stream id accepted packet.\n"); - return 0; - } - // send media packet request - send_media_packet_request(mms); - if (get_tcp_server_response(mms) == SC_PKT_MEDIA_PKT_FOLLOWS) { - mms->is_playing = 1; - } else { - dprintf(NULL, "Canot get media follows packet from server.\n"); - return 0; - } + result = mms_safe_send_recv(mms, send_media_packet_request, SC_PKT_MEDIA_PKT_FOLLOWS); + if (result) + return result; + else + mms->is_playing = 1; } return read_mms_packet(mms, buf, size); }

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[soc]: r5777 - mms/mmst.c
by spyfeng 26 Apr '10

26 Apr '10

Author: spyfeng Date: Mon Apr 26 18:24:08 2010 New Revision: 5777 Log: add parameter type for function pointer. Modified: mms/mmst.c Modified: mms/mmst.c ============================================================================== --- mms/mmst.c Sun Apr 25 10:12:44 2010 (r5776) +++ mms/mmst.c Mon Apr 26 18:24:08 2010 (r5777) @@ -353,7 +353,9 @@ static MMSSCPacketType get_tcp_server_re return packet_type; } -static int mms_safe_send_recv(MMSContext *mms, int (*send_fun)(), const MMSSCPacketType expect_type) +static int mms_safe_send_recv(MMSContext *mms, + int (*send_fun)(MMSContext *mms), + const MMSSCPacketType expect_type) { MMSSCPacketType type; if(send_fun) {

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[soc]: r5775 - mms/mmst.c
by spyfeng 25 Apr '10

25 Apr '10

Author: spyfeng Date: Sun Apr 25 08:10:48 2010 New Revision: 5775 Log: add comments to explain the checking condition. Modified: mms/mmst.c Modified: mms/mmst.c ============================================================================== --- mms/mmst.c Sun Apr 25 08:01:03 2010 (r5774) +++ mms/mmst.c Sun Apr 25 08:10:48 2010 (r5775) @@ -418,6 +418,9 @@ static int asf_header_parser(MMSContext } else if (!memcmp(p, ff_asf_stream_header, sizeof(ff_asf_guid))) { flags = AV_RL16(p + sizeof(ff_asf_guid)*3 + 24); stream_id = flags & 0x7F; + //The second condition is for checking CS_PKT_STREAM_ID_REQUEST packet size, + //we can calcuate the packet size by stream_num. + //Please see function send_stream_selection_request(). if (mms->stream_num < MAX_STREAMS && 46 + mms->stream_num * 6 < sizeof(mms->out_buffer)) { mms->streams[mms->stream_num].id = stream_id;

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[soc]: r5774 - mms/mmst.c
by spyfeng 25 Apr '10

25 Apr '10

Author: spyfeng Date: Sun Apr 25 08:01:03 2010 New Revision: 5774 Log: check mms->asf_packet_len <=0 and return failure value. Modified: mms/mmst.c Modified: mms/mmst.c ============================================================================== --- mms/mmst.c Tue Apr 20 17:50:20 2010 (r5773) +++ mms/mmst.c Sun Apr 25 08:01:03 2010 (r5774) @@ -409,9 +409,10 @@ static int asf_header_parser(MMSContext /* read packet size */ if (end - p > sizeof(ff_asf_guid) * 2 + 68) { mms->asf_packet_len = AV_RL32(p + sizeof(ff_asf_guid) * 2 + 64); - if (mms->asf_packet_len > sizeof(mms->in_buffer)) { + if (mms->asf_packet_len <= 0 || mms->asf_packet_len > sizeof(mms->in_buffer)) { dprintf(NULL,"Too large packet len:%d" " may overwrite in_buffer when padding", mms->asf_packet_len); + return -1; } } } else if (!memcmp(p, ff_asf_stream_header, sizeof(ff_asf_guid))) {

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G723.1 Frame Parser
by Mohamed Naufal 23 Apr '10

23 Apr '10

Hi Here is the code for a frame parser i wrote for the g723.1 codec as my GSoC qualification task #include <stdint.h> #include "libavutil/intreadwrite.h" #include "avcodec.h" #include "get_bits.h" #define PITCH_MIN 18 #define FRM_LEN 240 #define SFRM_LEN (FRM_LEN/4) #define AD_CB_SIZE_HIGH_RATE 85 ///< 85 entry codebook for Frame6k3s #define AD_CB_SIZE_LOW_RATE 170 ///< 170 entry codebook for Frame5k3s /* * G723.1 frame types */ typedef enum { Frame6k3, ///< 6.3 kbps frame rate Frame5k3, ///< 5.3 kbps frame rate FrameSID, ///< silence insertion descriptor frame FrameUntransmitted } FrameType; typedef enum { Rate6k3, Rate5k3 } Rate; /* * G723.1 unpacked data subframe */ typedef struct { uint16_t ad_cb_lag; ///< adaptive codebook lag uint16_t pitch_lag; uint16_t combined_gain; uint16_t pulse_pos; uint16_t pulse_sign; uint16_t grid_index; uint16_t amp_index; ///< for SID frames } G723_1_Subframe; /* * G723.1 unpacked data frame */ typedef struct { uint32_t lsp_index; G723_1_Subframe subframe[4]; } G723_1_Frame; typedef struct g723_1_context { G723_1_Frame frame; FrameType cur_frame_type; Rate cur_rate; int is_bad_frame; } G723_1_Context; /* * 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 void unpack_bitstream(G723_1_Context *p, const uint8_t *buf, int buf_size) { GetBitContext gb; G723_1_Frame *frame = &p->frame; uint32_t temp; 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; frame->lsp_index = get_bits(&gb, 24); if (p->cur_frame_type == FrameSID) { frame->subframe[0].amp_index = get_bits(&gb, 6); return; } // Extract the info common to both rates p->cur_rate = (p->cur_frame_type == 0) ? Rate6k3 : Rate5k3; temp = get_bits(&gb, 7); if (temp <= 123) { // test if forbidden code frame->subframe[0].pitch_lag = temp + PITCH_MIN; }else { p->is_bad_frame = 1; // transmission error return; } frame->subframe[1].ad_cb_lag = get_bits(&gb, 2); temp = get_bits(&gb, 7); if (temp <= 123) { frame->subframe[1].pitch_lag = temp + PITCH_MIN; }else { p->is_bad_frame = 1; return; } frame->subframe[3].ad_cb_lag = get_bits(&gb, 2); frame->subframe[0].ad_cb_lag = 1; frame->subframe[2].ad_cb_lag = 1; frame->subframe[0].combined_gain = get_bits(&gb, 12); frame->subframe[1].combined_gain = get_bits(&gb, 12); frame->subframe[2].combined_gain = get_bits(&gb, 12); frame->subframe[3].combined_gain = get_bits(&gb, 12); frame->subframe[0].grid_index = get_bits(&gb, 1); frame->subframe[1].grid_index = get_bits(&gb, 1); frame->subframe[2].grid_index = get_bits(&gb, 1); frame->subframe[3].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 13bit combined position index temp = get_bits(&gb, 13); frame->subframe[0].pulse_pos = (temp / 90) / 9; frame->subframe[1].pulse_pos = (temp / 90) % 9; frame->subframe[2].pulse_pos = (temp % 90) / 9; frame->subframe[3].pulse_pos = (temp % 90) % 9; frame->subframe[0].pulse_pos = (frame->subframe[0].pulse_pos << 16) + get_bits(&gb, 16); frame->subframe[1].pulse_pos = (frame->subframe[1].pulse_pos << 14) + get_bits(&gb, 14); frame->subframe[2].pulse_pos = (frame->subframe[2].pulse_pos << 16) + get_bits(&gb, 16); frame->subframe[3].pulse_pos = (frame->subframe[3].pulse_pos << 14) + get_bits(&gb, 14); frame->subframe[0].pulse_sign = get_bits(&gb, 6); frame->subframe[1].pulse_sign = get_bits(&gb, 5); frame->subframe[2].pulse_sign = get_bits(&gb, 6); frame->subframe[3].pulse_sign = get_bits(&gb, 5); }else { // Frame5k3 frame->subframe[0].pulse_pos = get_bits(&gb, 12); frame->subframe[1].pulse_pos = get_bits(&gb, 12); frame->subframe[2].pulse_pos = get_bits(&gb, 12); frame->subframe[3].pulse_pos = get_bits(&gb, 12); frame->subframe[0].pulse_sign = get_bits(&gb, 4); frame->subframe[1].pulse_sign = get_bits(&gb, 4); frame->subframe[2].pulse_sign = get_bits(&gb, 4); frame->subframe[3].pulse_sign = get_bits(&gb, 4); } } static int g723_1_parse_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; unpack_bitstream(p, buf, buf_size); if (p->is_bad_frame) { av_log_error(avctx, AV_LOG_ERROR, "Bad frame\n"); return AVERROR_INVALIDDATA; } return 0; } AVCodec g723_1_decoder = { .name = "g723_1", .type = CODEC_TYPE_AUDIO, .id = CODEC_ID_G723_1, .priv_data_size = sizeof(G723_1_Context), .decode = g723_1_parse_frame, .long_name = NULL_IF_CONFIG_SMALL("G.723.1"), }; Naufal

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[soc]: r5773 - mms/mmst.c
by spyfeng 20 Apr '10

20 Apr '10

Author: spyfeng Date: Tue Apr 20 17:50:20 2010 New Revision: 5773 Log: check stream numbers when parsering the header. Modified: mms/mmst.c Modified: mms/mmst.c ============================================================================== --- mms/mmst.c Tue Apr 20 17:39:15 2010 (r5772) +++ mms/mmst.c Tue Apr 20 17:50:20 2010 (r5773) @@ -417,7 +417,8 @@ static int asf_header_parser(MMSContext } else if (!memcmp(p, ff_asf_stream_header, sizeof(ff_asf_guid))) { flags = AV_RL16(p + sizeof(ff_asf_guid)*3 + 24); stream_id = flags & 0x7F; - if (mms->stream_num < MAX_STREAMS) { + if (mms->stream_num < MAX_STREAMS && + 46 + mms->stream_num * 6 < sizeof(mms->out_buffer)) { mms->streams[mms->stream_num].id = stream_id; mms->stream_num++; } else @@ -439,11 +440,6 @@ static int send_stream_selection_request // send the streams we want back... start_command_packet(mms, CS_PKT_STREAM_ID_REQUEST); bytestream_put_le32(&mms->write_out_ptr, mms->stream_num); // stream nums - if (mms->write_out_ptr - mms->out_buffer > - sizeof(mms->out_buffer) - 6 * mms->stream_num) { - dprintf("buffer will overflow for too many streams: %d.\n", mms->stream_num); - return AVERROR_IO; - } for(i= 0; i<mms->stream_num; i++) { bytestream_put_le16(&mms->write_out_ptr, 0xffff); // flags bytestream_put_le16(&mms->write_out_ptr, mms->streams[i].id); // stream id

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