From 002ca3e099f0d54cefab250baf027a7c006410d8 Mon Sep 17 00:00:00 2001 From: Michael Niedermayer Date: Mon, 24 Jun 2013 14:23:44 +0200 Subject: [PATCH 01/12] avio: Handle AVERROR_EOF in the same way as the return value 0 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This makes sure the ffurl_read_complete function actually returns the number of bytes read, as the documentation of the function says, even if the underlying protocol uses AVERROR_EOF instead of 0. Signed-off-by: Martin Storsjö (cherry picked from commit 5d876be87a115b93dd2e644049e3ada2cfb5ccb7) Signed-off-by: Luca Barbato --- libavformat/avio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavformat/avio.c b/libavformat/avio.c index a43b241399..ad39e6fdb0 100644 --- a/libavformat/avio.c +++ b/libavformat/avio.c @@ -238,7 +238,7 @@ static inline int retry_transfer_wrapper(URLContext *h, unsigned char *buf, int else av_usleep(1000); } else if (ret < 1) - return ret < 0 ? ret : len; + return (ret < 0 && ret != AVERROR_EOF) ? ret : len; if (ret) fast_retries = FFMAX(fast_retries, 2); len += ret; From a747cf8873ee8e4cb229b215fd3356f374488c12 Mon Sep 17 00:00:00 2001 From: Michael Niedermayer Date: Mon, 1 Jul 2013 23:38:08 +0200 Subject: [PATCH 02/12] rmdec: Use the AVIOContext given as parameter in rm_read_metadata() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This fixes crashes when playing back certain RealRTSP streams. When invoked from the RTP depacketizer, the full realmedia demuxer isn't invoked, but only certain functions from it, where a separate AVIOContext is passed in as parameter (for the buffer containing the data to parse). The functions called from within those entry points should only be using that parameter, not s->pb. In the depacketizer case, s is the RTSP context, where ->pb is null. Cc: libav-stable@libav.org Signed-off-by: Martin Storsjö (cherry picked from commit d35b6cd3775456a23b63e73316e244b671caa02f) Signed-off-by: Luca Barbato --- libavformat/rmdec.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c index 3cf2c97fe4..6495bdf17a 100644 --- a/libavformat/rmdec.c +++ b/libavformat/rmdec.c @@ -93,13 +93,13 @@ static int rm_read_extradata(AVIOContext *pb, AVCodecContext *avctx, unsigned si return 0; } -static void rm_read_metadata(AVFormatContext *s, int wide) +static void rm_read_metadata(AVFormatContext *s, AVIOContext *pb, int wide) { char buf[1024]; int i; for (i=0; ipb) : avio_r8(s->pb); - get_strl(s->pb, buf, sizeof(buf), len); + int len = wide ? avio_rb16(pb) : avio_r8(pb); + get_strl(pb, buf, sizeof(buf), len); av_dict_set(&s->metadata, ff_rm_metadata[i], buf, 0); } } @@ -129,7 +129,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb, int header_size = avio_rb16(pb); int64_t startpos = avio_tell(pb); avio_skip(pb, 14); - rm_read_metadata(s, 0); + rm_read_metadata(s, pb, 0); if ((startpos + header_size) >= avio_tell(pb) + 2) { // fourcc (should always be "lpcJ") avio_r8(pb); @@ -276,7 +276,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb, avio_r8(pb); avio_r8(pb); avio_r8(pb); - rm_read_metadata(s, 0); + rm_read_metadata(s, pb, 0); } } return 0; @@ -471,7 +471,7 @@ static int rm_read_header(AVFormatContext *s) flags = avio_rb16(pb); /* flags */ break; case MKTAG('C', 'O', 'N', 'T'): - rm_read_metadata(s, 1); + rm_read_metadata(s, pb, 1); break; case MKTAG('M', 'D', 'P', 'R'): st = avformat_new_stream(s, NULL); From fb1823e17807294ba0ab89b28f744ba73856009f Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Thu, 27 Jun 2013 03:19:05 +0200 Subject: [PATCH 03/12] vqavideo: check the version Prevent out of buffer write. Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org (cherry picked from commit c4abc9098cacb227dba39bac6aea16b2bceba0d0) Signed-off-by: Luca Barbato --- libavcodec/vqavideo.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/libavcodec/vqavideo.c b/libavcodec/vqavideo.c index 841210ce97..b4656b8891 100644 --- a/libavcodec/vqavideo.c +++ b/libavcodec/vqavideo.c @@ -135,6 +135,17 @@ static av_cold int vqa_decode_init(AVCodecContext *avctx) /* load up the VQA parameters from the header */ s->vqa_version = s->avctx->extradata[0]; + switch (s->vqa_version) { + case 1: + case 2: + break; + case 3: + av_log_missing_feature(avctx, "VQA Version 3", 0); + return AVERROR_PATCHWELCOME; + default: + av_log_missing_feature(avctx, "VQA Version", 1); + return AVERROR_PATCHWELCOME; + } s->width = AV_RL16(&s->avctx->extradata[6]); s->height = AV_RL16(&s->avctx->extradata[8]); if(av_image_check_size(s->width, s->height, 0, avctx)){ From e7800543fe804dde114e4d86d7ab053a5b5516da Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Thu, 27 Jun 2013 04:30:20 +0200 Subject: [PATCH 04/12] westwood_vqa: do not free extradata on error in read_header The extradata is already freed by avformat_open_input on failure. Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org (cherry picked from commit 76f5dfbfd902178df4a38221a68dc8540189345a) Signed-off-by: Luca Barbato --- libavformat/westwood_vqa.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/libavformat/westwood_vqa.c b/libavformat/westwood_vqa.c index 7c9cb0db81..77da375e70 100644 --- a/libavformat/westwood_vqa.c +++ b/libavformat/westwood_vqa.c @@ -106,7 +106,6 @@ static int wsvqa_read_header(AVFormatContext *s) header = (unsigned char *)st->codec->extradata; if (avio_read(pb, st->codec->extradata, VQA_HEADER_SIZE) != VQA_HEADER_SIZE) { - av_free(st->codec->extradata); return AVERROR(EIO); } st->codec->width = AV_RL16(&header[6]); @@ -132,7 +131,6 @@ static int wsvqa_read_header(AVFormatContext *s) * FINF has been skipped and the file will be ready to be demuxed */ do { if (avio_read(pb, scratch, VQA_PREAMBLE_SIZE) != VQA_PREAMBLE_SIZE) { - av_free(st->codec->extradata); return AVERROR(EIO); } chunk_tag = AV_RB32(&scratch[0]); From 86eec54c943981d7fade67bbde64cb097acea819 Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Thu, 27 Jun 2013 02:20:59 +0200 Subject: [PATCH 05/12] qdm2: use init_static_data (cherry picked from commit f054e309c58894450a5d18cce9799ef58aab9f14) Signed-off-by: Luca Barbato --- libavcodec/qdm2.c | 244 +++++++++++++++++++++++++--------------------- 1 file changed, 132 insertions(+), 112 deletions(-) diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 17729d18c9..767368cf91 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -221,116 +221,145 @@ static const uint16_t qdm2_vlc_offs[] = { static av_cold void qdm2_init_vlc(void) { - static int vlcs_initialized = 0; static VLC_TYPE qdm2_table[3838][2]; - if (!vlcs_initialized) { + vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; + vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; + init_vlc(&vlc_tab_level, 8, 24, + vlc_tab_level_huffbits, 1, 1, + vlc_tab_level_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; - vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; - init_vlc (&vlc_tab_level, 8, 24, - vlc_tab_level_huffbits, 1, 1, - vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; + vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; + init_vlc(&vlc_tab_diff, 8, 37, + vlc_tab_diff_huffbits, 1, 1, + vlc_tab_diff_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; - vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; - init_vlc (&vlc_tab_diff, 8, 37, - vlc_tab_diff_huffbits, 1, 1, - vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; + vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; + init_vlc(&vlc_tab_run, 5, 6, + vlc_tab_run_huffbits, 1, 1, + vlc_tab_run_huffcodes, 1, 1, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; - vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; - init_vlc (&vlc_tab_run, 5, 6, - vlc_tab_run_huffbits, 1, 1, - vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; + fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - + qdm2_vlc_offs[3]; + init_vlc(&fft_level_exp_alt_vlc, 8, 28, + fft_level_exp_alt_huffbits, 1, 1, + fft_level_exp_alt_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; - fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; - init_vlc (&fft_level_exp_alt_vlc, 8, 28, - fft_level_exp_alt_huffbits, 1, 1, - fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; + fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; + init_vlc(&fft_level_exp_vlc, 8, 20, + fft_level_exp_huffbits, 1, 1, + fft_level_exp_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; + fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - + qdm2_vlc_offs[5]; + init_vlc(&fft_stereo_exp_vlc, 6, 7, + fft_stereo_exp_huffbits, 1, 1, + fft_stereo_exp_huffcodes, 1, 1, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; - fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; - init_vlc (&fft_level_exp_vlc, 8, 20, - fft_level_exp_huffbits, 1, 1, - fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; + fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - + qdm2_vlc_offs[6]; + init_vlc(&fft_stereo_phase_vlc, 6, 9, + fft_stereo_phase_huffbits, 1, 1, + fft_stereo_phase_huffcodes, 1, 1, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; - fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; - init_vlc (&fft_stereo_exp_vlc, 6, 7, - fft_stereo_exp_huffbits, 1, 1, - fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_tone_level_idx_hi1.table = + &qdm2_table[qdm2_vlc_offs[7]]; + vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - + qdm2_vlc_offs[7]; + init_vlc(&vlc_tab_tone_level_idx_hi1, 8, 20, + vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, + vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; - fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; - init_vlc (&fft_stereo_phase_vlc, 6, 9, - fft_stereo_phase_huffbits, 1, 1, - fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_tone_level_idx_mid.table = + &qdm2_table[qdm2_vlc_offs[8]]; + vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - + qdm2_vlc_offs[8]; + init_vlc(&vlc_tab_tone_level_idx_mid, 8, 24, + vlc_tab_tone_level_idx_mid_huffbits, 1, 1, + vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; - vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; - init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, - vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, - vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_tone_level_idx_hi2.table = + &qdm2_table[qdm2_vlc_offs[9]]; + vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - + qdm2_vlc_offs[9]; + init_vlc(&vlc_tab_tone_level_idx_hi2, 8, 24, + vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, + vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; - vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; - init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, - vlc_tab_tone_level_idx_mid_huffbits, 1, 1, - vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; + vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; + init_vlc(&vlc_tab_type30, 6, 9, + vlc_tab_type30_huffbits, 1, 1, + vlc_tab_type30_huffcodes, 1, 1, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; - vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; - init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, - vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, - vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; + vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; + init_vlc(&vlc_tab_type34, 5, 10, + vlc_tab_type34_huffbits, 1, 1, + vlc_tab_type34_huffcodes, 1, 1, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; - vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; - init_vlc (&vlc_tab_type30, 6, 9, - vlc_tab_type30_huffbits, 1, 1, - vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_fft_tone_offset[0].table = + &qdm2_table[qdm2_vlc_offs[12]]; + vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - + qdm2_vlc_offs[12]; + init_vlc(&vlc_tab_fft_tone_offset[0], 8, 23, + vlc_tab_fft_tone_offset_0_huffbits, 1, 1, + vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; - vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; - init_vlc (&vlc_tab_type34, 5, 10, - vlc_tab_type34_huffbits, 1, 1, - vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_fft_tone_offset[1].table = + &qdm2_table[qdm2_vlc_offs[13]]; + vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - + qdm2_vlc_offs[13]; + init_vlc(&vlc_tab_fft_tone_offset[1], 8, 28, + vlc_tab_fft_tone_offset_1_huffbits, 1, 1, + vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; - vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; - init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, - vlc_tab_fft_tone_offset_0_huffbits, 1, 1, - vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_fft_tone_offset[2].table = + &qdm2_table[qdm2_vlc_offs[14]]; + vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - + qdm2_vlc_offs[14]; + init_vlc(&vlc_tab_fft_tone_offset[2], 8, 32, + vlc_tab_fft_tone_offset_2_huffbits, 1, 1, + vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; - vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; - init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, - vlc_tab_fft_tone_offset_1_huffbits, 1, 1, - vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); + vlc_tab_fft_tone_offset[3].table = + &qdm2_table[qdm2_vlc_offs[15]]; + vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - + qdm2_vlc_offs[15]; + init_vlc(&vlc_tab_fft_tone_offset[3], 8, 35, + vlc_tab_fft_tone_offset_3_huffbits, 1, 1, + vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; - vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; - init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, - vlc_tab_fft_tone_offset_2_huffbits, 1, 1, - vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; - vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; - init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, - vlc_tab_fft_tone_offset_3_huffbits, 1, 1, - vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; - vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; - init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, - vlc_tab_fft_tone_offset_4_huffbits, 1, 1, - vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - vlcs_initialized=1; - } + vlc_tab_fft_tone_offset[4].table = + &qdm2_table[qdm2_vlc_offs[16]]; + vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - + qdm2_vlc_offs[16]; + init_vlc(&vlc_tab_fft_tone_offset[4], 8, 38, + vlc_tab_fft_tone_offset_4_huffbits, 1, 1, + vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, + INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); } static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) @@ -1652,20 +1681,12 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) * * @param q context */ -static av_cold void qdm2_init(QDM2Context *q) { - static int initialized = 0; - - if (initialized != 0) - return; - initialized = 1; - +static av_cold void qdm2_init_static_data(AVCodec *codec) { qdm2_init_vlc(); ff_mpa_synth_init_float(ff_mpa_synth_window_float); softclip_table_init(); rnd_table_init(); init_noise_samples(); - - av_log(NULL, AV_LOG_DEBUG, "init done\n"); } @@ -1846,8 +1867,6 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); ff_mpadsp_init(&s->mpadsp); - qdm2_init(s); - avctx->sample_fmt = AV_SAMPLE_FMT_S16; avcodec_get_frame_defaults(&s->frame); @@ -1963,13 +1982,14 @@ static int qdm2_decode_frame(AVCodecContext *avctx, void *data, AVCodec ff_qdm2_decoder = { - .name = "qdm2", - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_QDM2, - .priv_data_size = sizeof(QDM2Context), - .init = qdm2_decode_init, - .close = qdm2_decode_close, - .decode = qdm2_decode_frame, - .capabilities = CODEC_CAP_DR1, - .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), + .name = "qdm2", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_QDM2, + .priv_data_size = sizeof(QDM2Context), + .init = qdm2_decode_init, + .init_static_data = qdm2_init_static_data, + .close = qdm2_decode_close, + .decode = qdm2_decode_frame, + .capabilities = CODEC_CAP_DR1, + .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), }; From 998a0389d307109af43e452441050281a09d540c Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Thu, 27 Jun 2013 02:49:15 +0200 Subject: [PATCH 06/12] qdm2: formatting cosmetics Apply the usual style plus drop few unnecessary return at the end of void functions. (cherry picked from commit 76efedeadb1f6bf79020c44a71dd0cee13d932ad) Signed-off-by: Luca Barbato --- libavcodec/qdm2.c | 391 ++++++++++++++++++++++++---------------------- 1 file changed, 204 insertions(+), 187 deletions(-) diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 767368cf91..297ca65bc1 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -219,6 +219,10 @@ static const uint16_t qdm2_vlc_offs[] = { 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, }; +static const int switchtable[23] = { + 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4 +}; + static av_cold void qdm2_init_vlc(void) { static VLC_TYPE qdm2_table[3838][2]; @@ -362,7 +366,7 @@ static av_cold void qdm2_init_vlc(void) INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); } -static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) +static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) { int value; @@ -370,29 +374,27 @@ static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) /* stage-2, 3 bits exponent escape sequence */ if (value-- == 0) - value = get_bits (gb, get_bits (gb, 3) + 1); + value = get_bits(gb, get_bits(gb, 3) + 1); /* stage-3, optional */ if (flag) { int tmp = vlc_stage3_values[value]; if ((value & ~3) > 0) - tmp += get_bits (gb, (value >> 2)); + tmp += get_bits(gb, (value >> 2)); value = tmp; } return value; } - -static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) +static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth) { - int value = qdm2_get_vlc (gb, vlc, 0, depth); + int value = qdm2_get_vlc(gb, vlc, 0, depth); return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); } - /** * QDM2 checksum * @@ -402,49 +404,50 @@ static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) * * @return 0 if checksum is OK */ -static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { +static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value) +{ int i; - for (i=0; i < length; i++) + for (i = 0; i < length; i++) value -= data[i]; return (uint16_t)(value & 0xffff); } - /** * Fill a QDM2SubPacket structure with packet type, size, and data pointer. * * @param gb bitreader context * @param sub_packet packet under analysis */ -static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) +static void qdm2_decode_sub_packet_header(GetBitContext *gb, + QDM2SubPacket *sub_packet) { - sub_packet->type = get_bits (gb, 8); + sub_packet->type = get_bits(gb, 8); if (sub_packet->type == 0) { sub_packet->size = 0; sub_packet->data = NULL; } else { - sub_packet->size = get_bits (gb, 8); + sub_packet->size = get_bits(gb, 8); - if (sub_packet->type & 0x80) { - sub_packet->size <<= 8; - sub_packet->size |= get_bits (gb, 8); - sub_packet->type &= 0x7f; - } + if (sub_packet->type & 0x80) { + sub_packet->size <<= 8; + sub_packet->size |= get_bits(gb, 8); + sub_packet->type &= 0x7f; + } - if (sub_packet->type == 0x7f) - sub_packet->type |= (get_bits (gb, 8) << 8); + if (sub_packet->type == 0x7f) + sub_packet->type |= (get_bits(gb, 8) << 8); - sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data + // FIXME: this depends on bitreader-internal data + sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; } - av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", - sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); + av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n", + sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); } - /** * Return node pointer to first packet of requested type in list. * @@ -452,7 +455,8 @@ static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub * @param type type of searched subpacket * @return node pointer for subpacket if found, else NULL */ -static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) +static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, + int type) { while (list != NULL && list->packet != NULL) { if (list->packet->type == type) @@ -462,14 +466,13 @@ static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int return NULL; } - /** * Replace 8 elements with their average value. * Called by qdm2_decode_superblock before starting subblock decoding. * * @param q context */ -static void average_quantized_coeffs (QDM2Context *q) +static void average_quantized_coeffs(QDM2Context *q) { int i, j, n, ch, sum; @@ -486,12 +489,11 @@ static void average_quantized_coeffs (QDM2Context *q) if (sum > 0) sum--; - for (j=0; j < 8; j++) + for (j = 0; j < 8; j++) q->quantized_coeffs[ch][i][j] = sum; } } - /** * Build subband samples with noise weighted by q->tone_level. * Called by synthfilt_build_sb_samples. @@ -499,7 +501,7 @@ static void average_quantized_coeffs (QDM2Context *q) * @param q context * @param sb subband index */ -static void build_sb_samples_from_noise (QDM2Context *q, int sb) +static void build_sb_samples_from_noise(QDM2Context *q, int sb) { int ch, j; @@ -508,14 +510,16 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) if (!q->nb_channels) return; - for (ch = 0; ch < q->nb_channels; ch++) + for (ch = 0; ch < q->nb_channels; ch++) { for (j = 0; j < 64; j++) { - q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; - q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; + q->sb_samples[ch][j * 2][sb] = + SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; + q->sb_samples[ch][j * 2 + 1][sb] = + SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; } + } } - /** * Called while processing data from subpackets 11 and 12. * Used after making changes to coding_method array. @@ -524,44 +528,62 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) * @param channels number of channels * @param coding_method q->coding_method[0][0][0] */ -static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) +static void fix_coding_method_array(int sb, int channels, + sb_int8_array coding_method) { - int j,k; + int j, k; int ch; int run, case_val; - static const int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4}; for (ch = 0; ch < channels; ch++) { for (j = 0; j < 64; ) { - if((coding_method[ch][sb][j] - 8) > 22) { - run = 1; + if ((coding_method[ch][sb][j] - 8) > 22) { + run = 1; case_val = 8; } else { - switch (switchtable[coding_method[ch][sb][j]-8]) { - case 0: run = 10; case_val = 10; break; - case 1: run = 1; case_val = 16; break; - case 2: run = 5; case_val = 24; break; - case 3: run = 3; case_val = 30; break; - case 4: run = 1; case_val = 30; break; - case 5: run = 1; case_val = 8; break; - default: run = 1; case_val = 8; break; + switch (switchtable[coding_method[ch][sb][j] - 8]) { + case 0: run = 10; + case_val = 10; + break; + case 1: run = 1; + case_val = 16; + break; + case 2: run = 5; + case_val = 24; + break; + case 3: run = 3; + case_val = 30; + break; + case 4: run = 1; + case_val = 30; + break; + case 5: run = 1; + case_val = 8; + break; + default: run = 1; + case_val = 8; + break; } } - for (k = 0; k < run; k++) - if (j + k < 128) - if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) + for (k = 0; k < run; k++) { + if (j + k < 128) { + if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) { if (k > 0) { - SAMPLES_NEEDED + SAMPLES_NEEDED //not debugged, almost never used - memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); - memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); + memset(&coding_method[ch][sb][j + k], case_val, + k *sizeof(int8_t)); + memset(&coding_method[ch][sb][j + k], case_val, + 3 * sizeof(int8_t)); } + } + } + } j += run; } } } - /** * Related to synthesis filter * Called by process_subpacket_10 @@ -569,7 +591,7 @@ static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_ * @param q context * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 */ -static void fill_tone_level_array (QDM2Context *q, int flag) +static void fill_tone_level_array(QDM2Context *q, int flag) { int i, sb, ch, sb_used; int tmp, tab; @@ -641,16 +663,14 @@ static void fill_tone_level_array (QDM2Context *q, int flag) } } } - - return; } - /** * Related to synthesis filter * Called by process_subpacket_11 * c is built with data from subpacket 11 - * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples + * Most of this function is used only if superblock_type_2_3 == 0, + * never seen it in samples. * * @param tone_level_idx * @param tone_level_idx_temp @@ -660,9 +680,12 @@ static void fill_tone_level_array (QDM2Context *q, int flag) * @param superblocktype_2_3 flag based on superblock packet type * @param cm_table_select q->cm_table_select */ -static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, - sb_int8_array coding_method, int nb_channels, - int c, int superblocktype_2_3, int cm_table_select) +static void fill_coding_method_array(sb_int8_array tone_level_idx, + sb_int8_array tone_level_idx_temp, + sb_int8_array coding_method, + int nb_channels, + int c, int superblocktype_2_3, + int cm_table_select) { int ch, sb, j; int tmp, acc, esp_40, comp; @@ -768,15 +791,14 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra for (j = 0; j < 64; j++) coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; } - - return; } - /** * - * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 - * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used + * Called by process_subpacket_11 to process more data from subpacket 11 + * with sb 0-8. + * Called by process_subpacket_12 to process data from subpacket 12 with + * sb 8-sb_used. * * @param q context * @param gb bitreader context @@ -784,7 +806,8 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra * @param sb_min lower subband processed (sb_min included) * @param sb_max higher subband processed (sb_max excluded) */ -static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) +static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, + int length, int sb_min, int sb_max) { int sb, j, k, n, ch, run, channels; int joined_stereo, zero_encoding, chs; @@ -964,16 +987,18 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l } // subband loop } - /** - * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). - * This is similar to process_subpacket_9, but for a single channel and for element [0] + * Init the first element of a channel in quantized_coeffs with data + * from packet 10 (quantized_coeffs[ch][0]). + * This is similar to process_subpacket_9, but for a single channel + * and for element [0] * same VLC tables as process_subpacket_9 are used. * * @param quantized_coeffs pointer to quantized_coeffs[ch][0] * @param gb bitreader context */ -static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb) +static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs, + GetBitContext *gb) { int i, k, run, level, diff; @@ -1000,11 +1025,11 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext } } - /** * Related to synthesis filter, process data from packet 10 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 - * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 + * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with + * data from packet 10 * * @param q context * @param gb bitreader context @@ -1072,29 +1097,29 @@ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb) * @param q context * @param node pointer to node with packet */ -static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) +static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) { GetBitContext gb; int i, j, k, n, ch, run, level, diff; - init_get_bits(&gb, node->packet->data, node->packet->size*8); + init_get_bits(&gb, node->packet->data, node->packet->size * 8); - n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function + n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; for (i = 1; i < n; i++) - for (ch=0; ch < q->nb_channels; ch++) { + for (ch = 0; ch < q->nb_channels; ch++) { level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); q->quantized_coeffs[ch][i][0] = level; for (j = 0; j < (8 - 1); ) { - run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; + run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); for (k = 1; k <= run; k++) - q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); + q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); level += diff; - j += run; + j += run; } } @@ -1103,14 +1128,13 @@ static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) q->quantized_coeffs[ch][0][i] = 0; } - /** * Process subpacket 10 if not null, else * * @param q context * @param node pointer to node with packet */ -static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node) +static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node) { GetBitContext gb; @@ -1123,14 +1147,13 @@ static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node) } } - /** * Process subpacket 11 * * @param q context * @param node pointer to node with packet */ -static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node) +static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node) { GetBitContext gb; int length = 0; @@ -1141,24 +1164,25 @@ static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node) } if (length >= 32) { - int c = get_bits (&gb, 13); + int c = get_bits(&gb, 13); if (c > 3) - fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, - q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); + fill_coding_method_array(q->tone_level_idx, + q->tone_level_idx_temp, q->coding_method, + q->nb_channels, 8 * c, + q->superblocktype_2_3, q->cm_table_select); } synthfilt_build_sb_samples(q, &gb, length, 0, 8); } - /** * Process subpacket 12 * * @param q context * @param node pointer to node with packet */ -static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node) +static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node) { GetBitContext gb; int length = 0; @@ -1177,7 +1201,7 @@ static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node) * @param q context * @param list list with synthesis filter packets (list D) */ -static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) +static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) { QDM2SubPNode *nodes[4]; @@ -1204,13 +1228,12 @@ static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) process_subpacket_12(q, NULL); } - /* * Decode superblock, fill packet lists. * * @param q context */ -static void qdm2_decode_super_block (QDM2Context *q) +static void qdm2_decode_super_block(QDM2Context *q) { GetBitContext gb; QDM2SubPacket header, *packet; @@ -1222,33 +1245,33 @@ static void qdm2_decode_super_block (QDM2Context *q) memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); q->sub_packets_B = 0; - sub_packets_D = 0; + sub_packets_D = 0; average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] - init_get_bits(&gb, q->compressed_data, q->compressed_size*8); + init_get_bits(&gb, q->compressed_data, q->compressed_size * 8); qdm2_decode_sub_packet_header(&gb, &header); if (header.type < 2 || header.type >= 8) { q->has_errors = 1; - av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); + av_log(NULL, AV_LOG_ERROR, "bad superblock type\n"); return; } q->superblocktype_2_3 = (header.type == 2 || header.type == 3); - packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); + packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); - init_get_bits(&gb, header.data, header.size*8); + init_get_bits(&gb, header.data, header.size * 8); if (header.type == 2 || header.type == 4 || header.type == 5) { - int csum = 257 * get_bits(&gb, 8); - csum += 2 * get_bits(&gb, 8); + int csum = 257 * get_bits(&gb, 8); + csum += 2 * get_bits(&gb, 8); csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); if (csum != 0) { q->has_errors = 1; - av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); + av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n"); return; } } @@ -1274,8 +1297,8 @@ static void qdm2_decode_super_block (QDM2Context *q) q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; /* seek to next block */ - init_get_bits(&gb, header.data, header.size*8); - skip_bits(&gb, next_index*8); + init_get_bits(&gb, header.data, header.size * 8); + skip_bits(&gb, next_index * 8); if (next_index >= header.size) break; @@ -1284,7 +1307,7 @@ static void qdm2_decode_super_block (QDM2Context *q) /* decode subpacket */ packet = &q->sub_packets[i]; qdm2_decode_sub_packet_header(&gb, packet); - next_index = packet->size + get_bits_count(&gb) / 8; + next_index = packet->size + get_bits_count(&gb) / 8; sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; if (packet->type == 0) @@ -1317,13 +1340,13 @@ static void qdm2_decode_super_block (QDM2Context *q) } else if (packet->type == 15) { SAMPLES_NEEDED_2("packet type 15") return; - } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { + } else if (packet->type >= 16 && packet->type < 48 && + !fft_subpackets[packet->type - 16]) { /* packets for FFT */ QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); } } // Packet bytes loop -/* **************************************************************** */ if (q->sub_packet_list_D[0].packet != NULL) { process_synthesis_subpackets(q, q->sub_packet_list_D); q->do_synth_filter = 1; @@ -1332,49 +1355,48 @@ static void qdm2_decode_super_block (QDM2Context *q) process_subpacket_11(q, NULL); process_subpacket_12(q, NULL); } -/* **************************************************************** */ } - -static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, - int offset, int duration, int channel, - int exp, int phase) +static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, + int offset, int duration, int channel, + int exp, int phase) { if (q->fft_coefs_min_index[duration] < 0) q->fft_coefs_min_index[duration] = q->fft_coefs_index; - q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); + q->fft_coefs[q->fft_coefs_index].sub_packet = + ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); q->fft_coefs[q->fft_coefs_index].channel = channel; - q->fft_coefs[q->fft_coefs_index].offset = offset; - q->fft_coefs[q->fft_coefs_index].exp = exp; - q->fft_coefs[q->fft_coefs_index].phase = phase; + q->fft_coefs[q->fft_coefs_index].offset = offset; + q->fft_coefs[q->fft_coefs_index].exp = exp; + q->fft_coefs[q->fft_coefs_index].phase = phase; q->fft_coefs_index++; } - -static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) +static void qdm2_fft_decode_tones(QDM2Context *q, int duration, + GetBitContext *gb, int b) { int channel, stereo, phase, exp; - int local_int_4, local_int_8, stereo_phase, local_int_10; + int local_int_4, local_int_8, stereo_phase, local_int_10; int local_int_14, stereo_exp, local_int_20, local_int_28; int n, offset; - local_int_4 = 0; + local_int_4 = 0; local_int_28 = 0; local_int_20 = 2; - local_int_8 = (4 - duration); + local_int_8 = (4 - duration); local_int_10 = 1 << (q->group_order - duration - 1); - offset = 1; + offset = 1; while (1) { if (q->superblocktype_2_3) { while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { offset = 1; if (n == 0) { - local_int_4 += local_int_10; + local_int_4 += local_int_10; local_int_28 += (1 << local_int_8); } else { - local_int_4 += 8*local_int_10; + local_int_4 += 8 * local_int_10; local_int_28 += (8 << local_int_8); } } @@ -1382,7 +1404,7 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext * } else { offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); while (offset >= (local_int_10 - 1)) { - offset += (1 - (local_int_10 - 1)); + offset += (1 - (local_int_10 - 1)); local_int_4 += local_int_10; local_int_28 += (1 << local_int_8); } @@ -1397,22 +1419,22 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext * if (q->nb_channels > 1) { channel = get_bits1(gb); - stereo = get_bits1(gb); + stereo = get_bits1(gb); } else { channel = 0; - stereo = 0; + stereo = 0; } - exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); + exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; - exp = (exp < 0) ? 0 : exp; + exp = (exp < 0) ? 0 : exp; - phase = get_bits(gb, 3); - stereo_exp = 0; + phase = get_bits(gb, 3); + stereo_exp = 0; stereo_phase = 0; if (stereo) { - stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); + stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); if (stereo_phase < 0) stereo_phase += 8; @@ -1421,17 +1443,18 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext * if (q->frequency_range > (local_int_14 + 1)) { int sub_packet = (local_int_20 + local_int_28); - qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); + qdm2_fft_init_coefficient(q, sub_packet, offset, duration, + channel, exp, phase); if (stereo) - qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); + qdm2_fft_init_coefficient(q, sub_packet, offset, duration, + 1 - channel, + stereo_exp, stereo_phase); } - offset++; } } - -static void qdm2_decode_fft_packets (QDM2Context *q) +static void qdm2_decode_fft_packets(QDM2Context *q) { int i, j, min, max, value, type, unknown_flag; GetBitContext gb; @@ -1441,18 +1464,18 @@ static void qdm2_decode_fft_packets (QDM2Context *q) /* reset minimum indexes for FFT coefficients */ q->fft_coefs_index = 0; - for (i=0; i < 5; i++) + for (i = 0; i < 5; i++) q->fft_coefs_min_index[i] = -1; /* process subpackets ordered by type, largest type first */ for (i = 0, max = 256; i < q->sub_packets_B; i++) { - QDM2SubPacket *packet= NULL; + QDM2SubPacket *packet = NULL; /* find subpacket with largest type less than max */ for (j = 0, min = 0; j < q->sub_packets_B; j++) { value = q->sub_packet_list_B[j].packet->type; if (value > min && value < max) { - min = value; + min = value; packet = q->sub_packet_list_B[j].packet; } } @@ -1463,11 +1486,13 @@ static void qdm2_decode_fft_packets (QDM2Context *q) if (!packet) return; - if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) + if (i == 0 && + (packet->type < 16 || packet->type >= 48 || + fft_subpackets[packet->type - 16])) return; /* decode FFT tones */ - init_get_bits (&gb, packet->data, packet->size*8); + init_get_bits(&gb, packet->data, packet->size * 8); if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) unknown_flag = 1; @@ -1482,13 +1507,13 @@ static void qdm2_decode_fft_packets (QDM2Context *q) if (duration >= 0 && duration < 4) qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); } else if (type == 31) { - for (j=0; j < 4; j++) + for (j = 0; j < 4; j++) qdm2_fft_decode_tones(q, j, &gb, unknown_flag); } else if (type == 46) { - for (j=0; j < 6; j++) + for (j = 0; j < 6; j++) q->fft_level_exp[j] = get_bits(&gb, 6); - for (j=0; j < 4; j++) - qdm2_fft_decode_tones(q, j, &gb, unknown_flag); + for (j = 0; j < 4; j++) + qdm2_fft_decode_tones(q, j, &gb, unknown_flag); } } // Loop on B packets @@ -1503,20 +1528,19 @@ static void qdm2_decode_fft_packets (QDM2Context *q) q->fft_coefs_max_index[j] = q->fft_coefs_index; } - -static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) +static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone) { - float level, f[6]; - int i; - QDM2Complex c; - const double iscale = 2.0*M_PI / 512.0; + float level, f[6]; + int i; + QDM2Complex c; + const double iscale = 2.0 * M_PI / 512.0; tone->phase += tone->phase_shift; /* calculate current level (maximum amplitude) of tone */ level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; - c.im = level * sin(tone->phase*iscale); - c.re = level * cos(tone->phase*iscale); + c.im = level * sin(tone->phase * iscale); + c.re = level * cos(tone->phase * iscale); /* generate FFT coefficients for tone */ if (tone->duration >= 3 || tone->cutoff >= 3) { @@ -1526,30 +1550,31 @@ static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) tone->complex[1].re -= c.re; } else { f[1] = -tone->table[4]; - f[0] = tone->table[3] - tone->table[0]; - f[2] = 1.0 - tone->table[2] - tone->table[3]; - f[3] = tone->table[1] + tone->table[4] - 1.0; - f[4] = tone->table[0] - tone->table[1]; - f[5] = tone->table[2]; + f[0] = tone->table[3] - tone->table[0]; + f[2] = 1.0 - tone->table[2] - tone->table[3]; + f[3] = tone->table[1] + tone->table[4] - 1.0; + f[4] = tone->table[0] - tone->table[1]; + f[5] = tone->table[2]; for (i = 0; i < 2; i++) { - tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; - tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); + tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += + c.re * f[i]; + tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += + c.im * ((tone->cutoff <= i) ? -f[i] : f[i]); } for (i = 0; i < 4; i++) { - tone->complex[i].re += c.re * f[i+2]; - tone->complex[i].im += c.im * f[i+2]; + tone->complex[i].re += c.re * f[i + 2]; + tone->complex[i].im += c.im * f[i + 2]; } } /* copy the tone if it has not yet died out */ if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { - memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); - q->fft_tone_end = (q->fft_tone_end + 1) % 1000; + memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); + q->fft_tone_end = (q->fft_tone_end + 1) % 1000; } } - -static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) +static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet) { int i, j, ch; const double iscale = 0.25 * M_PI; @@ -1620,29 +1645,27 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) } } - -static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) +static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet) { const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; - float *out = q->output_buffer + channel; + float *out = q->output_buffer + channel; int i; q->fft.complex[channel][0].re *= 2.0f; - q->fft.complex[channel][0].im = 0.0f; + q->fft.complex[channel][0].im = 0.0f; q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); /* add samples to output buffer */ for (i = 0; i < FFALIGN(q->fft_size, 8); i++) { out[0] += q->fft.complex[channel][i].re * gain; out[q->channels] += q->fft.complex[channel][i].im * gain; - out += 2 * q->channels; + out += 2 * q->channels; } } - /** * @param q context * @param index subpacket number */ -static void qdm2_synthesis_filter (QDM2Context *q, int index) +static void qdm2_synthesis_filter(QDM2Context *q, int index) { int i, k, ch, sb_used, sub_sampling, dither_state = 0; @@ -1651,7 +1674,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) for (ch = 0; ch < q->channels; ch++) for (i = 0; i < 8; i++) - for (k=sb_used; k < SBLIMIT; k++) + for (k = sb_used; k < SBLIMIT; k++) q->sb_samples[ch][(8 * index) + i][k] = 0; for (ch = 0; ch < q->nb_channels; ch++) { @@ -1659,10 +1682,10 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) for (i = 0; i < 8; i++) { ff_mpa_synth_filter_float(&q->mpadsp, - q->synth_buf[ch], &(q->synth_buf_offset[ch]), - ff_mpa_synth_window_float, &dither_state, - samples_ptr, q->nb_channels, - q->sb_samples[ch][(8 * index) + i]); + q->synth_buf[ch], &(q->synth_buf_offset[ch]), + ff_mpa_synth_window_float, &dither_state, + samples_ptr, q->nb_channels, + q->sb_samples[ch][(8 * index) + i]); samples_ptr += 32 * q->nb_channels; } } @@ -1675,7 +1698,6 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; } - /** * Init static data (does not depend on specific file) * @@ -1689,7 +1711,6 @@ static av_cold void qdm2_init_static_data(AVCodec *codec) { init_noise_samples(); } - /** * Init parameters from codec extradata */ @@ -1739,7 +1760,7 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) return -1; } - extradata = avctx->extradata; + extradata = avctx->extradata; extradata_size = avctx->extradata_size; while (extradata_size > 7) { @@ -1875,7 +1896,6 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) return 0; } - static av_cold int qdm2_decode_close(AVCodecContext *avctx) { QDM2Context *s = avctx->priv_data; @@ -1885,8 +1905,7 @@ static av_cold int qdm2_decode_close(AVCodecContext *avctx) return 0; } - -static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) +static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out) { int ch, i; const int frame_size = (q->frame_size * q->channels); @@ -1945,7 +1964,6 @@ static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) return 0; } - static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { @@ -1980,8 +1998,7 @@ static int qdm2_decode_frame(AVCodecContext *avctx, void *data, return s->checksum_size; } -AVCodec ff_qdm2_decoder = -{ +AVCodec ff_qdm2_decoder = { .name = "qdm2", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_QDM2, From 64bcb5d35013a4a41969b124a3dcf9a21aef9346 Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Thu, 27 Jun 2013 02:50:52 +0200 Subject: [PATCH 07/12] qdm2: check and reset dithering index per channel Checking per subband would have the index exceed the dithering noise table size. Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org (cherry picked from commit 744a11c996641888d477a3981d609e79eeb69ea9) Signed-off-by: Luca Barbato --- libavcodec/qdm2.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 297ca65bc1..697f852c12 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -825,8 +825,6 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, } for (sb = sb_min; sb < sb_max; sb++) { - FIX_NOISE_IDX(q->noise_idx); - channels = q->nb_channels; if (q->nb_channels <= 1 || sb < 12) @@ -850,6 +848,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, } for (ch = 0; ch < channels; ch++) { + FIX_NOISE_IDX(q->noise_idx); zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; type34_predictor = 0.0; type34_first = 1; From 8f3fe7c696866c965ebc4444aaa5487bb1cd53f1 Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Tue, 9 Jul 2013 01:03:13 +0200 Subject: [PATCH 08/12] atrac3: fix error handling decode_tonal_components returns a proper AVERROR. Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org (cherry picked from commit 874c8a17ac9b04fb7ac23d003e54e3662dd23b4e) Signed-off-by: Luca Barbato --- libavcodec/atrac3.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c index 910c15e473..632cc37a0b 100644 --- a/libavcodec/atrac3.c +++ b/libavcodec/atrac3.c @@ -664,8 +664,8 @@ static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb, snd->num_components = decode_tonal_components(gb, snd->components, snd->bands_coded); - if (snd->num_components == -1) - return -1; + if (snd->num_components < 0) + return snd->num_components; num_subbands = decode_spectrum(gb, snd->spectrum); From 67a8a1c20242de9481ca491aa0bbbfad9c8b6565 Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Tue, 9 Jul 2013 04:20:23 +0200 Subject: [PATCH 09/12] atrac3: set the getbits context the right buffer_end Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org (cherry picked from commit 22e76ec635bafdd1d1ec35581a7ac09e69e3c43e) Signed-off-by: Luca Barbato --- libavcodec/atrac3.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c index 632cc37a0b..b2052f7aa9 100644 --- a/libavcodec/atrac3.c +++ b/libavcodec/atrac3.c @@ -742,7 +742,7 @@ static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf, /* set the bitstream reader at the start of the second Sound Unit*/ - init_get_bits(&q->gb, ptr1, avctx->block_align * 8); + init_get_bits(&q->gb, ptr1, (avctx->block_align - i) * 8); /* Fill the Weighting coeffs delay buffer */ memmove(q->weighting_delay, &q->weighting_delay[2], From aa99cb15f6e4032efc1ade53f832a8788bdc758b Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Tue, 9 Jul 2013 04:44:26 +0200 Subject: [PATCH 10/12] atrac3: Error on impossible encoding/channel combinations Joint stereo encoded mono is impossible. Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org (cherry picked from commit 50cf5a7fb78846fc39b3ecdaa896a10bcd74da2a) Signed-off-by: Luca Barbato --- libavcodec/atrac3.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c index b2052f7aa9..68ce2475b6 100644 --- a/libavcodec/atrac3.c +++ b/libavcodec/atrac3.c @@ -941,9 +941,11 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx) if (q->coding_mode == STEREO) av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n"); - else if (q->coding_mode == JOINT_STEREO) + else if (q->coding_mode == JOINT_STEREO) { + if (avctx->channels != 2) + return AVERROR_INVALIDDATA; av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n"); - else { + } else { av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n", q->coding_mode); return AVERROR_INVALIDDATA; From 6d2a92c4678616fe342be0b606343ae3ade83641 Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Tue, 9 Jul 2013 09:18:16 +0200 Subject: [PATCH 11/12] imc: Catch a division by zero Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org (cherry picked from commit bbf6a4aa20bfe3d7869b2218e66063602dfb8aa7) Signed-off-by: Luca Barbato --- libavcodec/imc.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/libavcodec/imc.c b/libavcodec/imc.c index 5cff70fcb4..316dd77462 100644 --- a/libavcodec/imc.c +++ b/libavcodec/imc.c @@ -450,6 +450,10 @@ static int bit_allocation(IMCContext *q, IMCChannel *chctx, iacc += chctx->bandWidthT[i]; summa += chctx->bandWidthT[i] * chctx->flcoeffs4[i]; } + + if (!iacc) + return AVERROR_INVALIDDATA; + chctx->bandWidthT[BANDS - 1] = 0; summa = (summa * 0.5 - freebits) / iacc; From c02d4c1a98aef485be25228b33adb4ce357173e4 Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Sun, 7 Jul 2013 12:56:12 +0200 Subject: [PATCH 12/12] adpcm: Write the correct number of samples for ima-dk4 Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org (cherry picked from commit 12576afe206d35231ccd61f9033c5fdab6a11e80) Signed-off-by: Luca Barbato --- libavcodec/adpcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c index 3b0eb8b9d0..df6b9d300f 100644 --- a/libavcodec/adpcm.c +++ b/libavcodec/adpcm.c @@ -773,7 +773,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data, return AVERROR_INVALIDDATA; } } - for (n = nb_samples >> (1 - st); n > 0; n--) { + for (n = (nb_samples >> (1 - st)) - 1; n > 0; n--) { int v = bytestream2_get_byteu(&gb); *samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v >> 4 , 3); *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);