| Message ID | hK2YvSmkl_TUdaAXbc9eMwThZo3KVpk1rulfTdH0P_jBHv4fX8E3A-TSGlQ1oJRDbuM8jPTCOuQnU35Yg6bk_o-NR9_w_QSudDh_qR5t7Os=@protonmail.com |
|---|---|
| State | New |
| Headers | show |
| Series | [FFmpeg-devel,1/2] avformat/aeadec, avcodec/atrac1: Fix 8 and 4-channel ATRAC1 support | expand |
| Context | Check | Description |
|---|---|---|
| yinshiyou/make_loongarch64 | success | Make finished |
| yinshiyou/make_fate_loongarch64 | fail | Make fate failed |
| andriy/make_x86 | success | Make finished |
| andriy/make_fate_x86 | success | Make fate finished |
asivery via ffmpeg-devel: > I'm sending the patch again, so that it is correctly rebased on the current master. > Here are the two sample files required by the FATE test: > https://0x0.st/Xaw2.aea/boxboy333_house_music_multitrack.aea > https://0x0.st/Xaw_.pcm/boxboy333_house_music_multitrack.pcm > Please let me know if I need to provide anything else. > Why such a gigantic file? And why actually a multi-channel file given that there are no special coding modes for multi-channel files? - Andreas
On Sat, 15 Jun 2024 I was asked for samples by Anton Khirnov - this is why I wrote the test. If the test is not necessary after all, that patch doesn't need to be applied. I can also shorten the test file, if it's required. On Wednesday, July 3rd, 2024 at 4:49 PM, Andreas Rheinhardt <andreas.rheinhardt@outlook.com> wrote: > asivery via ffmpeg-devel: > > > I'm sending the patch again, so that it is correctly rebased on the current master. > > Here are the two sample files required by the FATE test: > > https://0x0.st/Xaw2.aea/boxboy333_house_music_multitrack.aea > > https://0x0.st/Xaw_.pcm/boxboy333_house_music_multitrack.pcm > > Please let me know if I need to provide anything else. > > > Why such a gigantic file? And why actually a multi-channel file given > that there are no special coding modes for multi-channel files? > > - Andreas > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
Quoting Andreas Rheinhardt (2024-07-03 16:49:55) > asivery via ffmpeg-devel: > > I'm sending the patch again, so that it is correctly rebased on the current master. > > Here are the two sample files required by the FATE test: > > https://0x0.st/Xaw2.aea/boxboy333_house_music_multitrack.aea > > https://0x0.st/Xaw_.pcm/boxboy333_house_music_multitrack.pcm > > Please let me know if I need to provide anything else. > > > > Why such a gigantic file? And why actually a multi-channel file given > that there are no special coding modes for multi-channel files? In what world is this gigantic? I find your obsession with FATE sample sizes to be very counterproductive. Storage is cheap, developer time is not.
From 22b988b046c1a1eb02033416710a3f3244f8859a Mon Sep 17 00:00:00 2001 From: asivery <asivery@protonmail.com> Date: Wed, 3 Jul 2024 16:24:41 +0200 Subject: [PATCH 1/2] avformat/aeadec, avcodec/atrac1: Fix 8 and 4-channel ATRAC1 support Signed-off-by: asivery <asivery@protonmail.com> --- libavcodec/atrac1.c | 12 ++++++------ libavformat/aeadec.c | 2 +- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c index cdcc7a669e..448bdfc7e9 100644 --- a/libavcodec/atrac1.c +++ b/libavcodec/atrac1.c @@ -49,7 +49,7 @@ #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit #define AT1_FRAME_SIZE AT1_SU_SIZE * 2 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 -#define AT1_MAX_CHANNELS 2 +#define AT1_MAX_CHANNELS 8 #define AT1_QMF_BANDS 3 #define IDX_LOW_BAND 0 @@ -339,7 +339,7 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx) AVFloatDSPContext *fdsp; int channels = avctx->ch_layout.nb_channels; float scale = -1.0 / (1 << 15); - int ret; + int ret, ch; avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; @@ -380,10 +380,10 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx) q->bands[2] = q->high; /* Prepare the mdct overlap buffers */ - q->SUs[0].spectrum[0] = q->SUs[0].spec1; - q->SUs[0].spectrum[1] = q->SUs[0].spec2; - q->SUs[1].spectrum[0] = q->SUs[1].spec1; - q->SUs[1].spectrum[1] = q->SUs[1].spec2; + for (ch = 0; ch < AT1_MAX_CHANNELS; ch++) { + q->SUs[ch].spectrum[0] = q->SUs[ch].spec1; + q->SUs[ch].spectrum[1] = q->SUs[ch].spec2; + } return 0; } diff --git a/libavformat/aeadec.c b/libavformat/aeadec.c index be18e7b725..0a3d09a89d 100644 --- a/libavformat/aeadec.c +++ b/libavformat/aeadec.c @@ -84,7 +84,7 @@ static int aea_read_header(AVFormatContext *s) st->codecpar->sample_rate = 44100; st->codecpar->bit_rate = 146000 * channels; - if (channels != 1 && channels != 2) { + if (channels < 1 || channels > 8) { av_log(s, AV_LOG_ERROR, "Channels %d not supported!\n", channels); return AVERROR_INVALIDDATA; } -- 2.45.2