From patchwork Fri Oct 9 13:04:19 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Anton Khirnov X-Patchwork-Id: 22793 Return-Path: X-Original-To: patchwork@ffaux-bg.ffmpeg.org Delivered-To: patchwork@ffaux-bg.ffmpeg.org Received: from ffbox0-bg.mplayerhq.hu (ffbox0-bg.ffmpeg.org [79.124.17.100]) by ffaux.localdomain (Postfix) with ESMTP id 967B744AB2F for ; Fri, 9 Oct 2020 16:09:47 +0300 (EEST) Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id 7F4A168BA0D; Fri, 9 Oct 2020 16:06:51 +0300 (EEST) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from mail.red.khirnov.net (red.khirnov.net [176.97.15.12]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id D2CC768B9D9 for ; Fri, 9 Oct 2020 16:06:38 +0300 (EEST) Received: from localhost (localhost [IPv6:::1]) by mail.red.khirnov.net (Postfix) with ESMTP id BA9B829625D for ; Fri, 9 Oct 2020 15:06:34 +0200 (CEST) Received: from mail.red.khirnov.net ([IPv6:::1]) by localhost (mail.red.khirnov.net [IPv6:::1]) (amavisd-new, port 10024) with ESMTP id rankwlRenf_v for ; Fri, 9 Oct 2020 15:06:34 +0200 (CEST) Received: from libav.daenerys.khirnov.net (libav.daenerys.khirnov.net [IPv6:2a00:c500:561:201::7]) (using TLSv1.3 with cipher TLS_AES_256_GCM_SHA384 (256/256 bits) key-exchange X25519 server-signature RSA-PSS (2048 bits) server-digest SHA256 client-signature RSA-PSS (2048 bits) client-digest SHA256) (Client CN "libav.daenerys.khirnov.net", Issuer "smtp.khirnov.net SMTP CA" (verified OK)) by mail.red.khirnov.net (Postfix) with ESMTPS id 8648929625B for ; Fri, 9 Oct 2020 15:06:31 +0200 (CEST) Received: by libav.daenerys.khirnov.net (Postfix, from userid 1000) id 37EA720E0235; Fri, 9 Oct 2020 15:06:26 +0200 (CEST) From: Anton Khirnov To: ffmpeg-devel@ffmpeg.org Date: Fri, 9 Oct 2020 15:04:19 +0200 Message-Id: <20201009130430.602-7-anton@khirnov.net> X-Mailer: git-send-email 2.28.0 In-Reply-To: <20201009130430.602-1-anton@khirnov.net> References: <20201009130430.602-1-anton@khirnov.net> MIME-Version: 1.0 Subject: [FFmpeg-devel] [PATCH 07/18] lavf: move AVStream.{*skip_samples.*_discard_sample} to AVStreamInternal X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Those are private fields, no reason to have them exposed in a public header. --- libavformat/avformat.h | 29 ----------------------------- libavformat/internal.h | 29 +++++++++++++++++++++++++++++ libavformat/mov.c | 12 ++++++------ libavformat/mp3dec.c | 8 ++++---- libavformat/swfdec.c | 2 +- libavformat/utils.c | 26 +++++++++++++------------- 6 files changed, 53 insertions(+), 53 deletions(-) diff --git a/libavformat/avformat.h b/libavformat/avformat.h index 977680ec83..8028b1b558 100644 --- a/libavformat/avformat.h +++ b/libavformat/avformat.h @@ -1100,35 +1100,6 @@ typedef struct AVStream { */ int skip_to_keyframe; - /** - * Number of samples to skip at the start of the frame decoded from the next packet. - */ - int skip_samples; - - /** - * If not 0, the number of samples that should be skipped from the start of - * the stream (the samples are removed from packets with pts==0, which also - * assumes negative timestamps do not happen). - * Intended for use with formats such as mp3 with ad-hoc gapless audio - * support. - */ - int64_t start_skip_samples; - - /** - * If not 0, the first audio sample that should be discarded from the stream. - * This is broken by design (needs global sample count), but can't be - * avoided for broken by design formats such as mp3 with ad-hoc gapless - * audio support. - */ - int64_t first_discard_sample; - - /** - * The sample after last sample that is intended to be discarded after - * first_discard_sample. Works on frame boundaries only. Used to prevent - * early EOF if the gapless info is broken (considered concatenated mp3s). - */ - int64_t last_discard_sample; - /** * An opaque field for libavformat internal usage. * Must not be accessed in any way by callers. diff --git a/libavformat/internal.h b/libavformat/internal.h index b1112fe463..12105aa7d0 100644 --- a/libavformat/internal.h +++ b/libavformat/internal.h @@ -225,6 +225,35 @@ struct AVStreamInternal { } *info; + /** + * Number of samples to skip at the start of the frame decoded from the next packet. + */ + int skip_samples; + + /** + * If not 0, the number of samples that should be skipped from the start of + * the stream (the samples are removed from packets with pts==0, which also + * assumes negative timestamps do not happen). + * Intended for use with formats such as mp3 with ad-hoc gapless audio + * support. + */ + int64_t start_skip_samples; + + /** + * If not 0, the first audio sample that should be discarded from the stream. + * This is broken by design (needs global sample count), but can't be + * avoided for broken by design formats such as mp3 with ad-hoc gapless + * audio support. + */ + int64_t first_discard_sample; + + /** + * The sample after last sample that is intended to be discarded after + * first_discard_sample. Works on frame boundaries only. Used to prevent + * early EOF if the gapless info is broken (considered concatenated mp3s). + */ + int64_t last_discard_sample; + /** * Number of internally decoded frames, used internally in libavformat, do not access * its lifetime differs from info which is why it is not in that structure. diff --git a/libavformat/mov.c b/libavformat/mov.c index 82fd1d74f6..6103678cdb 100644 --- a/libavformat/mov.c +++ b/libavformat/mov.c @@ -3550,7 +3550,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) } if (first_non_zero_audio_edit > 0) - st->skip_samples = msc->start_pad = 0; + st->internal->skip_samples = msc->start_pad = 0; } // While reordering frame index according to edit list we must handle properly @@ -3625,7 +3625,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) curr_cts < edit_list_media_time && curr_cts + frame_duration > edit_list_media_time && first_non_zero_audio_edit > 0) { packet_skip_samples = edit_list_media_time - curr_cts; - st->skip_samples += packet_skip_samples; + st->internal->skip_samples += packet_skip_samples; // Shift the index entry timestamp by packet_skip_samples to be correct. edit_list_dts_counter -= packet_skip_samples; @@ -3658,7 +3658,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) // Increment skip_samples for the first non-zero audio edit list if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && first_non_zero_audio_edit > 0 && st->codecpar->codec_id != AV_CODEC_ID_VORBIS) { - st->skip_samples += frame_duration; + st->internal->skip_samples += frame_duration; } } } @@ -3744,7 +3744,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) // Update av stream length, if it ends up shorter than the track's media duration st->duration = FFMIN(st->duration, edit_list_dts_entry_end - start_dts); - msc->start_pad = st->skip_samples; + msc->start_pad = st->internal->skip_samples; // Free the old index and the old CTTS structures av_free(e_old); @@ -7615,7 +7615,7 @@ static int mov_read_header(AVFormatContext *s) fix_timescale(mov, sc); if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && st->codecpar->codec_id == AV_CODEC_ID_AAC) { - st->skip_samples = sc->start_pad; + st->internal->skip_samples = sc->start_pad; } if (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO && sc->nb_frames_for_fps > 0 && sc->duration_for_fps > 0) av_reduce(&st->avg_frame_rate.num, &st->avg_frame_rate.den, @@ -8104,7 +8104,7 @@ static int mov_read_seek(AVFormatContext *s, int stream_index, int64_t sample_ti int64_t timestamp; MOVStreamContext *sc = s->streams[i]->priv_data; st = s->streams[i]; - st->skip_samples = (sample_time <= 0) ? sc->start_pad : 0; + st->internal->skip_samples = (sample_time <= 0) ? sc->start_pad : 0; if (stream_index == i) continue; diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c index b044679c02..5e7f273c6a 100644 --- a/libavformat/mp3dec.c +++ b/libavformat/mp3dec.c @@ -255,13 +255,13 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream *st, mp3->start_pad = v>>12; mp3-> end_pad = v&4095; - st->start_skip_samples = mp3->start_pad + 528 + 1; + st->internal->start_skip_samples = mp3->start_pad + 528 + 1; if (mp3->frames) { - st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * (int64_t)spf; - st->last_discard_sample = mp3->frames * (int64_t)spf; + st->internal->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * (int64_t)spf; + st->internal->last_discard_sample = mp3->frames * (int64_t)spf; } if (!st->start_time) - st->start_time = av_rescale_q(st->start_skip_samples, + st->start_time = av_rescale_q(st->internal->start_skip_samples, (AVRational){1, c->sample_rate}, st->time_base); av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3-> end_pad); diff --git a/libavformat/swfdec.c b/libavformat/swfdec.c index 2769a768de..fa11c050cd 100644 --- a/libavformat/swfdec.c +++ b/libavformat/swfdec.c @@ -292,7 +292,7 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt) return AVERROR(ENOMEM); ast->duration = avio_rl32(pb); // number of samples if (((v>>4) & 15) == 2) { // MP3 sound data record - ast->skip_samples = avio_rl16(pb); + ast->internal->skip_samples = avio_rl16(pb); len -= 2; } len -= 7; diff --git a/libavformat/utils.c b/libavformat/utils.c index ae7e91171e..9fd79ee540 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -1123,7 +1123,7 @@ static void update_initial_timestamps(AVFormatContext *s, int stream_index, if (st->start_time == AV_NOPTS_VALUE && pktl_it->pkt.pts != AV_NOPTS_VALUE) { st->start_time = pktl_it->pkt.pts; if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && st->codecpar->sample_rate) - st->start_time = av_sat_add64(st->start_time, av_rescale_q(st->skip_samples, (AVRational){1, st->codecpar->sample_rate}, st->time_base)); + st->start_time = av_sat_add64(st->start_time, av_rescale_q(st->internal->skip_samples, (AVRational){1, st->codecpar->sample_rate}, st->time_base)); } } @@ -1136,7 +1136,7 @@ static void update_initial_timestamps(AVFormatContext *s, int stream_index, st->start_time = pts; } if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && st->codecpar->sample_rate) - st->start_time = av_sat_add64(st->start_time, av_rescale_q(st->skip_samples, (AVRational){1, st->codecpar->sample_rate}, st->time_base)); + st->start_time = av_sat_add64(st->start_time, av_rescale_q(st->internal->skip_samples, (AVRational){1, st->codecpar->sample_rate}, st->time_base)); } } @@ -1638,25 +1638,25 @@ FF_ENABLE_DEPRECATION_WARNINGS if (ret >= 0) { AVStream *st = s->streams[pkt->stream_index]; int discard_padding = 0; - if (st->first_discard_sample && pkt->pts != AV_NOPTS_VALUE) { + if (st->internal->first_discard_sample && pkt->pts != AV_NOPTS_VALUE) { int64_t pts = pkt->pts - (is_relative(pkt->pts) ? RELATIVE_TS_BASE : 0); int64_t sample = ts_to_samples(st, pts); int duration = ts_to_samples(st, pkt->duration); int64_t end_sample = sample + duration; - if (duration > 0 && end_sample >= st->first_discard_sample && - sample < st->last_discard_sample) - discard_padding = FFMIN(end_sample - st->first_discard_sample, duration); + if (duration > 0 && end_sample >= st->internal->first_discard_sample && + sample < st->internal->last_discard_sample) + discard_padding = FFMIN(end_sample - st->internal->first_discard_sample, duration); } - if (st->start_skip_samples && (pkt->pts == 0 || pkt->pts == RELATIVE_TS_BASE)) - st->skip_samples = st->start_skip_samples; - if (st->skip_samples || discard_padding) { + if (st->internal->start_skip_samples && (pkt->pts == 0 || pkt->pts == RELATIVE_TS_BASE)) + st->internal->skip_samples = st->internal->start_skip_samples; + if (st->internal->skip_samples || discard_padding) { uint8_t *p = av_packet_new_side_data(pkt, AV_PKT_DATA_SKIP_SAMPLES, 10); if (p) { - AV_WL32(p, st->skip_samples); + AV_WL32(p, st->internal->skip_samples); AV_WL32(p + 4, discard_padding); - av_log(s, AV_LOG_DEBUG, "demuxer injecting skip %d / discard %d\n", st->skip_samples, discard_padding); + av_log(s, AV_LOG_DEBUG, "demuxer injecting skip %d / discard %d\n", st->internal->skip_samples, discard_padding); } - st->skip_samples = 0; + st->internal->skip_samples = 0; } if (st->internal->inject_global_side_data) { @@ -1890,7 +1890,7 @@ void ff_read_frame_flush(AVFormatContext *s) if (s->internal->inject_global_side_data) st->internal->inject_global_side_data = 1; - st->skip_samples = 0; + st->internal->skip_samples = 0; } }