From patchwork Tue Jan 9 01:16:55 2018 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Devin Heitmueller X-Patchwork-Id: 7220 Delivered-To: ffmpegpatchwork@gmail.com Received: by 10.2.78.2 with SMTP id r2csp3317960jaa; Mon, 8 Jan 2018 17:18:29 -0800 (PST) X-Google-Smtp-Source: ACJfBov2rp5+eShJgEJVFK8738XAXAIDZ6A3M/oGVBuE5RoLGh1c0ImGGThJFhCkG3OvEsaFmV5f X-Received: by 10.223.199.1 with SMTP id k1mr11863272wrg.183.1515460709849; Mon, 08 Jan 2018 17:18:29 -0800 (PST) ARC-Seal: i=1; a=rsa-sha256; t=1515460709; cv=none; d=google.com; s=arc-20160816; b=nqu9eaLWNSC07co15vsHLeyEj3pUjG4gC7OskXEbAUSDqKL1uu5n3GSTLCunYaPAfv SKTun9d1uCaoyPhR3rgzSFVlY6UR5NA+SzC0kg9mqjge2cK4hLrHIcLii5gj8mIDOAD/ e/Zeh+I7t6C5kzXdWV6ZfDijMGw+mi2Lk2U2kCuddmDHMSDe78OwTNqlZml4SKOB2Urq haIdACGTguvhzwLvorDUsIWh3xTy6emEqiEJygKCKp7Kaaxet5LgdDfLv4CL8XPbBqm1 O6uQbh8cz/P3WlPCw3ZxL7h/Um+Lv+mVk1D/RlrIxyv8ZOMlas4Fr8mkj3VMNy29jjaz R8PA== ARC-Message-Signature: i=1; a=rsa-sha256; c=relaxed/relaxed; d=google.com; s=arc-20160816; h=sender:errors-to:content-transfer-encoding:mime-version:cc:reply-to :list-subscribe:list-help:list-post:list-archive:list-unsubscribe :list-id:precedence:subject:references:in-reply-to:message-id:date :to:from:delivered-to:arc-authentication-results; bh=513Iw6pWZU4g6rrKEHy5OFa7/Rk1rDkB4uezszhJHMs=; b=BMBEiBaVDi3ENMWgggQProwTrOcKdaHCrstH6efZrZ6pb5/m4eE2InWSWZKhnqWh6I 2Mf/PqQAeZ5c5Rb+TWgu6QWXg5QxTBJ/+p9KjtznijKzZJKKZxD3YCNSSO6Dr4WC7l+i ZA7fBehj7laRUl/UH9Tgp6opNhvMjlws7mJ4TS7HzO43ZZSYj37p0Ft1C2S1KANg8kN3 27yoU13lwdgg8ussG6THRYk4PTlYAokcBQVcl6N0LyAjlTFGVKeCb0sAca8Jk7GSu3CS 5FIS+dBqgoH81E8vQcdkMZCB4axdsjSFFKopaLYrV3SoVthx5WljeXHo1gZSxZaRxZHy bSrA== ARC-Authentication-Results: i=1; mx.google.com; spf=pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) smtp.mailfrom=ffmpeg-devel-bounces@ffmpeg.org Return-Path: Received: from ffbox0-bg.mplayerhq.hu (ffbox0-bg.ffmpeg.org. [79.124.17.100]) by mx.google.com with ESMTP id v195si8638346wmf.11.2018.01.08.17.18.29; Mon, 08 Jan 2018 17:18:29 -0800 (PST) Received-SPF: pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) client-ip=79.124.17.100; Authentication-Results: mx.google.com; spf=pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) smtp.mailfrom=ffmpeg-devel-bounces@ffmpeg.org Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id 7C78168A0BE; Tue, 9 Jan 2018 03:17:31 +0200 (EET) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from was-smtp1.livetimenet.net (50-206-97-56-static.hfc.comcastbusiness.net [50.206.97.56]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id 64386689F61 for ; Tue, 9 Jan 2018 03:17:29 +0200 (EET) Received: by was-smtp1.livetimenet.net with esmtpsa (TLSv1:AES128-SHA:128) (Exim 4.84_2) (envelope-from ) id 1eYiXg-0008Fd-Fj; Mon, 08 Jan 2018 20:17:29 -0500 From: Devin Heitmueller To: ffmpeg-devel@ffmpeg.org Date: Mon, 8 Jan 2018 20:16:55 -0500 Message-Id: <20180109011658.72370-9-dheitmueller@ltnglobal.com> X-Mailer: git-send-email 2.13.2 In-Reply-To: <20180109011658.72370-1-dheitmueller@ltnglobal.com> References: <20180109011658.72370-1-dheitmueller@ltnglobal.com> X-Spam-Score: -1.9 (-) Subject: [FFmpeg-devel] [PATCH 08/11] decklink: Add support for compressed AC-3 output over SDI 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 Cc: jgreen@ltnglobal.com, Devin Heitmueller MIME-Version: 1.0 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Extend the decklink output to include support for compressed AC-3, encapsulated using the SMPTE ST 377:2015 standard. This functionality can be exercised by using the "copy" codec when the input audio stream is AC-3. For example: ./ffmpeg -i ~/foo.ts -codec:a copy -f decklink 'UltraStudio Mini Monitor' Note that the default behavior continues to be to do PCM output, which means without specifying the copy codec a stream containing AC-3 will be decoded and downmixed to stereo audio before output. Updated to reflect feedback from Aaron Levinson Signed-off-by: Devin Heitmueller --- libavdevice/decklink_enc.cpp | 100 ++++++++++++++++++++++++++++++++++++------- 1 file changed, 85 insertions(+), 15 deletions(-) diff --git a/libavdevice/decklink_enc.cpp b/libavdevice/decklink_enc.cpp index ff60050..c39fc0e 100644 --- a/libavdevice/decklink_enc.cpp +++ b/libavdevice/decklink_enc.cpp @@ -237,19 +237,32 @@ static int decklink_setup_audio(AVFormatContext *avctx, AVStream *st) av_log(avctx, AV_LOG_ERROR, "Only one audio stream is supported!\n"); return -1; } - if (c->sample_rate != 48000) { - av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate!" - " Only 48kHz is supported.\n"); - return -1; - } - if (c->channels != 2 && c->channels != 8 && c->channels != 16) { - av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels!" - " Only 2, 8 or 16 channels are supported.\n"); + + if (st->codecpar->codec_id == AV_CODEC_ID_AC3) { + /* Regardless of the number of channels in the codec, we're only + using 2 SDI audio channels at 48000Hz */ + ctx->channels = 2; + } else if (st->codecpar->codec_id == AV_CODEC_ID_PCM_S16LE) { + if (c->channels != 2 && c->channels != 8 && c->channels != 16) { + av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels!" + " Only 2, 8 or 16 channels are supported.\n"); + return -1; + } + if (c->sample_rate != bmdAudioSampleRate48kHz) { + av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate!" + " Only 48kHz is supported.\n"); + return -1; + } + ctx->channels = c->channels; + } else { + av_log(avctx, AV_LOG_ERROR, "Unsupported codec specified!" + " Only PCM_S16LE and AC-3 are supported.\n"); return -1; } + if (ctx->dlo->EnableAudioOutput(bmdAudioSampleRate48kHz, bmdAudioSampleType16bitInteger, - c->channels, + ctx->channels, bmdAudioOutputStreamTimestamped) != S_OK) { av_log(avctx, AV_LOG_ERROR, "Could not enable audio output!\n"); return -1; @@ -260,8 +273,7 @@ static int decklink_setup_audio(AVFormatContext *avctx, AVStream *st) } /* The device expects the sample rate to be fixed. */ - avpriv_set_pts_info(st, 64, 1, c->sample_rate); - ctx->channels = c->channels; + avpriv_set_pts_info(st, 64, 1, bmdAudioSampleRate48kHz); ctx->audio = 1; @@ -553,25 +565,83 @@ static int decklink_write_video_packet(AVFormatContext *avctx, AVPacket *pkt) return 0; } +static int create_s337_payload(AVPacket *pkt, enum AVCodecID codec_id, uint8_t **outbuf, int *outsize) +{ + uint8_t *s337_payload; + uint8_t *s337_payload_start; + int payload_size = (pkt->size + 4) * sizeof(uint16_t); + int i; + + /* Encapsulate AC3 syncframe into SMPTE 337 packet */ + s337_payload = (uint8_t *) av_mallocz(payload_size); + if (s337_payload == NULL) + return AVERROR(ENOMEM); + + /* Construct SMPTE S337 Burst preamble */ + s337_payload[0] = 0x72; /* Sync Word 1 */ + s337_payload[1] = 0xf8; /* Sync Word 1 */ + s337_payload[2] = 0x1f; /* Sync Word 1 */ + s337_payload[3] = 0x4e; /* Sync Word 1 */ + + if (codec_id == AV_CODEC_ID_AC3) { + s337_payload[4] = 0x01; + } else { + av_free(s337_payload); + return AVERROR(EINVAL); + } + + s337_payload[5] = 0x00; + uint16_t bitcount = pkt->size * 8; + s337_payload[6] = bitcount & 0xff; /* Length code */ + s337_payload[7] = bitcount >> 8; /* Length code */ + s337_payload_start = &s337_payload[8]; + for (i = 0; i < pkt->size; i += 2) { + s337_payload_start[0] = pkt->data[i+1]; + s337_payload_start[1] = pkt->data[i]; + s337_payload_start += 2; + } + + *outbuf = s337_payload; + *outsize = payload_size; + return 0; +} + static int decklink_write_audio_packet(AVFormatContext *avctx, AVPacket *pkt) { struct decklink_cctx *cctx = (struct decklink_cctx *)avctx->priv_data; struct decklink_ctx *ctx = (struct decklink_ctx *)cctx->ctx; - int sample_count = pkt->size / (ctx->channels << 1); + AVStream *st = avctx->streams[pkt->stream_index]; + int sample_count; buffercount_type buffered; + uint8_t *outbuf = NULL; + int ret = 0; ctx->dlo->GetBufferedAudioSampleFrameCount(&buffered); if (pkt->pts > 1 && !buffered) av_log(avctx, AV_LOG_WARNING, "There's no buffered audio." " Audio will misbehave!\n"); - if (ctx->dlo->ScheduleAudioSamples(pkt->data, sample_count, pkt->pts, + if (st->codecpar->codec_id == AV_CODEC_ID_AC3) { + /* Encapsulate AC3 syncframe into SMPTE 337 packet */ + ret = create_s337_payload(pkt, st->codecpar->codec_id, + &outbuf, &sample_count); + if (ret != 0) + return ret; + } else { + sample_count = pkt->size / (ctx->channels << 1); + outbuf = pkt->data; + } + + if (ctx->dlo->ScheduleAudioSamples(outbuf, sample_count, pkt->pts, bmdAudioSampleRate48kHz, NULL) != S_OK) { av_log(avctx, AV_LOG_ERROR, "Could not schedule audio samples.\n"); - return AVERROR(EIO); + ret = AVERROR(EIO); } - return 0; + if (st->codecpar->codec_id == AV_CODEC_ID_AC3) + av_free(outbuf); + + return ret; } extern "C" {