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The concat demuxer can join a list of video files into one continuous input for FFmpeg, but it only works when those files already share the same streams, codecs and time base. It does not turn mismatched clips into 4K at 60 fps. A reliable 4K60 file playlist therefore has three stages: inspect and prepare each file, join them, then send one encoder-compatible live stream to YouTube and check it in Live Control Room.
This guide assumes a 16:9 playlist targeting 2160p at 60 fps, a Linux, macOS or Windows machine with a recent FFmpeg build, and source files that are MP4 containers with H.264 video and AAC audio. Other containers and codecs can work, but the checks below still apply.
What the concat demuxer does and does not do
The concat demuxer reads a text list of files and presents them to FFmpeg as one input, adjusting timestamps so the files play back to back. The FFmpeg ffprobe documentation at ffprobe Documentation states the rule that matters here: “All files must have the same streams (same codecs, same time base, etc.).” If that rule is broken, you get stutters, freezes, timestamp errors or a stream that fails at the first file boundary.
Stream copy, which the concat demuxer is usually paired with, avoids decoding and re-encoding. That is fast and keeps the original quality, but it cannot filter, resize, change the frame rate or normalize anything. The FFmpeg manual at FFmpeg Documentation also means that an output frame-rate option in stream-copy mode does not change the packets. Writing -r 60 next to -c copy labels the output but does not make 24 fps or 30 fps packets into 60 fps packets.
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A community thread on the FFmpeg user list in September 2024 (FFmpeg user mailing list, 2024-09-22) starts with the question “Using ffmpeg to stream a playlist of videos,” where the poster describes files with “very different formats, sizes, and aspect ratios.” That is the usual reason this workflow fails. The thread is a useful troubleshooting pattern, not an official procedure, so the steps below rely on FFmpeg’s own documentation and YouTube’s Help pages.
Before you start: decide your target and check your upload
Settle three things before touching FFmpeg:
- Target format. 2160p (3840×2160), 60 fps, 16:9 with square pixels. If your sources are 1080p or 4K at 30 fps, the target still has to be 4K60 in the output, which means interpolated frames or duplicated frames and upscaled pixels. The result looks like 4K but carries the detail of the source.
- Upload capacity. YouTube’s streaming tips page (YouTube Help, streaming tips) recommends leaving headroom above your stream bitrate. The encoder settings page (YouTube Help, live encoder settings) recommends 20% room beyond the total of your primary and backup bitrates. At the 35 Mbps H.264 figure below, that means about 42 Mbps of sustained upload.
- Machine capacity. Real-time 4K60 H.264 encoding with libx264 is CPU-heavy. A file that plays smoothly in a media player can still fall behind in a live encode. Test the encode speed before going live (see the troubleshooting table).
Stage 1: Inspect every file
Run ffprobe on each file. Start with the video stream:
ffprobe -v error -select_streams v:0 -show_entries stream=codec_name,profile,width,height,pix_fmt,sample_aspect_ratio,avg_frame_rate,r_frame_rate,time_base -of default=nw=1 clip1.mp4Then the audio stream:
ffprobe -v error -select_streams a:0 -show_entries stream=codec_name,sample_rate,channels,channel_layout,time_base -of default=nw=1 clip1.mp4Then the container duration:
ffprobe -v error -show_entries format=duration -of default=nw=1:nk=1 clip1.mp4Compare the results across all files against this table.
| Property | Why it matters for a concat playlist | Must match across files? |
|---|---|---|
| Video codec_name and profile | The concat demuxer cannot join different codecs in stream-copy mode. | Yes |
| width and height | Different dimensions break a copied stream at the boundary; a live encoder needs one output size. | Yes, for copy. For re-encoding, the output size is set by you. |
| time_base | The FFmpeg documentation requires the same time base across files. | Yes, for copy |
| pix_fmt | Mixed pixel formats (for example yuv420p and yuv422p10le) produce wrong colours or errors. | Yes |
| sample_aspect_ratio | A non-square SAR changes the displayed shape of the picture. | Yes, should be 1:1 |
| avg_frame_rate | Mixed rates (24, 25, 30, 50, 60) cause judder or duplicated frames. | Yes, for copy; 60 is the target |
| Audio codec, sample_rate, channels | Mismatched audio streams fail the same way video does. | Yes |
| format duration | Incorrect duration metadata can create artifacts or gaps at file joins, per the ffprobe documentation. | Should be accurate for each file |
If the duration in the container disagrees with the real length, remuxing with ffmpeg -i bad.mp4 -c copy fixed.mp4 can rewrite the container. It does not change the stream data, so if the gaps remain, re-encode that file in Stage 2.
Stage 2: Prepare the files
Use the result of Stage 1 to choose one path.
Path A: every file already matches
If every file has the same codec, profile, dimensions, pixel format, SAR, time base, frame rate of 60 fps, and identical audio layout, skip to Stage 3. This is the only case where the concat demuxer with stream copy is a clean fit.
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Path B: normalize each file to the target
When the files differ, encode each one to a common format once, then concatenate the normalized copies. This is the FFmpeg FAQ’s approach when re-encoding is needed. The filter chain below letterboxes a 16:9 or 4:3 file into a 3840×2160 frame without stretching, converts the frame rate to 60, and fixes the pixel format and SAR:
ffmpeg -i clip1.mp4
-vf "scale=3840:2160:force_original_aspect_ratio=decrease,pad=3840:2160:(ow-iw)/2:(oh-ih)/2,fps=60,setsar=1,format=yuv420p"
-c:v libx264 -preset medium -profile:v high -b:v 35M -maxrate 35M -bufsize 70M
-g 120 -keyint_min 120 -sc_threshold 0
-c:a aac -b:a 160k -ar 48000 -ac 2
norm_clip1.mp4Run the same command for each file, changing only the input and output names. Then repeat the ffprobe checks from Stage 1 on each normalized file; they should now all report identical values.
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Notes on this step:
fps=60duplicates frames when the source is 30 fps and drops or blends frames when it is 50 or 59.94 fps. A 24 fps source becomes 60 fps by repeating frames in an uneven pattern, which can look judder-prone on pans. If the material matters, shoot or master at 60 fps.- Scaling a 1080p source to 2160p adds pixels but not detail. The output is the same size the live encoder will send, which is the point of this step.
- The audio settings (48 kHz, stereo AAC at 160 kbps) are a common, consistent choice. They are not a figure from YouTube’s encoder page, so change them if your ingest requires something else.
- The preset is a trade-off.
mediumgives a smaller, better file but takes longer to produce. This is an offline step, so it can be slower than the live encode.
Path C: one-pass filter-based concatenation
If you do not want intermediate files, the concat filter joins the inputs and re-encodes them in one command. This example joins two clips with video and audio:
ffmpeg -i clip1.mp4 -i clip2.mp4
-filter_complex "[0:v]scale=3840:2160:force_original_aspect_ratio=decrease,pad=3840:2160:(ow-iw)/2:(oh-ih)/2,fps=60,setsar=1,format=yuv420p[v0];[0:a]aresample=48000,aformat=channel_layouts=stereo[a0];[1:v]scale=3840:2160:force_original_aspect_ratio=decrease,pad=3840:2160:(ow-iw)/2:(oh-ih)/2,fps=60,setsar=1,format=yuv420p[v1];[1:a]aresample=48000,aformat=channel_layouts=stereo[a1];[v0][a0][v1][a1]concat=n=2:v=1:a=1[v][a]"
-map "[v]" -map "[a]"
-c:v libx264 -preset medium -b:v 35M -maxrate 35M -bufsize 70M -g 120 -keyint_min 120 -sc_threshold 0
-c:a aac -b:a 160k
joined.mp4This works for short playlists. For a long list, normalized files plus the concat demuxer (Path B) are easier to debug, because you can test each file on its own.
Stage 3: Build the playlist and test the join
The concat demuxer reads a plain text file in FFconcat format. Each line names one file:
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file 'norm_intro.mp4'
file 'norm_part1.mp4'
file 'norm_part2.mp4'Keep the list file in the same folder as the media and use relative names. FFmpeg’s default safe-path check then accepts the names with no extra options. If you use absolute paths such as file '/home/you/show/intro.mp4', you must add -safe 0 to the command. Only do this with a playlist you wrote yourself, because the option turns off a safety check on file names.
Before you stream, test the join to a file. This writes one continuous output you can inspect:
ffmpeg -f concat -i playlist.txt -c copy test_join.mp4Play test_join.mp4 and check each boundary for a freeze, a frame jump or a sound click. Then run the Stage 1 ffprobe commands on test_join.mp4; a clean join keeps the same codec, size and frame rate. If the join fails with a time-base or codec error, one of the files is still out of spec, so return to Stage 2.
Stage 4: Send the stream to YouTube
Get the ingest URL and stream key
In YouTube Studio, open the live streaming setup for the channel (the menu path is Create, then Go live; labels can change, so follow the current Studio screens). Copy the stream URL and the stream key. Treat the key like a password: anyone who has it can publish to your channel. The URL and key combine as <ingest-url>/<stream-key>.
Target settings
The YouTube encoder settings page lists RTMP and RTMPS ingest, the codecs H.264, H.265/HEVC and AV1, a maximum of 60 fps, constant bitrate (CBR), and keyframes every two seconds with a four-second maximum. For 2160p at 60 fps, the bitrates shown on the English (UK) version of that page when this guide was checked were:
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| Codec | 2160p60 bitrate listed on the page | Practical note |
|---|---|---|
| H.264 | Minimum not stated; 35 Mbps recommended | The most widely supported option for FFmpeg-based pipelines. |
| H.265 / HEVC | 10 Mbps minimum, 40 Mbps maximum | Check that your FFmpeg build and ingest path accept HEVC in your container before you rely on it. |
| AV1 | 10 Mbps minimum, 40 Mbps maximum | Same caution as HEVC; test first. |
Localized versions of the page and later revisions can show different values, so compare your choice against the live table before a long broadcast. The settings below use H.264 at 35 Mbps, the recommended figure.
- Keyframe interval: 2 seconds. At 60 fps that is 120 frames, so
-g 120. Never exceed 4 seconds (240 frames). - Bitrate control: CBR, which means
-b:v 35M -maxrate 35M -bufsize 70Mplus-x264-params nal-hrd=cbrso the encoder holds the rate. - Audio: AAC stereo at 48 kHz. Every live stream needs an audio track; a silent one is better than none.
Re-encode and stream
This command reads the playlist, paces the input in real time, loops it indefinitely, re-encodes to the settings above and sends the stream to YouTube:
ffmpeg -re -stream_loop -1 -f concat -i playlist.txt
-map 0:v:0 -map 0:a:0
-c:v libx264 -preset veryfast -profile:v high -pix_fmt yuv420p
-s 3840x2160 -r 60
-b:v 35M -minrate 35M -maxrate 35M -bufsize 70M
-g 120 -keyint_min 120 -sc_threshold 0 -x264-params nal-hrd=cbr
-c:a aac -b:a 160k -ar 48000 -ac 2
-f flv "<ingest-url>/<stream-key>"What each part does:
-remakes FFmpeg read the files at their native speed. Without it, FFmpeg reads as fast as the disk allows and floods the connection. It does not change resolution, frame rate or codec.-stream_loop -1is an input option, so it goes before-i. It loops the input indefinitely. Because it is applied to a concat list, confirm in a short test that the loop continues at the end of the list in your FFmpeg build. If it does not, repeat thefilelines in the list (for example, the whole sequence three times) as a fallback.-map 0:a:0fails if the input has no audio stream. Normalized files from Stage 2 always have one.-f flvis the container RTMP expects for H.264 and AAC. If your build reports an error for anrtmps://URL, check that it has TLS support, or use the RTMP URL shown in Studio.
Keep the terminal open. FFmpeg writes a progress line with speed=. A value near 1.0x means it keeps pace with real time. Values below 1.0x mean the stream is falling behind, which causes stutter on YouTube.
Stream copy: only when the files already meet the target
If every file is already H.264 with AAC audio at 2160p60, a CBR-like bitrate near 35 Mbps, and 2-second keyframes, you can skip the encode:
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One free scan finds every outdated or missing driver and matches the right update for your exact hardware.Free scan · exact hardware matchffmpeg -re -stream_loop -1 -f concat -i playlist.txt -c copy -f flv "<ingest-url>/<stream-key>"Copy mode keeps each file’s bitrate pattern. Most camera and editor exports are variable bitrate and do not match the listed range or the CBR recommendation, so check with ffprobe (-show_entries format=bit_rate for each file) before using copy. Copy mode also cannot correct a keyframe interval that is too long.
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Stage 5: Verify in Live Control Room
YouTube’s streaming tips page says to test before going live: “Make sure that you test before you start your live stream.” Start the stream with the commands above, then open the stream in YouTube Studio’s Live Control Room and check the following before relying on it:
- The stream health panel shows the incoming stream as connected and receiving data. Labels in Studio can change, so look for the status and health warnings rather than the exact wording.
- The resolution shows 2160p and the frame rate shows 60 fps. If it shows a lower value, the stream is not arriving at the size you intended.
- The reported bitrate stays near 35 Mbps. A large dip means the upload cannot keep up; a steady spike toward the top of the range can mean the bitrate control is off.
- No warnings about bitrate, keyframes or audio are open. Fix these before going live to viewers.
- Each file boundary plays without a freeze or audio click in the preview.
Run a test of at least 10 to 15 minutes, including a file boundary and at least one loop back to the start. Latency on 4K streams is higher than on 1080p settings; YouTube’s guidance treats this as normal, so do not judge the stream by a low-latency 1080p benchmark. Then stop the test and start the real broadcast.
Troubleshooting
| Symptom | Likely cause | Fix |
|---|---|---|
| “Unsafe file name” or refusal to open a listed file | The list uses absolute paths and -safe 0 is missing. |
Add -safe 0, or switch to relative paths in the same folder as the list. |
| Freeze, stutter or “non-monotonic DTS” warnings at a file boundary | Mismatched time base, frame rate or wrong duration metadata. | Normalize the offending file (Path B) and repeat the Stage 1 checks. |
| Colour shifts or a squashed picture at one clip | Different pixel format or sample aspect ratio. | Normalize with format=yuv420p and setsar=1. |
| Join fails with a codec or stream-layout error | Stream copy was used on files with different codecs or stream counts. | Re-encode every file to one codec, or use the concat filter (Path C). |
| Speed reads below 1.0x and YouTube shows dropped frames | The CPU cannot run a 4K60 libx264 encode in real time at the chosen preset. | Use a faster preset, a faster machine, or a hardware encoder. Hardware encoders need their own settings, which this guide does not cover; test any change in Live Control Room. |
| Stream stops at the end of the list | The loop option is not taking effect in this FFmpeg build. | Repeat the file lines in the list as a fallback. |
| Stream has no sound | Source files lack audio, or the audio was not normalized. | Add an audio track in Stage 2 (a silent track is acceptable), then retest. |
| Live Control Room flags low or high bitrate, or keyframe problems | Copy mode is passing through a variable-bitrate or long-keyframe source. | Switch to the re-encode command with CBR and -g 120. |
| Stream drops after a period of time | Upload is below the target, Wi-Fi is unstable, or the computer sleeps or restarts. | Check upload with a speed test during the broadcast, use wired networking, and disable sleep. Reconnect with the same command. |
| YouTube ends the stream or raises a copyright notice | The playlist contains material you do not have rights to broadcast. | Use only footage you own or are licensed to stream. The copyright safety checklist at https://streamneo.com/tools/copyright-safety-checklist lists the questions to answer before going live with recorded content. |
Costs and limits
The main cost of this method is the machine. A 4K60 encode keeps the CPU busy for as long as the stream runs, so the computer must stay on, awake and connected for the whole broadcast. Upload capacity is the second constraint: a 35 Mbps stream needs about 42 Mbps of sustained upload under the headroom recommendation above. Offline preparation time also grows with playlist length, since every mismatched file needs one normalization pass.
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- Upload your recording or build a playlist in StreamNeo.
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- Any quality up to 4K 60fps at one price per slot. Uploads stream as you made them, with no re-encode and no quality tiers. Normalize your files with Stage 2 first, because the cloud stream plays exactly what you upload.
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Start with the free day at https://app.streamneo.com/register, no card needed, and see whether a cloud loop fits your channel.
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