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Supports: TS
.ts (MPEG-2 Transport Stream) recordings. Concatenated HLS segments and DVR captures are accepted. Batch processing is supported.HH:MM:SS.sss to extract a single song or interview from a longer recording. Click Convert. Files are uploaded over an encrypted connection, processed on our servers, and deleted automatically after a few hours — no sign-up, no watermark, never shared..ts is the file extension for MPEG-2 Transport Stream, the container ATSC, DVB, and HLS use to deliver broadcast and streamed video. The audio inside a TS file is almost always lossy: ATSC mandates AC-3 (Dolby Digital), DVB allows AC-3 or MP2, and HLS encoders typically use AAC or AC-3. FLAC is a lossless codec — standardized as IETF RFC 9639 in December 2024 — that compresses PCM audio to roughly 50–70% of its uncompressed size with bit-exact decoding.
Converting TS to FLAC extracts the audio elementary stream, decodes it to PCM, and re-encodes losslessly. The resulting FLAC will be larger than the original AC-3/AAC payload because lossless storage of decoded PCM has more entropy than the lossy bitstream — the upside is no further generation loss in your library.
segment-001.ts … segment-NNN.ts from a .m3u8 playlist, FLAC gives you a single durable audio file you can tag and ship.| Property | TS (MPEG-2 Transport Stream) | FLAC (output) |
|---|---|---|
| Container vs. codec | Container — carries video + audio elementary streams | Audio codec + native container |
| Typical audio inside | AC-3, MP2, AAC, E-AC-3 (all lossy) | FLAC (lossless) |
| Lossless? | No — the audio payload is already lossy | Yes — bit-exact decode |
| Defined by | ISO/IEC 13818-1 (MPEG-2 Systems) | IETF RFC 9639 (Dec 2024) |
| 1 hr broadcast audio size | ~150–200 MB audio (within a 2–8 GB video file) | ~250–400 MB at 16-bit 48 kHz stereo |
| Bit depth | Codec-dependent | 4–32 bits per sample (RFC 9639) |
| Sample rate | Usually 48 kHz (ATSC/DVB), 44.1/48 kHz (HLS) | 1 Hz to 1,048,575 Hz |
| Metadata | PMT/SDT, no song tags | Vorbis comments, cover art |
| Editing | Needs demux first | Direct in any audio editor |
| Source codec inside TS | Where you'll see it | Typical bitrate | After conversion to FLAC |
|---|---|---|---|
| AC-3 (Dolby Digital) | ATSC over-the-air, Blu-ray menus, some DVB | 192–448 kbps | Decoded to PCM, re-encoded lossless. No quality recovery, no further loss. |
| MP2 (MPEG-1 Layer II) | DVB radio and TV in Europe | 192–384 kbps | Same — wraps the decoded PCM losslessly. |
| AAC-LC | HLS, IPTV, modern DVB | 128–256 kbps | Same. |
| E-AC-3 (Dolby Digital Plus) | ATSC 3.0, streaming | 96–256 kbps | Same; multichannel layouts preserved when "Original" channel is kept. |
No. The TS file already contains lossy audio (almost always AC-3, MP2, or AAC). FLAC stores whatever PCM samples come out of the decoder bit-exactly, but it cannot reconstruct frequencies and dynamics the original lossy encoder discarded. What you get is a lossless preservation of the broadcast audio at its current quality level — useful for archival because no further loss accumulates if you later edit or re-encode.
You're comparing a lossy bitstream to a near-uncompressed PCM representation. A 1-hour AC-3 5.1 track at 384 kbps is about 165 MB inside the TS; the same audio decoded to 48 kHz 16-bit stereo PCM and re-encoded to FLAC is roughly 270–330 MB. If you only care about lossy storage, convert TS to MP3 instead — you'll land around 60–90 MB at 192–256 kbps.
Both are MPEG-2 Transport Stream variants. .ts is the generic broadcast/HLS form; .m2ts (BDAV) is the Blu-ray variant with extra timestamp information per packet. The audio decoders are the same. If your file is from a Blu-ray rip, use the M2TS to FLAC page instead.
Yes, as long as the segments were joined correctly (e.g., with cat seg-*.ts > full.ts or via FFmpeg's concat demuxer) and the PAT/PMT tables at the start are intact. HLS audio is usually AAC-LC; the converter demuxes the audio elementary stream and encodes it to FLAC at the source sample rate.
Leave it on Original unless your downstream tool requires a specific rate. ATSC and DVB broadcasts standardize on 48000 Hz; HLS streams are typically 44100 or 48000 Hz. Forcing a different rate triggers resampling, which is a small but unnecessary precision loss for an archival workflow.
No. FLAC's compression levels (1–12 here, which maps to FFmpeg's -compression_level 0–12) only trade encoder CPU time for file size. Level 8 is a good default; level 12 saves perhaps 1–2% more space for substantially more time. The audio output is bit-identical at every level.
Yes. Open the Trim controls and set a start time and duration in seconds or HH:MM:SS.sss form. The trim happens in the time domain after decoding, so frame-accurate cuts are possible without any audio re-encoding penalty since FLAC is lossless.
VLC, foobar2000, Audacity, Reaper, and Audirvana all play FLAC natively. Apple added FLAC support in iOS 11 and macOS High Sierra (2017+). Android has supported FLAC since 3.1 (2011). Most modern car head units list FLAC under "Hi-Res Audio." If you specifically need MP3 for an older device, run FLAC to MP3 afterwards.
FLAC is the right call for archival. If you want a smaller lossy file for casual listening, target TS to AAC (good codec match, since AAC is often the source) or TS to WAV (uncompressed PCM — larger than FLAC, no advantage for storage, but useful if a tool requires WAV input).