Skip to content
←Back to Applications

APPLICATION

Speech & AudioMediaTop

Eleven v3: the commercial audio stack that covers speaking, listening and singing

elevenlabs-v3

ElevenLabs' flagship speech synthesis model, officially its most emotive and expressive: 70+ languages, a 5,000-character request cap, dramatic delivery and natural multi-speaker dialogue, with a separate Text to Dialogue API making multi-character dialogue an endpoint, not a prompt trick. It sits at a top row in audio for the stack behind it: the only commercial one covering speaking, listening and singing, with four TTS tiers, three ASR tiers including a Medical build and two music models. The TTS tiers are mutually exclusive: v3 is most expressive but non-realtime at 5,000 characters, Flash v2.5 is lowest latency (about 75ms, 50% cheaper per character) and Multilingual v2 is most stable for long text at 29 languages. On ASR, Scribe v2 covers 90+ languages with diarisation for 32 speakers. Latency figures are model-side only, excluding network and upstream LLM time, not SLAs. Limits: you split long text and own continuity across the seams; closed and not self-hostable, a hard barrier where data sovereignty matters. No speech board among the boards synced here, so listed alongside Gemini 3.8 Flash TTS without ranking. Graded C (vendor-stated): no WER recomputation or independent blind listening test.

C
CONFIDENCE
Vendor Claim
Official model card or keynote only, no independent re-test
70+
KEY METRIC
Language coverage
Vendor Claim · 2026-09
MATURITY
Product
research → demo → product → production
Editor's take

Graded C (vendor claim). The grade is constrained by the evidence form available across the whole audio domain: what can be verified is specification and pricing (language counts, character limits, latency tiers, feature lists), while "most expressive", "state-of-the-art speech recognition" and "35% fewer errors on clinical audio" are vendor statements. No WER was measured, no blind preference listening was run, and no prosodic continuity after long-form segmentation was measured, so do not upgrade them to A. With no third-party board in this domain at all, C is the only honest grade.

What earns it a place at the top of the ladder is not one model but stack completeness: four TTS tiers, three ASR tiers and two music tiers make it the only commercial vendor covering speaking, listening and music in one place. Real audio production was never only synthesis - a dubbed video needs TTS, subtitles need ASR word-level timestamps and diarisation, and a score needs a music model that can inpaint locally. Having all three behind one API and one account system saves integration cost rather than unit price. The Scribe v2 parameters deserve particular attention: 1000-term keyterm prompting, 65 entity types and 32-speaker diarisation is a configuration aimed squarely at enterprise meetings and professional transcription, not at a general demo.

Selection has to respect the trade-off in that table: expressiveness, latency and language coverage are mutually exclusive and no tier wins all three. v3 is the most expressive yet capped at 5,000 characters; the most stable long-form model, Multilingual v2, covers 29 languages; the ~75ms Flash v2.5 is narrower on both other axes. The right approach is one model per scenario, not one "best" model everywhere. Two risks must stay visible. First, every latency figure is model-side and excludes network and upstream LLM time, and in a real voice agent the LLM segment usually dominates, so they are not end-to-end SLAs. Second, it is closed with no self-hosting: audio must pass through ElevenLabs, which is a hard gate for data-sovereignty-sensitive work in medical, government and financial recording even with a Scribe v2 Medical tier. Compliance for voice cloning always stays with the operator - capability is not authorisation.

Text-to-SpeechSpeech RecognitionMusic GenerationElevenLabs
ElevenLabsOfficial site6 min read740

The problem it solves: audio as a complete production line rather than a TTS endpoint

Eleven v3 is ElevenLabs' flagship speech synthesis model, described officially as "our most emotionally rich, expressive speech synthesis model", with four properties: dramatic delivery and performance, 70+ languages, a 5,000 character limit per request, and support for natural multi-speaker dialogue.

We catalogue it at the top of the audio ladder not for that model alone but for what sits behind it: the only commercial audio stack that covers speaking, listening and music in one place - TTS (v3 / v3 Conversational / Multilingual v2 / Flash v2.5), ASR (Scribe v2 / Realtime / Medical) and music (Eleven Music v2.5 / v2). Competition in speech stopped being about whose timbre sounds most human; it is about who can put dubbing, subtitling and scoring into one API and one account system.

Tiering: expressiveness, latency and stability pull against each other

ModelLatencyLanguagesChars per requestUse
Eleven v3Non-realtime tier70+5,000 (about 5 minutes of audio)Audiobooks, podcasts, dubbing: when you need performance
Eleven v3 Conversational~280ms70+-Realtime dialogue agents, with audio tags for fine-grained control
Eleven Multilingual v2-2910,000The most stable tier for long-form generation
Eleven Flash v2.5~75ms3240,000Telephony and low-latency work; 50% lower price per character on the API

The most useful thing in that table is the trade-off it exposes: the most expressive model, v3, allows only 5,000 characters per request; the most stable long-form model, Multilingual v2, covers 29 languages; the lowest-latency Flash v2.5 is narrower on both expressiveness and language coverage. No tier wins all three axes, so the engineering answer is to pick per scenario rather than choose one "best" model and use it everywhere. The official guidance points the same way: v3 for multi-character audio experiences, long-form narration and emotional dialogue, shipped alongside a dedicated Text to Dialogue API that treats multi-speaker dialogue as an interface rather than a prompting trick.

Read the ~280ms and ~75ms figures with their scope in mind: both are model-side latencies as published (carrying footnotes in the docs), excluding network round trip and excluding the time your upstream LLM takes to emit text. In a real voice agent the LLM segment usually dominates.

The ASR side: Scribe v2's parameter density is worth more attention than the TTS

  • Accurate transcription in 90+ languages, with smart language detection.
  • Keyterm prompting, up to 1000 terms: feed in a domain vocabulary and proper nouns and terminology hold up. This is a hard requirement for enterprise audio (medical, legal, internal codenames) and exactly where generic ASR fails.
  • Entity detection across 65 entity types: the output is typed text, not just text.
  • Word-level timestamps plus speaker diarisation for up to 32 speakers: that number is clearly aimed at meetings and multi-party interviews.
  • Scribe v2 Realtime: ~150ms streaming transcription, same 90+ languages, word-level timestamps and entity detection.
  • Scribe v2 Medical: fine-tuned for clinical audio, with 35% fewer transcription errors on clinical audio than Scribe v2, equal accuracy on everyday speech, and identical features, languages, pricing and API. This is one of the few commercial audio products we have seen that quantifies what vertical fine-tuning buys.

The music side: Eleven Music v2.5

v2.5 is the current top tier, described as better quality and prompt adherence with richer melodies, deeper arrangements and more layered instruments. It keeps the same workflows as v2 and therefore the three capabilities that matter in production: composition plans, audio reference, and inpainting. v2 is positioned as studio-grade music from natural-language prompts in any style, with complete control over genre, style and structure.

Inpainting is the easiest of the three to underestimate and the biggest time saver: if four bars of a cue are wrong, regenerating the whole piece gambles the parts you already liked. Being able to edit locally is what makes it a production tool.

Boundaries

  • v3 caps at 5,000 characters per request, about five minutes of audio. Long form means segmenting it yourself, and prosodic continuity and timbre consistency across segment boundaries are your problem; the model does not guarantee cross-request consistency.
  • The three axes are mutually exclusive: expressiveness, latency and language coverage - see the table. There is no all-round tier.
  • Closed source, no self-hosting: no downloadable weights, no deployment in your own environment, and audio must pass through ElevenLabs. For data-sovereignty-sensitive work (medical, government, financial recordings) that is a hard gate even with a vertical Scribe v2 Medical tier.
  • Latency figures are model-side: ~280ms / ~150ms / ~75ms all carry official footnotes and exclude network and upstream LLM time, so they are not end-to-end SLAs.
  • "SOTA" and "35% fewer errors" are vendor statements: we did not recompute WER on our own test sets and ran no blind comparison against Gemini TTS, OpenAI TTS or the Whisper family.
  • Compliance for voice cloning sits with the operator: capability is not authorisation, and requirements for voice rights and synthetic-audio labelling differ by jurisdiction.
  • No third-party speech board: none of the twelve leaderboards this site syncs covers speech or audio, so this asset cannot be ranked.

Our verification status

Facts come from the ElevenLabs official Models documentation, read in full: the four TTS tiers (v3 at 70+ languages / 5,000 characters / multi-speaker dialogue; v3 Conversational at ~280ms with audio tags; Multilingual v2 at 29 languages / 10,000 characters; Flash v2.5 at ~75ms / 32 languages / 40,000 characters / 50% lower price per character), v3's stated use cases and the Text to Dialogue API, the three ASR tiers (Scribe v2 at 90+ languages / 1000 keyterms / 65 entity types / word-level timestamps / 32-speaker diarisation; Realtime at ~150ms; Medical at 35% fewer clinical errors), and the two music tiers (v2.5 with composition plans, audio reference and inpainting; v2 with genre and structure control).

Confidence is graded C (vendor claim), for the same reason as Gemini 3.8 Flash TTS: everything checkable here is specification and pricing, not a quality reading. "Most expressive", "state-of-the-art speech recognition" and "35% fewer errors on clinical audio" are vendor statements. We have not measured WER, not run blind preference listening, and not measured prosodic continuity after segmenting at the 5,000-character boundary. Reaching grade A would require WER and speaker-similarity measurement on a fixed multilingual test set, organised blind comparison, and a quantified measure of consistency decay over segmented long form.

We place it alongside Gemini 3.8 Flash TTS at the top of the audio ladder rather than ordering the two. Their strengths genuinely differ: ElevenLabs wins on stack completeness (commercial products across speaking, listening and music, with an especially dense ASR parameter set) and on latency tiering; Gemini 3.8 Flash TTS wins on voice design plus cloning plus inline emotion control inside one model, with broader language coverage (130 vs 70+) and published, lower audio-output pricing. With no third-party board in this domain, we state facts side by side and do not invent a ranking.

More in Speech & Audio

4
Speech & AudioMediaTopC

Gemini 3.8 Flash TTS: turning director-level voice direction into API parameters

Google's speech generation model (gemini-3.8-flash-tts), positioned for studio-grade voice fidelity, expressive acting and long-form stability, the most overlooked being the last: audiobook failure is not an ugly sentence but timbre drift by hour three. The design separates scopes: text is strictly a script while performance direction travels on another channel and is never read aloud, fixing "(sighs)" that used to be spoken; round style goes into speech_metadata.style, word-level emotion into pipe tags. Four voice sources: prebuilt, an extended library, voice design IDs (describe it, then reuse) and replication IDs; design and replication arrived only in 3.8. Specs: 130 languages with automatic language detection, WAV by default at 24 kHz, at most 2 speakers per request and only with prebuilt voices. Billing is per audio token at 25 tokens per second, $0.50 input and $9.00 output per million through 2026-12-31 then $1.00/$18.00. Limits: 24 kHz is not mastering grade; replication compliance rests with the user. No speech board among the 12 synced here, so listed alongside ElevenLabs v3 without ranking. Graded C (vendor-stated): no intelligibility or independent blind listening test.

All supportedVoice design + cloning + multi-speakerVendor Claim · 2026-09
ProductGoogleSite
Gemini 3.8 Flash TTS: turning director-level voice direction into API parameters
Speech & AudioMediaTopA

VoiceStudio: the commercial voice stack moved onto local hardware and opened to agents by default

An open source, fully local voice workstation (debpalash/VoiceStudio, AGPL-3.0, Python plus Electron) positioned as the local alternative to ElevenLabs: voice cloning, voice design, video dubbing, dictation, transcription and audiobook creation, with a claimed 646 languages. Created on 2026-04-09 and at 42,831 stars and 5,005 forks on 2026-09-29. It sits at the top self-hosted row in the audio domain, not because any model behind it is stronger, but because the abstraction layer is useful. Open voice capability is scattered across a dozen repositories with their own weight formats, device requirements, cloning interfaces and licences, so assembling a pipeline of clone a voice, transcribe, align, dub and produce an audiobook spends most of its effort on glue. VoiceStudio collects 17 TTS and 11 ASR engines into one registry and reports per engine which devices it runs on (CUDA, MPS, CPU), whether it can clone, and its max_ref_seconds and ref_strategy, on top of a workflow that actually runs. The agent surface matters most for the catalogue: a local API and an MCP server ship as first class product features, the README includes an installation prompt ready to paste into Claude Code, Codex or Cursor, and npx skills add installs it as a skill, which means coding agents can call dubbing and transcription as tools rather than needing a human at the interface. The reference audio policy is equally rigorous, with a 75 second ceiling while how much of a longer clip reaches the model depends on the engine and is reported per engine, an automatic or saved transcript ignored beyond the limit, and a hard [clone_ref_too_long] error when sending more than 20 seconds plus transcript text to the default OmniVoice. One measurement point needs stating plainly: the official docs/benchmarks.md contains a real harness with per stage profiling, refusal to start when memory is insufficient, and a rule that only RTF warm and peak memory on named hardware and versions are accepted with no estimated numbers allowed, but the results column reads No verified rows yet. therefore attach no performance figure at all and the card carries only the star count verified ourselves through the GitHub API, which is what earns the A grade. That A asserts the number is real and nothing about being better than ElevenLabs. Four boundaries: AGPL-3.0 is strong copyleft with a network clause, so embedding it in a service offered to outsiders obliges you to release your own server side source, and closed commercial use means either process isolation calling only the local API under your own legal review or assembling the pipeline from Apache-2.0 pieces such as CosyVoice 3 or Kokoro; per-model licences are not settled by the wrapper, since the Breeze-TTS-2 weights behind audio.cpp are research and non-commercial, Supertonic-3 and PocketTTS need extra permissions and GPT-SoVITS requires its own server, so check every model before commercial use; the 646 language figure is the repository own claim and real coverage depends on the engine chosen, from 25 European languages on Parakeet through 50 plus on FunASR to the wider Whisper family, unverified language by language; and desktop is Electron only now, with 0.5.3 the last Tauri release, so existing users must migrate. It sits alongside Eleven v4 on the ladder rather than replacing it, one representing the quality ceiling and convenience of a closed commercial stack and the other the data sovereignty and engine substitutability of local self-hosting.

42,831GitHub stars (2026-09-29, verified here via GitHub API)Confirmed · 2026-09
Productdebpalash (open source)SiteRepo
VoiceStudio: the commercial voice stack moved onto local hardware and opened to agents by default
Video GenerationMediaTopC

Kling 3.0 series: native audio, 4K frames, and parameterised camera language as the moat

Kuaishou Kling was among the first Chinese video lines to reach a level where it can take commercial work, and its lasting strength is not raw text-to-video but image-to-video plus camera control: lock composition and character in the image stage, then let the video model interpolate from that fixed keyframe, with first/last-frame constraints and push/pull/pan/truck/orbit/follow parameters producing predictable clips. The 3.0 series (official release-note banner: API fully available) brings three substantive jumps - audio generated natively (5s/10s/12s audio tiers), native 2K/4K for both image and video, and storyboarding plus element reference turning multi-shot coherence from an editing problem into something the model controls; the line-up is 3.0 / 3.0 Omni / 3.0 Turbo, Kling Image 3.0 and -omni, Native 4K Video and Motion Control. The arena reading, stated plainly: image-to-video kling-v3-pro is #19 (Elo 1354), behind MiniMax h3, gemini-omni, wan3.0 and seedance-2.5. No authoritative release date for the 3.0 series could be obtained, so the metric date is recorded only at the verifiable month, 2026-09. Closed, vendor API only; not independently benchmarked, graded C.

3.0 / Omni / TurboCurrent flagship seriesVendor Claim · 2026-09
ProductKuaishouSite
Kling 3.0 series: native audio, 4K frames, and parameterised camera language as the moat
Speech & AudioMediaTopC

Eleven v4: the generation that pushes cloning fidelity far enough to reproduce bad training data

The new generation speech synthesis model from ElevenLabs, shipping in two variants: Eleven v4 for highest quality and Eleven v4 Turbo for real time use at a median inference latency of about 100 ms, with an official footnote stating that application and network latency are excluded. The vendor positions it above v3 on output quality, voice accuracy, consistency, emotion, delivery, audio tags and language coverage, and recommends migrating. It sits at the top closed source voice row in the audio domain because this generation changed the optimisation target from sounding better to sounding more like the source: timbre, cadence, delivery and other mannerisms are reproduced together, and the vendor concedes that accuracy and personal preference are not always the same thing and that you may still prefer how a voice sounded in v3. The engineering consequence is hard, since the burden of data cleaning moves back to the user. The showcase samples are explicitly labelled raw, with no EQ, compression, normalisation, de-essing or plosive removal, and the vendor says that if you hear clipping or plosives they are very likely in the training data and the model simply captured them accurately. What is deliverable comes down to three numbers: about 100 ms on Turbo; 10,000 characters per request, roughly ten minutes of audio, against 5,000 on v3 so long form text carries twice as many segment seams; and a coverage claim of 90 plus languages against a FAQ that enumerates 87, a gap kept visible rather than smoothed away. Four boundaries matter. Cross language accent handling is a default behaviour change and not a toggle, so generating in a language that differs from the reference produces fluent native sounding speech in the target language, and the vendor calls a switchable version a research project with no timeline; any persona that depends on a native accent carried into a second language must be tested first. The control surface narrowed to Stability and Similarity, the Style and Speed sliders are gone, SSML is unsupported, and fine grained delivery moves to audio tags which the vendor admits are not perfect yet. Continuous model evolution is stated in writing, with training continuing after launch and behaviour possibly shifting over time, so teams treating a voice as a brand asset need periodic re-testing. Voice Design voices may also be less performative on v4. It is closed and not self-hostable, data must pass through ElevenLabs, and cloning compliance sits with the user; the self-hostable counterparts are CosyVoice 3, VibeVoice and Kokoro for speech and YuE2 for music under a non-commercial weight licence. This entry and the Eleven v3 already catalogued here are two generations of the same commercial stack rather than a replacement. Graded C (vendor claim): no verifiable third party benchmark, and no independent blind listening test.

~100msv4 Turbo median inference latency (model side, excl. app and network)Vendor Claim · 2026-09
ProductionElevenLabsSite
Eleven v4: the generation that pushes cloning fidelity far enough to reproduce bad training data

As an Amazon Associate, we earn from qualifying purchases.