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Fix the decoding issues #1768

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@bobqianic bobqianic commented Jan 14, 2024

  • Basic functionality
  • Rewrite whisper_wrap_segment
  • Rewrite L5717-L5805
  • Remove print_realtime This is too tricky
  • Remove hallucination by using token_nosp
  • Heuristic hallucination detection (Basic implementation)
  • Disable beam search when $temperature>0$
  • Fix tokenizer
  • Fix audio feature seeking mechanism
  • Use compression ratio instead of entropy Will be addressed in separate PRs
  • Code cleanup

@bobqianic bobqianic added the decoding Decoding related issues label Jan 17, 2024
@bobqianic
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I think that to completely avoid hallucination, the best approach is similar to using DTW to calculate token timestamps. By comparing these with cross-attention weights, we can definitely identify anomalies if there are any hallucinations.

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looks good to me, but I think @ggerganov needs to approve it

@ggerganov
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Did you run some tests?

@bobqianic
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bobqianic commented Feb 8, 2024

Did you run some tests?

I've done some initial testing, and the results are promising. However, I need a bit more time to conduct a comprehensive analysis. You can already notice the difference by testing a few audio files. Currently, I'm downloading the Common Voice Corpus 15.0, which is over 100GB, so completing the testing will take a little while.

There is a person who sent me a test file via Discord. Running large-v2 with master will generate a lot of duplicate content, but using this PR will be much better. The file is copyrighted, so I cannot make it public, but you can ask him for it privately.

#1724 (comment)

@jettoblack
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@bobqianic I'm very appreciative of this work and very excited to see this branch implemented, but getting some bad results with weird non-speech tokens at the beginning of many files, this problem does not happen in master branch.

Example 1:

wav file: https://www.dropbox.com/scl/fi/bdz7lx4khunq3kiauyus8/shermer.wav?rlkey=hzy02rkewjb4pwoamp9whch4b&dl=0

Command:

./main -m ggml-largev2.bin -f shermer.wav

Output of master branch @ 434b8f3 (current):

[00:00:00.000 --> 00:00:09.000] [music]
[00:00:09.000 --> 00:00:12.000] [applause]
[00:00:12.000 --> 00:00:14.000] Hey, I am Michael Shermer, the director of the Skeptic Society,
...

Output of this PR @ c0277e3:

[00:00:00.000 --> 00:00:07.000] Transcriber's Name Reviewer's Name
[00:00:12.340 --> 00:00:14.300] I am Michael Shermer, the director of the Skeptic Society,
...

Example 2 with translate fr to en:

wav file: https://www.dropbox.com/scl/fi/1go0yxkr10vwhfyxs76vz/french.wav?rlkey=312gc5qmw3r31ovh003410hyb&dl=0

Command:

./main -m ggml-largev2.bin -f french.wav -l fr -tr

Output of master branch @ 434b8f3 (current):

[00:00:00.000 --> 00:00:04.000] (Music)
[00:00:04.000 --> 00:00:07.000] (Applause)
[00:00:07.000 --> 00:00:20.000] I am a champion of France.
...

Output of this PR @ c0277e3:

[00:00:00.000 --> 00:00:17.000] Translation & subtitling by Quentin Dewaghe Traduction & sous-titrage par Quentin Dewaghe q.dewaghe.com
[00:00:17.000 --> 00:00:20.000] I'm a champion of France.
...

Any idea why these non-speech tokens like "Transcriber's Name Reviewer's Name" are being output as speech at the beginning? Thanks again.

@bobqianic
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Any idea why these non-speech tokens like "Transcriber's Name Reviewer's Name" are being output as speech at the beginning? Thanks again.

Thank you for letting me know. It seems the primary issue stems from my having suppressed non-speech tokens, which has resulted in symbols like ( and [ having a zero probability of appearing. While this approach enhances the overall quality, it clearly didn't account for situations like yours, which I hadn't anticipated. As mentioned, I'll conduct further tests and explore ways to address this issue.

@bobqianic
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@jettoblack I've added a heuristic for detecting repetitive hallucinations, which you can disable via parameters if you prefer. Additionally, I've removed the tokens ( ) [ and ] from the list of tokens to be suppressed, so they will remain unaffected even when suppression mode is enabled.

Output of this PR @ 476dff4:

[00:00:00.000 --> 00:00:17.000] [Music]
[00:00:17.000 --> 00:00:20.000] I am a champion of France.
...

@jettoblack
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@bobqianic The repetition heuristic seems to be working well so far. I'm seeing fewer hallucinations on silent intervals. I looked at the code and this is unrelated to the non-speech token changes, right?

I'm not so sure about the non-speech token changes. With your latest commit I see fewer cases of the problem I mentioned above, but it's still happening a lot. One example I got just now in the sg1.wav file I sent you previously on Discord:

[00:57:11.700 --> 00:57:14.700] (c) 2014 University of Georgia College of Agricultural and Environmental Sciences UGA Extension Office of Communications and Creative Services

A hallucination like ♪♪ or repeated text is far less objectionable than someone else's copyright notice or translator's notes which is what I'm getting a lot of.

This change also removes many useful tokens from the output, like quotation marks and music notes. Using the -nsnst option restores these tokens but that causes this issue to be much worse, and I've caught a lot more cases of it occurring in many files, including in the middle of files not just the beginning. If these were the only two options I'd leave suppression enabled, but master branch includes these useful tokens without this hallucination problem.

It might be helpful to compare the output of a branch with the other fixes of this PR excluding the non-speech token changes, or at least have a way to turn those completely off and go back to master branch behavior.

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bobqianic commented Feb 10, 2024

I looked at the code and this is unrelated to the non-speech token changes, right?

Yes. In situations where the model exhibits hallucinations with high confidence (avg_log_probs), this non-speech token approach will not be effective. The heuristic repetition check that I've implemented serves as a workaround for the compression ratio check. Implementing compression in C++ can be challenging without using third-party libraries. In the official implementation by OpenAI, both the compression ratio and non-speech tokens anti-hallucination mechanisms are utilized.

[00:57:11.700 --> 00:57:14.700] (c) 2014 University of Georgia College of Agricultural and Environmental Sciences UGA Extension Office of Communications and Creative Services

Which branch are you using? I can't find the hallucinations you mentioned.

large-v2

image

@jettoblack
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Which branch are you using? I can't find the hallucinations you mentioned.

I was using this PR @ 476dff4, unless I did something wrong, but this was on a Mac using the Metal gpu backend so that could make a difference. I'll retest on CPU and CUDA shortly and let you know.

@ukolovda
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ukolovda commented Feb 16, 2024

Hi!

@bobqianic new version is very robust!

On my test files, main branch emit 10 hallucinations on 26 WAV files (model ggml-large-v2, russian language).
With this PR it give only 2 hallucination. It is very fine result!!!

But example/server doesn't work at all, both CPU and CUDA versions. It returns empty text without any errors.
I try patch and append parameters (heuristic and other), but it not help. With --print-progress it print progress, but not result text.

Also it give the error on specific file:
500 Internal Server Error map::at

What can we do for fix it, how do you think?

Run server command:

/usr/src/whisper.cpp-bobqianic/server -m ../../models/ggml-large-v2.bin -l ru --print-progress --print-realtime -nt -nf

whisper_init_from_file_with_params_no_state: loading model from '../../models/ggml-large-v2.bin'
whisper_model_load: loading model
whisper_model_load: n_vocab       = 51865
whisper_model_load: n_audio_ctx   = 1500
whisper_model_load: n_audio_state = 1280
whisper_model_load: n_audio_head  = 20
whisper_model_load: n_audio_layer = 32
whisper_model_load: n_text_ctx    = 448
whisper_model_load: n_text_state  = 1280
whisper_model_load: n_text_head   = 20
whisper_model_load: n_text_layer  = 32
whisper_model_load: n_mels        = 80
whisper_model_load: ftype         = 1
whisper_model_load: qntvr         = 0
whisper_model_load: type          = 5 (large)
whisper_model_load: adding 1608 extra tokens
whisper_model_load: n_langs       = 99
ggml_init_cublas: GGML_CUDA_FORCE_MMQ:   no
ggml_init_cublas: CUDA_USE_TENSOR_CORES: yes
ggml_init_cublas: found 1 CUDA devices:
  Device 0: NVIDIA GeForce RTX 4070, compute capability 8.9, VMM: yes
whisper_backend_init: using CUDA backend
whisper_model_load:    CUDA0 total size =  3094.49 MB (3 buffers)
whisper_model_load: model size    = 3093.99 MB
whisper_backend_init: using CUDA backend
whisper_init_state: kv self size  =  220.20 MB
whisper_init_state: kv cross size =  245.76 MB
whisper_init_state: compute buffer (conv)   =   33.91 MB
whisper_init_state: compute buffer (encode) =  233.50 MB
whisper_init_state: compute buffer (cross)  =   10.15 MB
whisper_init_state: compute buffer (decode) =  108.99 MB

whisper server listening at http://127.0.0.1:8080

Received request: 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-14.wav
Successfully loaded 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-14.wav

system_info: n_threads = 4 / 12 | AVX = 1 | AVX2 = 1 | AVX512 = 0 | FMA = 1 | NEON = 0 | ARM_FMA = 0 | METAL = 0 | F16C = 1 | FP16_VA = 0 | WASM_SIMD = 0 | BLAS = 1 | SSE3 = 1 | SSSE3 = 1 | VSX = 0 | CUDA = 1 | COREML = 0 | OPENVINO = 0 | 

operator(): processing '0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-14.wav' (168960 samples, 10.6 sec), 4 threads, 1 processors, lang = ru, task = transcribe, timestamps = 0 ...

Running whisper.cpp inference on 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-14.wav
Received request: 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-15.wav
Successfully loaded 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-15.wav

system_info: n_threads = 4 / 12 | AVX = 1 | AVX2 = 1 | AVX512 = 0 | FMA = 1 | NEON = 0 | ARM_FMA = 0 | METAL = 0 | F16C = 1 | FP16_VA = 0 | WASM_SIMD = 0 | BLAS = 1 | SSE3 = 1 | SSSE3 = 1 | VSX = 0 | CUDA = 1 | COREML = 0 | OPENVINO = 0 | 

operator(): processing '0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-15.wav' (235200 samples, 14.7 sec), 4 threads, 1 processors, lang = ru, task = transcribe, timestamps = 0 ...

Running whisper.cpp inference on 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-15.wav

whisper_print_progress_callback: progress = 204%
Received request: 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-16.wav
Successfully loaded 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-16.wav

system_info: n_threads = 4 / 12 | AVX = 1 | AVX2 = 1 | AVX512 = 0 | FMA = 1 | NEON = 0 | ARM_FMA = 0 | METAL = 0 | F16C = 1 | FP16_VA = 0 | WASM_SIMD = 0 | BLAS = 1 | SSE3 = 1 | SSSE3 = 1 | VSX = 0 | CUDA = 1 | COREML = 0 | OPENVINO = 0 | 

operator(): processing '0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-16.wav' (512000 samples, 32.0 sec), 4 threads, 1 processors, lang = ru, task = transcribe, timestamps = 0 ...

Running whisper.cpp inference on 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-16.wav

whisper_print_progress_callback: progress =  93%

whisper_print_progress_callback: progress = 187%
Received request: 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-18.wav
Successfully loaded 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-18.wav

system_info: n_threads = 4 / 12 | AVX = 1 | AVX2 = 1 | AVX512 = 0 | FMA = 1 | NEON = 0 | ARM_FMA = 0 | METAL = 0 | F16C = 1 | FP16_VA = 0 | WASM_SIMD = 0 | BLAS = 1 | SSE3 = 1 | SSSE3 = 1 | VSX = 0 | CUDA = 1 | COREML = 0 | OPENVINO = 0 | 

operator(): processing '0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-18.wav' (115520 samples, 7.2 sec), 4 threads, 1 processors, lang = ru, task = transcribe, timestamps = 0 ...

Running whisper.cpp inference on 0f3657ce-6352-4cbb-a88f-b39dc6a37a34-1-18.wav

whisper_print_progress_callback: progress = 416%
...

Send file command:

curl localhost:8080/inference -H "Content-Type: multipart/form-data" -F file="@${filename}"

git diff result:

diff --git a/examples/server/server.cpp b/examples/server/server.cpp
index cf0157d..5030e87 100644
--- a/examples/server/server.cpp
+++ b/examples/server/server.cpp
@@ -64,6 +64,7 @@ struct whisper_params {
     float word_thold      =  0.01f;
     float entropy_thold   =  2.40f;
     float logprob_thold   = -1.00f;
+    float no_speech_thold =  0.60f;
     float temperature     =  0.00f;
     float temperature_inc =  0.20f;
 
@@ -78,6 +79,8 @@ struct whisper_params {
     bool print_realtime  = false;
     bool print_progress  = false;
     bool no_timestamps   = false;
+    bool suppress_nst    = true;  // suppress non speech tokens
+    bool heuristic       = true;
     bool use_gpu         = true;
 
     std::string language        = "en";
@@ -183,7 +186,10 @@ bool whisper_params_parse(int argc, char ** argv, whisper_params & params, serve
         else if (arg == "-wt"   || arg == "--word-thold")      { params.word_thold      = std::stof(argv[++i]); }
         else if (arg == "-et"   || arg == "--entropy-thold")   { params.entropy_thold   = std::stof(argv[++i]); }
         else if (arg == "-lpt"  || arg == "--logprob-thold")   { params.logprob_thold   = std::stof(argv[++i]); }
+        else if (arg == "-nst"  || arg == "--nospeech-thold")  { params.no_speech_thold = std::stof(argv[++i]); }
         // else if (arg == "-su"   || arg == "--speed-up")        { params.speed_up        = true; }
+        else if (arg == "-nsnst"|| arg == "--no-suppress-nst") { params.suppress_nst    = false; }
+        else if (arg == "-nh"   || arg == "--no-heuristic")    { params.heuristic       = false; }
         else if (arg == "-tr"   || arg == "--translate")       { params.translate       = true; }
         else if (arg == "-di"   || arg == "--diarize")         { params.diarize         = true; }
         else if (arg == "-tdrz" || arg == "--tinydiarize")     { params.tinydiarize     = true; }
@@ -726,6 +732,7 @@ int main(int argc, char ** argv) {
             wparams.max_len          = params.max_len == 0 ? 60 : params.max_len;
 
             wparams.speed_up         = params.speed_up;
+wparams.heuristic = params.heuristic;
 
             wparams.tdrz_enable      = params.tinydiarize; // [TDRZ]
 
@@ -738,8 +745,11 @@ int main(int argc, char ** argv) {
             wparams.temperature_inc  = params.temperature_inc;
             wparams.entropy_thold    = params.entropy_thold;
             wparams.logprob_thold    = params.logprob_thold;
+wparams.no_speech_thold = params.no_speech_thold;
 
             wparams.no_timestamps    = params.no_timestamps;
+wparams.suppress_non_speech_tokens = params.suppress_nst;
+
             wparams.token_timestamps = !params.no_timestamps && params.response_format == vjson_format;
 
             whisper_print_user_data user_data = { &params, &pcmf32s, 0 };

Thank you!

@felrock
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felrock commented Feb 17, 2024

Hello @ukolovda I took a look at this yesterday evening. Whats missing in server.cpp is what you mentioned:

  • heuristics
  • supress_nst
  • no_speech_thold

I got an output in the terminal by circumventing the print_realtime flag(instead of using a callback segment). So the model does in fact generate the output string but for some unknown reason whisper_full_n_segments(ctx) returns 0. I try to check this a bit more tomorrow.

@ukolovda
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I got an output in the terminal by circumventing the print_realtime flag(instead of using a callback segment). So the model does in fact generate the output string but for some unknown reason whisper_full_n_segments(ctx) returns 0.

Hello, @felrock !

Thank you!

@ukolovda
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ukolovda commented Feb 20, 2024

Append issue with zero-filled WAV.
#1881

@ukolovda
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ukolovda commented Feb 20, 2024

File from #1881 (zero filled WAV) give a gallucination in this version too.

$ ../whisper.cpp-bobqianic/main -m ./models/ggml-large-v3.bin -l ru --threads 8 -mc 0 samples/zeroes.wav
whisper_init_from_file_with_params_no_state: loading model from './models/ggml-large-v3.bin'
whisper_model_load: loading model
whisper_model_load: n_vocab       = 51866
whisper_model_load: n_audio_ctx   = 1500
whisper_model_load: n_audio_state = 1280
whisper_model_load: n_audio_head  = 20
whisper_model_load: n_audio_layer = 32
whisper_model_load: n_text_ctx    = 448
whisper_model_load: n_text_state  = 1280
whisper_model_load: n_text_head   = 20
whisper_model_load: n_text_layer  = 32
whisper_model_load: n_mels        = 128
whisper_model_load: ftype         = 1
whisper_model_load: qntvr         = 0
whisper_model_load: type          = 5 (large v3)
whisper_model_load: adding 1609 extra tokens
whisper_model_load: n_langs       = 100
ggml_init_cublas: GGML_CUDA_FORCE_MMQ:   no
ggml_init_cublas: CUDA_USE_TENSOR_CORES: yes
ggml_init_cublas: found 1 CUDA devices:
  Device 0: NVIDIA GeForce RTX 4070, compute capability 8.9, VMM: yes
whisper_backend_init: using CUDA backend
whisper_model_load:    CUDA0 total size =  3094,86 MB (3 buffers)
whisper_model_load: model size    = 3094,36 MB
whisper_backend_init: using CUDA backend
whisper_init_state: kv self size  =  220,20 MB
whisper_init_state: kv cross size =  245,76 MB
whisper_init_state: compute buffer (conv)   =   35,50 MB
whisper_init_state: compute buffer (encode) =  233,50 MB
whisper_init_state: compute buffer (cross)  =   10,15 MB
whisper_init_state: compute buffer (decode) =  108,99 MB

system_info: n_threads = 8 / 12 | AVX = 1 | AVX2 = 1 | AVX512 = 0 | FMA = 1 | NEON = 0 | ARM_FMA = 0 | METAL = 0 | F16C = 1 | FP16_VA = 0 | WASM_SIMD = 0 | BLAS = 1 | SSE3 = 1 | SSSE3 = 1 | VSX = 0 | CUDA = 1 | COREML = 0 | OPENVINO = 0 | 

run: processing 'samples/zeroes.wav' (19200 samples, 1,2 sec), 8 threads, 1 processors, 5 beams + best of 5, lang = ru, task = transcribe, timestamps = 1 ...


[00:00:00.000 --> 00:00:29.980]   Продолжение следует...


whisper_print_timings:     load time =   781,61 ms
whisper_print_timings:     fallbacks =   0 p /   0 h
whisper_print_timings:      mel time =     4,81 ms
whisper_print_timings:   sample time =    28,10 ms /    79 runs (    0,36 ms per run)
whisper_print_timings:   encode time =   162,31 ms /     1 runs (  162,31 ms per run)
whisper_print_timings:   decode time =     0,00 ms /     1 runs (    0,00 ms per run)
whisper_print_timings:   batchd time =   482,89 ms /    77 runs (    6,27 ms per run)
whisper_print_timings:   prompt time =     0,00 ms /     1 runs (    0,00 ms per run)
whisper_print_timings:    total time =  1502,74 ms

@linmi
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linmi commented Feb 21, 2024

-output-json-full has problems with the output format.

  • Language: Chinese
    CleanShot 2024-02-21 at 16 41 35@2x

@thewh1teagle
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What's the status of this PR? is it safe to use?
I experience decoding issues
thewh1teagle/vibe#34

@jwijffels
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jwijffels commented Apr 5, 2024

I'm thinking about including this pull request in the R wrapper at audio.whisper . There the current approach to handle some of the hallucinations is to use R packages audio.vadwebrtc or audio.vadsilero to detect silences or general non-voiced signals and either

  • instead of looping over different files in the main loop, loop over the detected non-silence sections in the audio.
  • or create a new audio file with only the voiced audio and recompute the timestamps later on by adding what was left out

I haven't looked into the extreme details on this pull request (only skimmed through the logic which was changed in main.cpp and whisper.cpp) but would it make sense already to incorporate this pull request in audio.whisper or are there a lot of changes to be expected here or is this pull request going to be split into a BPE change (#1854) and a change regarding how to handle non-speech?

@ronyfadel
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@bobqianic are you pursuing this at the moment?

@bobqianic
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@bobqianic are you pursuing this at the moment?

No, at least not in May. I'm really tied up with a lot of things this month.

@bygreencn
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I'm thinking about including this pull request in the R wrapper at audio.whisper . There the current approach to handle some of the hallucinations is to use R packages audio.vadwebrtc or audio.vadsilero to detect silences or general non-voiced signals and either

  • instead of looping over different files in the main loop, loop over the detected non-silence sections in the audio.
  • or create a new audio file with only the voiced audio and recompute the timestamps later on by adding what was left out

I haven't looked into the extreme details on this pull request (only skimmed through the logic which was changed in main.cpp and whisper.cpp) but would it make sense already to incorporate this pull request in audio.whisper or are there a lot of changes to be expected here or is this pull request going to be split into a BPE change (#1854) and a change regarding how to handle non-speech?

The best way to include Silero Voice Activity into whisper.cpp is to add thirdparty package of onnxruntime1.12.1 dll, then call silero onnx model. My branch had added it. Even VAD, the hallucinations on silent intervals is also happenning.

@IntendedConsequence
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The best way to include Silero Voice Activity into whisper.cpp is to add thirdparty package of onnxruntime1.12.1 dll, then call silero onnx model. My branch had added it. Even VAD, the hallucinations on silent intervals is also happenning.

I recommend considering a previous Silero VAD version, namely v3.1. The current version v4 (at the moment of writing) often hallucinates speech on lengthy chunks of silent or near-silent audio segments.
snakers4/silero-vad#369
snakers4/silero-vad#396

But you have to add a heavyweight dependency like onnxruntime just to run a 750KB model. The smallest size I could possibly reduce onnxruntime.dll to was about 2.2MB, which is still 3x the size of silero weights, and requires a lengthy custom build of onnxruntime from source with reduced operator set configs and other size reduction options. And prebuilt redistributables are easily 5-9 MB or more.

I have a working Silero v3.1 implementation in pure C, but as much as I would like to suggest it as an option, the code is quite bad, I wrote it as a personal project for learning low level neural nets.

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