#!/bin/bash
set -e
echo -e "\e[1;31monly if you see 'Test success !!!', the cli testing is successful\e[0m"
# Audio classification
wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/cat.wav https://paddlespeech.bj.bcebos.com/PaddleAudio/dog.wav
paddlespeech cls --input ./cat.wav --topk 10
# Punctuation_restoration
paddlespeech text --input 今天的天气真不错啊你下午有空吗我想约你一起去吃饭 --model ernie_linear_p3_wudao_fast
[ASR] support wav2vec2 command line and demo (#2658)
* wav2vec2_cli
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* Update RESULTS.md
* Update RESULTS.md
* Update base_commands.py
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
2 years ago
# Speech SSL
Cherry-pick to r1.4 branch (#3798)
* [TTS]add Diffsinger with opencpop dataset (#3005)
* Update requirements.txt
* fix vits reduce_sum's input/output dtype, test=tts (#3028)
* [TTS] add opencpop PWGAN example (#3031)
* add opencpop voc, test=tts
* soft link
* Update textnorm_test_cases.txt
* [TTS] add opencpop HIFIGAN example (#3038)
* add opencpop voc, test=tts
* soft link
* add opencpop hifigan, test=tts
* update
* fix dtype diff of last expand_v2 op of VITS (#3041)
* [ASR]add squeezeformer model (#2755)
* add squeezeformer model
* change CodeStyle, test=asr
* change CodeStyle, test=asr
* fix subsample rate error, test=asr
* merge classes as required, test=asr
* change CodeStyle, test=asr
* fix missing code, test=asr
* split code to new file, test=asr
* remove rel_shift, test=asr
* Update README.md
* Update README_cn.md
* Update README.md
* Update README_cn.md
* Update README.md
* fix input dtype of elementwise_mul op from bool to int64 (#3054)
* [TTS] add svs frontend (#3062)
* [TTS]clean starganv2 vc model code and add docstring (#2987)
* clean code
* add docstring
* [Doc] change define asr server config to chunk asr config, test=doc (#3067)
* Update README.md
* Update README_cn.md
* get music score, test=doc (#3070)
* [TTS]fix elementwise_floordiv's fill_constant (#3075)
* fix elementwise_floordiv's fill_constant
* add float converter for min_value in attention
* fix paddle2onnx's install version, install the newest paddle2onnx in run.sh (#3084)
* [TTS] update svs_music_score.md (#3085)
* rm unused dep, test=tts (#3097)
* Update bug-report-tts.md (#3120)
* [TTS]Fix VITS lite infer (#3098)
* [TTS]add starganv2 vc trainer (#3143)
* add starganv2 vc trainer
* fix StarGANv2VCUpdater and losses
* fix StarGANv2VCEvaluator
* add some typehint
* [TTS]【Hackathon + No.190】 + 模型复现:iSTFTNet (#3006)
* iSTFTNet implementation based on hifigan, not affect the function and execution of HIFIGAN
* modify the comment in iSTFT.yaml
* add the comments in hifigan
* iSTFTNet implementation based on hifigan, not affect the function and execution of HIFIGAN
* modify the comment in iSTFT.yaml
* add the comments in hifigan
* add iSTFTNet.md
* modify the format of iSTFTNet.md
* modify iSTFT.yaml and hifigan.py
* Format code using pre-commit
* modify hifigan.py,delete the unused self.istft_layer_id , move the self.output_conv behind else, change conv_post to output_conv
* update iSTFTNet_csmsc_ckpt.zip download link
* modify iSTFTNet.md
* modify hifigan.py and iSTFT.yaml
* modify iSTFTNet.md
* add function for generating srt file (#3123)
* add function for generating srt file
在原来websocket_client.py的基础上,增加了由wav或mp3格式的音频文件生成对应srt格式字幕文件的功能
* add function for generating srt file
在原来websocket_client.py的基础上,增加了由wav或mp3格式的音频文件生成对应srt格式字幕文件的功能
* keep origin websocket_client.py
恢复原本的websocket_client.py文件
* add generating subtitle function into README
* add generate subtitle funciton into README
* add subtitle generation function
* add subtitle generation function
* fix example/aishell local/train.sh if condition bug, test=asr (#3146)
* fix some preprocess bugs (#3155)
* add amp for U2 conformer.
* fix scaler save
* fix scaler save and load.
* mv scaler.unscale_ blow grad_clip.
* [TTS]add StarGANv2VC preprocess (#3163)
* [TTS] [黑客松]Add JETS (#3109)
* Update quick_start.md (#3175)
* [BUG] Fix progress bar unit. (#3177)
* Update quick_start_cn.md (#3176)
* [TTS]StarGANv2 VC fix some trainer bugs, add add reset_parameters (#3182)
* VITS learning rate revised, test=tts
* VITS learning rate revised, test=tts
* [s2t] mv dataset into paddlespeech.dataset (#3183)
* mv dataset into paddlespeech.dataset
* add aidatatang
* fix import
* Fix some typos. (#3178)
* [s2t] move s2t data preprocess into paddlespeech.dataset (#3189)
* move s2t data preprocess into paddlespeech.dataset
* avg model, compute wer, format rsl into paddlespeech.dataset
* fix format rsl
* fix avg ckpts
* Update pretrained model in README (#3193)
* [TTS]Fix losses of StarGAN v2 VC (#3184)
* VITS learning rate revised, test=tts
* VITS learning rate revised, test=tts
* add new aishell model for better CER.
* add readme
* [s2t] fix cli args to config (#3194)
* fix cli args to config
* fix train cli
* Update README.md
* [ASR] Support Hubert, fintuned on the librispeech dataset (#3088)
* librispeech hubert, test=asr
* librispeech hubert, test=asr
* hubert decode
* review
* copyright, notes, example related
* hubert cli
* pre-commit format
* fix conflicts
* fix conflicts
* doc related
* doc and train config
* librispeech.py
* support hubert cli
* [ASR] fix asr 0-d tensor. (#3214)
* Update README.md
* Update README.md
* fix: 🐛 修复服务端 python ASREngine 无法使用conformer_talcs模型 (#3230)
* fix: 🐛 fix python ASREngine not pass codeswitch
* docs: 📝 Update Docs
* 修改模型判断方式
* Adding WavLM implementation
* fix model m5s
* Code clean up according to comments in https://github.com/PaddlePaddle/PaddleSpeech/pull/3242
* fix error in tts/st
* Changed the path for the uploaded weight
* Update phonecode.py
# 固话的正则 错误修改
参考https://github.com/speechio/chinese_text_normalization/blob/master/python/cn_tn.py
固化的正则为:
pattern = re.compile(r"\D((0(10|2[1-3]|[3-9]\d{2})-?)?[1-9]\d{6,7})\D")
* Adapted wavlmASR model to pretrained weights and CLI
* Changed the MD5 of the pretrained tar file due to bug fixes
* Deleted examples/librispeech/asr5/format_rsl.py
* Update released_model.md
* Code clean up for CIs
* Fixed the transpose usages ignored before
* Update setup.py
* refactor mfa scripts
* Final cleaning; Modified SSL/infer.py and README for wavlm inclusion in model options
* updating readme and readme_cn
* remove tsinghua pypi
* Update setup.py (#3294)
* Update setup.py
* refactor rhy
* fix ckpt
* add dtype param for arange API. (#3302)
* add scripts for tts code switch
* add t2s assets
* more comment on tts frontend
* fix librosa==0.8.1 numpy==1.23.5 for paddleaudio align with this version
* move ssl into t2s.frontend; fix spk_id for 0-D tensor;
* add ssml unit test
* add en_frontend file
* add mix frontend test
* fix long text oom using ssml; filter comma; update polyphonic
* remove print
* hotfix english G2P
* en frontend unit text
* fix profiler (#3323)
* old grad clip has 0d tensor problem, fix it (#3334)
* update to py3.8
* remove fluid.
* add roformer
* fix bugs
* add roformer result
* support position interpolation for langer attention context windown length.
* RoPE with position interpolation
* rope for streaming decoding
* update result
* fix rotary embeding
* Update README.md
* fix weight decay
* fix develop view confict with model's
* Add XPU support for SpeedySpeech (#3502)
* Add XPU support for SpeedySpeech
* fix typos
* update description of nxpu
* Add XPU support for FastSpeech2 (#3514)
* Add XPU support for FastSpeech2
* optimize
* Update ge2e_clone.py (#3517)
修复在windows上的多空格错误
* Fix Readme. (#3527)
* Update README.md
* Update README_cn.md
* Update README_cn.md
* Update README.md
* FIX: Added missing imports
* FIX: Fixed the implementation of a special method
* 【benchmark】add max_mem_reserved for benchmark (#3604)
* fix profiler
* add max_mem_reserved for benchmark
* fix develop bug function:view to reshape (#3633)
* 【benchmark】fix gpu_mem unit (#3634)
* fix profiler
* add max_mem_reserved for benchmark
* fix benchmark
* 增加文件编码读取 (#3606)
Fixed #3605
* bugfix: audio_len should be 1D, no 0D, which will raise list index out (#3490)
of range error in the following decode process
Co-authored-by: Luzhenhui <luzhenhui@mqsz.com>
* Update README.md (#3532)
Fixed a typo
* fixed version for paddlepaddle. (#3701)
* fixed version for paddlepaddle.
* fix code style
* 【Fix Speech Issue No.5】issue 3444 transformation import error (#3779)
* fix paddlespeech.s2t.transform.transformation import error
* fix paddlespeech.s2t.transform import error
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug (#3786)
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug
* 【test】add cli test readme (#3784)
* add cli test readme
* fix code style
* 【test】fix test cli bug (#3793)
* add cli test readme
* fix code style
* fix bug
* Update setup.py (#3795)
* adapt view behavior change, fix KeyError. (#3794)
* adapt view behavior change, fix KeyError.
* fix readme demo run error.
* fixed opencc version
---------
Co-authored-by: liangym <34430015+lym0302@users.noreply.github.com>
Co-authored-by: TianYuan <white-sky@qq.com>
Co-authored-by: 夜雨飘零 <yeyupiaoling@foxmail.com>
Co-authored-by: zxcd <228587199@qq.com>
Co-authored-by: longRookie <68834517+longRookie@users.noreply.github.com>
Co-authored-by: twoDogy <128727742+twoDogy@users.noreply.github.com>
Co-authored-by: lemondy <lemondy9@gmail.com>
Co-authored-by: ljhzxc <33015549+ljhzxc@users.noreply.github.com>
Co-authored-by: PiaoYang <495384481@qq.com>
Co-authored-by: WongLaw <mailoflawrence@gmail.com>
Co-authored-by: Hui Zhang <zhtclz@foxmail.com>
Co-authored-by: Shuangchi He <34329208+Yulv-git@users.noreply.github.com>
Co-authored-by: TianHao Zhang <32243340+Zth9730@users.noreply.github.com>
Co-authored-by: guanyc <guanyc@gmail.com>
Co-authored-by: jiamingkong <kinetical@live.com>
Co-authored-by: zoooo0820 <zoooo0820@qq.com>
Co-authored-by: shuishu <990941859@qq.com>
Co-authored-by: LixinGuo <18510030324@126.com>
Co-authored-by: gmm <38800877+mmglove@users.noreply.github.com>
Co-authored-by: Wang Huan <wanghuan29@baidu.com>
Co-authored-by: Kai Song <50285351+USTCKAY@users.noreply.github.com>
Co-authored-by: skyboooox <zcj924@gmail.com>
Co-authored-by: fazledyn-or <ataf@openrefactory.com>
Co-authored-by: luyao-cv <1367355728@qq.com>
Co-authored-by: Color_yr <402067010@qq.com>
Co-authored-by: JeffLu <luzhenhui@gmail.com>
Co-authored-by: Luzhenhui <luzhenhui@mqsz.com>
Co-authored-by: satani99 <42287151+satani99@users.noreply.github.com>
Co-authored-by: mjxs <52824616+kk-2000@users.noreply.github.com>
Co-authored-by: Mattheliu <leonliuzx@outlook.com>
5 months ago
wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/en.wav
[ASR] support wav2vec2 command line and demo (#2658)
* wav2vec2_cli
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* Update RESULTS.md
* Update RESULTS.md
* Update base_commands.py
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
* wav2vec2 demo update: support different optimizer and lr_schedular, align mdoel, update input type, test=asr
2 years ago
paddlespeech ssl --task asr --lang en --input ./en.wav
paddlespeech ssl --task vector --lang en --input ./en.wav
# Speech_recognition
Cherry-pick to r1.4 branch (#3798)
* [TTS]add Diffsinger with opencpop dataset (#3005)
* Update requirements.txt
* fix vits reduce_sum's input/output dtype, test=tts (#3028)
* [TTS] add opencpop PWGAN example (#3031)
* add opencpop voc, test=tts
* soft link
* Update textnorm_test_cases.txt
* [TTS] add opencpop HIFIGAN example (#3038)
* add opencpop voc, test=tts
* soft link
* add opencpop hifigan, test=tts
* update
* fix dtype diff of last expand_v2 op of VITS (#3041)
* [ASR]add squeezeformer model (#2755)
* add squeezeformer model
* change CodeStyle, test=asr
* change CodeStyle, test=asr
* fix subsample rate error, test=asr
* merge classes as required, test=asr
* change CodeStyle, test=asr
* fix missing code, test=asr
* split code to new file, test=asr
* remove rel_shift, test=asr
* Update README.md
* Update README_cn.md
* Update README.md
* Update README_cn.md
* Update README.md
* fix input dtype of elementwise_mul op from bool to int64 (#3054)
* [TTS] add svs frontend (#3062)
* [TTS]clean starganv2 vc model code and add docstring (#2987)
* clean code
* add docstring
* [Doc] change define asr server config to chunk asr config, test=doc (#3067)
* Update README.md
* Update README_cn.md
* get music score, test=doc (#3070)
* [TTS]fix elementwise_floordiv's fill_constant (#3075)
* fix elementwise_floordiv's fill_constant
* add float converter for min_value in attention
* fix paddle2onnx's install version, install the newest paddle2onnx in run.sh (#3084)
* [TTS] update svs_music_score.md (#3085)
* rm unused dep, test=tts (#3097)
* Update bug-report-tts.md (#3120)
* [TTS]Fix VITS lite infer (#3098)
* [TTS]add starganv2 vc trainer (#3143)
* add starganv2 vc trainer
* fix StarGANv2VCUpdater and losses
* fix StarGANv2VCEvaluator
* add some typehint
* [TTS]【Hackathon + No.190】 + 模型复现:iSTFTNet (#3006)
* iSTFTNet implementation based on hifigan, not affect the function and execution of HIFIGAN
* modify the comment in iSTFT.yaml
* add the comments in hifigan
* iSTFTNet implementation based on hifigan, not affect the function and execution of HIFIGAN
* modify the comment in iSTFT.yaml
* add the comments in hifigan
* add iSTFTNet.md
* modify the format of iSTFTNet.md
* modify iSTFT.yaml and hifigan.py
* Format code using pre-commit
* modify hifigan.py,delete the unused self.istft_layer_id , move the self.output_conv behind else, change conv_post to output_conv
* update iSTFTNet_csmsc_ckpt.zip download link
* modify iSTFTNet.md
* modify hifigan.py and iSTFT.yaml
* modify iSTFTNet.md
* add function for generating srt file (#3123)
* add function for generating srt file
在原来websocket_client.py的基础上,增加了由wav或mp3格式的音频文件生成对应srt格式字幕文件的功能
* add function for generating srt file
在原来websocket_client.py的基础上,增加了由wav或mp3格式的音频文件生成对应srt格式字幕文件的功能
* keep origin websocket_client.py
恢复原本的websocket_client.py文件
* add generating subtitle function into README
* add generate subtitle funciton into README
* add subtitle generation function
* add subtitle generation function
* fix example/aishell local/train.sh if condition bug, test=asr (#3146)
* fix some preprocess bugs (#3155)
* add amp for U2 conformer.
* fix scaler save
* fix scaler save and load.
* mv scaler.unscale_ blow grad_clip.
* [TTS]add StarGANv2VC preprocess (#3163)
* [TTS] [黑客松]Add JETS (#3109)
* Update quick_start.md (#3175)
* [BUG] Fix progress bar unit. (#3177)
* Update quick_start_cn.md (#3176)
* [TTS]StarGANv2 VC fix some trainer bugs, add add reset_parameters (#3182)
* VITS learning rate revised, test=tts
* VITS learning rate revised, test=tts
* [s2t] mv dataset into paddlespeech.dataset (#3183)
* mv dataset into paddlespeech.dataset
* add aidatatang
* fix import
* Fix some typos. (#3178)
* [s2t] move s2t data preprocess into paddlespeech.dataset (#3189)
* move s2t data preprocess into paddlespeech.dataset
* avg model, compute wer, format rsl into paddlespeech.dataset
* fix format rsl
* fix avg ckpts
* Update pretrained model in README (#3193)
* [TTS]Fix losses of StarGAN v2 VC (#3184)
* VITS learning rate revised, test=tts
* VITS learning rate revised, test=tts
* add new aishell model for better CER.
* add readme
* [s2t] fix cli args to config (#3194)
* fix cli args to config
* fix train cli
* Update README.md
* [ASR] Support Hubert, fintuned on the librispeech dataset (#3088)
* librispeech hubert, test=asr
* librispeech hubert, test=asr
* hubert decode
* review
* copyright, notes, example related
* hubert cli
* pre-commit format
* fix conflicts
* fix conflicts
* doc related
* doc and train config
* librispeech.py
* support hubert cli
* [ASR] fix asr 0-d tensor. (#3214)
* Update README.md
* Update README.md
* fix: 🐛 修复服务端 python ASREngine 无法使用conformer_talcs模型 (#3230)
* fix: 🐛 fix python ASREngine not pass codeswitch
* docs: 📝 Update Docs
* 修改模型判断方式
* Adding WavLM implementation
* fix model m5s
* Code clean up according to comments in https://github.com/PaddlePaddle/PaddleSpeech/pull/3242
* fix error in tts/st
* Changed the path for the uploaded weight
* Update phonecode.py
# 固话的正则 错误修改
参考https://github.com/speechio/chinese_text_normalization/blob/master/python/cn_tn.py
固化的正则为:
pattern = re.compile(r"\D((0(10|2[1-3]|[3-9]\d{2})-?)?[1-9]\d{6,7})\D")
* Adapted wavlmASR model to pretrained weights and CLI
* Changed the MD5 of the pretrained tar file due to bug fixes
* Deleted examples/librispeech/asr5/format_rsl.py
* Update released_model.md
* Code clean up for CIs
* Fixed the transpose usages ignored before
* Update setup.py
* refactor mfa scripts
* Final cleaning; Modified SSL/infer.py and README for wavlm inclusion in model options
* updating readme and readme_cn
* remove tsinghua pypi
* Update setup.py (#3294)
* Update setup.py
* refactor rhy
* fix ckpt
* add dtype param for arange API. (#3302)
* add scripts for tts code switch
* add t2s assets
* more comment on tts frontend
* fix librosa==0.8.1 numpy==1.23.5 for paddleaudio align with this version
* move ssl into t2s.frontend; fix spk_id for 0-D tensor;
* add ssml unit test
* add en_frontend file
* add mix frontend test
* fix long text oom using ssml; filter comma; update polyphonic
* remove print
* hotfix english G2P
* en frontend unit text
* fix profiler (#3323)
* old grad clip has 0d tensor problem, fix it (#3334)
* update to py3.8
* remove fluid.
* add roformer
* fix bugs
* add roformer result
* support position interpolation for langer attention context windown length.
* RoPE with position interpolation
* rope for streaming decoding
* update result
* fix rotary embeding
* Update README.md
* fix weight decay
* fix develop view confict with model's
* Add XPU support for SpeedySpeech (#3502)
* Add XPU support for SpeedySpeech
* fix typos
* update description of nxpu
* Add XPU support for FastSpeech2 (#3514)
* Add XPU support for FastSpeech2
* optimize
* Update ge2e_clone.py (#3517)
修复在windows上的多空格错误
* Fix Readme. (#3527)
* Update README.md
* Update README_cn.md
* Update README_cn.md
* Update README.md
* FIX: Added missing imports
* FIX: Fixed the implementation of a special method
* 【benchmark】add max_mem_reserved for benchmark (#3604)
* fix profiler
* add max_mem_reserved for benchmark
* fix develop bug function:view to reshape (#3633)
* 【benchmark】fix gpu_mem unit (#3634)
* fix profiler
* add max_mem_reserved for benchmark
* fix benchmark
* 增加文件编码读取 (#3606)
Fixed #3605
* bugfix: audio_len should be 1D, no 0D, which will raise list index out (#3490)
of range error in the following decode process
Co-authored-by: Luzhenhui <luzhenhui@mqsz.com>
* Update README.md (#3532)
Fixed a typo
* fixed version for paddlepaddle. (#3701)
* fixed version for paddlepaddle.
* fix code style
* 【Fix Speech Issue No.5】issue 3444 transformation import error (#3779)
* fix paddlespeech.s2t.transform.transformation import error
* fix paddlespeech.s2t.transform import error
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug (#3786)
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug
* 【test】add cli test readme (#3784)
* add cli test readme
* fix code style
* 【test】fix test cli bug (#3793)
* add cli test readme
* fix code style
* fix bug
* Update setup.py (#3795)
* adapt view behavior change, fix KeyError. (#3794)
* adapt view behavior change, fix KeyError.
* fix readme demo run error.
* fixed opencc version
---------
Co-authored-by: liangym <34430015+lym0302@users.noreply.github.com>
Co-authored-by: TianYuan <white-sky@qq.com>
Co-authored-by: 夜雨飘零 <yeyupiaoling@foxmail.com>
Co-authored-by: zxcd <228587199@qq.com>
Co-authored-by: longRookie <68834517+longRookie@users.noreply.github.com>
Co-authored-by: twoDogy <128727742+twoDogy@users.noreply.github.com>
Co-authored-by: lemondy <lemondy9@gmail.com>
Co-authored-by: ljhzxc <33015549+ljhzxc@users.noreply.github.com>
Co-authored-by: PiaoYang <495384481@qq.com>
Co-authored-by: WongLaw <mailoflawrence@gmail.com>
Co-authored-by: Hui Zhang <zhtclz@foxmail.com>
Co-authored-by: Shuangchi He <34329208+Yulv-git@users.noreply.github.com>
Co-authored-by: TianHao Zhang <32243340+Zth9730@users.noreply.github.com>
Co-authored-by: guanyc <guanyc@gmail.com>
Co-authored-by: jiamingkong <kinetical@live.com>
Co-authored-by: zoooo0820 <zoooo0820@qq.com>
Co-authored-by: shuishu <990941859@qq.com>
Co-authored-by: LixinGuo <18510030324@126.com>
Co-authored-by: gmm <38800877+mmglove@users.noreply.github.com>
Co-authored-by: Wang Huan <wanghuan29@baidu.com>
Co-authored-by: Kai Song <50285351+USTCKAY@users.noreply.github.com>
Co-authored-by: skyboooox <zcj924@gmail.com>
Co-authored-by: fazledyn-or <ataf@openrefactory.com>
Co-authored-by: luyao-cv <1367355728@qq.com>
Co-authored-by: Color_yr <402067010@qq.com>
Co-authored-by: JeffLu <luzhenhui@gmail.com>
Co-authored-by: Luzhenhui <luzhenhui@mqsz.com>
Co-authored-by: satani99 <42287151+satani99@users.noreply.github.com>
Co-authored-by: mjxs <52824616+kk-2000@users.noreply.github.com>
Co-authored-by: Mattheliu <leonliuzx@outlook.com>
5 months ago
wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/zh.wav https://paddlespeech.bj.bcebos.com/PaddleAudio/ch_zh_mix.wav
paddlespeech asr --input ./zh.wav
paddlespeech asr --model conformer_aishell --input ./zh.wav
paddlespeech asr --model conformer_online_aishell --input ./zh.wav
paddlespeech asr --model conformer_online_wenetspeech --input ./zh.wav
paddlespeech asr --model conformer_online_multicn --input ./zh.wav
paddlespeech asr --model conformer_u2pp_online_wenetspeech --lang zh --input zh.wav
paddlespeech asr --model transformer_librispeech --lang en --input ./en.wav
paddlespeech asr --model deepspeech2offline_aishell --input ./zh.wav
paddlespeech asr --model deepspeech2online_wenetspeech --input ./zh.wav
paddlespeech asr --model deepspeech2online_aishell --input ./zh.wav
paddlespeech asr --model deepspeech2offline_librispeech --lang en --input ./en.wav
paddlespeech asr --model conformer_talcs --lang zh_en --codeswitch True --input ./ch_zh_mix.wav
# Support editing num_decoding_left_chunks
paddlespeech asr --model conformer_online_wenetspeech --num_decoding_left_chunks 3 --input ./zh.wav
# long audio restriction
{
wget -c https://paddlespeech.bj.bcebos.com/datasets/single_wav/zh/test_long_audio_01.wav
paddlespeech asr --model deepspeech2online_wenetspeech --input test_long_audio_01.wav -y
if [ $? -ne 255 ] ; then
echo -e "\e[1;31mTime restriction not passed\e[0m"
exit 1
fi
} &&
{
echo -e "\033[32mTime restriction passed\033[0m"
}
# Text To Speech
paddlespeech tts --input "你好,欢迎使用百度飞桨深度学习框架!"
paddlespeech tts --am speedyspeech_csmsc --input "你好,欢迎使用百度飞桨深度学习框架!"
paddlespeech tts --voc mb_melgan_csmsc --input "你好,欢迎使用百度飞桨深度学习框架!"
paddlespeech tts --voc style_melgan_csmsc --input "你好,欢迎使用百度飞桨深度学习框架!"
paddlespeech tts --voc pwgan_csmsc --input "你好,欢迎使用百度飞桨深度学习框架!"
paddlespeech tts --am fastspeech2_aishell3 --voc pwgan_aishell3 --input "你好,欢迎使用百度飞桨深度学习框架!" --spk_id 0
paddlespeech tts --am fastspeech2_aishell3 --voc hifigan_aishell3 --input "你好,欢迎使用百度飞桨深度学习框架!" --spk_id 0
paddlespeech tts --am fastspeech2_ljspeech --voc pwgan_ljspeech --lang en --input "Life was like a box of chocolates, you never know what you're gonna get."
paddlespeech tts --am fastspeech2_ljspeech --voc hifigan_ljspeech --lang en --input "Life was like a box of chocolates, you never know what you're gonna get."
paddlespeech tts --am fastspeech2_vctk --voc pwgan_vctk --input "Life was like a box of chocolates, you never know what you're gonna get." --lang en --spk_id 0
paddlespeech tts --am fastspeech2_vctk --voc hifigan_vctk --input "Life was like a box of chocolates, you never know what you're gonna get." --lang en --spk_id 0
paddlespeech tts --am tacotron2_csmsc --input "你好,欢迎使用百度飞桨深度学习框架!"
paddlespeech tts --am tacotron2_csmsc --voc wavernn_csmsc --input "你好,欢迎使用百度飞桨深度学习框架!"
paddlespeech tts --am tacotron2_ljspeech --voc pwgan_ljspeech --lang en --input "Life was like a box of chocolates, you never know what you're gonna get."
paddlespeech tts --am fastspeech2_male --voc pwgan_male --lang zh --input "你好,欢迎使用百度飞桨深度学习框架!"
paddlespeech tts --am fastspeech2_male --voc pwgan_male --lang en --input "Life was like a box of chocolates, you never know what you're gonna get."
paddlespeech tts --am fastspeech2_canton --voc pwgan_aishell3 --input "各个国家有各个国家嘅国歌" --lang canton --spk_id 10
# mix tts
# The `am` must be `fastspeech2_mix`!
# The `lang` must be `mix`!
# The voc must be chinese datasets' voc now!
# spk 174 is csmcc, spk 175 is ljspeech
paddlespeech tts --am fastspeech2_mix --voc hifigan_csmsc --lang mix --input "热烈欢迎您在 Discussions 中提交问题,并在 Issues 中指出发现的 bug。此外, 我们非常希望您参与到 Paddle Speech 的开发中!" --spk_id 174 --output mix_spk174.wav
paddlespeech tts --am fastspeech2_mix --voc hifigan_aishell3 --lang mix --input "热烈欢迎您在 Discussions 中提交问题,并在 Issues 中指出发现的 bug。此外, 我们非常希望您参与到 Paddle Speech 的开发中!" --spk_id 174 --output mix_spk174_aishell3.wav
paddlespeech tts --am fastspeech2_mix --voc pwgan_csmsc --lang mix --input "我们的声学模型使用了 Fast Speech Two, 声码器使用了 Parallel Wave GAN and Hifi GAN." --spk_id 175 --output mix_spk175_pwgan.wav
paddlespeech tts --am fastspeech2_mix --voc hifigan_csmsc --lang mix --input "我们的声学模型使用了 Fast Speech Two, 声码器使用了 Parallel Wave GAN and Hifi GAN." --spk_id 175 --output mix_spk175.wav
# male mix tts
paddlespeech tts --am fastspeech2_male --voc pwgan_male --lang mix --input "我们的声学模型使用了 Fast Speech Two, 声码器使用了 Parallel Wave GAN and Hifi GAN." --output male_mix_fs2_pwgan.wav
# Speech Translation (only support linux)
paddlespeech st --input ./en.wav
# Speaker Verification
wget -c https://paddlespeech.bj.bcebos.com/vector/audio/85236145389.wav
paddlespeech vector --task spk --input 85236145389.wav
# batch process
echo -e "1 欢迎光临。\n2 谢谢惠顾。" | paddlespeech tts
echo -e "demo1 85236145389.wav \n demo2 85236145389.wav" > vec.job
paddlespeech vector --task spk --input vec.job
echo -e "demo3 85236145389.wav \n demo4 85236145389.wav" | paddlespeech vector --task spk
rm 85236145389.wav
rm vec.job
# shell pipeline
paddlespeech asr --input ./zh.wav | paddlespeech text --task punc
# stats
paddlespeech stats --task asr
paddlespeech stats --task tts
paddlespeech stats --task cls
paddlespeech stats --task text
paddlespeech stats --task vector
paddlespeech stats --task st
# whisper text recognize
paddlespeech whisper --task transcribe --input ./zh.wav
# whisper recognize text and translate to English
paddlespeech whisper --task translate --input ./zh.wav
Cherry-pick to r1.4 branch (#3798)
* [TTS]add Diffsinger with opencpop dataset (#3005)
* Update requirements.txt
* fix vits reduce_sum's input/output dtype, test=tts (#3028)
* [TTS] add opencpop PWGAN example (#3031)
* add opencpop voc, test=tts
* soft link
* Update textnorm_test_cases.txt
* [TTS] add opencpop HIFIGAN example (#3038)
* add opencpop voc, test=tts
* soft link
* add opencpop hifigan, test=tts
* update
* fix dtype diff of last expand_v2 op of VITS (#3041)
* [ASR]add squeezeformer model (#2755)
* add squeezeformer model
* change CodeStyle, test=asr
* change CodeStyle, test=asr
* fix subsample rate error, test=asr
* merge classes as required, test=asr
* change CodeStyle, test=asr
* fix missing code, test=asr
* split code to new file, test=asr
* remove rel_shift, test=asr
* Update README.md
* Update README_cn.md
* Update README.md
* Update README_cn.md
* Update README.md
* fix input dtype of elementwise_mul op from bool to int64 (#3054)
* [TTS] add svs frontend (#3062)
* [TTS]clean starganv2 vc model code and add docstring (#2987)
* clean code
* add docstring
* [Doc] change define asr server config to chunk asr config, test=doc (#3067)
* Update README.md
* Update README_cn.md
* get music score, test=doc (#3070)
* [TTS]fix elementwise_floordiv's fill_constant (#3075)
* fix elementwise_floordiv's fill_constant
* add float converter for min_value in attention
* fix paddle2onnx's install version, install the newest paddle2onnx in run.sh (#3084)
* [TTS] update svs_music_score.md (#3085)
* rm unused dep, test=tts (#3097)
* Update bug-report-tts.md (#3120)
* [TTS]Fix VITS lite infer (#3098)
* [TTS]add starganv2 vc trainer (#3143)
* add starganv2 vc trainer
* fix StarGANv2VCUpdater and losses
* fix StarGANv2VCEvaluator
* add some typehint
* [TTS]【Hackathon + No.190】 + 模型复现:iSTFTNet (#3006)
* iSTFTNet implementation based on hifigan, not affect the function and execution of HIFIGAN
* modify the comment in iSTFT.yaml
* add the comments in hifigan
* iSTFTNet implementation based on hifigan, not affect the function and execution of HIFIGAN
* modify the comment in iSTFT.yaml
* add the comments in hifigan
* add iSTFTNet.md
* modify the format of iSTFTNet.md
* modify iSTFT.yaml and hifigan.py
* Format code using pre-commit
* modify hifigan.py,delete the unused self.istft_layer_id , move the self.output_conv behind else, change conv_post to output_conv
* update iSTFTNet_csmsc_ckpt.zip download link
* modify iSTFTNet.md
* modify hifigan.py and iSTFT.yaml
* modify iSTFTNet.md
* add function for generating srt file (#3123)
* add function for generating srt file
在原来websocket_client.py的基础上,增加了由wav或mp3格式的音频文件生成对应srt格式字幕文件的功能
* add function for generating srt file
在原来websocket_client.py的基础上,增加了由wav或mp3格式的音频文件生成对应srt格式字幕文件的功能
* keep origin websocket_client.py
恢复原本的websocket_client.py文件
* add generating subtitle function into README
* add generate subtitle funciton into README
* add subtitle generation function
* add subtitle generation function
* fix example/aishell local/train.sh if condition bug, test=asr (#3146)
* fix some preprocess bugs (#3155)
* add amp for U2 conformer.
* fix scaler save
* fix scaler save and load.
* mv scaler.unscale_ blow grad_clip.
* [TTS]add StarGANv2VC preprocess (#3163)
* [TTS] [黑客松]Add JETS (#3109)
* Update quick_start.md (#3175)
* [BUG] Fix progress bar unit. (#3177)
* Update quick_start_cn.md (#3176)
* [TTS]StarGANv2 VC fix some trainer bugs, add add reset_parameters (#3182)
* VITS learning rate revised, test=tts
* VITS learning rate revised, test=tts
* [s2t] mv dataset into paddlespeech.dataset (#3183)
* mv dataset into paddlespeech.dataset
* add aidatatang
* fix import
* Fix some typos. (#3178)
* [s2t] move s2t data preprocess into paddlespeech.dataset (#3189)
* move s2t data preprocess into paddlespeech.dataset
* avg model, compute wer, format rsl into paddlespeech.dataset
* fix format rsl
* fix avg ckpts
* Update pretrained model in README (#3193)
* [TTS]Fix losses of StarGAN v2 VC (#3184)
* VITS learning rate revised, test=tts
* VITS learning rate revised, test=tts
* add new aishell model for better CER.
* add readme
* [s2t] fix cli args to config (#3194)
* fix cli args to config
* fix train cli
* Update README.md
* [ASR] Support Hubert, fintuned on the librispeech dataset (#3088)
* librispeech hubert, test=asr
* librispeech hubert, test=asr
* hubert decode
* review
* copyright, notes, example related
* hubert cli
* pre-commit format
* fix conflicts
* fix conflicts
* doc related
* doc and train config
* librispeech.py
* support hubert cli
* [ASR] fix asr 0-d tensor. (#3214)
* Update README.md
* Update README.md
* fix: 🐛 修复服务端 python ASREngine 无法使用conformer_talcs模型 (#3230)
* fix: 🐛 fix python ASREngine not pass codeswitch
* docs: 📝 Update Docs
* 修改模型判断方式
* Adding WavLM implementation
* fix model m5s
* Code clean up according to comments in https://github.com/PaddlePaddle/PaddleSpeech/pull/3242
* fix error in tts/st
* Changed the path for the uploaded weight
* Update phonecode.py
# 固话的正则 错误修改
参考https://github.com/speechio/chinese_text_normalization/blob/master/python/cn_tn.py
固化的正则为:
pattern = re.compile(r"\D((0(10|2[1-3]|[3-9]\d{2})-?)?[1-9]\d{6,7})\D")
* Adapted wavlmASR model to pretrained weights and CLI
* Changed the MD5 of the pretrained tar file due to bug fixes
* Deleted examples/librispeech/asr5/format_rsl.py
* Update released_model.md
* Code clean up for CIs
* Fixed the transpose usages ignored before
* Update setup.py
* refactor mfa scripts
* Final cleaning; Modified SSL/infer.py and README for wavlm inclusion in model options
* updating readme and readme_cn
* remove tsinghua pypi
* Update setup.py (#3294)
* Update setup.py
* refactor rhy
* fix ckpt
* add dtype param for arange API. (#3302)
* add scripts for tts code switch
* add t2s assets
* more comment on tts frontend
* fix librosa==0.8.1 numpy==1.23.5 for paddleaudio align with this version
* move ssl into t2s.frontend; fix spk_id for 0-D tensor;
* add ssml unit test
* add en_frontend file
* add mix frontend test
* fix long text oom using ssml; filter comma; update polyphonic
* remove print
* hotfix english G2P
* en frontend unit text
* fix profiler (#3323)
* old grad clip has 0d tensor problem, fix it (#3334)
* update to py3.8
* remove fluid.
* add roformer
* fix bugs
* add roformer result
* support position interpolation for langer attention context windown length.
* RoPE with position interpolation
* rope for streaming decoding
* update result
* fix rotary embeding
* Update README.md
* fix weight decay
* fix develop view confict with model's
* Add XPU support for SpeedySpeech (#3502)
* Add XPU support for SpeedySpeech
* fix typos
* update description of nxpu
* Add XPU support for FastSpeech2 (#3514)
* Add XPU support for FastSpeech2
* optimize
* Update ge2e_clone.py (#3517)
修复在windows上的多空格错误
* Fix Readme. (#3527)
* Update README.md
* Update README_cn.md
* Update README_cn.md
* Update README.md
* FIX: Added missing imports
* FIX: Fixed the implementation of a special method
* 【benchmark】add max_mem_reserved for benchmark (#3604)
* fix profiler
* add max_mem_reserved for benchmark
* fix develop bug function:view to reshape (#3633)
* 【benchmark】fix gpu_mem unit (#3634)
* fix profiler
* add max_mem_reserved for benchmark
* fix benchmark
* 增加文件编码读取 (#3606)
Fixed #3605
* bugfix: audio_len should be 1D, no 0D, which will raise list index out (#3490)
of range error in the following decode process
Co-authored-by: Luzhenhui <luzhenhui@mqsz.com>
* Update README.md (#3532)
Fixed a typo
* fixed version for paddlepaddle. (#3701)
* fixed version for paddlepaddle.
* fix code style
* 【Fix Speech Issue No.5】issue 3444 transformation import error (#3779)
* fix paddlespeech.s2t.transform.transformation import error
* fix paddlespeech.s2t.transform import error
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug (#3786)
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug
* 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug
* 【test】add cli test readme (#3784)
* add cli test readme
* fix code style
* 【test】fix test cli bug (#3793)
* add cli test readme
* fix code style
* fix bug
* Update setup.py (#3795)
* adapt view behavior change, fix KeyError. (#3794)
* adapt view behavior change, fix KeyError.
* fix readme demo run error.
* fixed opencc version
---------
Co-authored-by: liangym <34430015+lym0302@users.noreply.github.com>
Co-authored-by: TianYuan <white-sky@qq.com>
Co-authored-by: 夜雨飘零 <yeyupiaoling@foxmail.com>
Co-authored-by: zxcd <228587199@qq.com>
Co-authored-by: longRookie <68834517+longRookie@users.noreply.github.com>
Co-authored-by: twoDogy <128727742+twoDogy@users.noreply.github.com>
Co-authored-by: lemondy <lemondy9@gmail.com>
Co-authored-by: ljhzxc <33015549+ljhzxc@users.noreply.github.com>
Co-authored-by: PiaoYang <495384481@qq.com>
Co-authored-by: WongLaw <mailoflawrence@gmail.com>
Co-authored-by: Hui Zhang <zhtclz@foxmail.com>
Co-authored-by: Shuangchi He <34329208+Yulv-git@users.noreply.github.com>
Co-authored-by: TianHao Zhang <32243340+Zth9730@users.noreply.github.com>
Co-authored-by: guanyc <guanyc@gmail.com>
Co-authored-by: jiamingkong <kinetical@live.com>
Co-authored-by: zoooo0820 <zoooo0820@qq.com>
Co-authored-by: shuishu <990941859@qq.com>
Co-authored-by: LixinGuo <18510030324@126.com>
Co-authored-by: gmm <38800877+mmglove@users.noreply.github.com>
Co-authored-by: Wang Huan <wanghuan29@baidu.com>
Co-authored-by: Kai Song <50285351+USTCKAY@users.noreply.github.com>
Co-authored-by: skyboooox <zcj924@gmail.com>
Co-authored-by: fazledyn-or <ataf@openrefactory.com>
Co-authored-by: luyao-cv <1367355728@qq.com>
Co-authored-by: Color_yr <402067010@qq.com>
Co-authored-by: JeffLu <luzhenhui@gmail.com>
Co-authored-by: Luzhenhui <luzhenhui@mqsz.com>
Co-authored-by: satani99 <42287151+satani99@users.noreply.github.com>
Co-authored-by: mjxs <52824616+kk-2000@users.noreply.github.com>
Co-authored-by: Mattheliu <leonliuzx@outlook.com>
5 months ago
# to change model English-Only model
paddlespeech whisper --lang en --size base --task transcribe --input ./en.wav
echo -e "\033[32mTest success !!!\033[0m"