🗣️
aspeak
A simple text-to-speech client using azure TTS API(trial).
TL;DR: This program uses trial auth token of Azure Cognitive Services to do speech synthesis for you.
You can try the Azure TTS API online: https://azure.microsoft.com/en-us/services/cognitive-services/text-to-speech
Installation
$ pip install --upgrade aspeak
Limitations
Since we are using Azure Cognitive Services, there are some limitations:
| Quota | Free (F0)3 |
|---|---|
| Max number of transactions per certain time period per Speech service resource | |
| Real-time API. Prebuilt neural voices and custom neural voices. | 20 transactions per 60 seconds |
| Adjustable | No4 |
| HTTP-specific quotas | |
| Max audio length produced per request | 10 min |
Max total number of distinct <voice> and <audio> tags in SSML |
50 |
| Websocket specific quotas | |
| Max audio length produced per turn | 10 min |
Max total number of distinct <voice> and <audio> tags in SSML |
50 |
| Max SSML message size per turn | 64 KB |
This table is copied from Azure Cognitive Services documentation
And the limitations may be subject to change. The table above might become outdated in the future. Please refer to the latest Azure Cognitive Services documentation for the latest information.
Attention: If the result audio is longer than 10 minutes, the audio will be truncated to 10 minutes and the program will not report an error.
Using aspeak as a Python library
See DEVELOP.md for more details. You can find examples in src/examples.
Usage
usage: aspeak [-h] [-V | -L | -Q | [-t [TEXT] [-p PITCH] [-r RATE] [-S STYLE] [-R ROLE] [-d STYLE_DEGREE] | -s [SSML]]]
[-f FILE] [-e ENCODING] [-o OUTPUT_PATH] [-l LOCALE] [-v VOICE]
[--mp3 [-q QUALITY] | --ogg [-q QUALITY] | --webm [-q QUALITY] | --wav [-q QUALITY] | -F FORMAT]
This program uses trial auth token of Azure Cognitive Services to do speech synthesis for you
options:
-h, --help show this help message and exit
-V, --version show program's version number and exit
-L, --list-voices list available voices, you can combine this argument with -v and -l
-Q, --list-qualities-and-formats
list available qualities and formats
-t [TEXT], --text [TEXT]
Text to speak. Left blank when reading from file/stdin
-s [SSML], --ssml [SSML]
SSML to speak. Left blank when reading from file/stdin
-f FILE, --file FILE Text/SSML file to speak, default to `-`(stdin)
-e ENCODING, --encoding ENCODING
Text/SSML file encoding, default to "utf-8"(Not for stdin!)
-o OUTPUT_PATH, --output OUTPUT_PATH
Output file path, wav format by default
--mp3 Use mp3 format for output. (Only works when outputting to a file)
--ogg Use ogg format for output. (Only works when outputting to a file)
--webm Use webm format for output. (Only works when outputting to a file)
--wav Use wav format for output
-F FORMAT, --format FORMAT
Set output audio format (experts only)
-l LOCALE, --locale LOCALE
Locale to use, default to en-US
-v VOICE, --voice VOICE
Voice to use
-q QUALITY, --quality QUALITY
Output quality, default to 0
Options for --text:
-p PITCH, --pitch PITCH
Set pitch, default to 0. Valid values include floats(will be converted to percentages), percentages such as 20% and -10%, absolute values like 300Hz, and
relative values like -20Hz, +2st and string values like x-low. See the documentation for more details.
-r RATE, --rate RATE Set speech rate, default to 0. Valid values include floats(will be converted to percentages), percentages like -20%, floats with postfix "f" (e.g. 2f means
doubling the default speech rate), and string values like x-slow. See the documentation for more details.
-S STYLE, --style STYLE
Set speech style, default to "general"
-R {Girl,Boy,YoungAdultFemale,YoungAdultMale,OlderAdultFemale,OlderAdultMale,SeniorFemale,SeniorMale}, --role {Girl,Boy,YoungAdultFemale,YoungAdultMale,OlderAdultFemale,OlderAdultMale,SeniorFemale,SeniorMale}
Specifies the speaking role-play. This only works for some Chinese voices!
-d {values in range 0.01-2 (inclusive)}, --style-degree {values in range 0.01-2 (inclusive)}
Specifies the intensity of the speaking style.This only works for some Chinese voices!
Attention: If the result audio is longer than 10 minutes, the audio will be truncated to 10 minutes and the program will not report an error. Unreasonable high/low values for pitch
and rate will be clipped to reasonable values by Azure Cognitive Services.Please refer to the documentation for other limitations at
https://github.com/kxxt/aspeak/blob/main/README.md#limitations
- If you don't specify
-o, we will use your default speaker. - If you don't specify
-tor-s, we will assume-tis provided. - You must specify voice if you want to use special options for
--text.
Special Note for Pitch and Rate
rate: The speaking rate of the voice.- If you use a float value (say
0.5), the value will be multiplied by 100% and become50.00%. - You can use the following values as well:
x-slow,slow,medium,fast,x-fast,default. - You can also use percentage values directly:
+10%. - You can also use a relative float value (with
fpostfix),1.2f:- According to the Azure documentation,
- A relative value, expressed as a number that acts as a multiplier of the default.
- For example, a value of
1fresults in no change in the rate. A value of0.5fresults in a halving of the rate. A value of3fresults in a tripling of the rate.
- If you use a float value (say
pitch: The pitch of the voice.- If you use a float value (say
-0.5), the value will be multiplied by 100% and become-50.00%. - You can also use the following values as well:
x-low,low,medium,high,x-high,default. - You can also use percentage values directly:
+10%. - You can also use a relative value, (e.g.
-2stor+80Hz):- According to the Azure documentation,
- A relative value, expressed as a number preceded by "+" or "-" and followed by "Hz" or "st" that specifies an amount to change the pitch.
- The "st" indicates the change unit is semitone, which is half of a tone (a half step) on the standard diatonic scale.
- You can also use an absolute value: e.g.
600Hz
- If you use a float value (say
Note: Unreasonable high/low values will be clipped to reasonable values by Azure Cognitive Services.
About Custom Style Degree and Role
According to the Azure documentation , style degree specifies the intensity of the speaking style. It is a floating point number between 0.01 and 2, inclusive.
At the time of writing, style degree adjustments are supported for Chinese (Mandarin, Simplified) neural voices.
According to the Azure documentation , role specifies the speaking role-play. The voice acts as a different age and gender, but the voice name isn't changed.
At the time of writing, role adjustments are supported for these Chinese (Mandarin, Simplified) neural voices: zh-CN-XiaomoNeural, zh-CN-XiaoxuanNeural, zh-CN-YunxiNeural, and zh-CN-YunyeNeural.
Examples
Speak "Hello, world!" to default speaker.
$ aspeak -t "Hello, world"
List all available voices.
$ aspeak -L
List all available voices for Chinese.
$ aspeak -L -l zh-CN
Get information about a voice.
$ aspeak -L -v en-US-SaraNeural
Output
Microsoft Server Speech Text to Speech Voice (en-US, SaraNeural)
Display Name: Sara
Local Name: Sara @ en-US
Locale: English (United States)
Gender: Female
ID: en-US-SaraNeural
Styles: ['cheerful', 'angry', 'sad']
Voice Type: Neural
Status: GA
Save synthesized speech to a file.
$ aspeak -t "Hello, world" -o output.wav
If you prefer mp3/ogg/webm, you can use --mp3/--ogg/--webm option.
$ aspeak -t "Hello, world" -o output.mp3 --mp3
$ aspeak -t "Hello, world" -o output.ogg --ogg
$ aspeak -t "Hello, world" -o output.webm --webm
List available quality levels and formats
$ aspeak -Q
Output
Available qualities:
Qualities for wav:
-2: Riff8Khz16BitMonoPcm
-1: Riff16Khz16BitMonoPcm
0: Riff24Khz16BitMonoPcm
1: Riff24Khz16BitMonoPcm
Qualities for mp3:
-3: Audio16Khz32KBitRateMonoMp3
-2: Audio16Khz64KBitRateMonoMp3
-1: Audio16Khz128KBitRateMonoMp3
0: Audio24Khz48KBitRateMonoMp3
1: Audio24Khz96KBitRateMonoMp3
2: Audio24Khz160KBitRateMonoMp3
3: Audio48Khz96KBitRateMonoMp3
4: Audio48Khz192KBitRateMonoMp3
Qualities for ogg:
-1: Ogg16Khz16BitMonoOpus
0: Ogg24Khz16BitMonoOpus
1: Ogg48Khz16BitMonoOpus
Qualities for webm:
-1: Webm16Khz16BitMonoOpus
0: Webm24Khz16BitMonoOpus
1: Webm24Khz16Bit24KbpsMonoOpus
Available formats:
- Riff8Khz16BitMonoPcm
- Riff16Khz16BitMonoPcm
- Audio16Khz128KBitRateMonoMp3
- Raw24Khz16BitMonoPcm
- Raw48Khz16BitMonoPcm
- Raw16Khz16BitMonoPcm
- Audio24Khz160KBitRateMonoMp3
- Ogg24Khz16BitMonoOpus
- Audio16Khz64KBitRateMonoMp3
- Raw8Khz8BitMonoALaw
- Audio24Khz16Bit48KbpsMonoOpus
- Ogg16Khz16BitMonoOpus
- Riff8Khz8BitMonoALaw
- Riff8Khz8BitMonoMULaw
- Audio48Khz192KBitRateMonoMp3
- Raw8Khz16BitMonoPcm
- Audio24Khz48KBitRateMonoMp3
- Raw24Khz16BitMonoTrueSilk
- Audio24Khz16Bit24KbpsMonoOpus
- Audio24Khz96KBitRateMonoMp3
- Webm24Khz16BitMonoOpus
- Ogg48Khz16BitMonoOpus
- Riff48Khz16BitMonoPcm
- Webm24Khz16Bit24KbpsMonoOpus
- Raw8Khz8BitMonoMULaw
- Audio16Khz16Bit32KbpsMonoOpus
- Audio16Khz32KBitRateMonoMp3
- Riff24Khz16BitMonoPcm
- Raw16Khz16BitMonoTrueSilk
- Audio48Khz96KBitRateMonoMp3
- Webm16Khz16BitMonoOpus
Increase/Decrease audio qualities
# Less than default quality.
$ aspeak -t "Hello, world" -o output.mp3 --mp3 -q=-1
# Best quality for mp3
$ aspeak -t "Hello, world" -o output.mp3 --mp3 -q=3
Read text from file and speak it.
$ cat input.txt | aspeak
or
$ aspeak -f input.txt
with custom encoding:
$ aspeak -f input.txt -e gbk
Read from stdin and speak it.
$ aspeak
or (more verbose)
$ aspeak -f -
maybe you prefer:
$ aspeak -l zh-CN << EOF
我能吞下玻璃而不伤身体。
EOF
Speak Chinese.
$ aspeak -t "你好，世界！" -l zh-CN
Use a custom voice.
$ aspeak -t "你好，世界！" -v zh-CN-YunjianNeural
Custom pitch, rate and style
$ aspeak -t "你好，世界！" -v zh-CN-XiaoxiaoNeural -p 1.5 -r 0.5 -S sad
$ aspeak -t "你好，世界！" -v zh-CN-XiaoxiaoNeural -p=-10% -r=+5% -S cheerful
$ aspeak -t "你好，世界！" -v zh-CN-XiaoxiaoNeural -p=+40Hz -r=1.2f -S fearful
$ aspeak -t "你好，世界！" -v zh-CN-XiaoxiaoNeural -p=high -r=x-slow -S calm
$ aspeak -t "你好，世界！" -v zh-CN-XiaoxiaoNeural -p=+1st -r=-7% -S lyrical
Advanced Usage
Use a custom audio format for output
Note: When outputing to default speaker, using a non-wav format may lead to white noises.
$ aspeak -t "Hello World" -F Riff48Khz16BitMonoPcm -o high-quality.wav
About This Application
- I found Azure TTS can synthesize nearly authentic human voice, which is very interesting
😆 . - I wrote this program to learn Azure Cognitive Services.
- And I use this program daily, because
espeakandfestivaloutputs terrible😨 audio.- But I respect
🙌 their maintainers' work, both are good open source software and they can be used off-line.
- But I respect
- I hope you like it
❤️ .
Vulnerable Library - certifi-2022.9.24-py3-none-any.whl
CVSS | Dependency | Type | Fixed in (certifi version) | Remediation Available |
| ------------- | ------------- | ----- | ----- | ----- | ------------- | --- |
| 
Medium | 6.8 | certifi-2022.9.24-py3-none-any.whl | Direct | certifi - 2022.12.07 | ❌ |
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