path: root/python-streamlit-webrtc.spec

%global _empty_manifest_terminate_build 0
Name:		python-streamlit-webrtc
Version:	0.45.0
Release:	1
Summary:	please add a summary manually as the author left a blank one
License:	MIT
URL:		https://github.com/whitphx/streamlit-webrtc
Source0:	https://mirrors.nju.edu.cn/pypi/web/packages/41/1f/c82b7b3c72e5dc35217fed5a914d09214f9e0805e114614dbd63e4905fbb/streamlit_webrtc-0.45.0.tar.gz
BuildArch:	noarch

Requires:	python3-streamlit
Requires:	python3-aiortc
Requires:	python3-typing_extensions
Requires:	python3-packaging

%description
Create `app.py` with the content below.
```py
from streamlit_webrtc import webrtc_streamer
webrtc_streamer(key="sample")
```
Unlike other Streamlit components, `webrtc_streamer()` requires the `key` argument as a unique identifier. Set an arbitrary string to it.
Then run it with Streamlit and open http://localhost:8501/.
```shell
$ streamlit run app.py
```
You see the app view, so click the "START" button.
Then, video and audio streaming starts. If asked for permissions to access the camera and microphone, allow it.
![Basic example of streamlit-webrtc](./docs/images/streamlit_webrtc_basic.gif)
Next, edit `app.py` as below and run it again.
```py
from streamlit_webrtc import webrtc_streamer
import av
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    flipped = img[::-1,:,:]
    return av.VideoFrame.from_ndarray(flipped, format="bgr24")
webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
```
Now the video is vertically flipped.
![Vertically flipping example](./docs/images/streamlit_webrtc_flipped.gif)
As an example above, you can edit the video frames by defining a callback that receives and returns a frame and passing it to the `video_frame_callback` argument (or `audio_frame_callback` for audio manipulation).
The input and output frames are the instance of [`av.VideoFrame`](https://pyav.org/docs/develop/api/video.html#av.video.frame.VideoFrame) (or [`av.AudioFrame`](https://pyav.org/docs/develop/api/audio.html#av.audio.frame.AudioFrame) when dealing with audio) of [`PyAV` library](https://pyav.org/).
You can inject any kinds of image (or audio) processing inside the callback.
See examples above for more applications.
### Pass parameters to the callback
You can also pass parameters to the callback.
In the example below, a boolean `flip` flag is used to turn on/off the image flipping.
```python
import streamlit as st
from streamlit_webrtc import webrtc_streamer
import av
flip = st.checkbox("Flip")
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    flipped = img[::-1,:,:] if flip else img
    return av.VideoFrame.from_ndarray(flipped, format="bgr24")
webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
```
### Pull values from the callback
Sometimes we want to read the values generated in the callback from the outer scope.
Note that the callback is executed in a forked thread running independently of the main script, so we have to take care of the following points and need some tricks for implementation like the example below (See also the section below for some limitations in the callback due to multi-threading).
* Thread-safety
  * Passing the values between inside and outside the callback must be thread-safe.
* Using a loop to poll the values
  * During media streaming, while the callback continues to be called, the main script execution stops at the bottom as usual. So we need to use a loop to keep the main script running and get the values from the callback in the outer scope.
The following example is to pass the image frames from the callback to the outer scope and continuously process it in the loop. In this example, a simple image analysis (calculating the histogram like [this OpenCV tutorial](https://docs.opencv.org/4.x/d1/db7/tutorial_py_histogram_begins.html)) is done on the image frames.
[`threading.Lock`](https://docs.python.org/3/library/threading.html#lock-objects) is one standard way to control variable accesses across threads.
A dict object `img_container` here is a mutable container shared by the callback and the outer scope and the `lock` object is used at assigning and reading the values to/from the container for thread-safety.
```python
import threading
import cv2
import streamlit as st
from matplotlib import pyplot as plt
from streamlit_webrtc import webrtc_streamer
lock = threading.Lock()
img_container = {"img": None}
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    with lock:
        img_container["img"] = img
    return frame
ctx = webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
fig_place = st.empty()
fig, ax = plt.subplots(1, 1)
while ctx.state.playing:
    with lock:
        img = img_container["img"]
    if img is None:
        continue
    gray = cv2.cvtColor(img, cv2.COLOR_BGR2GRAY)
    ax.cla()
    ax.hist(gray.ravel(), 256, [0, 256])
    fig_place.pyplot(fig)
```
## Callback limitations
The callbacks are executed in forked threads different from the main one, so there are some limitations:
* Streamlit methods (`st.*` such as `st.write()`) do not work inside the callbacks.
* Variables inside the callbacks cannot be directly referred to from the outside.
* The `global` keyword does not work expectedly in the callbacks.
* You have to care about thread-safety when accessing the same objects both from outside and inside the callbacks as stated in the section above.
## Class-based callbacks
Until v0.37, the class-based callbacks were the standard.
See the [old version of the README](https://github.com/whitphx/streamlit-webrtc/blob/v0.37.0/README.md#quick-tutorial) about it.
## Serving from remote host
When deploying apps to remote servers, there are some things you need to be aware of.
### HTTPS
`streamlit-webrtc` uses [`getUserMedia()`](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia) API to access local media devices, and this method does not work in an insecure context.
[This document](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia#privacy_and_security) says
> A secure context is, in short, a page loaded using HTTPS or the file:/// URL scheme, or a page loaded from localhost.
So, when hosting your app on a remote server, it must be served via HTTPS if your app is using webcam or microphone.
If not, you will encounter an error when starting using the device. For example, it's something like below on Chrome.
> Error: navigator.mediaDevices is undefined. It seems the current document is not loaded securely.
[Streamlit Cloud](https://streamlit.io/cloud) is a recommended way for HTTPS serving. You can easily deploy Streamlit apps with it, and most importantly for this topic, it serves the apps via HTTPS automatically by defualt.
### Configure the STUN server
To deploy the app to the cloud, we have to configure the *STUN* server via the `rtc_configuration` argument on `webrtc_streamer()` like below.
```python
webrtc_streamer(
    # ...
    rtc_configuration={  # Add this config
        "iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]
    }
    # ...
)
```
This configuration is necessary to establish the media streaming connection when the server is on a remote host.
`streamlit_webrtc` uses WebRTC for its video and audio streaming. It has to access a "STUN server" in the global network for the remote peers (precisely, peers over the NATs) to establish WebRTC connections.
As we don't see the details about STUN servers here, please google it if interested with keywords such as STUN, TURN, or NAT traversal, or read these articles ([1](https://towardsdatascience.com/developing-web-based-real-time-video-audio-processing-apps-quickly-with-streamlit-7c7bcd0bc5a8#1cec), [2](https://dev.to/whitphx/python-webrtc-basics-with-aiortc-48id), [3](https://www.3cx.com/pbx/what-is-a-stun-server/)).
The example above is configured to use `stun.l.google.com:19302`, which is a free STUN server provided by Google.
You can also use any other STUN servers.
For example, [one user reported](https://github.com/whitphx/streamlit-webrtc/issues/283#issuecomment-889753789) that the Google's STUN server had a huge delay when using from China network, and the problem was solved by changing the STUN server.
For those who know about the browser WebRTC API: The value of the rtc_configuration argument will be passed to the [`RTCPeerConnection`](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection) constructor on the frontend.
### Configure the TURN server if necessary
Even if the STUN server is properly configured, media streaming may not work in some network environments.
For example, in some office or public networks, there are firewalls which drop the WebRTC packets.
In such environments, setting up a [TURN server](https://webrtc.org/getting-started/turn-server) is a solution. See https://github.com/whitphx/streamlit-webrtc/issues/335#issuecomment-897326755.
## Logging
For logging, this library uses the standard `logging` module and follows the practice described in [the official logging tutorial](https://docs.python.org/3/howto/logging.html#advanced-logging-tutorial). Then the logger names are the same as the module names - `streamlit_webrtc` or `streamlit_webrtc.*`.
So you can get the logger instance with `logging.getLogger("streamlit_webrtc")` through which you can control the logs from this library.
For example, if you want to set the log level on this library's logger as WARNING, you can use the following code.
```python
st_webrtc_logger = logging.getLogger("streamlit_webrtc")
st_webrtc_logger.setLevel(logging.WARNING)
```
In practice, `aiortc`, a third-party package this library is internally using, also emits many INFO level logs and you may want to control its logs too.
You can do it in the same way as below.
```python
aioice_logger = logging.getLogger("aioice")
aioice_logger.setLevel(logging.WARNING)
```
## API changes
Currently there is no documentation about the interface. See the examples in [./pages/*.py](./pages) for the usage.
The API is not finalized yet and can be changed without backward compatibility in the future releases until v1.0.
### For users since versions `<0.20`
`VideoTransformerBase` and its `transform` method have been marked as deprecated in v0.20.0. Please use `VideoProcessorBase#recv()` instead.
Note that the signature of the `recv` method is different from the `transform` in that the `recv` has to return an instance of `av.VideoFrame` or `av.AudioFrame`.
Also, `webrtc_streamer()`'s `video_transformer_factory` and `async_transform` arguments are deprecated, so use `video_processor_factory` and `async_processing` respectively.
See the samples in [app.py](./app.py) for their usage.
## Resources
* [Developing web-based real-time video/audio processing apps quickly with Streamlit](https://www.whitphx.info/posts/20211231-streamlit-webrtc-video-app-tutorial/)
  * A tutorial for real-time video app development using `streamlit-webrtc`.
  * Crosspost on dev.to: https://dev.to/whitphx/developing-web-based-real-time-videoaudio-processing-apps-quickly-with-streamlit-4k89
* [New Component: streamlit-webrtc, a new way to deal with real-time media streams (Streamlit Community)](https://discuss.streamlit.io/t/new-component-streamlit-webrtc-a-new-way-to-deal-with-real-time-media-streams/8669)
  * This is a forum topic where `streamlit-webrtc` has been introduced and discussed about.
## Support the project
[![ko-fi](https://ko-fi.com/img/githubbutton_sm.svg)](https://ko-fi.com/D1D2ERWFG)
<a href="https://www.buymeacoffee.com/whitphx" target="_blank" rel="noreferrer"><img src="https://cdn.buymeacoffee.com/buttons/v2/default-yellow.png" alt="Buy Me A Coffee" width="180" height="50" ></a>
[![GitHub Sponsors](https://img.shields.io/github/sponsors/whitphx?label=Sponsor%20me%20on%20GitHub%20Sponsors&style=social)](https://github.com/sponsors/whitphx)

%package -n python3-streamlit-webrtc
Summary:	please add a summary manually as the author left a blank one
Provides:	python-streamlit-webrtc
BuildRequires:	python3-devel
BuildRequires:	python3-setuptools
BuildRequires:	python3-pip
%description -n python3-streamlit-webrtc
Create `app.py` with the content below.
```py
from streamlit_webrtc import webrtc_streamer
webrtc_streamer(key="sample")
```
Unlike other Streamlit components, `webrtc_streamer()` requires the `key` argument as a unique identifier. Set an arbitrary string to it.
Then run it with Streamlit and open http://localhost:8501/.
```shell
$ streamlit run app.py
```
You see the app view, so click the "START" button.
Then, video and audio streaming starts. If asked for permissions to access the camera and microphone, allow it.
![Basic example of streamlit-webrtc](./docs/images/streamlit_webrtc_basic.gif)
Next, edit `app.py` as below and run it again.
```py
from streamlit_webrtc import webrtc_streamer
import av
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    flipped = img[::-1,:,:]
    return av.VideoFrame.from_ndarray(flipped, format="bgr24")
webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
```
Now the video is vertically flipped.
![Vertically flipping example](./docs/images/streamlit_webrtc_flipped.gif)
As an example above, you can edit the video frames by defining a callback that receives and returns a frame and passing it to the `video_frame_callback` argument (or `audio_frame_callback` for audio manipulation).
The input and output frames are the instance of [`av.VideoFrame`](https://pyav.org/docs/develop/api/video.html#av.video.frame.VideoFrame) (or [`av.AudioFrame`](https://pyav.org/docs/develop/api/audio.html#av.audio.frame.AudioFrame) when dealing with audio) of [`PyAV` library](https://pyav.org/).
You can inject any kinds of image (or audio) processing inside the callback.
See examples above for more applications.
### Pass parameters to the callback
You can also pass parameters to the callback.
In the example below, a boolean `flip` flag is used to turn on/off the image flipping.
```python
import streamlit as st
from streamlit_webrtc import webrtc_streamer
import av
flip = st.checkbox("Flip")
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    flipped = img[::-1,:,:] if flip else img
    return av.VideoFrame.from_ndarray(flipped, format="bgr24")
webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
```
### Pull values from the callback
Sometimes we want to read the values generated in the callback from the outer scope.
Note that the callback is executed in a forked thread running independently of the main script, so we have to take care of the following points and need some tricks for implementation like the example below (See also the section below for some limitations in the callback due to multi-threading).
* Thread-safety
  * Passing the values between inside and outside the callback must be thread-safe.
* Using a loop to poll the values
  * During media streaming, while the callback continues to be called, the main script execution stops at the bottom as usual. So we need to use a loop to keep the main script running and get the values from the callback in the outer scope.
The following example is to pass the image frames from the callback to the outer scope and continuously process it in the loop. In this example, a simple image analysis (calculating the histogram like [this OpenCV tutorial](https://docs.opencv.org/4.x/d1/db7/tutorial_py_histogram_begins.html)) is done on the image frames.
[`threading.Lock`](https://docs.python.org/3/library/threading.html#lock-objects) is one standard way to control variable accesses across threads.
A dict object `img_container` here is a mutable container shared by the callback and the outer scope and the `lock` object is used at assigning and reading the values to/from the container for thread-safety.
```python
import threading
import cv2
import streamlit as st
from matplotlib import pyplot as plt
from streamlit_webrtc import webrtc_streamer
lock = threading.Lock()
img_container = {"img": None}
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    with lock:
        img_container["img"] = img
    return frame
ctx = webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
fig_place = st.empty()
fig, ax = plt.subplots(1, 1)
while ctx.state.playing:
    with lock:
        img = img_container["img"]
    if img is None:
        continue
    gray = cv2.cvtColor(img, cv2.COLOR_BGR2GRAY)
    ax.cla()
    ax.hist(gray.ravel(), 256, [0, 256])
    fig_place.pyplot(fig)
```
## Callback limitations
The callbacks are executed in forked threads different from the main one, so there are some limitations:
* Streamlit methods (`st.*` such as `st.write()`) do not work inside the callbacks.
* Variables inside the callbacks cannot be directly referred to from the outside.
* The `global` keyword does not work expectedly in the callbacks.
* You have to care about thread-safety when accessing the same objects both from outside and inside the callbacks as stated in the section above.
## Class-based callbacks
Until v0.37, the class-based callbacks were the standard.
See the [old version of the README](https://github.com/whitphx/streamlit-webrtc/blob/v0.37.0/README.md#quick-tutorial) about it.
## Serving from remote host
When deploying apps to remote servers, there are some things you need to be aware of.
### HTTPS
`streamlit-webrtc` uses [`getUserMedia()`](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia) API to access local media devices, and this method does not work in an insecure context.
[This document](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia#privacy_and_security) says
> A secure context is, in short, a page loaded using HTTPS or the file:/// URL scheme, or a page loaded from localhost.
So, when hosting your app on a remote server, it must be served via HTTPS if your app is using webcam or microphone.
If not, you will encounter an error when starting using the device. For example, it's something like below on Chrome.
> Error: navigator.mediaDevices is undefined. It seems the current document is not loaded securely.
[Streamlit Cloud](https://streamlit.io/cloud) is a recommended way for HTTPS serving. You can easily deploy Streamlit apps with it, and most importantly for this topic, it serves the apps via HTTPS automatically by defualt.
### Configure the STUN server
To deploy the app to the cloud, we have to configure the *STUN* server via the `rtc_configuration` argument on `webrtc_streamer()` like below.
```python
webrtc_streamer(
    # ...
    rtc_configuration={  # Add this config
        "iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]
    }
    # ...
)
```
This configuration is necessary to establish the media streaming connection when the server is on a remote host.
`streamlit_webrtc` uses WebRTC for its video and audio streaming. It has to access a "STUN server" in the global network for the remote peers (precisely, peers over the NATs) to establish WebRTC connections.
As we don't see the details about STUN servers here, please google it if interested with keywords such as STUN, TURN, or NAT traversal, or read these articles ([1](https://towardsdatascience.com/developing-web-based-real-time-video-audio-processing-apps-quickly-with-streamlit-7c7bcd0bc5a8#1cec), [2](https://dev.to/whitphx/python-webrtc-basics-with-aiortc-48id), [3](https://www.3cx.com/pbx/what-is-a-stun-server/)).
The example above is configured to use `stun.l.google.com:19302`, which is a free STUN server provided by Google.
You can also use any other STUN servers.
For example, [one user reported](https://github.com/whitphx/streamlit-webrtc/issues/283#issuecomment-889753789) that the Google's STUN server had a huge delay when using from China network, and the problem was solved by changing the STUN server.
For those who know about the browser WebRTC API: The value of the rtc_configuration argument will be passed to the [`RTCPeerConnection`](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection) constructor on the frontend.
### Configure the TURN server if necessary
Even if the STUN server is properly configured, media streaming may not work in some network environments.
For example, in some office or public networks, there are firewalls which drop the WebRTC packets.
In such environments, setting up a [TURN server](https://webrtc.org/getting-started/turn-server) is a solution. See https://github.com/whitphx/streamlit-webrtc/issues/335#issuecomment-897326755.
## Logging
For logging, this library uses the standard `logging` module and follows the practice described in [the official logging tutorial](https://docs.python.org/3/howto/logging.html#advanced-logging-tutorial). Then the logger names are the same as the module names - `streamlit_webrtc` or `streamlit_webrtc.*`.
So you can get the logger instance with `logging.getLogger("streamlit_webrtc")` through which you can control the logs from this library.
For example, if you want to set the log level on this library's logger as WARNING, you can use the following code.
```python
st_webrtc_logger = logging.getLogger("streamlit_webrtc")
st_webrtc_logger.setLevel(logging.WARNING)
```
In practice, `aiortc`, a third-party package this library is internally using, also emits many INFO level logs and you may want to control its logs too.
You can do it in the same way as below.
```python
aioice_logger = logging.getLogger("aioice")
aioice_logger.setLevel(logging.WARNING)
```
## API changes
Currently there is no documentation about the interface. See the examples in [./pages/*.py](./pages) for the usage.
The API is not finalized yet and can be changed without backward compatibility in the future releases until v1.0.
### For users since versions `<0.20`
`VideoTransformerBase` and its `transform` method have been marked as deprecated in v0.20.0. Please use `VideoProcessorBase#recv()` instead.
Note that the signature of the `recv` method is different from the `transform` in that the `recv` has to return an instance of `av.VideoFrame` or `av.AudioFrame`.
Also, `webrtc_streamer()`'s `video_transformer_factory` and `async_transform` arguments are deprecated, so use `video_processor_factory` and `async_processing` respectively.
See the samples in [app.py](./app.py) for their usage.
## Resources
* [Developing web-based real-time video/audio processing apps quickly with Streamlit](https://www.whitphx.info/posts/20211231-streamlit-webrtc-video-app-tutorial/)
  * A tutorial for real-time video app development using `streamlit-webrtc`.
  * Crosspost on dev.to: https://dev.to/whitphx/developing-web-based-real-time-videoaudio-processing-apps-quickly-with-streamlit-4k89
* [New Component: streamlit-webrtc, a new way to deal with real-time media streams (Streamlit Community)](https://discuss.streamlit.io/t/new-component-streamlit-webrtc-a-new-way-to-deal-with-real-time-media-streams/8669)
  * This is a forum topic where `streamlit-webrtc` has been introduced and discussed about.
## Support the project
[![ko-fi](https://ko-fi.com/img/githubbutton_sm.svg)](https://ko-fi.com/D1D2ERWFG)
<a href="https://www.buymeacoffee.com/whitphx" target="_blank" rel="noreferrer"><img src="https://cdn.buymeacoffee.com/buttons/v2/default-yellow.png" alt="Buy Me A Coffee" width="180" height="50" ></a>
[![GitHub Sponsors](https://img.shields.io/github/sponsors/whitphx?label=Sponsor%20me%20on%20GitHub%20Sponsors&style=social)](https://github.com/sponsors/whitphx)

%package help
Summary:	Development documents and examples for streamlit-webrtc
Provides:	python3-streamlit-webrtc-doc
%description help
Create `app.py` with the content below.
```py
from streamlit_webrtc import webrtc_streamer
webrtc_streamer(key="sample")
```
Unlike other Streamlit components, `webrtc_streamer()` requires the `key` argument as a unique identifier. Set an arbitrary string to it.
Then run it with Streamlit and open http://localhost:8501/.
```shell
$ streamlit run app.py
```
You see the app view, so click the "START" button.
Then, video and audio streaming starts. If asked for permissions to access the camera and microphone, allow it.
![Basic example of streamlit-webrtc](./docs/images/streamlit_webrtc_basic.gif)
Next, edit `app.py` as below and run it again.
```py
from streamlit_webrtc import webrtc_streamer
import av
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    flipped = img[::-1,:,:]
    return av.VideoFrame.from_ndarray(flipped, format="bgr24")
webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
```
Now the video is vertically flipped.
![Vertically flipping example](./docs/images/streamlit_webrtc_flipped.gif)
As an example above, you can edit the video frames by defining a callback that receives and returns a frame and passing it to the `video_frame_callback` argument (or `audio_frame_callback` for audio manipulation).
The input and output frames are the instance of [`av.VideoFrame`](https://pyav.org/docs/develop/api/video.html#av.video.frame.VideoFrame) (or [`av.AudioFrame`](https://pyav.org/docs/develop/api/audio.html#av.audio.frame.AudioFrame) when dealing with audio) of [`PyAV` library](https://pyav.org/).
You can inject any kinds of image (or audio) processing inside the callback.
See examples above for more applications.
### Pass parameters to the callback
You can also pass parameters to the callback.
In the example below, a boolean `flip` flag is used to turn on/off the image flipping.
```python
import streamlit as st
from streamlit_webrtc import webrtc_streamer
import av
flip = st.checkbox("Flip")
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    flipped = img[::-1,:,:] if flip else img
    return av.VideoFrame.from_ndarray(flipped, format="bgr24")
webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
```
### Pull values from the callback
Sometimes we want to read the values generated in the callback from the outer scope.
Note that the callback is executed in a forked thread running independently of the main script, so we have to take care of the following points and need some tricks for implementation like the example below (See also the section below for some limitations in the callback due to multi-threading).
* Thread-safety
  * Passing the values between inside and outside the callback must be thread-safe.
* Using a loop to poll the values
  * During media streaming, while the callback continues to be called, the main script execution stops at the bottom as usual. So we need to use a loop to keep the main script running and get the values from the callback in the outer scope.
The following example is to pass the image frames from the callback to the outer scope and continuously process it in the loop. In this example, a simple image analysis (calculating the histogram like [this OpenCV tutorial](https://docs.opencv.org/4.x/d1/db7/tutorial_py_histogram_begins.html)) is done on the image frames.
[`threading.Lock`](https://docs.python.org/3/library/threading.html#lock-objects) is one standard way to control variable accesses across threads.
A dict object `img_container` here is a mutable container shared by the callback and the outer scope and the `lock` object is used at assigning and reading the values to/from the container for thread-safety.
```python
import threading
import cv2
import streamlit as st
from matplotlib import pyplot as plt
from streamlit_webrtc import webrtc_streamer
lock = threading.Lock()
img_container = {"img": None}
def video_frame_callback(frame):
    img = frame.to_ndarray(format="bgr24")
    with lock:
        img_container["img"] = img
    return frame
ctx = webrtc_streamer(key="example", video_frame_callback=video_frame_callback)
fig_place = st.empty()
fig, ax = plt.subplots(1, 1)
while ctx.state.playing:
    with lock:
        img = img_container["img"]
    if img is None:
        continue
    gray = cv2.cvtColor(img, cv2.COLOR_BGR2GRAY)
    ax.cla()
    ax.hist(gray.ravel(), 256, [0, 256])
    fig_place.pyplot(fig)
```
## Callback limitations
The callbacks are executed in forked threads different from the main one, so there are some limitations:
* Streamlit methods (`st.*` such as `st.write()`) do not work inside the callbacks.
* Variables inside the callbacks cannot be directly referred to from the outside.
* The `global` keyword does not work expectedly in the callbacks.
* You have to care about thread-safety when accessing the same objects both from outside and inside the callbacks as stated in the section above.
## Class-based callbacks
Until v0.37, the class-based callbacks were the standard.
See the [old version of the README](https://github.com/whitphx/streamlit-webrtc/blob/v0.37.0/README.md#quick-tutorial) about it.
## Serving from remote host
When deploying apps to remote servers, there are some things you need to be aware of.
### HTTPS
`streamlit-webrtc` uses [`getUserMedia()`](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia) API to access local media devices, and this method does not work in an insecure context.
[This document](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia#privacy_and_security) says
> A secure context is, in short, a page loaded using HTTPS or the file:/// URL scheme, or a page loaded from localhost.
So, when hosting your app on a remote server, it must be served via HTTPS if your app is using webcam or microphone.
If not, you will encounter an error when starting using the device. For example, it's something like below on Chrome.
> Error: navigator.mediaDevices is undefined. It seems the current document is not loaded securely.
[Streamlit Cloud](https://streamlit.io/cloud) is a recommended way for HTTPS serving. You can easily deploy Streamlit apps with it, and most importantly for this topic, it serves the apps via HTTPS automatically by defualt.
### Configure the STUN server
To deploy the app to the cloud, we have to configure the *STUN* server via the `rtc_configuration` argument on `webrtc_streamer()` like below.
```python
webrtc_streamer(
    # ...
    rtc_configuration={  # Add this config
        "iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]
    }
    # ...
)
```
This configuration is necessary to establish the media streaming connection when the server is on a remote host.
`streamlit_webrtc` uses WebRTC for its video and audio streaming. It has to access a "STUN server" in the global network for the remote peers (precisely, peers over the NATs) to establish WebRTC connections.
As we don't see the details about STUN servers here, please google it if interested with keywords such as STUN, TURN, or NAT traversal, or read these articles ([1](https://towardsdatascience.com/developing-web-based-real-time-video-audio-processing-apps-quickly-with-streamlit-7c7bcd0bc5a8#1cec), [2](https://dev.to/whitphx/python-webrtc-basics-with-aiortc-48id), [3](https://www.3cx.com/pbx/what-is-a-stun-server/)).
The example above is configured to use `stun.l.google.com:19302`, which is a free STUN server provided by Google.
You can also use any other STUN servers.
For example, [one user reported](https://github.com/whitphx/streamlit-webrtc/issues/283#issuecomment-889753789) that the Google's STUN server had a huge delay when using from China network, and the problem was solved by changing the STUN server.
For those who know about the browser WebRTC API: The value of the rtc_configuration argument will be passed to the [`RTCPeerConnection`](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection) constructor on the frontend.
### Configure the TURN server if necessary
Even if the STUN server is properly configured, media streaming may not work in some network environments.
For example, in some office or public networks, there are firewalls which drop the WebRTC packets.
In such environments, setting up a [TURN server](https://webrtc.org/getting-started/turn-server) is a solution. See https://github.com/whitphx/streamlit-webrtc/issues/335#issuecomment-897326755.
## Logging
For logging, this library uses the standard `logging` module and follows the practice described in [the official logging tutorial](https://docs.python.org/3/howto/logging.html#advanced-logging-tutorial). Then the logger names are the same as the module names - `streamlit_webrtc` or `streamlit_webrtc.*`.
So you can get the logger instance with `logging.getLogger("streamlit_webrtc")` through which you can control the logs from this library.
For example, if you want to set the log level on this library's logger as WARNING, you can use the following code.
```python
st_webrtc_logger = logging.getLogger("streamlit_webrtc")
st_webrtc_logger.setLevel(logging.WARNING)
```
In practice, `aiortc`, a third-party package this library is internally using, also emits many INFO level logs and you may want to control its logs too.
You can do it in the same way as below.
```python
aioice_logger = logging.getLogger("aioice")
aioice_logger.setLevel(logging.WARNING)
```
## API changes
Currently there is no documentation about the interface. See the examples in [./pages/*.py](./pages) for the usage.
The API is not finalized yet and can be changed without backward compatibility in the future releases until v1.0.
### For users since versions `<0.20`
`VideoTransformerBase` and its `transform` method have been marked as deprecated in v0.20.0. Please use `VideoProcessorBase#recv()` instead.
Note that the signature of the `recv` method is different from the `transform` in that the `recv` has to return an instance of `av.VideoFrame` or `av.AudioFrame`.
Also, `webrtc_streamer()`'s `video_transformer_factory` and `async_transform` arguments are deprecated, so use `video_processor_factory` and `async_processing` respectively.
See the samples in [app.py](./app.py) for their usage.
## Resources
* [Developing web-based real-time video/audio processing apps quickly with Streamlit](https://www.whitphx.info/posts/20211231-streamlit-webrtc-video-app-tutorial/)
  * A tutorial for real-time video app development using `streamlit-webrtc`.
  * Crosspost on dev.to: https://dev.to/whitphx/developing-web-based-real-time-videoaudio-processing-apps-quickly-with-streamlit-4k89
* [New Component: streamlit-webrtc, a new way to deal with real-time media streams (Streamlit Community)](https://discuss.streamlit.io/t/new-component-streamlit-webrtc-a-new-way-to-deal-with-real-time-media-streams/8669)
  * This is a forum topic where `streamlit-webrtc` has been introduced and discussed about.
## Support the project
[![ko-fi](https://ko-fi.com/img/githubbutton_sm.svg)](https://ko-fi.com/D1D2ERWFG)
<a href="https://www.buymeacoffee.com/whitphx" target="_blank" rel="noreferrer"><img src="https://cdn.buymeacoffee.com/buttons/v2/default-yellow.png" alt="Buy Me A Coffee" width="180" height="50" ></a>
[![GitHub Sponsors](https://img.shields.io/github/sponsors/whitphx?label=Sponsor%20me%20on%20GitHub%20Sponsors&style=social)](https://github.com/sponsors/whitphx)

%prep
%autosetup -n streamlit-webrtc-0.45.0

%build
%py3_build

%install
%py3_install
install -d -m755 %{buildroot}/%{_pkgdocdir}
if [ -d doc ]; then cp -arf doc %{buildroot}/%{_pkgdocdir}; fi
if [ -d docs ]; then cp -arf docs %{buildroot}/%{_pkgdocdir}; fi
if [ -d example ]; then cp -arf example %{buildroot}/%{_pkgdocdir}; fi
if [ -d examples ]; then cp -arf examples %{buildroot}/%{_pkgdocdir}; fi
pushd %{buildroot}
if [ -d usr/lib ]; then
	find usr/lib -type f -printf "/%h/%f\n" >> filelist.lst
fi
if [ -d usr/lib64 ]; then
	find usr/lib64 -type f -printf "/%h/%f\n" >> filelist.lst
fi
if [ -d usr/bin ]; then
	find usr/bin -type f -printf "/%h/%f\n" >> filelist.lst
fi
if [ -d usr/sbin ]; then
	find usr/sbin -type f -printf "/%h/%f\n" >> filelist.lst
fi
touch doclist.lst
if [ -d usr/share/man ]; then
	find usr/share/man -type f -printf "/%h/%f.gz\n" >> doclist.lst
fi
popd
mv %{buildroot}/filelist.lst .
mv %{buildroot}/doclist.lst .

%files -n python3-streamlit-webrtc -f filelist.lst
%dir %{python3_sitelib}/*

%files help -f doclist.lst
%{_docdir}/*

%changelog
* Fri May 05 2023 Python_Bot <Python_Bot@openeuler.org> - 0.45.0-1
- Package Spec generated