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chore: dispatch starting_recording and stop_recording. (#342)

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Co-authored-by: Ming Xu <albertxu@amazon.com>
This commit is contained in:
AlbertMingXu
2025-06-10 06:36:32 +08:00
committed by GitHub
parent 6875d6610a
commit 8780265659
4 changed files with 12 additions and 1 deletions
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2
backend/fastrtc/webrtc.py
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@@ -57,7 +57,7 @@ class WebRTC(Component, WebRTCConnectionMixin):
Demos: video_identity_2 Demos: video_identity_2
""" """
EVENTS = ["tick", "state_change", "submit"] EVENTS = ["tick", "state_change", "submit", "start_recording", "stop_recording"]
data_model = WebRTCModel data_model = WebRTCModel
def __init__( def __init__(

2
frontend/Index.svelte
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@@ -208,6 +208,8 @@
{pulse_color} {pulse_color}
{button_labels} {button_labels}
{variant} {variant}
on:start_recording={() => gradio.dispatch("start_recording")}
on:stop_recording={() => gradio.dispatch("stop_recording")}
on:tick={() => gradio.dispatch("tick")} on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)} on:error={({ detail }) => gradio.dispatch("error", detail)}
on:warning={({ detail }) => gradio.dispatch("warning", detail)} on:warning={({ detail }) => gradio.dispatch("warning", detail)}

7
frontend/shared/InteractiveAudio.svelte
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@@ -104,6 +104,8 @@
error: string; error: string;
play: undefined; play: undefined;
stop: undefined; stop: undefined;
start_recording: undefined;
stop_recording: undefined;
}>(); }>();
async function access_mic(): Promise<void> { async function access_mic(): Promise<void> {
@@ -147,6 +149,7 @@
async function start_stream(): Promise<void> { async function start_stream(): Promise<void> {
if (stream_state === "open") { if (stream_state === "open") {
dispatch("stop_recording");
stop(pc); stop(pc);
stream_state = "closed"; stream_state = "closed";
_time_limit = null; _time_limit = null;
@@ -154,6 +157,8 @@
await server.quit_output_stream({ webrtc_id: _webrtc_id }); await server.quit_output_stream({ webrtc_id: _webrtc_id });
return; return;
} }
dispatch("start_recording");
_webrtc_id = Math.random().toString(36).substring(2); _webrtc_id = Math.random().toString(36).substring(2);
value.webrtc_id = _webrtc_id; value.webrtc_id = _webrtc_id;
stream_state = "waiting"; stream_state = "waiting";
@@ -325,6 +330,8 @@
bind:this={audio_player} bind:this={audio_player}
on:ended={() => dispatch("stop")} on:ended={() => dispatch("stop")}
on:play={() => dispatch("play")} on:play={() => dispatch("play")}
on:start_recording
on:stop_recording
/> />
{#if variant === "textbox"} {#if variant === "textbox"}
<TextboxWithMic <TextboxWithMic

2
frontend/shared/Webcam.svelte
View File

@@ -146,6 +146,7 @@
async function start_webrtc(): Promise<void> { async function start_webrtc(): Promise<void> {
if (stream_state === "closed") { if (stream_state === "closed") {
dispatch("start_recording");
await server.turn().then((rtc_configuration_) => { await server.turn().then((rtc_configuration_) => {
if (rtc_configuration_.error) { if (rtc_configuration_.error) {
dispatch("error", rtc_configuration_.error); dispatch("error", rtc_configuration_.error);
@@ -207,6 +208,7 @@
stream_state = "closed"; stream_state = "closed";
}); });
} else { } else {
dispatch("stop_recording");
stop(pc); stop(pc);
stream_state = "closed"; stream_state = "closed";
_time_limit = null; _time_limit = null;