feat: add public apis to set video/audio bitrate

components/esp_webrtc/include/esp_webrtc.h

@@ -298,6 +298,40 @@ int esp_webrtc_stop(esp_webrtc_handle_t rtc_handle);
  */
 int esp_webrtc_close(esp_webrtc_handle_t rtc_handle);
 
+/**
+ * @brief  Set video bitrate for encoder
+ *
+ * @note   This can be called before or after WebRTC starts. If called before start,
+ *         the bitrate will be applied when the capture sink is created.
+ *         If called during streaming, the bitrate is updated immediately without restart.
+ *
+ * @param[in]  rtc_handle  WebRTC handle
+ * @param[in]  bps         Target bitrate in bits per second (e.g., 700000 for 700 kbps)
+ *
+ * @return
+ *      - 0                         On success
+ *      - ESP_PEER_ERR_INVALID_ARG  Invalid argument
+ *      - Others                    Error from capture sink
+ */
+int esp_webrtc_set_video_bitrate(esp_webrtc_handle_t rtc_handle, uint32_t bps);
+
+/**
+ * @brief  Set audio bitrate for encoder
+ *
+ * @note   This can be called before or after WebRTC starts. If called before start,
+ *         the bitrate will be applied when the capture sink is created.
+ *         If called during streaming, the bitrate is updated immediately without restart.
+ *
+ * @param[in]  rtc_handle  WebRTC handle
+ * @param[in]  bps         Target bitrate in bits per second (e.g., 64000 for 64 kbps)
+ *
+ * @return
+ *      - 0                         On success
+ *      - ESP_PEER_ERR_INVALID_ARG  Invalid argument
+ *      - Others                    Error from capture sink
+ */
+int esp_webrtc_set_audio_bitrate(esp_webrtc_handle_t rtc_handle, uint32_t bps);
+
 #ifdef __cplusplus
 }
 #endif

components/esp_webrtc/src/esp_webrtc.c

@@ -97,6 +97,12 @@ typedef struct {
     uint8_t  vid_send_num;
     uint8_t  aud_recv_num;
     uint8_t  vid_recv_num;
+
+    // Pending bitrate settings (applied when sink is created)
+    uint32_t pending_video_bitrate_bps;
+    bool     video_bitrate_pending;
+    uint32_t pending_audio_bitrate_bps;
+    bool     audio_bitrate_pending;
 } webrtc_t;
 
 static const char *TAG = "webrtc";
@@ -598,6 +604,30 @@ static int pc_start(webrtc_t *rtc, esp_peer_ice_server_cfg_t *server_info, int s
     }
     esp_capture_sink_setup(rtc->media_provider.capture, 0, &sink_cfg, &rtc->capture_path);
     esp_capture_sink_enable(rtc->capture_path, ESP_CAPTURE_RUN_MODE_ALWAYS);
+
+    // Apply pending bitrate settings now that sink exists
+    if (rtc->video_bitrate_pending && rtc->capture_path) {
+        int bitrate_ret = esp_capture_sink_set_bitrate(rtc->capture_path, ESP_CAPTURE_STREAM_TYPE_VIDEO, 
+                                                        rtc->pending_video_bitrate_bps);
+        if (bitrate_ret == ESP_CAPTURE_ERR_OK) {
+            ESP_LOGI(TAG, "Applied pending video bitrate: %lu bps", (unsigned long)rtc->pending_video_bitrate_bps);
+        } else {
+            ESP_LOGW(TAG, "Failed to apply pending video bitrate: %d", bitrate_ret);
+        }
+        rtc->video_bitrate_pending = false;
+    }
+
+    if (rtc->audio_bitrate_pending && rtc->capture_path) {
+        int bitrate_ret = esp_capture_sink_set_bitrate(rtc->capture_path, ESP_CAPTURE_STREAM_TYPE_AUDIO, 
+                                                        rtc->pending_audio_bitrate_bps);
+        if (bitrate_ret == ESP_CAPTURE_ERR_OK) {
+            ESP_LOGI(TAG, "Applied pending audio bitrate: %lu bps", (unsigned long)rtc->pending_audio_bitrate_bps);
+        } else {
+            ESP_LOGW(TAG, "Failed to apply pending audio bitrate: %d", bitrate_ret);
+        }
+        rtc->audio_bitrate_pending = false;
+    }
+
     return ret;
 }
 
@@ -948,3 +978,57 @@ int esp_webrtc_close(esp_webrtc_handle_t handle)
     free(rtc);
     return ESP_PEER_ERR_NONE;
 }
+
+int esp_webrtc_set_video_bitrate(esp_webrtc_handle_t handle, uint32_t bps)
+{
+    if (handle == NULL) {
+        return ESP_PEER_ERR_INVALID_ARG;
+    }
+
+    webrtc_t *rtc = (webrtc_t *)handle;
+
+    // If capture path exists, apply immediately
+    if (rtc->capture_path) {
+        int ret = esp_capture_sink_set_bitrate(rtc->capture_path, ESP_CAPTURE_STREAM_TYPE_VIDEO, bps);
+        if (ret == ESP_CAPTURE_ERR_OK) {
+            ESP_LOGI(TAG, "Set video bitrate: %lu bps", (unsigned long)bps);
+            return 0;
+        } else {
+            ESP_LOGW(TAG, "Failed to set video bitrate: %d", ret);
+            return ret;
+        }
+    }
+
+    // Otherwise, store as pending
+    rtc->pending_video_bitrate_bps = bps;
+    rtc->video_bitrate_pending = true;
+    ESP_LOGI(TAG, "Video bitrate %lu bps will be applied when WebRTC starts", (unsigned long)bps);
+    return 0;
+}
+
+int esp_webrtc_set_audio_bitrate(esp_webrtc_handle_t handle, uint32_t bps)
+{
+    if (handle == NULL) {
+        return ESP_PEER_ERR_INVALID_ARG;
+    }
+
+    webrtc_t *rtc = (webrtc_t *)handle;
+
+    // If capture path exists, apply immediately
+    if (rtc->capture_path) {
+        int ret = esp_capture_sink_set_bitrate(rtc->capture_path, ESP_CAPTURE_STREAM_TYPE_AUDIO, bps);
+        if (ret == ESP_CAPTURE_ERR_OK) {
+            ESP_LOGI(TAG, "Set audio bitrate: %lu bps", (unsigned long)bps);
+            return 0;
+        } else {
+            ESP_LOGW(TAG, "Failed to set audio bitrate: %d", ret);
+            return ret;
+        }
+    }
+
+    // Otherwise, store as pending
+    rtc->pending_audio_bitrate_bps = bps;
+    rtc->audio_bitrate_pending = true;
+    ESP_LOGI(TAG, "Audio bitrate %lu bps will be applied when WebRTC starts", (unsigned long)bps);
+    return 0;
+}