commit
387bc00a20
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@ -70,6 +70,7 @@ void hexdump(const void *mem, uint32_t len, uint8_t cols = 16) {
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}
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void InitI2SSpeakerOrMic(int mode) {
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Serial.printf("InitI2sSpeakerOrMic %d\n", mode);
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esp_err_t err = ESP_OK;
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i2s_driver_uninstall(SPEAKER_I2S_NUMBER);
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@ -136,17 +137,19 @@ void webSocketEvent(WStype_t type, uint8_t * payload, size_t length) {
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break;
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case WStype_TEXT:
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Serial.printf("[WSc] get text: %s\n", payload);
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if ((char)payload[0] == 's'){
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Serial.println("start");
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speaker_offset = 0;
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InitI2SSpeakerOrMic(MODE_SPK);
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{
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std::string str(payload, payload + length);
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bool isAudio = str.find("\"audio\"") != std::string::npos;
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if (isAudio && str.find("\"start\"") != std::string::npos) {
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Serial.println("start playback");
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speaker_offset = 0;
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InitI2SSpeakerOrMic(MODE_SPK);
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} else if (isAudio && str.find("\"end\"") != std::string::npos) {
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Serial.println("end playback");
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// speaker_play(speakerdata0, speaker_offset);
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// speaker_offset = 0;
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}
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}
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if ((char)payload[0] == 'e'){
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Serial.println("end");
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// speaker_play(speakerdata0, speaker_offset);
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// speaker_offset = 0;
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}
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// send message to server
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// webSocket.sendTXT("message here");
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break;
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@ -180,12 +183,12 @@ void websocket_setup() {
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Serial.println("connecting to WiFi");
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}
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Serial.println("connected to WiFi");
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webSocket.begin(COMPUTER_IP, 9001, "/");
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webSocket.begin(COMPUTER_IP, 8000, "/");
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webSocket.onEvent(webSocketEvent);
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// webSocket.setAuthorization("user", "Password");
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webSocket.setReconnectInterval(5000);
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}
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void setup() {
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M5.begin(true, false, true);
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M5.dis.drawpix(0, CRGB(128, 128, 0));
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@ -208,17 +211,19 @@ void loop() {
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button.loop();
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if (button.justPressed()) {
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Serial.println("Recording...");
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webSocket.sendTXT("s");
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webSocket.sendTXT("{\"role\": \"user\", \"type\": \"audio\", \"format\": \"bytes.raw\", \"start\": true}");
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InitI2SSpeakerOrMic(MODE_MIC);
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recording = true;
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data_offset = 0;
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Serial.println("Recording ready.");
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} else if (button.justReleased()) {
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Serial.println("Stopped recording.");
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webSocket.sendTXT("e");
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webSocket.sendTXT("{\"role\": \"user\", \"type\": \"audio\", \"format\": \"bytes.raw\", \"end\": true}");
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flush_microphone();
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recording = false;
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}
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if (recording) {
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data_offset = 0;
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} else if (recording) {
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Serial.printf("Reading chunk at %d...\n", data_offset);
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size_t bytes_read;
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i2s_read(
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SPEAKER_I2S_NUMBER,
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@ -226,13 +231,13 @@ void loop() {
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DATA_SIZE, &bytes_read, (100 / portTICK_RATE_MS)
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);
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data_offset += bytes_read;
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Serial.printf("Read %d bytes in chunk.\n", bytes_read);
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if (data_offset > 1024*10) {
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if (data_offset > 1024*9) {
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flush_microphone();
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}
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}
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M5.update();
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webSocket.loop();
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}
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@ -43,6 +43,18 @@ def configure_interpreter(interpreter: OpenInterpreter):
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Remember: You can run Python code. Be very concise. Ensure that you actually run code every time! THIS IS IMPORTANT. You NEED to write code. **Help the user by being very concise in your answers.** Do not break down tasks excessively, just into simple, few minute steps. Don't assume the user lives their life in a certain way— pick very general tasks if you're breaking a task down.
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ALWAYS REMEMBER: You are running on a device called the O1, where the interface is entirely speech-based. Keep your responses succint in light of this!
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IF YOU NEED TO THINK ABOUT A PROBLEM: (such as "Here's the plan:"), WRITE IT IN THE COMMENTS of the code block!
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For example:
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> User: What is 432/7?
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> Assistant: Let me use Python to calculate that.
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> Assistant Python function call:
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> # Here's the plan:
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> # 1. Divide the numbers
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> # 2. Round it to 3 digits.
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> print(round(432/7, 3))
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> Assistant: 432 / 7 is 61.714.
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Use the following functions (assume they're imported) to complete your goals whenever possible:
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{{
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import sys
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@ -70,7 +82,9 @@ print(output)
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""".strip()
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interpreter.custom_instructions = system_message
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# interpreter.custom_instructions = system_message
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interpreter.system_message = system_message
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interpreter.llm.supports_functions = True
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### LLM SETTINGS
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@ -201,7 +201,7 @@ async def listener():
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accumulated_text = ""
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for chunk in interpreter.chat(messages, stream=True, display=False):
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for chunk in interpreter.chat(messages, stream=True, display=True):
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logger.debug("Got chunk:", chunk)
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@ -212,7 +212,7 @@ async def listener():
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if os.getenv('TTS_RUNNER') == "server":
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# Speak full sentences out loud
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if chunk["role"] == "assistant" and "content" in chunk:
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if chunk["role"] == "assistant" and "content" in chunk and chunk["type"] == "message":
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accumulated_text += chunk["content"]
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sentences = split_into_sentences(accumulated_text)
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@ -241,7 +241,7 @@ async def listener():
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# Check if it's just an end flag. We ignore those.
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temp_message = await from_user.get()
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if temp_message == {'role': 'user', 'type': 'message', 'end': True}:
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if type(temp_message) is dict and temp_message.get("role") == "user" and temp_message.get("end"):
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# Yup. False alarm.
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continue
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else:
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@ -251,8 +251,9 @@ async def listener():
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with open(conversation_history_path, 'w') as file:
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json.dump(interpreter.messages, file, indent=4)
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logger.info("New user message recieved. Breaking.")
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break
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# TODO: is triggering seemingly randomly
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#logger.info("New user message recieved. Breaking.")
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#break
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# Also check if there's any new computer messages
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if not from_computer.empty():
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@ -25,6 +25,8 @@ def convert_mime_type_to_format(mime_type: str) -> str:
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return "wav"
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if mime_type == "audio/webm":
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return "webm"
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if mime_type == "audio/raw":
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return "dat"
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return mime_type
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@ -43,7 +45,16 @@ def export_audio_to_wav_ffmpeg(audio: bytearray, mime_type: str) -> str:
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# Export to wav
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output_path = os.path.join(temp_dir, f"output_{datetime.now().strftime('%Y%m%d%H%M%S%f')}.wav")
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ffmpeg.input(input_path).output(output_path, acodec='pcm_s16le', ac=1, ar='16k').run()
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print(mime_type, input_path, output_path)
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if mime_type == "audio/raw":
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ffmpeg.input(
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input_path,
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f='s16le',
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ar='16000',
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ac=1,
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).output(output_path).run()
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else:
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ffmpeg.input(input_path).output(output_path, acodec='pcm_s16le', ac=1, ar='16k').run()
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try:
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yield output_path
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@ -6,6 +6,7 @@ from pydub import AudioSegment
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from dotenv import load_dotenv
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load_dotenv() # take environment variables from .env.
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import ffmpeg
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import tempfile
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from openai import OpenAI
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import os
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@ -28,11 +29,17 @@ def stream_tts(text):
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input=text,
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response_format="opus"
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)
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with tempfile.NamedTemporaryFile(suffix=".opus") as temp_file:
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with tempfile.NamedTemporaryFile(suffix=".opus", delete=False) as temp_file:
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response.stream_to_file(temp_file.name)
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audio_bytes = temp_file.read()
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file_type = "bytes.opus"
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# TODO: hack to format audio correctly for device
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outfile = tempfile.gettempdir() + "/" + "raw.dat"
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ffmpeg.input(temp_file.name).output(outfile, f="s16le", ar="16000", ac="1").run()
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with open(outfile, "rb") as f:
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audio_bytes = f.read()
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file_type = "bytes.raw"
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print(outfile, len(audio_bytes))
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os.remove(outfile)
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else:
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with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file:
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Loading…
Reference in New Issue