Merge pull request #195 from google/gbg/speaker-app

speaker app

apps/speaker/logo.svg

@@ -0,0 +1,42 @@
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+<svg height="100%" style="fill-rule:nonzero;clip-rule:evenodd;stroke-linecap:round;stroke-linejoin:round;" xmlns:xlink="http://www.w3.org/1999/xlink" xmlns="http://www.w3.org/2000/svg" xml:space="preserve" width="100%" xmlns:vectornator="http://vectornator.io" version="1.1" viewBox="0 0 745 744.634">
+<metadata>
+<vectornator:setting key="DimensionsVisible" value="1"/>
+<vectornator:setting key="PencilOnly" value="0"/>
+<vectornator:setting key="SnapToPoints" value="0"/>
+<vectornator:setting key="OutlineMode" value="0"/>
+<vectornator:setting key="CMYKEnabledKey" value="0"/>
+<vectornator:setting key="RulersVisible" value="1"/>
+<vectornator:setting key="SnapToEdges" value="0"/>
+<vectornator:setting key="GuidesVisible" value="1"/>
+<vectornator:setting key="DisplayWhiteBackground" value="0"/>
+<vectornator:setting key="doHistoryDisabled" value="0"/>
+<vectornator:setting key="SnapToGuides" value="1"/>
+<vectornator:setting key="TimeLapseWatermarkDisabled" value="0"/>
+<vectornator:setting key="Units" value="Pixels"/>
+<vectornator:setting key="DynamicGuides" value="0"/>
+<vectornator:setting key="IsolateActiveLayer" value="0"/>
+<vectornator:setting key="SnapToGrid" value="0"/>
+</metadata>
+<defs/>
+<g id="Layer 1" vectornator:layerName="Layer 1">
+<path stroke="#000000" stroke-width="18.6464" d="M368.753+729.441L58.8847+550.539L58.8848+192.734L368.753+13.8313L678.621+192.734L678.621+550.539L368.753+729.441Z" fill="#0082fc" stroke-linecap="butt" fill-opacity="0.307489" opacity="1" stroke-linejoin="round"/>
+<g opacity="1">
+<g opacity="1">
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+</g>
+<path stroke="#000000" stroke-width="0.1" d="M103.161+309.167L226.956+443.903L366.671+309.604L103.161+309.167Z" fill="#0082fc" stroke-linecap="round" opacity="1" stroke-linejoin="round"/>
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+<path stroke="#000000" stroke-width="20" d="M522.045+154.808L229.559+448.882L83.8397+300.104L653.666+302.936L511.759+444.785L223.101+156.114" fill="none" stroke-linecap="round" opacity="1" stroke-linejoin="round"/>
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+<g opacity="1">
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+</g>
+</g>
+</g>
+</svg>

apps/speaker/speaker.css

@@ -0,0 +1,76 @@
+body, h1, h2, h3, h4, h5, h6 {
+    font-family: sans-serif;
+}
+
+#controlsDiv {
+    margin: 6px;
+}
+
+#connectionText {
+    background-color: rgb(239, 89, 75);
+    border: none;
+    border-radius: 4px;
+    padding: 8px;
+    display: inline-block;
+    margin: 4px;
+}
+
+#startButton {
+    padding: 4px;
+    margin: 6px;
+}
+
+#fftCanvas {
+    border-radius: 16px;
+    margin: 6px;
+}
+
+#bandwidthCanvas {
+    border: grey;
+    border-style: solid;
+    border-radius: 8px;
+    margin: 6px;
+}
+
+#streamStateText {
+    background-color: rgb(93, 165, 93);
+    border: none;
+    border-radius: 8px;
+    padding: 10px 20px;
+    display: inline-block;
+    margin: 6px;
+}
+
+#connectionStateText {
+    background-color: rgb(112, 146, 206);
+    border: none;
+    border-radius: 8px;
+    padding: 10px 20px;
+    display: inline-block;
+    margin: 6px;
+}
+
+#propertiesTable {
+    border: grey;
+    border-style: solid;
+    border-radius: 4px;
+    padding: 4px;
+    margin: 6px;
+    margin-left: 0px;
+}
+
+th, td {
+    padding-left: 6px;
+    padding-right: 6px;
+}
+
+.properties td:nth-child(even) {
+    background-color: #D6EEEE;
+    font-family: monospace;
+}
+
+.properties td:nth-child(odd) {
+    font-weight: bold;
+}
+
+.properties tr td:nth-child(2) { width: 150px; }

apps/speaker/speaker.html

@@ -0,0 +1,34 @@
+<!DOCTYPE html>
+<html>
+<head>
+  <title>Bumble Speaker</title>
+  <script type="text/javascript" src="speaker.js"></script>
+  <link rel="stylesheet" href="speaker.css">
+</head>
+<body>
+  <h1><img src="logo.svg" width=100 height=100 style="vertical-align:middle" alt=""/>Bumble Virtual Speaker</h1>
+  <div id="connectionText"></div>
+  <div id="speaker">
+    <table><tr>
+      <td>
+        <table id="propertiesTable" class="properties">
+          <tr><td>Codec</td><td><span id="codecText"></span></td></tr>
+          <tr><td>Packets</td><td><span id="packetsReceivedText"></span></td></tr>
+          <tr><td>Bytes</td><td><span id="bytesReceivedText"></span></td></tr>
+        </table>
+      </td>
+      <td>
+        <canvas id="bandwidthCanvas" width="500", height="100">Bandwidth Graph</canvas>
+      </td>
+    </tr></table>
+    <span id="streamStateText">IDLE</span>
+    <span id="connectionStateText">NOT CONNECTED</span>
+    <div id="controlsDiv">
+        <button id="audioOnButton">Audio On</button>
+        <span id="audioSupportMessageText"></span>
+    </div>
+    <canvas id="fftCanvas" width="1024", height="300">Audio Frequencies Animation</canvas>
+    <audio id="audio"></audio>
+  </div>
+</body>
+</html>

apps/speaker/speaker.js

@@ -0,0 +1,315 @@
+(function () {
+    'use strict';
+
+const channelUrl = ((window.location.protocol === "https:") ? "wss://" : "ws://") + window.location.host + "/channel";
+let channelSocket;
+let connectionText;
+let codecText;
+let packetsReceivedText;
+let bytesReceivedText;
+let streamStateText;
+let connectionStateText;
+let controlsDiv;
+let audioOnButton;
+let mediaSource;
+let sourceBuffer;
+let audioElement;
+let audioContext;
+let audioAnalyzer;
+let audioFrequencyBinCount;
+let audioFrequencyData;
+let packetsReceived = 0;
+let bytesReceived = 0;
+let audioState = "stopped";
+let streamState = "IDLE";
+let audioSupportMessageText;
+let fftCanvas;
+let fftCanvasContext;
+let bandwidthCanvas;
+let bandwidthCanvasContext;
+let bandwidthBinCount;
+let bandwidthBins = [];
+
+const FFT_WIDTH = 800;
+const FFT_HEIGHT = 256;
+const BANDWIDTH_WIDTH = 500;
+const BANDWIDTH_HEIGHT = 100;
+
+function hexToBytes(hex) {
+    return Uint8Array.from(hex.match(/.{1,2}/g).map((byte) => parseInt(byte, 16)));
+}
+
+function init() {
+    initUI();
+    initMediaSource();
+    initAudioElement();
+    initAnalyzer();
+
+    connect();
+}
+
+function initUI() {
+    controlsDiv = document.getElementById("controlsDiv");
+    controlsDiv.style.visibility = "hidden";
+    connectionText = document.getElementById("connectionText");
+    audioOnButton = document.getElementById("audioOnButton");
+    codecText = document.getElementById("codecText");
+    packetsReceivedText = document.getElementById("packetsReceivedText");
+    bytesReceivedText = document.getElementById("bytesReceivedText");
+    streamStateText = document.getElementById("streamStateText");
+    connectionStateText = document.getElementById("connectionStateText");
+    audioSupportMessageText = document.getElementById("audioSupportMessageText");
+
+    audioOnButton.onclick = () => startAudio();
+
+    setConnectionText("");
+
+    requestAnimationFrame(onAnimationFrame);
+}
+
+function initMediaSource() {
+    mediaSource = new MediaSource();
+    mediaSource.onsourceopen = onMediaSourceOpen;
+    mediaSource.onsourceclose = onMediaSourceClose;
+    mediaSource.onsourceended = onMediaSourceEnd;
+}
+
+function initAudioElement() {
+    audioElement = document.getElementById("audio");
+    audioElement.src = URL.createObjectURL(mediaSource);
+    // audioElement.controls = true;
+}
+
+function initAnalyzer() {
+    fftCanvas = document.getElementById("fftCanvas");
+    fftCanvas.width = FFT_WIDTH
+    fftCanvas.height = FFT_HEIGHT
+    fftCanvasContext = fftCanvas.getContext('2d');
+    fftCanvasContext.fillStyle = "rgb(0, 0, 0)";
+    fftCanvasContext.fillRect(0, 0, FFT_WIDTH, FFT_HEIGHT);
+
+    bandwidthCanvas = document.getElementById("bandwidthCanvas");
+    bandwidthCanvas.width = BANDWIDTH_WIDTH
+    bandwidthCanvas.height = BANDWIDTH_HEIGHT
+    bandwidthCanvasContext = bandwidthCanvas.getContext('2d');
+    bandwidthCanvasContext.fillStyle = "rgb(255, 255, 255)";
+    bandwidthCanvasContext.fillRect(0, 0, BANDWIDTH_WIDTH, BANDWIDTH_HEIGHT);
+}
+
+function startAnalyzer() {
+    // FFT
+    if (audioElement.captureStream !== undefined) {
+        audioContext = new AudioContext();
+        audioAnalyzer = audioContext.createAnalyser();
+        audioAnalyzer.fftSize = 128;
+        audioFrequencyBinCount = audioAnalyzer.frequencyBinCount;
+        audioFrequencyData = new Uint8Array(audioFrequencyBinCount);
+        const stream = audioElement.captureStream();
+        const source = audioContext.createMediaStreamSource(stream);
+        source.connect(audioAnalyzer);
+    }
+
+    // Bandwidth
+    bandwidthBinCount = BANDWIDTH_WIDTH / 2;
+    bandwidthBins = [];
+}
+
+function setConnectionText(message) {
+    connectionText.innerText = message;
+    if (message.length == 0) {
+        connectionText.style.display = "none";
+    } else {
+        connectionText.style.display = "inline-block";
+    }
+}
+
+function setStreamState(state) {
+    streamState = state;
+    streamStateText.innerText = streamState;
+}
+
+function onAnimationFrame() {
+    // FFT
+    if (audioAnalyzer !== undefined) {
+        audioAnalyzer.getByteFrequencyData(audioFrequencyData);
+        fftCanvasContext.fillStyle = "rgb(0, 0, 0)";
+        fftCanvasContext.fillRect(0, 0, FFT_WIDTH, FFT_HEIGHT);
+        const barCount = audioFrequencyBinCount;
+        const barWidth = (FFT_WIDTH / audioFrequencyBinCount) - 1;
+        for (let bar = 0; bar < barCount; bar++) {
+            const barHeight = audioFrequencyData[bar];
+            fftCanvasContext.fillStyle = `rgb(${barHeight / 256 * 200 + 50}, 50, ${50 + 2 * bar})`;
+            fftCanvasContext.fillRect(bar * (barWidth + 1), FFT_HEIGHT - barHeight, barWidth, barHeight);
+        }
+    }
+
+    // Bandwidth
+    bandwidthCanvasContext.fillStyle = "rgb(255, 255, 255)";
+    bandwidthCanvasContext.fillRect(0, 0, BANDWIDTH_WIDTH, BANDWIDTH_HEIGHT);
+    bandwidthCanvasContext.fillStyle = `rgb(100, 100, 100)`;
+    for (let t = 0; t < bandwidthBins.length; t++) {
+        const lineHeight = (bandwidthBins[t] / 1000) * BANDWIDTH_HEIGHT;
+        bandwidthCanvasContext.fillRect(t * 2, BANDWIDTH_HEIGHT - lineHeight, 2, lineHeight);
+    }
+
+    // Display again at the next frame
+    requestAnimationFrame(onAnimationFrame);
+}
+
+function onMediaSourceOpen() {
+    console.log(this.readyState);
+    sourceBuffer = mediaSource.addSourceBuffer("audio/aac");
+}
+
+function onMediaSourceClose() {
+    console.log(this.readyState);
+}
+
+function onMediaSourceEnd() {
+    console.log(this.readyState);
+}
+
+async function startAudio() {
+    try {
+        console.log("starting audio...");
+        audioOnButton.disabled = true;
+        audioState = "starting";
+        await audioElement.play();
+        console.log("audio started");
+        audioState = "playing";
+        startAnalyzer();
+    } catch(error) {
+        console.error(`play failed: ${error}`);
+        audioState = "stopped";
+        audioOnButton.disabled = false;
+    }
+}
+
+function onAudioPacket(packet) {
+    if (audioState != "stopped") {
+        // Queue the audio packet.
+        sourceBuffer.appendBuffer(packet);
+    }
+
+    packetsReceived += 1;
+    packetsReceivedText.innerText = packetsReceived;
+    bytesReceived += packet.byteLength;
+    bytesReceivedText.innerText = bytesReceived;
+
+    bandwidthBins[bandwidthBins.length] = packet.byteLength;
+    if (bandwidthBins.length > bandwidthBinCount) {
+        bandwidthBins.shift();
+    }
+}
+
+function onChannelOpen() {
+    console.log('channel OPEN');
+    setConnectionText("");
+    controlsDiv.style.visibility = "visible";
+
+    // Handshake with the backend.
+    sendMessage({
+        type: "hello"
+    });
+}
+
+function onChannelClose() {
+    console.log('channel CLOSED');
+    setConnectionText("Connection to CLI app closed, restart it and reload this page.");
+    controlsDiv.style.visibility = "hidden";
+}
+
+function onChannelError(error) {
+    console.log(`channel ERROR: ${error}`);
+    setConnectionText(`Connection to CLI app error ({${error}}), restart it and reload this page.`);
+    controlsDiv.style.visibility = "hidden";
+}
+
+function onChannelMessage(message) {
+    if (typeof message.data === 'string' || message.data instanceof String) {
+        // JSON message.
+        const jsonMessage = JSON.parse(message.data);
+        console.log(`channel MESSAGE: ${message.data}`);
+
+        // Dispatch the message.
+        const handlerName = `on${jsonMessage.type.charAt(0).toUpperCase()}${jsonMessage.type.slice(1)}Message`
+        const handler = messageHandlers[handlerName];
+        if (handler !== undefined) {
+            const params = jsonMessage.params;
+            if (params === undefined) {
+                params = {};
+            }
+            handler(params);
+        } else {
+            console.warn(`unhandled message: ${jsonMessage.type}`)
+        }
+    } else {
+        // BINARY audio data.
+        onAudioPacket(message.data);
+    }
+}
+
+function onHelloMessage(params) {
+    codecText.innerText = params.codec;
+    if (params.codec != "aac") {
+        audioOnButton.disabled = true;
+        audioSupportMessageText.innerText = "Only AAC can be played, audio will be disabled";
+        audioSupportMessageText.style.display = "inline-block";
+    } else {
+        audioSupportMessageText.innerText = "";
+        audioSupportMessageText.style.display = "none";
+    }
+    if (params.streamState) {
+        setStreamState(params.streamState);
+    }
+}
+
+function onStartMessage(params) {
+    setStreamState("STARTED");
+}
+
+function onStopMessage(params) {
+    setStreamState("STOPPED");
+}
+
+function onSuspendMessage(params) {
+    setStreamState("SUSPENDED");
+}
+
+function onConnectionMessage(params) {
+    connectionStateText.innerText = `CONNECTED: ${params.peer_name} (${params.peer_address})`;
+}
+
+function onDisconnectionMessage(params) {
+    connectionStateText.innerText = "DISCONNECTED";
+}
+
+function sendMessage(message) {
+    channelSocket.send(JSON.stringify(message));
+}
+
+function connect() {
+    console.log("connecting to CLI app");
+
+    channelSocket = new WebSocket(channelUrl);
+    channelSocket.binaryType = "arraybuffer";
+    channelSocket.onopen = onChannelOpen;
+    channelSocket.onclose = onChannelClose;
+    channelSocket.onerror = onChannelError;
+    channelSocket.onmessage = onChannelMessage;
+}
+
+const messageHandlers = {
+    onHelloMessage,
+    onStartMessage,
+    onStopMessage,
+    onSuspendMessage,
+    onConnectionMessage,
+    onDisconnectionMessage
+}
+
+window.onload = (event) => {
+    init();
+}
+
+}());

apps/speaker/speaker.py

@@ -0,0 +1,747 @@
+# Copyright 2021-2023 Google LLC
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#      https://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+# -----------------------------------------------------------------------------
+# Imports
+# -----------------------------------------------------------------------------
+from __future__ import annotations
+import asyncio
+import asyncio.subprocess
+from importlib import resources
+import enum
+import json
+import os
+import logging
+import pathlib
+import subprocess
+from typing import Dict, List, Optional
+import weakref
+
+import click
+import aiohttp
+from aiohttp import web
+
+import bumble
+from bumble.colors import color
+from bumble.core import BT_BR_EDR_TRANSPORT, CommandTimeoutError
+from bumble.device import Connection, Device, DeviceConfiguration
+from bumble.hci import HCI_StatusError
+from bumble.pairing import PairingConfig
+from bumble.sdp import ServiceAttribute
+from bumble.transport import open_transport
+from bumble.avdtp import (
+    AVDTP_AUDIO_MEDIA_TYPE,
+    Listener,
+    MediaCodecCapabilities,
+    MediaPacket,
+    Protocol,
+)
+from bumble.a2dp import (
+    MPEG_2_AAC_LC_OBJECT_TYPE,
+    make_audio_sink_service_sdp_records,
+    A2DP_SBC_CODEC_TYPE,
+    A2DP_MPEG_2_4_AAC_CODEC_TYPE,
+    SBC_MONO_CHANNEL_MODE,
+    SBC_DUAL_CHANNEL_MODE,
+    SBC_SNR_ALLOCATION_METHOD,
+    SBC_LOUDNESS_ALLOCATION_METHOD,
+    SBC_STEREO_CHANNEL_MODE,
+    SBC_JOINT_STEREO_CHANNEL_MODE,
+    SbcMediaCodecInformation,
+    AacMediaCodecInformation,
+)
+from bumble.utils import AsyncRunner
+from bumble.codecs import AacAudioRtpPacket
+
+
+# -----------------------------------------------------------------------------
+# Logging
+# -----------------------------------------------------------------------------
+logger = logging.getLogger(__name__)
+
+
+# -----------------------------------------------------------------------------
+# Constants
+# -----------------------------------------------------------------------------
+DEFAULT_UI_PORT = 7654
+
+# -----------------------------------------------------------------------------
+class AudioExtractor:
+    @staticmethod
+    def create(codec: str):
+        if codec == 'aac':
+            return AacAudioExtractor()
+        if codec == 'sbc':
+            return SbcAudioExtractor()
+
+    def extract_audio(self, packet: MediaPacket) -> bytes:
+        raise NotImplementedError()
+
+
+# -----------------------------------------------------------------------------
+class AacAudioExtractor:
+    def extract_audio(self, packet: MediaPacket) -> bytes:
+        return AacAudioRtpPacket(packet.payload).to_adts()
+
+
+# -----------------------------------------------------------------------------
+class SbcAudioExtractor:
+    def extract_audio(self, packet: MediaPacket) -> bytes:
+        # header = packet.payload[0]
+        # fragmented = header >> 7
+        # start = (header >> 6) & 0x01
+        # last = (header >> 5) & 0x01
+        # number_of_frames = header & 0x0F
+
+        # TODO: support fragmented payloads
+        return packet.payload[1:]
+
+
+# -----------------------------------------------------------------------------
+class Output:
+    async def start(self) -> None:
+        pass
+
+    async def stop(self) -> None:
+        pass
+
+    async def suspend(self) -> None:
+        pass
+
+    async def on_connection(self, connection: Connection) -> None:
+        pass
+
+    async def on_disconnection(self, reason: int) -> None:
+        pass
+
+    def on_rtp_packet(self, packet: MediaPacket) -> None:
+        pass
+
+
+# -----------------------------------------------------------------------------
+class FileOutput(Output):
+    filename: str
+    codec: str
+    extractor: AudioExtractor
+
+    def __init__(self, filename, codec):
+        self.filename = filename
+        self.codec = codec
+        self.file = open(filename, 'wb')
+        self.extractor = AudioExtractor.create(codec)
+
+    def on_rtp_packet(self, packet: MediaPacket) -> None:
+        self.file.write(self.extractor.extract_audio(packet))
+
+
+# -----------------------------------------------------------------------------
+class QueuedOutput(Output):
+    MAX_QUEUE_SIZE = 32768
+
+    packets: asyncio.Queue
+    extractor: AudioExtractor
+    packet_pump_task: Optional[asyncio.Task]
+    started: bool
+
+    def __init__(self, extractor):
+        self.extractor = extractor
+        self.packets = asyncio.Queue()
+        self.packet_pump_task = None
+        self.started = False
+
+    async def start(self):
+        if self.started:
+            return
+
+        self.packet_pump_task = asyncio.create_task(self.pump_packets())
+
+    async def pump_packets(self):
+        while True:
+            packet = await self.packets.get()
+            await self.on_audio_packet(packet)
+
+    async def on_audio_packet(self, packet: bytes) -> None:
+        pass
+
+    def on_rtp_packet(self, packet: MediaPacket) -> None:
+        if self.packets.qsize() > self.MAX_QUEUE_SIZE:
+            logger.debug("queue full, dropping")
+            return
+
+        self.packets.put_nowait(self.extractor.extract_audio(packet))
+
+
+# -----------------------------------------------------------------------------
+class WebSocketOutput(QueuedOutput):
+    def __init__(self, codec, send_audio, send_message):
+        super().__init__(AudioExtractor.create(codec))
+        self.send_audio = send_audio
+        self.send_message = send_message
+
+    async def on_connection(self, connection: Connection) -> None:
+        try:
+            await connection.request_remote_name()
+        except HCI_StatusError:
+            pass
+        peer_name = '' if connection.peer_name is None else connection.peer_name
+        peer_address = str(connection.peer_address).replace('/P', '')
+        await self.send_message(
+            'connection',
+            peer_address=peer_address,
+            peer_name=peer_name,
+        )
+
+    async def on_disconnection(self, reason) -> None:
+        await self.send_message('disconnection')
+
+    async def on_audio_packet(self, packet: bytes) -> None:
+        await self.send_audio(packet)
+
+    async def start(self):
+        await super().start()
+        await self.send_message('start')
+
+    async def stop(self):
+        await super().stop()
+        await self.send_message('stop')
+
+    async def suspend(self):
+        await super().suspend()
+        await self.send_message('suspend')
+
+
+# -----------------------------------------------------------------------------
+class FfplayOutput(QueuedOutput):
+    MAX_QUEUE_SIZE = 32768
+
+    subprocess: Optional[asyncio.subprocess.Process]
+    ffplay_task: Optional[asyncio.Task]
+
+    def __init__(self) -> None:
+        super().__init__(AacAudioExtractor())
+        self.subprocess = None
+        self.ffplay_task = None
+
+    async def start(self):
+        if self.started:
+            return
+
+        await super().start()
+
+        self.subprocess = await asyncio.create_subprocess_shell(
+            'ffplay -acodec aac pipe:0',
+            stdin=asyncio.subprocess.PIPE,
+            stdout=asyncio.subprocess.PIPE,
+            stderr=asyncio.subprocess.PIPE,
+        )
+
+        self.ffplay_task = asyncio.create_task(self.monitor_ffplay())
+
+    async def stop(self):
+        # TODO
+        pass
+
+    async def suspend(self):
+        # TODO
+        pass
+
+    async def monitor_ffplay(self):
+        async def read_stream(name, stream):
+            while True:
+                data = await stream.read()
+                logger.debug(f'{name}:', data)
+
+        await asyncio.wait(
+            [
+                asyncio.create_task(
+                    read_stream('[ffplay stdout]', self.subprocess.stdout)
+                ),
+                asyncio.create_task(
+                    read_stream('[ffplay stderr]', self.subprocess.stderr)
+                ),
+                asyncio.create_task(self.subprocess.wait()),
+            ]
+        )
+        logger.debug("FFPLAY done")
+
+    async def on_audio_packet(self, packet):
+        try:
+            self.subprocess.stdin.write(packet)
+        except Exception:
+            logger.warning('!!!! exception while sending audio to ffplay pipe')
+
+
+# -----------------------------------------------------------------------------
+class UiServer:
+    speaker: weakref.ReferenceType[Speaker]
+    port: int
+
+    def __init__(self, speaker: Speaker, port: int) -> None:
+        self.speaker = weakref.ref(speaker)
+        self.port = port
+        self.channel_socket = None
+
+    async def start_http(self) -> None:
+        """Start the UI HTTP server."""
+
+        app = web.Application()
+        app.add_routes(
+            [
+                web.get('/', self.get_static),
+                web.get('/speaker.html', self.get_static),
+                web.get('/speaker.js', self.get_static),
+                web.get('/speaker.css', self.get_static),
+                web.get('/logo.svg', self.get_static),
+                web.get('/channel', self.get_channel),
+            ]
+        )
+
+        runner = web.AppRunner(app)
+        await runner.setup()
+        site = web.TCPSite(runner, 'localhost', self.port)
+        print('UI HTTP server at ' + color(f'http://127.0.0.1:{self.port}', 'green'))
+        await site.start()
+
+    async def get_static(self, request):
+        path = request.path
+        if path == '/':
+            path = '/speaker.html'
+        if path.endswith('.html'):
+            content_type = 'text/html'
+        elif path.endswith('.js'):
+            content_type = 'text/javascript'
+        elif path.endswith('.css'):
+            content_type = 'text/css'
+        elif path.endswith('.svg'):
+            content_type = 'image/svg+xml'
+        else:
+            content_type = 'text/plain'
+        text = (
+            resources.files("bumble.apps.speaker")
+            .joinpath(pathlib.Path(path).relative_to('/'))
+            .read_text(encoding="utf-8")
+        )
+        return aiohttp.web.Response(text=text, content_type=content_type)
+
+    async def get_channel(self, request):
+        ws = web.WebSocketResponse()
+        await ws.prepare(request)
+
+        # Process messages until the socket is closed.
+        self.channel_socket = ws
+        async for message in ws:
+            if message.type == aiohttp.WSMsgType.TEXT:
+                logger.debug(f'<<< received message: {message.data}')
+                await self.on_message(message.data)
+            elif message.type == aiohttp.WSMsgType.ERROR:
+                logger.debug(
+                    f'channel connection closed with exception {ws.exception()}'
+                )
+
+        self.channel_socket = None
+        logger.debug('--- channel connection closed')
+
+        return ws
+
+    async def on_message(self, message_str: str):
+        # Parse the message as JSON
+        message = json.loads(message_str)
+
+        # Dispatch the message
+        message_type = message['type']
+        message_params = message.get('params', {})
+        handler = getattr(self, f'on_{message_type}_message')
+        if handler:
+            await handler(**message_params)
+
+    async def on_hello_message(self):
+        await self.send_message(
+            'hello',
+            bumble_version=bumble.__version__,
+            codec=self.speaker().codec,
+            streamState=self.speaker().stream_state.name,
+        )
+        if connection := self.speaker().connection:
+            await self.send_message(
+                'connection',
+                peer_address=str(connection.peer_address).replace('/P', ''),
+                peer_name=connection.peer_name,
+            )
+
+    async def send_message(self, message_type: str, **kwargs) -> None:
+        if self.channel_socket is None:
+            return
+
+        message = {'type': message_type, 'params': kwargs}
+        await self.channel_socket.send_json(message)
+
+    async def send_audio(self, data: bytes) -> None:
+        if self.channel_socket is None:
+            return
+
+        try:
+            await self.channel_socket.send_bytes(data)
+        except Exception as error:
+            logger.warning(f'exception while sending audio packet: {error}')
+
+
+# -----------------------------------------------------------------------------
+class Speaker:
+    class StreamState(enum.Enum):
+        IDLE = 0
+        STOPPED = 1
+        STARTED = 2
+        SUSPENDED = 3
+
+    def __init__(self, device_config, transport, codec, discover, outputs, ui_port):
+        self.device_config = device_config
+        self.transport = transport
+        self.codec = codec
+        self.discover = discover
+        self.ui_port = ui_port
+        self.device = None
+        self.connection = None
+        self.listener = None
+        self.packets_received = 0
+        self.bytes_received = 0
+        self.stream_state = Speaker.StreamState.IDLE
+        self.outputs = []
+        for output in outputs:
+            if output == '@ffplay':
+                self.outputs.append(FfplayOutput())
+                continue
+
+            # Default to FileOutput
+            self.outputs.append(FileOutput(output, codec))
+
+        # Create an HTTP server for the UI
+        self.ui_server = UiServer(speaker=self, port=ui_port)
+
+    def sdp_records(self) -> Dict[int, List[ServiceAttribute]]:
+        service_record_handle = 0x00010001
+        return {
+            service_record_handle: make_audio_sink_service_sdp_records(
+                service_record_handle
+            )
+        }
+
+    def codec_capabilities(self) -> MediaCodecCapabilities:
+        if self.codec == 'aac':
+            return self.aac_codec_capabilities()
+
+        if self.codec == 'sbc':
+            return self.sbc_codec_capabilities()
+
+        raise RuntimeError('unsupported codec')
+
+    def aac_codec_capabilities(self) -> MediaCodecCapabilities:
+        return MediaCodecCapabilities(
+            media_type=AVDTP_AUDIO_MEDIA_TYPE,
+            media_codec_type=A2DP_MPEG_2_4_AAC_CODEC_TYPE,
+            media_codec_information=AacMediaCodecInformation.from_lists(
+                object_types=[MPEG_2_AAC_LC_OBJECT_TYPE],
+                sampling_frequencies=[48000, 44100],
+                channels=[1, 2],
+                vbr=1,
+                bitrate=256000,
+            ),
+        )
+
+    def sbc_codec_capabilities(self) -> MediaCodecCapabilities:
+        return MediaCodecCapabilities(
+            media_type=AVDTP_AUDIO_MEDIA_TYPE,
+            media_codec_type=A2DP_SBC_CODEC_TYPE,
+            media_codec_information=SbcMediaCodecInformation.from_lists(
+                sampling_frequencies=[48000, 44100, 32000, 16000],
+                channel_modes=[
+                    SBC_MONO_CHANNEL_MODE,
+                    SBC_DUAL_CHANNEL_MODE,
+                    SBC_STEREO_CHANNEL_MODE,
+                    SBC_JOINT_STEREO_CHANNEL_MODE,
+                ],
+                block_lengths=[4, 8, 12, 16],
+                subbands=[4, 8],
+                allocation_methods=[
+                    SBC_LOUDNESS_ALLOCATION_METHOD,
+                    SBC_SNR_ALLOCATION_METHOD,
+                ],
+                minimum_bitpool_value=2,
+                maximum_bitpool_value=53,
+            ),
+        )
+
+    async def dispatch_to_outputs(self, function):
+        for output in self.outputs:
+            await function(output)
+
+    def on_bluetooth_connection(self, connection):
+        print(f'Connection: {connection}')
+        self.connection = connection
+        connection.on('disconnection', self.on_bluetooth_disconnection)
+        AsyncRunner.spawn(
+            self.dispatch_to_outputs(lambda output: output.on_connection(connection))
+        )
+
+    def on_bluetooth_disconnection(self, reason):
+        print(f'Disconnection ({reason})')
+        self.connection = None
+        AsyncRunner.spawn(self.advertise())
+        AsyncRunner.spawn(
+            self.dispatch_to_outputs(lambda output: output.on_disconnection(reason))
+        )
+
+    def on_avdtp_connection(self, protocol):
+        print('Audio Stream Open')
+
+        # Add a sink endpoint to the server
+        sink = protocol.add_sink(self.codec_capabilities())
+        sink.on('start', self.on_sink_start)
+        sink.on('stop', self.on_sink_stop)
+        sink.on('suspend', self.on_sink_suspend)
+        sink.on('configuration', lambda: self.on_sink_configuration(sink.configuration))
+        sink.on('rtp_packet', self.on_rtp_packet)
+        sink.on('rtp_channel_open', self.on_rtp_channel_open)
+        sink.on('rtp_channel_close', self.on_rtp_channel_close)
+
+        # Listen for close events
+        protocol.on('close', self.on_avdtp_close)
+
+        # Discover all endpoints on the remote device is requested
+        if self.discover:
+            AsyncRunner.spawn(self.discover_remote_endpoints(protocol))
+
+    def on_avdtp_close(self):
+        print("Audio Stream Closed")
+
+    def on_sink_start(self):
+        print("Sink Started\u001b[0K")
+        self.stream_state = self.StreamState.STARTED
+        AsyncRunner.spawn(self.dispatch_to_outputs(lambda output: output.start()))
+
+    def on_sink_stop(self):
+        print("Sink Stopped\u001b[0K")
+        self.stream_state = self.StreamState.STOPPED
+        AsyncRunner.spawn(self.dispatch_to_outputs(lambda output: output.stop()))
+
+    def on_sink_suspend(self):
+        print("Sink Suspended\u001b[0K")
+        self.stream_state = self.StreamState.SUSPENDED
+        AsyncRunner.spawn(self.dispatch_to_outputs(lambda output: output.suspend()))
+
+    def on_sink_configuration(self, config):
+        print("Sink Configuration:")
+        print('\n'.join(["  " + str(capability) for capability in config]))
+
+    def on_rtp_channel_open(self):
+        print("RTP Channel Open")
+
+    def on_rtp_channel_close(self):
+        print("RTP Channel Closed")
+        self.stream_state = self.StreamState.IDLE
+
+    def on_rtp_packet(self, packet):
+        self.packets_received += 1
+        self.bytes_received += len(packet.payload)
+        print(
+            f'[{self.bytes_received} bytes in {self.packets_received} packets] {packet}',
+            end='\r',
+        )
+
+        for output in self.outputs:
+            output.on_rtp_packet(packet)
+
+    async def advertise(self):
+        await self.device.set_discoverable(True)
+        await self.device.set_connectable(True)
+
+    async def connect(self, address):
+        # Connect to the source
+        print(f'=== Connecting to {address}...')
+        connection = await self.device.connect(address, transport=BT_BR_EDR_TRANSPORT)
+        print(f'=== Connected to {connection.peer_address}')
+
+        # Request authentication
+        print('*** Authenticating...')
+        await connection.authenticate()
+        print('*** Authenticated')
+
+        # Enable encryption
+        print('*** Enabling encryption...')
+        await connection.encrypt()
+        print('*** Encryption on')
+
+        protocol = await Protocol.connect(connection)
+        self.listener.set_server(connection, protocol)
+        self.on_avdtp_connection(protocol)
+
+    async def discover_remote_endpoints(self, protocol):
+        endpoints = await protocol.discover_remote_endpoints()
+        print(f'@@@ Found {len(endpoints)} endpoints')
+        for endpoint in endpoints:
+            print('@@@', endpoint)
+
+    async def run(self, connect_address):
+        await self.ui_server.start_http()
+        self.outputs.append(
+            WebSocketOutput(
+                self.codec, self.ui_server.send_audio, self.ui_server.send_message
+            )
+        )
+
+        async with await open_transport(self.transport) as (hci_source, hci_sink):
+            # Create a device
+            device_config = DeviceConfiguration()
+            if self.device_config:
+                device_config.load_from_file(self.device_config)
+            else:
+                device_config.name = "Bumble Speaker"
+                device_config.class_of_device = 0x240414
+                device_config.keystore = "JsonKeyStore"
+
+            device_config.classic_enabled = True
+            device_config.le_enabled = False
+            self.device = Device.from_config_with_hci(
+                device_config, hci_source, hci_sink
+            )
+
+            # Setup the SDP to expose the sink service
+            self.device.sdp_service_records = self.sdp_records()
+
+            # Don't require MITM when pairing.
+            self.device.pairing_config_factory = lambda connection: PairingConfig(
+                mitm=False
+            )
+
+            # Start the controller
+            await self.device.power_on()
+
+            # Print some of the config/properties
+            print("Speaker Name:", color(device_config.name, 'yellow'))
+            print(
+                "Speaker Bluetooth Address:",
+                color(
+                    self.device.public_address.to_string(with_type_qualifier=False),
+                    'yellow',
+                ),
+            )
+
+            # Listen for Bluetooth connections
+            self.device.on('connection', self.on_bluetooth_connection)
+
+            # Create a listener to wait for AVDTP connections
+            self.listener = Listener(Listener.create_registrar(self.device))
+            self.listener.on('connection', self.on_avdtp_connection)
+
+            print(f'Speaker ready to play, codec={color(self.codec, "cyan")}')
+
+            if connect_address:
+                # Connect to the source
+                try:
+                    await self.connect(connect_address)
+                except CommandTimeoutError:
+                    print(color("Connection timed out", "red"))
+                    return
+            else:
+                # Start being discoverable and connectable
+                print("Waiting for connection...")
+                await self.advertise()
+
+            await hci_source.wait_for_termination()
+
+        for output in self.outputs:
+            await output.stop()
+
+
+# -----------------------------------------------------------------------------
+@click.group()
+@click.pass_context
+def speaker_cli(ctx, device_config):
+    ctx.ensure_object(dict)
+    ctx.obj['device_config'] = device_config
+
+
+@click.command()
+@click.option(
+    '--codec', type=click.Choice(['sbc', 'aac']), default='aac', show_default=True
+)
+@click.option(
+    '--discover', is_flag=True, help='Discover remote endpoints once connected'
+)
+@click.option(
+    '--output',
+    multiple=True,
+    metavar='NAME',
+    help=(
+        'Send audio to this named output '
+        '(may be used more than once for multiple outputs)'
+    ),
+)
+@click.option(
+    '--ui-port',
+    'ui_port',
+    metavar='HTTP_PORT',
+    default=DEFAULT_UI_PORT,
+    show_default=True,
+    help='HTTP port for the UI server',
+)
+@click.option(
+    '--connect',
+    'connect_address',
+    metavar='ADDRESS_OR_NAME',
+    help='Address or name to connect to',
+)
+@click.option('--device-config', metavar='FILENAME', help='Device configuration file')
+@click.argument('transport')
+def speaker(
+    transport, codec, connect_address, discover, output, ui_port, device_config
+):
+    """Run the speaker."""
+
+    # ffplay only works with AAC for now
+    if codec != 'aac' and '@ffplay' in output:
+        print(
+            color(
+                f'{codec} not supported with @ffplay output, '
+                '@ffplay output will be skipped',
+                'yellow',
+            )
+        )
+        output = list(filter(lambda x: x != '@ffplay', output))
+
+    if '@ffplay' in output:
+        # Check if ffplay is installed
+        try:
+            subprocess.run(['ffplay', '-version'], capture_output=True, check=True)
+        except FileNotFoundError:
+            print(
+                color('ffplay not installed, @ffplay output will be disabled', 'yellow')
+            )
+            output = list(filter(lambda x: x != '@ffplay', output))
+
+    asyncio.run(
+        Speaker(device_config, transport, codec, discover, output, ui_port).run(
+            connect_address
+        )
+    )
+
+
+# -----------------------------------------------------------------------------
+def main():
+    logging.basicConfig(level=os.environ.get('BUMBLE_LOGLEVEL', 'WARNING').upper())
+    speaker()
+
+
+# -----------------------------------------------------------------------------
+if __name__ == "__main__":
+    main()  # pylint: disable=no-value-for-parameter

bumble/a2dp.py

@@ -432,6 +432,7 @@ class AacMediaCodecInformation(
                 cls.SAMPLING_FREQUENCY_BITS[x] for x in sampling_frequencies
             ),
             channels=sum(cls.CHANNELS_BITS[x] for x in channels),
+            rfa=0,
             vbr=vbr,
             bitrate=bitrate,
         )

bumble/avdtp.py

@@ -1207,7 +1207,7 @@ class DelayReport_Reject(Simple_Reject):
 
 
 # -----------------------------------------------------------------------------
-class Protocol:
+class Protocol(EventEmitter):
     SINGLE_PACKET = 0
     START_PACKET = 1
     CONTINUE_PACKET = 2
@@ -1234,6 +1234,7 @@ class Protocol:
         return protocol
 
     def __init__(self, l2cap_channel, version=(1, 3)):
+        super().__init__()
         self.l2cap_channel = l2cap_channel
         self.version = version
         self.rtx_sig_timer = AVDTP_DEFAULT_RTX_SIG_TIMER
@@ -1250,6 +1251,7 @@ class Protocol:
         # Register to receive PDUs from the channel
         l2cap_channel.sink = self.on_pdu
         l2cap_channel.on('open', self.on_l2cap_channel_open)
+        l2cap_channel.on('close', self.on_l2cap_channel_close)
 
     def get_local_endpoint_by_seid(self, seid):
         if 0 < seid <= len(self.local_endpoints):
@@ -1392,11 +1394,18 @@ class Protocol:
 
     def on_l2cap_connection(self, channel):
         # Forward the channel to the endpoint that's expecting it
-        if self.channel_acceptor:
+        if self.channel_acceptor is None:
-            self.channel_acceptor.on_l2cap_connection(channel)
+            logger.warning(color('!!! l2cap connection with no acceptor', 'red'))
+            return
+        self.channel_acceptor.on_l2cap_connection(channel)
 
     def on_l2cap_channel_open(self):
         logger.debug(color('<<< L2CAP channel open', 'magenta'))
+        self.emit('open')
+
+    def on_l2cap_channel_close(self):
+        logger.debug(color('<<< L2CAP channel close', 'magenta'))
+        self.emit('close')
 
     def send_message(self, transaction_label, message):
         logger.debug(
@@ -1651,6 +1660,10 @@ class Listener(EventEmitter):
     def set_server(self, connection, server):
         self.servers[connection.handle] = server
 
+    def remove_server(self, connection):
+        if connection.handle in self.servers:
+            del self.servers[connection.handle]
+
     def __init__(self, registrar, version=(1, 3)):
         super().__init__()
         self.version = version
@@ -1669,11 +1682,17 @@ class Listener(EventEmitter):
         else:
             # This is a new command/response channel
             def on_channel_open():
+                logger.debug('setting up new Protocol for the connection')
                 server = Protocol(channel, self.version)
                 self.set_server(channel.connection, server)
                 self.emit('connection', server)
 
+            def on_channel_close():
+                logger.debug('removing Protocol for the connection')
+                self.remove_server(channel.connection)
+
             channel.on('open', on_channel_open)
+            channel.on('close', on_channel_close)
 
 
 # -----------------------------------------------------------------------------
@@ -1967,11 +1986,12 @@ class DiscoveredStreamEndPoint(StreamEndPoint, StreamEndPointProxy):
 
 
 # -----------------------------------------------------------------------------
-class LocalStreamEndPoint(StreamEndPoint):
+class LocalStreamEndPoint(StreamEndPoint, EventEmitter):
     def __init__(
         self, protocol, seid, media_type, tsep, capabilities, configuration=None
     ):
-        super().__init__(seid, media_type, tsep, 0, capabilities)
+        StreamEndPoint.__init__(self, seid, media_type, tsep, 0, capabilities)
+        EventEmitter.__init__(self)
         self.protocol = protocol
         self.configuration = configuration if configuration is not None else []
         self.stream = None
@@ -1988,40 +2008,47 @@ class LocalStreamEndPoint(StreamEndPoint):
     def on_reconfigure_command(self, command):
         pass
 
+    def on_set_configuration_command(self, configuration):
+        logger.debug(
+            '<<< received configuration: '
+            f'{",".join([str(capability) for capability in configuration])}'
+        )
+        self.configuration = configuration
+        self.emit('configuration')
+
     def on_get_configuration_command(self):
         return Get_Configuration_Response(self.configuration)
 
     def on_open_command(self):
-        pass
+        self.emit('open')
 
     def on_start_command(self):
-        pass
+        self.emit('start')
 
     def on_suspend_command(self):
-        pass
+        self.emit('suspend')
 
     def on_close_command(self):
-        pass
+        self.emit('close')
 
     def on_abort_command(self):
-        pass
+        self.emit('abort')
 
     def on_rtp_channel_open(self):
-        pass
+        self.emit('rtp_channel_open')
 
     def on_rtp_channel_close(self):
-        pass
+        self.emit('rtp_channel_close')
 
 
 # -----------------------------------------------------------------------------
-class LocalSource(LocalStreamEndPoint, EventEmitter):
+class LocalSource(LocalStreamEndPoint):
     def __init__(self, protocol, seid, codec_capabilities, packet_pump):
         capabilities = [
             ServiceCapabilities(AVDTP_MEDIA_TRANSPORT_SERVICE_CATEGORY),
             codec_capabilities,
         ]
-        LocalStreamEndPoint.__init__(
+        super().__init__(
-            self,
             protocol,
             seid,
             codec_capabilities.media_type,
@@ -2029,14 +2056,13 @@ class LocalSource(LocalStreamEndPoint, EventEmitter):
             capabilities,
             capabilities,
         )
-        EventEmitter.__init__(self)
         self.packet_pump = packet_pump
 
     async def start(self):
         if self.packet_pump:
             return await self.packet_pump.start(self.stream.rtp_channel)
 
-        self.emit('start', self.stream.rtp_channel)
+        self.emit('start')
 
     async def stop(self):
         if self.packet_pump:
@@ -2044,11 +2070,6 @@ class LocalSource(LocalStreamEndPoint, EventEmitter):
 
         self.emit('stop')
 
-    def on_set_configuration_command(self, configuration):
-        # For now, blindly accept the configuration
-        logger.debug(f'<<< received source configuration: {configuration}')
-        self.configuration = configuration
-
     def on_start_command(self):
         asyncio.create_task(self.start())
 
@@ -2057,30 +2078,28 @@ class LocalSource(LocalStreamEndPoint, EventEmitter):
 
 
 # -----------------------------------------------------------------------------
-class LocalSink(LocalStreamEndPoint, EventEmitter):
+class LocalSink(LocalStreamEndPoint):
     def __init__(self, protocol, seid, codec_capabilities):
         capabilities = [
             ServiceCapabilities(AVDTP_MEDIA_TRANSPORT_SERVICE_CATEGORY),
             codec_capabilities,
         ]
-        LocalStreamEndPoint.__init__(
+        super().__init__(
-            self,
             protocol,
             seid,
             codec_capabilities.media_type,
             AVDTP_TSEP_SNK,
             capabilities,
         )
-        EventEmitter.__init__(self)
-
-    def on_set_configuration_command(self, configuration):
-        # For now, blindly accept the configuration
-        logger.debug(f'<<< received sink configuration: {configuration}')
-        self.configuration = configuration
 
     def on_rtp_channel_open(self):
         logger.debug(color('<<< RTP channel open', 'magenta'))
         self.stream.rtp_channel.sink = self.on_avdtp_packet
+        super().on_rtp_channel_open()
+
+    def on_rtp_channel_close(self):
+        logger.debug(color('<<< RTP channel close', 'magenta'))
+        super().on_rtp_channel_close()
 
     def on_avdtp_packet(self, packet):
         rtp_packet = MediaPacket.from_bytes(packet)

bumble/codecs.py

@@ -0,0 +1,381 @@
+# Copyright 2023 Google LLC
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#      https://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+# -----------------------------------------------------------------------------
+# Imports
+# -----------------------------------------------------------------------------
+from __future__ import annotations
+from dataclasses import dataclass
+
+
+# -----------------------------------------------------------------------------
+class BitReader:
+    """Simple but not optimized bit stream reader."""
+
+    data: bytes
+    bytes_position: int
+    bit_position: int
+    cache: int
+    bits_cached: int
+
+    def __init__(self, data: bytes):
+        self.data = data
+        self.byte_position = 0
+        self.bit_position = 0
+        self.cache = 0
+        self.bits_cached = 0
+
+    def read(self, bits: int) -> int:
+        """ "Read up to 32 bits."""
+
+        if bits > 32:
+            raise ValueError('maximum read size is 32')
+
+        if self.bits_cached >= bits:
+            # We have enough bits.
+            self.bits_cached -= bits
+            self.bit_position += bits
+            return (self.cache >> self.bits_cached) & ((1 << bits) - 1)
+
+        # Read more cache, up to 32 bits
+        feed_bytes = self.data[self.byte_position : self.byte_position + 4]
+        feed_size = len(feed_bytes)
+        feed_int = int.from_bytes(feed_bytes, byteorder='big')
+        if 8 * feed_size + self.bits_cached < bits:
+            raise ValueError('trying to read past the data')
+        self.byte_position += feed_size
+
+        # Combine the new cache and the old cache
+        cache = self.cache & ((1 << self.bits_cached) - 1)
+        new_bits = bits - self.bits_cached
+        self.bits_cached = 8 * feed_size - new_bits
+        result = (feed_int >> self.bits_cached) | (cache << new_bits)
+        self.cache = feed_int
+
+        self.bit_position += bits
+        return result
+
+    def read_bytes(self, count: int):
+        if self.bit_position + 8 * count > 8 * len(self.data):
+            raise ValueError('not enough data')
+
+        if self.bit_position % 8:
+            # Not byte aligned
+            result = bytearray(count)
+            for i in range(count):
+                result[i] = self.read(8)
+            return bytes(result)
+
+        # Byte aligned
+        self.byte_position = self.bit_position // 8
+        self.bits_cached = 0
+        self.cache = 0
+        offset = self.bit_position // 8
+        self.bit_position += 8 * count
+        return self.data[offset : offset + count]
+
+    def bits_left(self) -> int:
+        return (8 * len(self.data)) - self.bit_position
+
+    def skip(self, bits: int) -> None:
+        # Slow, but simple...
+        while bits:
+            if bits > 32:
+                self.read(32)
+                bits -= 32
+            else:
+                self.read(bits)
+                break
+
+
+# -----------------------------------------------------------------------------
+class AacAudioRtpPacket:
+    """AAC payload encapsulated in an RTP packet payload"""
+
+    @staticmethod
+    def latm_value(reader: BitReader) -> int:
+        bytes_for_value = reader.read(2)
+        value = 0
+        for _ in range(bytes_for_value + 1):
+            value = value * 256 + reader.read(8)
+        return value
+
+    @staticmethod
+    def program_config_element(reader: BitReader):
+        raise ValueError('program_config_element not supported')
+
+    @dataclass
+    class GASpecificConfig:
+        def __init__(
+            self, reader: BitReader, channel_configuration: int, audio_object_type: int
+        ) -> None:
+            # GASpecificConfig - ISO/EIC 14496-3 Table 4.1
+            frame_length_flag = reader.read(1)
+            depends_on_core_coder = reader.read(1)
+            if depends_on_core_coder:
+                self.core_coder_delay = reader.read(14)
+            extension_flag = reader.read(1)
+            if not channel_configuration:
+                AacAudioRtpPacket.program_config_element(reader)
+            if audio_object_type in (6, 20):
+                self.layer_nr = reader.read(3)
+            if extension_flag:
+                if audio_object_type == 22:
+                    num_of_sub_frame = reader.read(5)
+                layer_length = reader.read(11)
+                if audio_object_type in (17, 19, 20, 23):
+                    aac_section_data_resilience_flags = reader.read(1)
+                    aac_scale_factor_data_resilience_flags = reader.read(1)
+                    aac_spectral_data_resilience_flags = reader.read(1)
+                extension_flag_3 = reader.read(1)
+                if extension_flag_3 == 1:
+                    raise ValueError('extensionFlag3 == 1 not supported')
+
+    @staticmethod
+    def audio_object_type(reader: BitReader):
+        # GetAudioObjectType - ISO/EIC 14496-3 Table 1.16
+        audio_object_type = reader.read(5)
+        if audio_object_type == 31:
+            audio_object_type = 32 + reader.read(6)
+
+        return audio_object_type
+
+    @dataclass
+    class AudioSpecificConfig:
+        audio_object_type: int
+        sampling_frequency_index: int
+        sampling_frequency: int
+        channel_configuration: int
+        sbr_present_flag: int
+        ps_present_flag: int
+        extension_audio_object_type: int
+        extension_sampling_frequency_index: int
+        extension_sampling_frequency: int
+        extension_channel_configuration: int
+
+        SAMPLING_FREQUENCIES = [
+            96000,
+            88200,
+            64000,
+            48000,
+            44100,
+            32000,
+            24000,
+            22050,
+            16000,
+            12000,
+            11025,
+            8000,
+            7350,
+        ]
+
+        def __init__(self, reader: BitReader) -> None:
+            # AudioSpecificConfig - ISO/EIC 14496-3 Table 1.15
+            self.audio_object_type = AacAudioRtpPacket.audio_object_type(reader)
+            self.sampling_frequency_index = reader.read(4)
+            if self.sampling_frequency_index == 0xF:
+                self.sampling_frequency = reader.read(24)
+            else:
+                self.sampling_frequency = self.SAMPLING_FREQUENCIES[
+                    self.sampling_frequency_index
+                ]
+            self.channel_configuration = reader.read(4)
+            self.sbr_present_flag = -1
+            self.ps_present_flag = -1
+            if self.audio_object_type in (5, 29):
+                self.extension_audio_object_type = 5
+                self.sbc_present_flag = 1
+                if self.audio_object_type == 29:
+                    self.ps_present_flag = 1
+                self.extension_sampling_frequency_index = reader.read(4)
+                if self.extension_sampling_frequency_index == 0xF:
+                    self.extension_sampling_frequency = reader.read(24)
+                else:
+                    self.extension_sampling_frequency = self.SAMPLING_FREQUENCIES[
+                        self.extension_sampling_frequency_index
+                    ]
+                self.audio_object_type = AacAudioRtpPacket.audio_object_type(reader)
+                if self.audio_object_type == 22:
+                    self.extension_channel_configuration = reader.read(4)
+            else:
+                self.extension_audio_object_type = 0
+
+            if self.audio_object_type in (1, 2, 3, 4, 6, 7, 17, 19, 20, 21, 22, 23):
+                ga_specific_config = AacAudioRtpPacket.GASpecificConfig(
+                    reader, self.channel_configuration, self.audio_object_type
+                )
+            else:
+                raise ValueError(
+                    f'audioObjectType {self.audio_object_type} not supported'
+                )
+
+            # if self.extension_audio_object_type != 5 and bits_to_decode >= 16:
+            #     sync_extension_type = reader.read(11)
+            #     if sync_extension_type == 0x2B7:
+            #         self.extension_audio_object_type = AacAudioRtpPacket.audio_object_type(reader)
+            #         if self.extension_audio_object_type == 5:
+            #             self.sbr_present_flag = reader.read(1)
+            #             if self.sbr_present_flag:
+            #                 self.extension_sampling_frequency_index = reader.read(4)
+            #                 if self.extension_sampling_frequency_index == 0xF:
+            #                     self.extension_sampling_frequency = reader.read(24)
+            #                 else:
+            #                     self.extension_sampling_frequency = self.SAMPLING_FREQUENCIES[self.extension_sampling_frequency_index]
+            #                 if bits_to_decode >= 12:
+            #                     sync_extension_type = reader.read(11)
+            #                     if sync_extension_type == 0x548:
+            #                         self.ps_present_flag = reader.read(1)
+            #         elif self.extension_audio_object_type == 22:
+            #             self.sbr_present_flag = reader.read(1)
+            #             if self.sbr_present_flag:
+            #                 self.extension_sampling_frequency_index = reader.read(4)
+            #                 if self.extension_sampling_frequency_index == 0xF:
+            #                     self.extension_sampling_frequency = reader.read(24)
+            #                 else:
+            #                     self.extension_sampling_frequency = self.SAMPLING_FREQUENCIES[self.extension_sampling_frequency_index]
+            #             self.extension_channel_configuration = reader.read(4)
+
+    @dataclass
+    class StreamMuxConfig:
+        other_data_present: int
+        other_data_len_bits: int
+        audio_specific_config: AacAudioRtpPacket.AudioSpecificConfig
+
+        def __init__(self, reader: BitReader) -> None:
+            # StreamMuxConfig - ISO/EIC 14496-3 Table 1.42
+            audio_mux_version = reader.read(1)
+            if audio_mux_version == 1:
+                audio_mux_version_a = reader.read(1)
+            else:
+                audio_mux_version_a = 0
+            if audio_mux_version_a != 0:
+                raise ValueError('audioMuxVersionA != 0 not supported')
+            if audio_mux_version == 1:
+                tara_buffer_fullness = AacAudioRtpPacket.latm_value(reader)
+            stream_cnt = 0
+            all_streams_same_time_framing = reader.read(1)
+            num_sub_frames = reader.read(6)
+            num_program = reader.read(4)
+            if num_program != 0:
+                raise ValueError('num_program != 0 not supported')
+            num_layer = reader.read(3)
+            if num_layer != 0:
+                raise ValueError('num_layer != 0 not supported')
+            if audio_mux_version == 0:
+                self.audio_specific_config = AacAudioRtpPacket.AudioSpecificConfig(
+                    reader
+                )
+            else:
+                asc_len = AacAudioRtpPacket.latm_value(reader)
+                marker = reader.bit_position
+                self.audio_specific_config = AacAudioRtpPacket.AudioSpecificConfig(
+                    reader
+                )
+                audio_specific_config_len = reader.bit_position - marker
+                if asc_len < audio_specific_config_len:
+                    raise ValueError('audio_specific_config_len > asc_len')
+                asc_len -= audio_specific_config_len
+                reader.skip(asc_len)
+            frame_length_type = reader.read(3)
+            if frame_length_type == 0:
+                latm_buffer_fullness = reader.read(8)
+            elif frame_length_type == 1:
+                frame_length = reader.read(9)
+            else:
+                raise ValueError(f'frame_length_type {frame_length_type} not supported')
+
+            self.other_data_present = reader.read(1)
+            if self.other_data_present:
+                if audio_mux_version == 1:
+                    self.other_data_len_bits = AacAudioRtpPacket.latm_value(reader)
+                else:
+                    self.other_data_len_bits = 0
+                    while True:
+                        self.other_data_len_bits *= 256
+                        other_data_len_esc = reader.read(1)
+                        self.other_data_len_bits += reader.read(8)
+                        if other_data_len_esc == 0:
+                            break
+            crc_check_present = reader.read(1)
+            if crc_check_present:
+                crc_checksum = reader.read(8)
+
+    @dataclass
+    class AudioMuxElement:
+        payload: bytes
+        stream_mux_config: AacAudioRtpPacket.StreamMuxConfig
+
+        def __init__(self, reader: BitReader, mux_config_present: int):
+            if mux_config_present == 0:
+                raise ValueError('muxConfigPresent == 0 not supported')
+
+            # AudioMuxElement - ISO/EIC 14496-3 Table 1.41
+            use_same_stream_mux = reader.read(1)
+            if use_same_stream_mux:
+                raise ValueError('useSameStreamMux == 1 not supported')
+            self.stream_mux_config = AacAudioRtpPacket.StreamMuxConfig(reader)
+
+            # We only support:
+            # allStreamsSameTimeFraming == 1
+            # audioMuxVersionA == 0,
+            # numProgram == 0
+            # numSubFrames == 0
+            # numLayer == 0
+
+            mux_slot_length_bytes = 0
+            while True:
+                tmp = reader.read(8)
+                mux_slot_length_bytes += tmp
+                if tmp != 255:
+                    break
+
+            self.payload = reader.read_bytes(mux_slot_length_bytes)
+
+            if self.stream_mux_config.other_data_present:
+                reader.skip(self.stream_mux_config.other_data_len_bits)
+
+            # ByteAlign
+            while reader.bit_position % 8:
+                reader.read(1)
+
+    def __init__(self, data: bytes) -> None:
+        # Parse the bit stream
+        reader = BitReader(data)
+        self.audio_mux_element = self.AudioMuxElement(reader, mux_config_present=1)
+
+    def to_adts(self):
+        # pylint: disable=line-too-long
+        sampling_frequency_index = (
+            self.audio_mux_element.stream_mux_config.audio_specific_config.sampling_frequency_index
+        )
+        channel_configuration = (
+            self.audio_mux_element.stream_mux_config.audio_specific_config.channel_configuration
+        )
+        frame_size = len(self.audio_mux_element.payload)
+        return (
+            bytes(
+                [
+                    0xFF,
+                    0xF1,  # 0xF9 (MPEG2)
+                    0x40
+                    | (sampling_frequency_index << 2)
+                    | (channel_configuration >> 2),
+                    ((channel_configuration & 0x3) << 6) | ((frame_size + 7) >> 11),
+                    ((frame_size + 7) >> 3) & 0xFF,
+                    (((frame_size + 7) << 5) & 0xFF) | 0x1F,
+                    0xFC,
+                ]
+            )
+            + self.audio_mux_element.payload
+        )

bumble/device.py

@@ -954,12 +954,16 @@ class Device(CompositeEventEmitter):
         config.load_from_file(filename)
         return cls(config=config)
 
+    @classmethod
+    def from_config_with_hci(cls, config, hci_source, hci_sink):
+        host = Host(controller_source=hci_source, controller_sink=hci_sink)
+        return cls(config=config, host=host)
+
     @classmethod
     def from_config_file_with_hci(cls, filename, hci_source, hci_sink):
         config = DeviceConfiguration()
         config.load_from_file(filename)
-        host = Host(controller_source=hci_source, controller_sink=hci_sink)
+        return cls.from_config_with_hci(config, hci_source, hci_sink)
-        return cls(config=config, host=host)
 
     def __init__(
         self,
@@ -2441,7 +2445,7 @@ class Device(CompositeEventEmitter):
 
             if result.status != HCI_COMMAND_STATUS_PENDING:
                 logger.warning(
-                    'HCI_Set_Connection_Encryption_Command failed: '
+                    'HCI_Remote_Name_Request_Command failed: '
                     f'{HCI_Constant.error_name(result.status)}'
                 )
                 raise HCI_StatusError(result)

bumble/hci.py

@@ -62,7 +62,7 @@ def map_null_terminated_utf8_string(utf8_bytes):
     try:
         terminator = utf8_bytes.find(0)
         if terminator < 0:
-            return utf8_bytes
+            terminator = len(utf8_bytes)
         return utf8_bytes[0:terminator].decode('utf8')
     except UnicodeDecodeError:
         return utf8_bytes
@@ -1795,6 +1795,16 @@ class Address:
     def to_bytes(self):
         return self.address_bytes
 
+    def to_string(self, with_type_qualifier=True):
+        '''
+        String representation of the address, MSB first, with an optional type
+        qualifier.
+        '''
+        result = ':'.join([f'{x:02X}' for x in reversed(self.address_bytes)])
+        if not with_type_qualifier or not self.is_public:
+            return result
+        return result + '/P'
+
     def __bytes__(self):
         return self.to_bytes()
 
@@ -1808,13 +1818,7 @@ class Address:
         )
 
     def __str__(self):
-        '''
+        return self.to_string()
-        String representation of the address, MSB first
-        '''
-        result = ':'.join([f'{x:02X}' for x in reversed(self.address_bytes)])
-        if not self.is_public:
-            return result
-        return result + '/P'
 
 
 # Predefined address values
@@ -5373,7 +5377,7 @@ class HCI_AclDataPacket:
     def __str__(self):
         return (
             f'{color("ACL", "blue")}: '
-            f'handle=0x{self.connection_handle:04x}'
+            f'handle=0x{self.connection_handle:04x}, '
             f'pb={self.pb_flag}, bc={self.bc_flag}, '
             f'data_total_length={self.data_total_length}, '
             f'data={self.data.hex()}'

bumble/host.py

@@ -62,6 +62,7 @@ from .hci import (
     HCI_Read_Local_Version_Information_Command,
     HCI_Reset_Command,
     HCI_Set_Event_Mask_Command,
+    map_null_terminated_utf8_string,
 )
 from .core import (
     BT_BR_EDR_TRANSPORT,
@@ -887,7 +888,12 @@ class Host(AbortableEventEmitter):
         if event.status != HCI_SUCCESS:
             self.emit('remote_name_failure', event.bd_addr, event.status)
         else:
-            self.emit('remote_name', event.bd_addr, event.remote_name)
+            utf8_name = event.remote_name
+            terminator = utf8_name.find(0)
+            if terminator >= 0:
+                utf8_name = utf8_name[0:terminator]
+
+            self.emit('remote_name', event.bd_addr, utf8_name)
 
     def on_hci_remote_host_supported_features_notification_event(self, event):
         self.emit(

docs/mkdocs/mkdocs.yml

@@ -44,6 +44,7 @@ nav:
     - Overview: apps_and_tools/index.md
     - Console: apps_and_tools/console.md
     - Bench: apps_and_tools/bench.md
+    - Speaker: apps_and_tools/speaker.md
     - HCI Bridge: apps_and_tools/hci_bridge.md
     - Golden Gate Bridge: apps_and_tools/gg_bridge.md
     - Show: apps_and_tools/show.md

docs/mkdocs/src/apps_and_tools/index.md

@@ -11,4 +11,5 @@ These include:
   * [HCI Bridge](hci_bridge.md) - a HCI transport bridge to connect two HCI transports and filter/snoop the HCI packets
   * [Golden Gate Bridge](gg_bridge.md) - a bridge between GATT and UDP to use with the Golden Gate "stack tool"
   * [Show](show.md) - Parse a file with HCI packets and print the details of each packet in a human readable form
+  * [Speaker](speaker.md) - Virtual Bluetooth speaker, with a command line and browser-based UI.
   * [Link Relay](link_relay.md) - WebSocket relay for virtual RemoteLink instances to communicate with each other.

docs/mkdocs/src/apps_and_tools/speaker.md

@@ -0,0 +1,86 @@
+SPEAKER APP
+===========
+
+![logo](../images/speaker_screenshot.png){ width=400 height=320 }
+
+The Speaker app is virtual Bluetooth speaker (A2DP sink).
+The app runs as a command-line executable, but also offers an optional simple
+web-browser-based user interface.
+
+# General Usage
+You can invoke the app either as `bumble-speaker` when installed as command
+from `pip`, or `python3 apps/speaker/speaker.py` when running from a source
+distribution.
+
+```
+Usage: speaker.py [OPTIONS] TRANSPORT
+
+  Run the speaker.
+
+Options:
+  --codec [sbc|aac]          [default: aac]
+  --discover                 Discover remote endpoints once connected
+  --output NAME              Send audio to this named output (may be used more
+                             than once for multiple outputs)
+  --ui-port HTTP_PORT        HTTP port for the UI server  [default: 7654]
+  --connect ADDRESS_OR_NAME  Address or name to connect to
+  --device-config FILENAME   Device configuration file
+  --help                     Show this message and exit.
+```
+
+# Connection
+By default, the virtual speaker will wait for another device (like a phone or
+computer) to connect to it (and possibly pair). Alternatively, the speaker can
+be told to initiate a connection to a remote device, using the `--connect` 
+option.
+
+# Outputs
+The speaker can have one or more outputs. By default, the only output is a text
+display on the console, as well as a browser-based user interface if connected.
+In addition, a file output can be used, in which case the received audio data is
+saved to a specified file.
+Finally, if the host computer on which your are running the application has `ffplay`
+as an available command line executable, the `@ffplay` output can be selected, in 
+which case the received audio will be played on the computer's builtin speakers via
+a pipe to `ffplay`. (see the [ffplay documentation](https://www.ffmpeg.org/ffplay.html)
+for details)
+
+# Web User Interface
+When the speaker app starts, it prints out on the console the local URL at which you
+may point a browser (Chrome recommended for full functionality). The console line
+specifying the local UI URL will look like:
+```
+UI HTTP server at http://127.0.0.1:7654
+```
+
+By default, the web UI will show the status of the connection, as well as a realtime
+graph of the received audio bandwidth.
+In order to also hear the received audio, you need to click the `Audio on` button
+(this is due to the fact that most browsers will require some user interface with the
+page before granting access to the audio output APIs).
+
+# Examples
+
+In the following examples, we use a single USB Bluetooth controllers `usb:0`. Other 
+transports can be used of course.
+
+!!! example "Start the speaker and wait for a connection"
+    ```
+    $ bumble-speaker usb:0
+    ```
+
+!!! example "Start the speaker and save the AAC audio to a file named `audio.aac`."
+    ```
+    $ bumble-speaker --output audio.aac usb:0
+    ```
+
+!!! example "Start the speaker and save the SBC audio to a file named `audio.sbc`."
+    ```
+    $ bumble-speaker --codec sbc --output audio.sbc usb:0
+    ```
+
+!!! example "Start the speaker and connect it to a phone at address `B8:7B:C5:05:57:ED`."
+    ```
+    $ bumble-speaker --connect B8:7B:C5:05:57:ED usb:0
+    ```
+

examples/run_classic_connect.py

@@ -23,7 +23,7 @@ from bumble.colors import color
 
 from bumble.device import Device
 from bumble.transport import open_transport_or_link
-from bumble.core import BT_BR_EDR_TRANSPORT, BT_L2CAP_PROTOCOL_ID
+from bumble.core import BT_BR_EDR_TRANSPORT, BT_L2CAP_PROTOCOL_ID, CommandTimeoutError
 from bumble.sdp import (
     Client as SDP_Client,
     SDP_PUBLIC_BROWSE_ROOT,
@@ -48,62 +48,70 @@ async def main():
         # Create a device
         device = Device.from_config_file_with_hci(sys.argv[1], hci_source, hci_sink)
         device.classic_enabled = True
+        device.le_enabled = False
         await device.power_on()
 
-    async def connect(target_address):
+        async def connect(target_address):
-        print(f'=== Connecting to {target_address}...')
+            print(f'=== Connecting to {target_address}...')
-        connection = await device.connect(target_address, transport=BT_BR_EDR_TRANSPORT)
+            try:
-        print(f'=== Connected to {connection.peer_address}!')
+                connection = await device.connect(
-
+                    target_address, transport=BT_BR_EDR_TRANSPORT
-        # Connect to the SDP Server
-        sdp_client = SDP_Client(device)
-        await sdp_client.connect(connection)
-
-        # List all services in the root browse group
-        service_record_handles = await sdp_client.search_services(
-            [SDP_PUBLIC_BROWSE_ROOT]
-        )
-        print(color('\n==================================', 'blue'))
-        print(color('SERVICES:', 'yellow'), service_record_handles)
-
-        # For each service in the root browse group, get all its attributes
-        for service_record_handle in service_record_handles:
-            attributes = await sdp_client.get_attributes(
-                service_record_handle, [SDP_ALL_ATTRIBUTES_RANGE]
-            )
-            print(color(f'SERVICE {service_record_handle:04X} attributes:', 'yellow'))
-            for attribute in attributes:
-                print('  ', attribute.to_string(with_colors=True))
-
-        # Search for services with an L2CAP service attribute
-        search_result = await sdp_client.search_attributes(
-            [BT_L2CAP_PROTOCOL_ID], [SDP_ALL_ATTRIBUTES_RANGE]
-        )
-        print(color('\n==================================', 'blue'))
-        print(color('SEARCH RESULTS:', 'yellow'))
-        for attribute_list in search_result:
-            print(color('SERVICE:', 'green'))
-            print(
-                '  '
-                + '\n  '.join(
-                    [
-                        attribute.to_string(with_colors=True)
-                        for attribute in attribute_list
-                    ]
                 )
+            except CommandTimeoutError:
+                print('!!! Connection timed out')
+                return
+            print(f'=== Connected to {connection.peer_address}!')
+
+            # Connect to the SDP Server
+            sdp_client = SDP_Client(device)
+            await sdp_client.connect(connection)
+
+            # List all services in the root browse group
+            service_record_handles = await sdp_client.search_services(
+                [SDP_PUBLIC_BROWSE_ROOT]
             )
+            print(color('\n==================================', 'blue'))
+            print(color('SERVICES:', 'yellow'), service_record_handles)
 
-        await sdp_client.disconnect()
+            # For each service in the root browse group, get all its attributes
-        await hci_source.wait_for_termination()
+            for service_record_handle in service_record_handles:
+                attributes = await sdp_client.get_attributes(
+                    service_record_handle, [SDP_ALL_ATTRIBUTES_RANGE]
+                )
+                print(
+                    color(f'SERVICE {service_record_handle:04X} attributes:', 'yellow')
+                )
+                for attribute in attributes:
+                    print('  ', attribute.to_string(with_colors=True))
 
-    # Connect to a peer
+            # Search for services with an L2CAP service attribute
-    target_addresses = sys.argv[3:]
+            search_result = await sdp_client.search_attributes(
-    await asyncio.wait(
+                [BT_L2CAP_PROTOCOL_ID], [SDP_ALL_ATTRIBUTES_RANGE]
-        [
+            )
-            asyncio.create_task(connect(target_address))
+            print(color('\n==================================', 'blue'))
-            for target_address in target_addresses
+            print(color('SEARCH RESULTS:', 'yellow'))
-        ]
+            for attribute_list in search_result:
-    )
+                print(color('SERVICE:', 'green'))
+                print(
+                    '  '
+                    + '\n  '.join(
+                        [
+                            attribute.to_string(with_colors=True)
+                            for attribute in attribute_list
+                        ]
+                    )
+                )
+
+            await sdp_client.disconnect()
+
+        # Connect to a peer
+        target_addresses = sys.argv[3:]
+        await asyncio.wait(
+            [
+                asyncio.create_task(connect(target_address))
+                for target_address in target_addresses
+            ]
+        )
 
 
 # -----------------------------------------------------------------------------

examples/speaker.json

@@ -0,0 +1,5 @@
+{
+    "name": "Bumble Speaker",
+    "class_of_device": 2360324,
+    "keystore": "JsonKeyStore"
+}

setup.cfg

@@ -30,22 +30,23 @@ package_dir =
     bumble.apps = apps
 include-package-data = True
 install_requires =
+    aiohttp >= 3.8.4; platform_system!='Emscripten'
     appdirs >= 1.4
+    bt-test-interfaces >= 0.0.2
     click >= 7.1.2; platform_system!='Emscripten'
     cryptography == 35; platform_system!='Emscripten'
     grpcio == 1.51.1; platform_system!='Emscripten'
+    humanize >= 4.6.0
     libusb1 >= 2.0.1; platform_system!='Emscripten'
     libusb-package == 1.0.26.1; platform_system!='Emscripten'
     prompt_toolkit >= 3.0.16; platform_system!='Emscripten'
+    prettytable >= 3.6.0
     protobuf >= 3.12.4
     pyee >= 8.2.2
     pyserial-asyncio >= 0.5; platform_system!='Emscripten'
     pyserial >= 3.5; platform_system!='Emscripten'
     pyusb >= 1.2; platform_system!='Emscripten'
     websockets >= 8.1; platform_system!='Emscripten'
-    prettytable >= 3.6.0
-    humanize >= 4.6.0
-    bt-test-interfaces >= 0.0.2
 
 [options.entry_points]
 console_scripts =
@@ -61,6 +62,7 @@ console_scripts =
     bumble-usb-probe = bumble.apps.usb_probe:main
     bumble-link-relay = bumble.apps.link_relay.link_relay:main
     bumble-bench = bumble.apps.bench:main
+    bumble-speaker = bumble.apps.speaker.speaker:main
     bumble-pandora-server = bumble.apps.pandora_server:main
 
 [options.package_data]

speaker.html

@@ -0,0 +1,28 @@
+<!DOCTYPE html>
+<html>
+  <head>
+    <title>Audio WAV Player</title>
+  </head>
+  <body>
+    <h1>Audio WAV Player</h1>
+    <audio id="audioPlayer" controls>
+      <source src="" type="audio/wav">
+    </audio>
+
+    <script>
+      const audioPlayer = document.getElementById('audioPlayer');
+      const ws = new WebSocket('ws://localhost:8080');
+
+      let mediaSource = new MediaSource();
+      audioPlayer.src = URL.createObjectURL(mediaSource);
+
+      mediaSource.addEventListener('sourceopen', function(event) {
+        const sourceBuffer = mediaSource.addSourceBuffer('audio/wav');
+
+        ws.onmessage = function(event) {
+          sourceBuffer.appendBuffer(event.data);
+        };
+      });
+    </script>
+  </body>
+</html>

tests/codecs_test.py

@@ -0,0 +1,67 @@
+# Copyright 2021-2023 Google LLC
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#      https://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+# -----------------------------------------------------------------------------
+# Imports
+# -----------------------------------------------------------------------------
+import pytest
+from bumble.codecs import AacAudioRtpPacket, BitReader
+
+
+# -----------------------------------------------------------------------------
+def test_reader():
+    reader = BitReader(b'')
+    with pytest.raises(ValueError):
+        reader.read(1)
+
+    reader = BitReader(b'hello')
+    with pytest.raises(ValueError):
+        reader.read(40)
+
+    reader = BitReader(bytes([0xFF]))
+    assert reader.read(1) == 1
+    with pytest.raises(ValueError):
+        reader.read(10)
+
+    reader = BitReader(bytes([0x78]))
+    value = 0
+    for _ in range(8):
+        value = (value << 1) | reader.read(1)
+    assert value == 0x78
+
+    data = bytes([x & 0xFF for x in range(66 * 100)])
+    reader = BitReader(data)
+    value = 0
+    for _ in range(100):
+        for bits in range(1, 33):
+            value = value << bits | reader.read(bits)
+    assert value == int.from_bytes(data, byteorder='big')
+
+
+def test_aac_rtp():
+    # pylint: disable=line-too-long
+    packet_data = bytes.fromhex(
+        '47fc0000b090800300202066000198000de120000000000000000000000000000000000000000000001c'
+    )
+    packet = AacAudioRtpPacket(packet_data)
+    adts = packet.to_adts()
+    assert adts == bytes.fromhex(
+        'fff1508004fffc2066000198000de120000000000000000000000000000000000000000000001c'
+    )
+
+
+# -----------------------------------------------------------------------------
+if __name__ == '__main__':
+    test_reader()
+    test_aac_rtp()