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Add binary protocol v2 & v3 to websocket

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This commit is contained in:
Terrence
2025-04-28 16:29:33 +08:00
parent 1f03d770cc
commit 5f8389c135
5 changed files with 86 additions and 13 deletions
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2
main/ota.cc
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@ -154,6 +154,8 @@ bool Ota::CheckVersion() {
cJSON_ArrayForEach(item, websocket) {
if (item->type == cJSON_String) {
settings.SetString(item->string, item->valuestring);
} else if (item->type == cJSON_Number) {
settings.SetInt(item->string, item->valueint);
}
}
has_websocket_config_ = true;

5
main/protocols/mqtt_protocol.cc
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@ -207,6 +207,11 @@ bool MqttProtocol::OpenAudioChannel() {
}
udp_ = Board::GetInstance().CreateUdp();
udp_->OnMessage([this](const std::string& data) {
/*
* UDP Encrypted OPUS Packet Format:
* |type 1u|flags 1u|payload_len 2u|ssrc 4u|timestamp 4u|sequence 4u|
* |payload payload_len|
*/
if (data.size() < sizeof(aes_nonce_)) {
ESP_LOGE(TAG, "Invalid audio packet size: %zu", data.size());
return;

15
main/protocols/protocol.h
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@ -5,6 +5,21 @@
#include <string>
#include <functional>
#include <chrono>
#include <vector>
struct AudioStreamPacket {
uint32_t timestamp;
std::vector<uint8_t> payload;
};
struct BinaryProtocol2 {
uint16_t version;
uint16_t type; // Message type (0: OPUS, 1: JSON)
uint32_t reserved; // Reserved for future use
uint32_t timestamp; // Timestamp in milliseconds (used for server-side AEC)
uint32_t payload_size; // Payload size in bytes
uint8_t payload[]; // Payload data
} __attribute__((packed));
struct BinaryProtocol3 {
uint8_t type;

64
main/protocols/websocket_protocol.cc
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@ -33,9 +33,37 @@ void WebsocketProtocol::SendAudio(const std::vector<uint8_t>& data) {
return;
}
if (version_ == 2) {
std::string packet;
packet.resize(sizeof(BinaryProtocol2) + data.size());
auto bp2 = (BinaryProtocol2*)packet.data();
bp2->version = htons(version_);
bp2->type = 0;
bp2->reserved = 0;
bp2->timestamp = htonl(0);
bp2->payload_size = htonl(data.size());
memcpy(bp2->payload, data.data(), data.size());
busy_sending_audio_ = true;
websocket_->Send(packet.data(), packet.size(), true);
busy_sending_audio_ = false;
} else if (version_ == 3) {
std::string packet;
packet.resize(sizeof(BinaryProtocol3) + data.size());
auto bp3 = (BinaryProtocol3*)packet.data();
bp3->type = 0;
bp3->reserved = 0;
bp3->payload_size = htons(data.size());
memcpy(bp3->payload, data.data(), data.size());
busy_sending_audio_ = true;
websocket_->Send(packet.data(), packet.size(), true);
busy_sending_audio_ = false;
} else {
busy_sending_audio_ = true;
websocket_->Send(data.data(), data.size(), true);
busy_sending_audio_ = false;
}
}
bool WebsocketProtocol::SendText(const std::string& text) {
@ -71,26 +99,48 @@ bool WebsocketProtocol::OpenAudioChannel() {
Settings settings("websocket", false);
std::string url = settings.GetString("url");
std::string token = settings.GetString("token");
int version = settings.GetInt("version");
if (version != 0) {
version_ = version;
}
busy_sending_audio_ = false;
error_occurred_ = false;
// If token not starts with "Bearer " or "bearer ", add it
if (token.empty() || (token.find("Bearer ") != 0 && token.find("bearer ") != 0)) {
websocket_ = Board::GetInstance().CreateWebSocket();
if (!token.empty()) {
// If token not has a space, add "Bearer " prefix
if (token.find(" ") == std::string::npos) {
token = "Bearer " + token;
}
websocket_ = Board::GetInstance().CreateWebSocket();
websocket_->SetHeader("Authorization", token.c_str());
websocket_->SetHeader("Protocol-Version", "1");
}
websocket_->SetHeader("Protocol-Version", std::to_string(version_).c_str());
websocket_->SetHeader("Device-Id", SystemInfo::GetMacAddress().c_str());
websocket_->SetHeader("Client-Id", Board::GetInstance().GetUuid().c_str());
websocket_->OnData([this](const char* data, size_t len, bool binary) {
if (binary) {
if (on_incoming_audio_ != nullptr) {
if (version_ == 2) {
BinaryProtocol2* bp2 = (BinaryProtocol2*)data;
bp2->version = ntohs(bp2->version);
bp2->type = ntohs(bp2->type);
bp2->timestamp = ntohl(bp2->timestamp);
bp2->payload_size = ntohl(bp2->payload_size);
auto payload = (uint8_t*)bp2->payload;
on_incoming_audio_(std::vector<uint8_t>(payload, payload + bp2->payload_size));
} else if (version_ == 3) {
BinaryProtocol3* bp3 = (BinaryProtocol3*)data;
bp3->type = bp3->type;
bp3->payload_size = ntohs(bp3->payload_size);
auto payload = (uint8_t*)bp3->payload;
on_incoming_audio_(std::vector<uint8_t>(payload, payload + bp3->payload_size));
} else {
on_incoming_audio_(std::vector<uint8_t>((uint8_t*)data, (uint8_t*)data + len));
}
}
} else {
// Parse JSON data
auto root = cJSON_Parse(data);
@ -118,7 +168,7 @@ bool WebsocketProtocol::OpenAudioChannel() {
}
});
ESP_LOGI(TAG, "Connecting to websocket server: %s with token: %s", url.c_str(), token.c_str());
ESP_LOGI(TAG, "Connecting to websocket server: %s with version: %d", url.c_str(), version_);
if (!websocket_->Connect(url.c_str())) {
ESP_LOGE(TAG, "Failed to connect to websocket server");
SetError(Lang::Strings::SERVER_NOT_FOUND);
@ -129,7 +179,7 @@ bool WebsocketProtocol::OpenAudioChannel() {
// keys: message type, version, audio_params (format, sample_rate, channels)
std::string message = "{";
message += "\"type\":\"hello\",";
message += "\"version\": 1,";
message += "\"version\": " + std::to_string(version_) + ",";
message += "\"transport\":\"websocket\",";
message += "\"audio_params\":{";
message += "\"format\":\"opus\", \"sample_rate\":16000, \"channels\":1, \"frame_duration\":" + std::to_string(OPUS_FRAME_DURATION_MS);

1
main/protocols/websocket_protocol.h
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@ -24,6 +24,7 @@ public:
private:
EventGroupHandle_t event_group_handle_;
WebSocket* websocket_ = nullptr;
int version_ = 1;
void ParseServerHello(const cJSON* root);
bool SendText(const std::string& text) override;