tirtc-devtools-cli 0.3.0 → 0.3.2

package/README.md

@@ -65,8 +65,9 @@ prepared asset 会按完整音视频轨循环，任一轨到达源文件末尾
 `device start` 的启动成功条件是 device listener 已就绪；客户端是否已经扫码连接不属于
 device 启动失败条件。没有 client 时命令会继续常驻，直到用户结束进程或显式
 `--duration-ms` 到期，summary 中 `media_send` 会保持未开始或标记为跳过。有 client
-连接时 driver 会为该连接送流；client 断开后，device 继续等待下一次连接，并为新连接
-重新挂载音视频输入和从源资产开头送出新的音视频包。同一时刻最多接受一个 client。
+连接时 driver 会把该连接挂到共享音视频输入上；client 断开后，device 会清理该连接，
+并继续接受新的连接。若旧连接的断开回调尚未到达，新连接也会立即挂载并收到同一组
+音视频输入的数据。
 
 native role 失败时，`reason_code` 与 `failed_stage` 仍是主失败事实；`log_upload`
 只说明失败现场日志是否已上传。上传成功时，CLI JSON 与 `summary.json` 会包含

package/dist/cli/src/index.js

@@ -125,6 +125,10 @@ device.command('start')
     .option('--endpoint <url>', 'TiRTC endpoint；不传时读取 TIRTC_ENDPOINT')
     .option('--source <path>', 'prepared asset root、manifest_path 或 encoded track；MP4 先运行 assets prepare')
     .option('--video-codec <codec>', 'h264|h265|mjpeg', 'h264')
+    .option('--audio-codec <codec>', 'g711a|aac', 'g711a')
+    .option('--audio-sample-rate <hz>', '8000|16000', '8000')
+    .option('--audio-channels <count>', '1|2', '1')
+    .option('--exit-after-first-session', '首个 client 会话完成后主动正常退出并写出 summary')
     .option('--duration-ms <ms>', '可选自动结束时长；默认持续运行直到用户结束进程')
     .option('--connect-timeout-ms <ms>', 'service ready 最大等待')
     .option('--first-packet-timeout-ms <ms>', '首包最大等待')

package/dist/cli/src/role_driver.d.ts

@@ -8,6 +8,10 @@ type DeviceCommandOptions = {
     endpoint?: string;
     source?: string;
     videoCodec?: string;
+    audioCodec?: string;
+    audioSampleRate?: string;
+    audioChannels?: string;
+    exitAfterFirstSession?: boolean;
     durationMs?: string;
     connectTimeoutMs?: string;
     firstPacketTimeoutMs?: string;

package/dist/cli/src/role_driver.js

@@ -39,6 +39,9 @@ function rolePreflightError(reasonCode, detail) {
 function roleUsageError(message) {
     return new RoleCommandError('invalid_request', 'config', usageExitCode, message);
 }
+function roleUsageReasonError(reasonCode, message) {
+    return new RoleCommandError(reasonCode, 'config', usageExitCode, reasonCode + ': ' + message);
+}
 function pathExists(filePath) {
     return fs_1.default.existsSync(filePath);
 }
@@ -270,6 +273,27 @@ function codecOrDefault(raw) {
     }
     return codec;
 }
+function audioCodecOrDefault(raw) {
+    const codec = raw?.trim() || 'g711a';
+    if (codec !== 'g711a' && codec !== 'aac') {
+        throw roleUsageReasonError('audio_codec_unsupported', 'audio-codec must be g711a or aac');
+    }
+    return codec;
+}
+function audioSampleRateOrDefault(raw) {
+    const sampleRate = raw === undefined ? 8000 : Number(raw);
+    if (sampleRate !== 8000 && sampleRate !== 16000) {
+        throw roleUsageReasonError('audio_format_unsupported', 'audio-sample-rate must be 8000 or 16000');
+    }
+    return sampleRate;
+}
+function audioChannelsOrDefault(raw) {
+    const channels = raw === undefined ? 1 : Number(raw);
+    if (channels !== 1 && channels !== 2) {
+        throw roleUsageReasonError('audio_format_unsupported', 'audio-channels must be 1 or 2');
+    }
+    return channels;
+}
 function consumerOrDefault(raw) {
     const consumer = raw?.trim() || 'frame_dump';
     if (consumer !== 'frame_dump' && consumer !== 'packet_dump') {
@@ -302,6 +326,9 @@ function printEnvelope(options, code, message, data) {
 }
 function buildDeviceRequest(roots, artifactRoot, options) {
     const codec = codecOrDefault(options.videoCodec);
+    const audioCodec = audioCodecOrDefault(options.audioCodec);
+    const audioSampleRateHz = audioSampleRateOrDefault(options.audioSampleRate);
+    const audioChannels = audioChannelsOrDefault(options.audioChannels);
     const executionId = 'cli-device-' + codec + '-' + executionSuffix();
     const caseId = 'devtools-cli-device.' + codec;
     const deviceIdentity = resolveDeviceIdentity(options);
@@ -328,9 +355,15 @@ function buildDeviceRequest(roots, artifactRoot, options) {
         media: {
             source: { kind: 'encoded_asset', path: options.source ?? resolveAssetRoot(roots) },
             video: { codec },
+            audio: {
+                codec: audioCodec,
+                sample_rate_hz: audioSampleRateHz,
+                channels: audioChannels,
+            },
         },
         output: { consumer: 'frame_dump', video: { frame_limit: defaultFrameLimit } },
         run: {
+            exit_after_first_session: options.exitAfterFirstSession === true,
             duration_ms: parseOptionalPositiveInt(options.durationMs, '--duration-ms'),
             connect_timeout_ms: parsePositiveInt(options.connectTimeoutMs, defaultConnectTimeoutMs, '--connect-timeout-ms'),
             first_packet_timeout_ms: parsePositiveInt(options.firstPacketTimeoutMs, defaultFirstPacketTimeoutMs, '--first-packet-timeout-ms'),
@@ -372,11 +405,15 @@ function buildClientRequest(roots, artifactRoot, options) {
     const executionId = 'cli-client-' + executionSuffix();
     const identityOverride = Boolean(bootstrap && (options.targetDeviceId || options.token));
     const videoCodec = codecOrDefault(bootstrap?.video_codec);
+    const audioCodec = audioCodecOrDefault(bootstrap?.audio_codec);
+    const audioSampleRateHz = audioSampleRateOrDefault(bootstrap?.audio_sample_rate_hz ?? bootstrap?.sample_rate_hz);
+    const audioChannels = audioChannelsOrDefault(bootstrap?.audio_channels ?? bootstrap?.channels);
     return {
         schema_version: 1,
         execution_id: executionId,
         case_id: 'devtools-cli-client',
         role: 'client',
+        require_audio: bootstrap?.require_audio ?? true,
         endpoint,
         identity: {
             device_id: targetDeviceId,
@@ -392,6 +429,11 @@ function buildClientRequest(roots, artifactRoot, options) {
         media: {
             source: { kind: 'encoded_asset', path: resolveAssetRoot(roots) },
             video: { codec: videoCodec },
+            audio: {
+                codec: audioCodec,
+                sample_rate_hz: audioSampleRateHz,
+                channels: audioChannels,
+            },
         },
         output: {
             consumer: consumerOrDefault(options.consumer),

package/dist/cli/src/role_live_log.js

@@ -83,7 +83,7 @@ function startRoleLiveLog(options) {
     let processedLineCount = 0;
     let stopped = false;
     const log = (message) => {
-        emit('[device] ' + message);
+        emit(new Date().toISOString() + ' [device] ' + message);
     };
     const media = nestedRecord(request, 'media');
     const codec = payloadString(nestedRecord(media, 'video'), 'codec') ??
@@ -145,6 +145,15 @@ function startRoleLiveLog(options) {
                 log('client connected session=' + (session ?? 'unknown'));
                 break;
             }
+            case 'connection.state.changed': {
+                const parts = ['connection state changed'];
+                appendField(parts, 'session', sessionIndex(payload));
+                appendField(parts, 'state', payloadString(payload, 'state'));
+                appendField(parts, 'state_code', payloadNumber(payload, 'state_code'));
+                appendField(parts, 'error', payloadNumber(payload, 'error'));
+                log(parts.join(' '));
+                break;
+            }
             case 'connection.session.start': {
                 const session = sessionIndex(payload);
                 if (session !== undefined) {
@@ -157,7 +166,7 @@ function startRoleLiveLog(options) {
                 const parts = ['audio input started'];
                 appendField(parts, 'session', sessionIndex(payload));
                 appendField(parts, 'stream_id', payloadNumber(payload, 'stream_id'));
-                appendField(parts, 'codec', payloadString(payload, 'codec'));
+                appendField(parts, 'codec', payloadString(payload, 'audio_codec') ?? payloadString(payload, 'codec'));
                 appendField(parts, 'sample_rate_hz', payloadNumber(payload, 'sample_rate_hz'));
                 log(parts.join(' '));
                 break;
@@ -219,6 +228,9 @@ function startRoleLiveLog(options) {
                 appendField(parts, 'session', session);
                 appendField(parts, 'first_audio', payloadBoolean(payload, 'sent_first_audio'));
                 appendField(parts, 'first_video', payloadBoolean(payload, 'sent_first_video'));
+                appendField(parts, 'exit_reason', payloadString(payload, 'exit_reason'));
+                appendField(parts, 'submit_error_track', payloadString(payload, 'submit_error_track'));
+                appendField(parts, 'submit_error_status', payloadNumber(payload, 'submit_error_status'));
                 appendField(parts, 'elapsed', elapsed === undefined ? undefined : formatDurationMs(elapsed));
                 log(parts.join(' '));
                 break;

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "tirtc-devtools-cli",
-  "version": "0.3.0",
+  "version": "0.3.2",
   "private": false,
   "main": "dist/cli/src/index.js",
   "types": "dist/cli/src/index.d.ts",

package/vendor/runtime/linux-x64/include/tirtc/audio.h

@@ -42,7 +42,11 @@ typedef enum TirtcAudioOutputState {
 } TirtcAudioOutputState;
 
 typedef struct TirtcAudioInputOptions {
-  TirtcAudioSampleRate sample_rate;
+  TirtcMediaCodec codec;
+
+  uint32_t sample_rate_hz;
+
+  uint32_t channels;
 
   TirtcAudioAecMode aec_mode;
 
@@ -61,9 +65,27 @@ typedef struct TirtcAudioInputOptions {
 typedef struct TirtcAudioEncodedInputOptions {
   TirtcMediaCodec codec;
 
-  TirtcAudioSampleRate sample_rate;
+  uint32_t sample_rate_hz;
+
+  uint32_t channels;
 } TirtcAudioEncodedInputOptions;
 
+typedef struct TirtcAudioEncodedInputFrame {
+  TirtcMediaCodec codec;
+
+  uint32_t sample_rate_hz;
+
+  uint32_t channels;
+
+  uint32_t samples_per_channel;
+
+  int64_t pts_us;
+
+  const void* data;
+
+  size_t data_bytes;
+} TirtcAudioEncodedInputFrame;
+
 typedef struct TirtcAudioOutputOptions {
   uint32_t volume_percent;
 
@@ -91,9 +113,30 @@ typedef struct TirtcPendingDurationMetrics {
   uint64_t decoded_duration_ms;
 } TirtcPendingDurationMetrics;
 
+typedef struct TirtcOutputLatencyStageMetrics {
+  uint64_t latest_ms;
+  uint64_t average_ms;
+  uint64_t peak_ms;
+  uint64_t sample_count;
+  uint64_t unavailable_count;
+} TirtcOutputLatencyStageMetrics;
+
+typedef struct TirtcOutputLocalLatencyMetrics {
+  uint32_t window_duration_ms;
+  TirtcOutputLatencyStageMetrics total;
+  TirtcOutputLatencyStageMetrics buffer;
+  TirtcOutputLatencyStageMetrics decode_or_ready;
+  TirtcOutputLatencyStageMetrics output;
+} TirtcOutputLocalLatencyMetrics;
+
 typedef struct TirtcAudioOutputMetricsSnapshot {
   TirtcOutputStutterMetrics stutter;
   TirtcPendingDurationMetrics pending;
+  double input_bitrate_kbps;
+  double input_packet_rate;
+  double render_callback_rate;
+  uint32_t rate_window_duration_ms;
+  TirtcOutputLocalLatencyMetrics local_latency;
 } TirtcAudioOutputMetricsSnapshot;
 
 /**
@@ -105,6 +148,10 @@ typedef struct TirtcAudioOutputMetricsSnapshot {
  */
 typedef struct TirtcAudioOutputDebugSnapshot {
   TirtcMediaCodec codec;
+
+  uint32_t sample_rate_hz;
+
+  uint32_t channels;
 } TirtcAudioOutputDebugSnapshot;
 
 /**
@@ -191,7 +238,7 @@ TirtcError tirtc_audio_encoded_input_attach(TirtcAudioEncodedInput* input, Tirtc
 TirtcError tirtc_audio_encoded_input_detach(TirtcAudioEncodedInput* input, TirtcConn* connection);
 
 TirtcError tirtc_audio_encoded_input_submit_frame(TirtcAudioEncodedInput* input,
-                                                  const TirtcAudioEncodedFrame* frame);
+                                                  const TirtcAudioEncodedInputFrame* frame);
 
 void tirtc_audio_encoded_input_destroy(TirtcAudioEncodedInput* input);
 

package/vendor/runtime/linux-x64/include/tirtc/audio_codec.h

@@ -16,6 +16,12 @@ TirtcError tirtc_audio_g711a_encode(const TirtcAudioPcmFrame* pcm_frame, void* o
 TirtcError tirtc_audio_g711a_decode(const TirtcAudioEncodedFrame* in_frame, void* out_buffer,
                                     size_t out_buffer_capacity, TirtcAudioPcmFrame* out_frame);
 
+TirtcError tirtc_audio_aac_encode(const TirtcAudioPcmFrame* pcm_frame, void* out_buffer,
+                                  size_t out_buffer_capacity, TirtcAudioEncodedFrame* out_frame);
+
+TirtcError tirtc_audio_aac_decode(const TirtcAudioEncodedFrame* in_frame, void* out_buffer,
+                                  size_t out_buffer_capacity, TirtcAudioPcmFrame* out_frame);
+
 #ifdef __cplusplus
 }
 #endif

package/vendor/runtime/linux-x64/include/tirtc/audio_format.h

@@ -0,0 +1,20 @@
+#ifndef TIRTC_MEDIA_CONTRACT_AUDIO_FORMAT_H_
+#define TIRTC_MEDIA_CONTRACT_AUDIO_FORMAT_H_
+
+#include <stdint.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef struct TirtcAudioFormat {
+  uint32_t sample_rate_hz;
+  uint32_t channels;
+  uint32_t bits_per_sample;
+} TirtcAudioFormat;
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // TIRTC_MEDIA_CONTRACT_AUDIO_FORMAT_H_

package/vendor/runtime/linux-x64/include/tirtc/audio_frame.h

@@ -4,7 +4,7 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "tirtc/audio_sample_rate.h"
+#include "tirtc/audio_format.h"
 #include "tirtc/media_codec.h"
 
 #ifdef __cplusplus
@@ -12,17 +12,16 @@ extern "C" {
 #endif
 
 typedef struct TirtcAudioPcmFrame {
-  TirtcAudioSampleRate sample_rate;
+  TirtcAudioFormat format;
   uint32_t samples_per_channel;
   int64_t pts_us;
-  /* Mutable PCM payload for capture, processing, and render-fill paths. */
   void* data;
   size_t data_bytes;
 } TirtcAudioPcmFrame;
 
 typedef struct TirtcAudioEncodedFrame {
   TirtcMediaCodec codec;
-  TirtcAudioSampleRate sample_rate;
+  TirtcAudioFormat format;
   uint32_t samples_per_channel;
   int64_t pts_us;
   const void* data;

package/vendor/runtime/linux-x64/include/tirtc/audio_io.h

@@ -14,8 +14,7 @@ typedef struct TirtcAudioAin TirtcAudioAin;
 typedef struct TirtcAudioAout TirtcAudioAout;
 
 typedef struct TirtcAudioIoConfig {
-  TirtcAudioSampleRate sample_rate;
-  /* AIN uses fixed 10 ms mono blocks; AOUT may use larger callback-sized multiples. */
+  TirtcAudioFormat format;
   uint32_t samples_per_buffer;
 } TirtcAudioIoConfig;
 
@@ -29,7 +28,7 @@ typedef void (*TirtcAudioAinOnFrameFn)(TirtcAudioAin* ain, const TirtcAudioPcmFr
                                        void* user_data);
 
 /*
- * AOUT pre-fills sample_rate, samples_per_channel, data, and data_bytes before
+ * AOUT pre-fills format, samples_per_channel, data, and data_bytes before
  * calling render. The callback contract expects a non-negative rendered PCM
  * byte count. Backends defensively clamp negative returns to zero and zero-fill
  * any remaining buffer space.

package/vendor/runtime/linux-x64/include/tirtc/audio_processing.h

@@ -2,6 +2,7 @@
 #define TIRTC_AUDIO_PROCESSING_H_
 
 #include "tirtc/audio_frame.h"
+#include "tirtc/audio_sample_rate.h"
 #include "tirtc/error.h"
 
 #ifdef __cplusplus

package/vendor/runtime/linux-x64/include/tirtc/av.h

@@ -46,6 +46,12 @@ typedef struct TirtcVideoOutputMetricsSnapshot {
   TirtcVideoOutputStartupMetrics startup;
   TirtcOutputStutterMetrics stutter;
   TirtcPendingDurationMetrics pending;
+  double input_bitrate_kbps;
+  double input_fps;
+  double decoded_fps;
+  double render_fps;
+  uint32_t rate_window_duration_ms;
+  TirtcOutputLocalLatencyMetrics local_latency;
 } TirtcVideoOutputMetricsSnapshot;
 
 /**
@@ -53,8 +59,7 @@ typedef struct TirtcVideoOutputMetricsSnapshot {
  *
  * The snapshot is a caller-owned value copy. `resolved_decoder_backend` is the runtime route that
  * has actually locked, so AUTO fallback reports the concrete backend family instead of the request.
- * Zero values mean no current fact or only cached zero-state is available. Additive changes require
- * a new V2 struct/function instead of extending this layout.
+ * Zero values mean no current fact or only cached zero-state is available.
  */
 typedef struct TirtcVideoOutputDebugSnapshot {
   TirtcMediaCodec codec;

package/vendor/runtime/linux-x64/include/tirtc/media_codec.h

@@ -8,6 +8,7 @@ extern "C" {
 typedef enum TirtcMediaCodec {
   TIRTC_MEDIA_CODEC_NONE = 0,
   TIRTC_MEDIA_CODEC_AUDIO_G711A = 1,
+  TIRTC_MEDIA_CODEC_AUDIO_AAC = 2,
   /* Reserved shared identifiers; concrete video implementations live outside media-contract. */
   TIRTC_MEDIA_CODEC_VIDEO_H264 = 65,
   TIRTC_MEDIA_CODEC_VIDEO_H265 = 66,

package/vendor/runtime/linux-x64/include/tirtc/media_downlink.h

@@ -66,9 +66,30 @@ typedef struct TirtcMediaDownlinkPendingMetricsSnapshot {
   uint64_t decoded_pending_duration_ms;
 } TirtcMediaDownlinkPendingMetricsSnapshot;
 
+typedef struct TirtcMediaDownlinkLatencyStageMetricsSnapshot {
+  uint64_t latest_ms;
+  uint64_t average_ms;
+  uint64_t peak_ms;
+  uint64_t sample_count;
+  uint64_t unavailable_count;
+} TirtcMediaDownlinkLatencyStageMetricsSnapshot;
+
+typedef struct TirtcMediaDownlinkLocalLatencyMetricsSnapshot {
+  uint32_t window_duration_ms;
+  TirtcMediaDownlinkLatencyStageMetricsSnapshot total;
+  TirtcMediaDownlinkLatencyStageMetricsSnapshot buffer;
+  TirtcMediaDownlinkLatencyStageMetricsSnapshot decode_or_ready;
+  TirtcMediaDownlinkLatencyStageMetricsSnapshot output;
+} TirtcMediaDownlinkLocalLatencyMetricsSnapshot;
+
 typedef struct TirtcMediaDownlinkAudioMetricsSnapshot {
   TirtcMediaDownlinkStutterMetricsSnapshot stutter;
   TirtcMediaDownlinkPendingMetricsSnapshot pending;
+  double input_bitrate_kbps;
+  double input_packet_rate;
+  double render_callback_rate;
+  uint32_t rate_window_duration_ms;
+  TirtcMediaDownlinkLocalLatencyMetricsSnapshot local_latency;
 } TirtcMediaDownlinkAudioMetricsSnapshot;
 
 typedef struct TirtcMediaDownlinkVideoStartupMetricsSnapshot {
@@ -84,6 +105,12 @@ typedef struct TirtcMediaDownlinkVideoMetricsSnapshot {
   TirtcMediaDownlinkVideoStartupMetricsSnapshot startup;
   TirtcMediaDownlinkStutterMetricsSnapshot stutter;
   TirtcMediaDownlinkPendingMetricsSnapshot pending;
+  double input_bitrate_kbps;
+  double input_fps;
+  double decoded_fps;
+  double render_fps;
+  uint32_t rate_window_duration_ms;
+  TirtcMediaDownlinkLocalLatencyMetricsSnapshot local_latency;
 } TirtcMediaDownlinkVideoMetricsSnapshot;
 
 /**
@@ -94,6 +121,7 @@ typedef struct TirtcMediaDownlinkVideoMetricsSnapshot {
  */
 typedef struct TirtcMediaDownlinkAudioDebugSnapshot {
   TirtcMediaCodec codec;
+  TirtcAudioFormat format;
 } TirtcMediaDownlinkAudioDebugSnapshot;
 
 /**

package/vendor/runtime/linux-x64/include/tirtc/media_uplink.h

@@ -29,7 +29,8 @@ typedef enum TirtcMediaUplinkState {
 } TirtcMediaUplinkState;
 
 typedef struct TirtcMediaUplinkAudioOptions {
-  TirtcAudioSampleRate sample_rate;
+  TirtcMediaCodec codec;
+  TirtcAudioFormat format;
   TirtcAudioAecMode aec_mode;
   TirtcAudioAgcLevel agc_level;
   TirtcAudioAnsLevel ans_level;

package/vendor/runtime/linux-x64/include/tirtc/transport.h

@@ -139,7 +139,7 @@ typedef struct TirtcTransportConnServiceOptions {
    * This is a service-level capacity contract. It applies to the listener as a
    * whole, not to any single connection and not to any individual stream.
    *
-   * The current public contract requires a non-zero value.
+   * Set to 0 to leave the listener capacity to the active transport backend.
    */
   uint32_t max_connections;
 } TirtcTransportConnServiceOptions;

package/vendor/runtime/linux-x64/include/tirtc/trp.h

@@ -30,6 +30,7 @@ typedef struct TirtcConnServiceStartOptions {
 
   const char* device_secret_key;
 
+  /** Set to 0 to leave accepted-connection capacity to the active backend. */
   uint32_t max_connections;
 } TirtcConnServiceStartOptions;
 

package/vendor/runtime/linux-x64/manifest.txt

@@ -1,28 +1,29 @@
 platform=linux-x64
 profile=full
-staged_at_utc=2026-05-07T08:43:43Z
+staged_at_utc=2026-05-13T16:35:31Z
 source_sdk=/Users/allenfeng/Development/Repositories/tirtc-nexus/tirtc-matrix/.build/sdk/linux-x64
 
-59ba10e0b7cc679d625d3e8c536e3e234b3173091b9859956252fe3dd631b774  include/tirtc/audio.h
-6d972ccfe150a3b4f2d7f18fa92b3ade9210c1c8bb754d061ac6b7997b59e2cb  include/tirtc/audio_codec.h
-7bacbdb2d8bb10d6444036a8fef42f2a8e3ea34dfc38e165ee678d61f189db41  include/tirtc/audio_frame.h
-fb8ea35263167c33487bd3aa503c26ed2dfbf97877f7bda92c10065cd5c97c13  include/tirtc/audio_io.h
+04c3139f26d53aa2fb0a586a21ffb481a001c01e0412cd3414a952c9a9d336ba  include/tirtc/audio.h
+ce7ccffa054250a2dab0e2128e4701a66c7168e50e8c470dcff3bf609ff0959c  include/tirtc/audio_codec.h
+0b463d8adfcf45acb3f95fe3e2d94b37a26aaf37d293eb2b18274ddcb161f767  include/tirtc/audio_format.h
+7cac5b8c0cef28526ee00800fa8b0d1e60897ab971a0801899cb0333c9189fde  include/tirtc/audio_frame.h
+6daadba731521c01a3cf11c5d8b2ef797238737c939e2da3045fc83ad89af04a  include/tirtc/audio_io.h
 c2e1f31dcc75be461c577d18b1cebe32774f212d51cb4dd2a5b5a9bfe62b693e  include/tirtc/audio_io_android.h
 878bb18220ac62304ec95807f31e83e050c8dac14811f873454f9b045028694a  include/tirtc/audio_io_apple.h
 0569eb1d043ee7fa8d45ca43143eca1a22e4c4bf1507ca7e045e010a3ecbc288  include/tirtc/audio_io_harmony.h
 51cbc911fe9f9834046f0e0a1a7cdd814a8e194a615894a8b4d11f9e5f095610  include/tirtc/audio_io_windows.h
-21f60729117260a44af22c1af986ef17d22673b102b7b7a035f492d0665cce16  include/tirtc/audio_processing.h
+5acfd19fdcd8c9d96e4aee7c6f1d33a9211c0a5b0c0ab56d96e1b4dee48f3aa9  include/tirtc/audio_processing.h
 0ca7c3c630b1242f51a0fd8154097c0a332b4c816a5707090e4381719852998c  include/tirtc/audio_sample_rate.h
-58807ba56770f38f47aec995661cadf9d3c1d6ea7bfc940767f1ce67317c7546  include/tirtc/av.h
+07ca1b210a6e334c1474f25446477681472154568ecf74557cb3dc9b3a730662  include/tirtc/av.h
 b3e110ce04057d3be696b120d1abeaf6141711a1c63ca7f9bdf65ed5ec6e16d7  include/tirtc/error.h
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