電視音訊

電視輸入架構 (TIF) 管理員可與音訊轉送 API 搭配運作,以支援彈性音訊功能 路徑變更當晶片系統 (SoC) 實作電視硬體抽象層 (HAL) 時, 電視輸入裝置 (HDMI IN、 Tuner 等) 會提供 TvInputHardwareInfo,指定音訊類型和位址的 AudioPort 資訊。

  • 實體音訊輸入/輸出裝置有對應的 AudioPort。
  • 軟體音訊輸出/輸入串流會以 AudioMixPort (其子項類別) 表示 AudioPort)。

TIF 隨後會將 AudioPort 資訊用於音訊路由 API。

Android TV 輸入架構 (TIF)

圖 1. 電視輸入架構 (TIF)

需求條件

SoC 必須實作下列音訊路由 API 支援的音訊 HAL:

音訊連接埠
  • 電視音訊輸入有對應的音訊來源通訊埠實作。
  • 電視音訊輸出有對應的音訊接收器通訊埠實作。
  • 可在任何電視輸入音訊通訊埠與任何電視輸出音訊通訊埠之間建立音訊修補。
預設輸入方式 AudioRecord (使用 DEFAULT 輸入來源建立) 必須區隔出以下項目的虛擬空值輸入來源: 在 Android TV 上取得音訊:AUDIO_DEVICE_IN_DEFAULT。
裝置回送 需要支援 AUDIO_DEVICE_IN_LOOPBACK 輸入,該輸入是所有音訊輸出的完整組合 所有電視輸出裝置 (11Khz、16 位元單聲道或 48Khz、16 位元單聲道)。僅適用於音訊擷取。

電視音訊裝置

Android 支援下列音訊裝置輸入/輸出音訊裝置。

system/media/audio/include/system/audio.h

注意:在 Android 5.1 以下版本中, 這個檔案是:system/core/include/system/audio.h

/* output devices */
AUDIO_DEVICE_OUT_AUX_DIGITAL  = 0x400,
AUDIO_DEVICE_OUT_HDMI   = AUDIO_DEVICE_OUT_AUX_DIGITAL,
/* HDMI Audio Return Channel */
AUDIO_DEVICE_OUT_HDMI_ARC   = 0x40000,
/* S/PDIF out */
AUDIO_DEVICE_OUT_SPDIF    = 0x80000,
/* input devices */
AUDIO_DEVICE_IN_AUX_DIGITAL   = AUDIO_DEVICE_BIT_IN | 0x20,
AUDIO_DEVICE_IN_HDMI      = AUDIO_DEVICE_IN_AUX_DIGITAL,
/* TV tuner input */
AUDIO_DEVICE_IN_TV_TUNER    = AUDIO_DEVICE_BIT_IN | 0x4000,
/* S/PDIF in */
AUDIO_DEVICE_IN_SPDIF   = AUDIO_DEVICE_BIT_IN | 0x10000,
AUDIO_DEVICE_IN_LOOPBACK    = AUDIO_DEVICE_BIT_IN | 0x40000,

音訊 HAL 擴充功能

音訊轉送 API 的 Audio HAL 擴充功能定義如下:

system/media/audio/include/system/audio.h

注意:在 Android 5.1 以下版本中, 這個檔案是:system/core/include/system/audio.h

/* audio port configuration structure used to specify a particular configuration of an audio port */
struct audio_port_config {
    audio_port_handle_t      id;           /* port unique ID */
    audio_port_role_t        role;         /* sink or source */
    audio_port_type_t        type;         /* device, mix ... */
    unsigned int             config_mask;  /* e.g. AUDIO_PORT_CONFIG_ALL */
    unsigned int             sample_rate;  /* sampling rate in Hz */
    audio_channel_mask_t     channel_mask; /* channel mask if applicable */
    audio_format_t           format;       /* format if applicable */
    struct audio_gain_config gain;         /* gain to apply if applicable */
    union {
        struct audio_port_config_device_ext  device;  /* device specific info */
        struct audio_port_config_mix_ext     mix;     /* mix specific info */
        struct audio_port_config_session_ext session; /* session specific info */
    } ext;
};
struct audio_port {
    audio_port_handle_t      id;                /* port unique ID */
    audio_port_role_t        role;              /* sink or source */
    audio_port_type_t        type;              /* device, mix ... */
    unsigned int             num_sample_rates;  /* number of sampling rates in following array */
    unsigned int             sample_rates[AUDIO_PORT_MAX_SAMPLING_RATES];
    unsigned int             num_channel_masks; /* number of channel masks in following array */
    audio_channel_mask_t     channel_masks[AUDIO_PORT_MAX_CHANNEL_MASKS];
    unsigned int             num_formats;       /* number of formats in following array */
    audio_format_t           formats[AUDIO_PORT_MAX_FORMATS];
    unsigned int             num_gains;         /* number of gains in following array */
    struct audio_gain        gains[AUDIO_PORT_MAX_GAINS];
    struct audio_port_config active_config;     /* current audio port configuration */
    union {
        struct audio_port_device_ext  device;
        struct audio_port_mix_ext     mix;
        struct audio_port_session_ext session;
    } ext;
};

hardware/libhardware/include/hardware/audio.h

struct audio_hw_device {
  :
    /**
     * Routing control
     */

    /* Creates an audio patch between several source and sink ports.
     * The handle is allocated by the HAL and should be unique for this
     * audio HAL module. */
    int (*create_audio_patch)(struct audio_hw_device *dev,
                               unsigned int num_sources,
                               const struct audio_port_config *sources,
                               unsigned int num_sinks,
                               const struct audio_port_config *sinks,
                               audio_patch_handle_t *handle);

    /* Release an audio patch */
    int (*release_audio_patch)(struct audio_hw_device *dev,
                               audio_patch_handle_t handle);

    /* Fills the list of supported attributes for a given audio port.
     * As input, "port" contains the information (type, role, address etc...)
     * needed by the HAL to identify the port.
     * As output, "port" contains possible attributes (sampling rates, formats,
     * channel masks, gain controllers...) for this port.
     */
    int (*get_audio_port)(struct audio_hw_device *dev,
                          struct audio_port *port);

    /* Set audio port configuration */
    int (*set_audio_port_config)(struct audio_hw_device *dev,
                         const struct audio_port_config *config);

正在測試 DEVICE_IN_LOOPBACK

如要測試 DEVICE_IN_LOOPBACK 以進行電視監控,請使用下列測試程式碼。執行 測試,擷取的音訊會儲存至 /sdcard/record_loopback.raw,方便您聆聽 並使用 FFmpeg

<uses-permission android:name="android.permission.MODIFY_AUDIO_ROUTING" />
<uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE" />

   AudioRecord mRecorder;
   Handler mHandler = new Handler();
   int mMinBufferSize = AudioRecord.getMinBufferSize(RECORD_SAMPLING_RATE,
           AudioFormat.CHANNEL_IN_MONO,
           AudioFormat.ENCODING_PCM_16BIT);;
   static final int RECORD_SAMPLING_RATE = 48000;
   public void doCapture() {
       mRecorder = new AudioRecord(MediaRecorder.AudioSource.DEFAULT, RECORD_SAMPLING_RATE,
               AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, mMinBufferSize * 10);
       AudioManager am = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
       ArrayList<AudioPort> audioPorts = new ArrayList<AudioPort>();
       am.listAudioPorts(audioPorts);
       AudioPortConfig srcPortConfig = null;
       AudioPortConfig sinkPortConfig = null;
       for (AudioPort audioPort : audioPorts) {
           if (srcPortConfig == null
                   && audioPort.role() == AudioPort.ROLE_SOURCE
                   && audioPort instanceof AudioDevicePort) {
               AudioDevicePort audioDevicePort = (AudioDevicePort) audioPort;
               if (audioDevicePort.type() == AudioManager.DEVICE_IN_LOOPBACK) {
                   srcPortConfig = audioPort.buildConfig(48000, AudioFormat.CHANNEL_IN_DEFAULT,
                           AudioFormat.ENCODING_DEFAULT, null);
                   Log.d(LOG_TAG, "Found loopback audio source port : " + audioPort);
               }
           }
           else if (sinkPortConfig == null
                   && audioPort.role() == AudioPort.ROLE_SINK
                   && audioPort instanceof AudioMixPort) {
               sinkPortConfig = audioPort.buildConfig(48000, AudioFormat.CHANNEL_OUT_DEFAULT,
                       AudioFormat.ENCODING_DEFAULT, null);
               Log.d(LOG_TAG, "Found recorder audio mix port : " + audioPort);
           }
       }
       if (srcPortConfig != null && sinkPortConfig != null) {
           AudioPatch[] patches = new AudioPatch[] { null };
           int status = am.createAudioPatch(
                   patches,
                   new AudioPortConfig[] { srcPortConfig },
                   new AudioPortConfig[] { sinkPortConfig });
           Log.d(LOG_TAG, "Result of createAudioPatch(): " + status);
       }
       mRecorder.startRecording();
       processAudioData();
       mRecorder.stop();
       mRecorder.release();
   }
   private void processAudioData() {
       OutputStream rawFileStream = null;
       byte data[] = new byte[mMinBufferSize];
       try {
           rawFileStream = new BufferedOutputStream(
                   new FileOutputStream(new File("/sdcard/record_loopback.raw")));
       } catch (FileNotFoundException e) {
           Log.d(LOG_TAG, "Can't open file.", e);
       }
       long startTimeMs = System.currentTimeMillis();
       while (System.currentTimeMillis() - startTimeMs < 5000) {
           int nbytes = mRecorder.read(data, 0, mMinBufferSize);
           if (nbytes <= 0) {
               continue;
           }
           try {
               rawFileStream.write(data);
           } catch (IOException e) {
               Log.e(LOG_TAG, "Error on writing raw file.", e);
           }
       }
       try {
           rawFileStream.close();
       } catch (IOException e) {
       }
       Log.d(LOG_TAG, "Exit audio recording.");
   }

/sdcard/record_loopback.raw 中找出擷取的音訊檔案,並使用以下工具聆聽該檔案: FFmpeg

adb pull /sdcard/record_loopback.raw
ffmpeg -f s16le -ar 48k -ac 1 -i record_loopback.raw record_loopback.wav
ffplay record_loopback.wav

用途

本節說明電視音訊的常見用途。

配備喇叭輸出的電視調諧器

電視調諧器啟用時,音訊轉送 API 會在調音器之間建立音訊修補程式 和預設輸出內容 (例如喇叭)。調音器輸出內容不需要解碼,但最終會以 音訊輸出與 Software output_stream 混合。

Android TV 調諧器音訊修補程式

圖 2. 電視調諧器 (內建喇叭輸出器) 的音訊修補程式。

直播電視時的 HDMI 輸出端

使用者正在觀看電視直播,然後切換到 HDMI 音訊輸出裝置 (Intent.ACTION_HDMI_AUDIO_PLUG) ,直接在 Google Cloud 控制台實際操作。所有 output_streams 的輸出裝置都會變更為 HDMI_OUT 連接埠,TIF 管理員也會變更 現有調諧器音訊修補程式的接收器通訊埠插入 HDMI_OUT 連接埠。

Android TV HDMI-OUT 音訊修補程式

圖 3. 直播電視的 HDMI OUT 音訊修補程式。