//打开流
static int adev_open_output_stream(struct audio_hw_device *dev,
                                   audio_io_handle_t handle,
                                   audio_devices_t devices,
                                   audio_output_flags_t flags,
                                   struct audio_config *config,
                                   struct audio_stream_out **stream_out,
                                   const char *address)
{
    struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev);
    ALOGD("adev_open_output_stream(address=%s)", address);
    struct submix_stream_out *out;
    (void)handle;
    (void)devices;
    (void)flags;

    *stream_out = NULL;

    // Make sure it's possible to open the device given the current audio config.
    submix_sanitize_config(config, false);

    int route_idx = -1;

    pthread_mutex_lock(&rsxadev->lock);

    status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx);
    if (res != OK) {
        ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address);
        pthread_mutex_unlock(&rsxadev->lock);
        return res;
    }

    if (!submix_open_validate_l(rsxadev, route_idx, config, false)) {
        ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address);
        pthread_mutex_unlock(&rsxadev->lock);
        return -EINVAL;
    }

    out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out));
    if (!out) {
        pthread_mutex_unlock(&rsxadev->lock);
        return -ENOMEM;
    }

    // Initialize the function pointer tables (v-tables).
    out->stream.common.get_sample_rate = out_get_sample_rate;
    out->stream.common.set_sample_rate = out_set_sample_rate;
    out->stream.common.get_buffer_size = out_get_buffer_size;
    out->stream.common.get_channels = out_get_channels;
    out->stream.common.get_format = out_get_format;
    out->stream.common.set_format = out_set_format;
    out->stream.common.standby = out_standby;
    out->stream.common.dump = out_dump;
    //stream
    out->stream.common.set_parameters = out_set_parameters;
    out->stream.common.get_parameters = out_get_parameters;
    out->stream.common.add_audio_effect = out_add_audio_effect;
    out->stream.common.remove_audio_effect = out_remove_audio_effect;
    out->stream.get_latency = out_get_latency;
    out->stream.set_volume = out_set_volume;
    out->stream.write = out_write;
    out->stream.get_render_position = out_get_render_position;
    out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
    out->stream.get_presentation_position = out_get_presentation_position;

    // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so
    // that it's recreated.
    if ((rsxadev->routes[route_idx].rsxSink != NULL
            && rsxadev->routes[route_idx].rsxSink->isShutdown())) {
        submix_audio_device_release_pipe_l(rsxadev, route_idx);
    }

    // Store a pointer to the device from the output stream.
    out->dev = rsxadev;
    // Initialize the pipe.
    const size_t pipeSizeInFrames = pipe_size_in_frames(config->sample_rate);
    ALOGI("adev_open_output_stream(): about to create pipe at index %d, rate %u, pipe size %zu",
          route_idx, config->sample_rate, pipeSizeInFrames);
    submix_audio_device_create_pipe_l(rsxadev, config, pipeSizeInFrames,
            DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx);
#if LOG_STREAMS_TO_FILES
    out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY,
                       LOG_STREAM_FILE_PERMISSIONS);
    ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s",
             strerror(errno));
    ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd);
#endif // LOG_STREAMS_TO_FILES
    // Return the output stream.
    *stream_out = &out->stream;

    pthread_mutex_unlock(&rsxadev->lock);
    return 0;
}

    /**
     * set/get audio stream parameters. The function accepts a list of
     * parameter key value pairs in the form: key1=value1;key2=value2;...
     *
     * Some keys are reserved for standard parameters (See AudioParameter class)
     *
     * If the implementation does not accept a parameter change while
     * the output is active but the parameter is acceptable otherwise, it must
     * return -ENOSYS.
     *
     * The audio flinger will put the stream in standby and then change the
     * parameter value.
     */
    int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs);

//frameworks/av/services/audioflinger/Threads.cpp
// A differential check on the timestamps to see if there is a change in the
// timestamp frame position between the last call to checkRunningTimestamp.   
// 判断是否还在running
bool AudioFlinger::PlaybackThread::checkRunningTimestamp()
{
    uint64_t position = 0;
    struct timespec unused;
    const status_t ret = mOutput->getPresentationPosition(&position, &unused);
    if (ret == NO_ERROR) {
        if (position != mLastCheckedTimestampPosition) {
            mLastCheckedTimestampPosition = position;
            return true;
        }
    }
    return false;
}

//frameworks/av/services/audioflinger/Threads.h
// Always query HAL timestamp and update timestamp verifier. In standby or pause,
// HAL may be draining some small duration buffered data for fade out.

status_t    threadloop_getHalTimestamp_l(ExtendedTimestamp *timestamp) const override {
       // For DIRECT and OFFLOAD threads, query the output sink directly.
       if (mOutput != nullptr) {
           uint64_t uposition64;
           struct timespec time;
           if (mOutput->getPresentationPosition(
               &uposition64, &time) == OK) {
               timestamp->mPosition[ExtendedTimestamp::LOCATION_KERNEL]
                   = (int64_t)uposition64;
               timestamp->mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]
                   = audio_utils_ns_from_timespec(&time);
               return NO_ERROR;
           }
       }
       return INVALID_OPERATION;
   }

static void android_media_MediaSync_native_updateQueuedAudioData(
        JNIEnv *env, jobject thiz, jint sizeInBytes, jlong presentationTimeUs) {
    sp<JMediaSync> sync = getMediaSync(env, thiz);
    if (sync == NULL) {
        throwExceptionAsNecessary(env, INVALID_OPERATION);
        return;
    }

    status_t err = sync->updateQueuedAudioData(sizeInBytes, presentationTimeUs);
    if (err != NO_ERROR) {
        throwExceptionAsNecessary(env, err);
        return;
    }
}

status_t MediaSync::updateQueuedAudioData(
        size_t sizeInBytes, int64_t presentationTimeUs) {
    if (sizeInBytes == 0) {
        return OK;
    }

    Mutex::Autolock lock(mMutex);

    if (mAudioTrack == NULL) {
        ALOGW("updateQueuedAudioData: audioTrack has NOT been configured.");
        return INVALID_OPERATION;
    }

    //计算帧数
    int64_t numFrames = sizeInBytes / mAudioTrack->frameSize();
    //最大媒体时间
    int64_t maxMediaTimeUs = presentationTimeUs
            + getDurationIfPlayedAtNativeSampleRate_l(numFrames);

    //本地时间
    int64_t nowUs = ALooper::GetNowUs();
    //当前媒体播放的媒体时间
    int64_t nowMediaUs = presentationTimeUs
        	//透过帧算出开始时间
            - getDurationIfPlayedAtNativeSampleRate_l(mNumFramesWritten)
        	//透过AudioTrack的getTimestamp计算已经播放的duration, 加上上面算出的开始时间,即为当前媒体时间
            + getPlayedOutAudioDurationMedia_l(nowUs);

    mNumFramesWritten += numFrames;

    int64_t oldRealTime = -1;
    if (mNextBufferItemMediaUs != -1) {
        oldRealTime = getRealTime(mNextBufferItemMediaUs, nowUs);
    }

    mMediaClock->updateAnchor(nowMediaUs, nowUs, maxMediaTimeUs);
    mHasAudio = true;

    if (oldRealTime != -1) {
        int64_t newRealTime = getRealTime(mNextBufferItemMediaUs, nowUs);
        if (newRealTime >= oldRealTime) {
            return OK;
        }
    }

    mNextBufferItemMediaUs = -1;
    onDrainVideo_l();
    return OK;
}

//计算当前播放的duration
int64_t MediaSync::getPlayedOutAudioDurationMedia_l(int64_t nowUs) {
    CHECK(mAudioTrack != NULL);

    uint32_t numFramesPlayed;
    int64_t numFramesPlayedAtUs;
    AudioTimestamp ts;

    status_t res = mAudioTrack->getTimestamp(ts);
    if (res == OK) {
        // case 1: mixing audio tracks.
        //帧数据
        numFramesPlayed = ts.mPosition;
        //当前本地时间的present timestamp
        numFramesPlayedAtUs = ts.mTime.tv_sec * 1000000LL + ts.mTime.tv_nsec / 1000;
        //ALOGD("getTimestamp: OK %d %lld",
        //      numFramesPlayed, (long long)numFramesPlayedAtUs);
    } else if (res == WOULD_BLOCK) {
        // case 2: transitory state on start of a new track
        numFramesPlayed = 0;
        numFramesPlayedAtUs = nowUs;
        //ALOGD("getTimestamp: WOULD_BLOCK %d %lld",
        //      numFramesPlayed, (long long)numFramesPlayedAtUs);
    } else {
        // case 3: transitory at new track or audio fast tracks.
        res = mAudioTrack->getPosition(&numFramesPlayed);
        CHECK_EQ(res, (status_t)OK);
        numFramesPlayedAtUs = nowUs;
        numFramesPlayedAtUs += 1000LL * mAudioTrack->latency() / 2; /* XXX */
        //ALOGD("getPosition: %d %lld", numFramesPlayed, (long long)numFramesPlayedAtUs);
    }

    //can't be negative until 12.4 hrs, test.
    //CHECK_EQ(numFramesPlayed & (1 << 31), 0);
    int64_t durationUs =
        //帧数据计算的时间
        getDurationIfPlayedAtNativeSampleRate_l(numFramesPlayed)
        //从numFramesPlayedAtUs到nowUs的偏移时间
            + nowUs - numFramesPlayedAtUs;
    if (durationUs < 0) {
        // Occurs when numFramesPlayed position is very small and the following:
        // (1) In case 1, the time nowUs is computed before getTimestamp() is
        //     called and numFramesPlayedAtUs is greater than nowUs by time more
        //     than numFramesPlayed.
        // (2) In case 3, using getPosition and adding mAudioTrack->latency()
        //     to numFramesPlayedAtUs, by a time amount greater than
        //     numFramesPlayed.
        //
        // Both of these are transitory conditions.
        ALOGV("getPlayedOutAudioDurationMedia_l: negative duration %lld "
              "set to zero", (long long)durationUs);
        durationUs = 0;
    }
    ALOGV("getPlayedOutAudioDurationMedia_l(%lld) nowUs(%lld) frames(%u) "
          "framesAt(%lld)",
          (long long)durationUs, (long long)nowUs, numFramesPlayed,
          (long long)numFramesPlayedAtUs);
    return durationUs;
}