spice-gtk音频数据播放流程

1、 channel-playback.c源文件中 SPICE_MSG_PLAYBACK_DATA的 set_handler处理回调函数（解析数据->解码数据->发送信号）

/* coroutine context */
static void playback_handle_data(SpiceChannel *channel, SpiceMsgIn *in)
{
    SpicePlaybackChannelPrivate *c = SPICE_PLAYBACK_CHANNEL(channel)->priv;
    SpiceMsgPlaybackPacket *packet = spice_msg_in_parsed(in);
#ifdef DEBUG
    CHANNEL_DEBUG(channel, "%s: time %u data %p size %d", __FUNCTION__,
                  packet->time, packet->data, packet->data_size);
#endif

    //判断时间是否过时
    if (spice_mmtime_diff(c->last_time, packet->time) > 0)
        g_warn_if_reached();
    c->last_time = packet->time;
    uint8_t *data = packet->data;
    int n = packet->data_size;
    uint8_t pcm[SND_CODEC_MAX_FRAME_SIZE * 2 * 2];

    //是否解码数据
    if (c->mode != SPICE_AUDIO_DATA_MODE_RAW) {
        n = sizeof(pcm);
        data = pcm;
        if (snd_codec_decode(c->codec, packet->data, packet->data_size,pcm, &n) != SND_CODEC_OK) {
            g_warning("snd_codec_decode() error");
            return;
        }
    }

    g_coroutine_signal_emit(channel, signals[SPICE_PLAYBACK_DATA], 0, data, n);
    if ((c->frame_count++ % 100) == 0) {
        g_coroutine_signal_emit(channel, signals[SPICE_PLAYBACK_GET_DELAY], 0);
    }
}

2、 spice-gstaudio.c源 文件中的 connect_channel 函数 连接了信号

static gboolean connect_channel(SpiceAudio *audio, SpiceChannel *channel)
{
    SpiceGstaudio *gstaudio = SPICE_GSTAUDIO(audio);
    SpiceGstaudioPrivate *p = gstaudio->priv;
    if (SPICE_IS_PLAYBACK_CHANNEL(channel)) {
        g_return_val_if_fail(p->pchannel == NULL, FALSE);
        p->pchannel = channel;
        g_object_weak_ref(G_OBJECT(p->pchannel), channel_weak_notified, audio);
        spice_g_signal_connect_object(channel, "playback-start", G_CALLBACK(playback_start), gstaudio, 0);
        spice_g_signal_connect_object(channel, "playback-data", G_CALLBACK(playback_data), gstaudio, 0);
        spice_g_signal_connect_object(channel, "playback-stop", G_CALLBACK(playback_stop), gstaudio, G_CONNECT_SWAPPED);
        spice_g_signal_connect_object(channel, "notify::volume", G_CALLBACK(playback_volume_changed), gstaudio, 0);
        spice_g_signal_connect_object(channel, "notify::mute", G_CALLBACK(playback_mute_changed), gstaudio, 0);
        return TRUE;
    }
    ...
}

3、s pice-gstaudio.c源 文件中的 playback_data函数将PCM的音频数据放到pipeline的appsrc中，开始播放

static void playback_data(SpicePlaybackChannel *channel,
                          gpointer *audio, gint size,
                          gpointer data)
{
    SpiceGstaudio *gstaudio = data;
    SpiceGstaudioPrivate *p = gstaudio->priv;
    GstBuffer *buf;
    g_return_if_fail(p != NULL);
    if(p->playback.queue){
        uint32_t queue_current_buffers = 0;
        g_object_get(p->playback.queue,"current-level-buffers",&queue_current_buffers,NULL);
        printf("1---------------queue_current_buffers:%u------\n",queue_current_buffers);
    }

    audio = g_memdup(audio, size); /* TODO: try to avoid memory copy */
    buf = gst_buffer_new_wrapped(audio, size);
    gst_app_src_push_buffer(GST_APP_SRC(p->playback.src), buf);
}

4、sp ice-gstaudio.c源 文件中的 playback_start 函数创建pipeline

static void playback_start(SpicePlaybackChannel *channel, gint format, gint channels,
                           gint frequency, gpointer data)
{
    SpiceGstaudio *gstaudio = data;
    SpiceGstaudioPrivate *p = gstaudio->priv;
    g_return_if_fail(p != NULL);
    g_return_if_fail(format == SPICE_AUDIO_FMT_S16);

    if (p->playback.pipe &&
        (p->playback.rate != frequency ||
         p->playback.channels != channels)) {
        playback_stop(gstaudio);
        g_clear_pointer(&p->playback.pipe, gst_object_unref);
    }

    if (!p->playback.pipe) {
        GError *error = NULL;
        gchar *audio_caps =
            g_strdup_printf("audio/x-raw,format=\"S16LE\",channels=%d,rate=%d,"
                            "layout=interleaved", channels, frequency);
        gchar *pipeline = g_strdup (g_getenv("SPICE_GST_AUDIOSINK"));
        if (pipeline == NULL){
            //pipeline = g_strdup_printf("appsrc is-live=1 do-timestamp=0 format=time caps=\"%s\" name=\"appsrc\" ! queue max-size-buffers=2 name=\"queue\" ! "
                                       //"audioconvert ! audioresample ! autoaudiosink name=\"audiosink\"", audio_caps);
            pipeline = g_strdup_printf("appsrc is-live=1 do-timestamp=0 format=time caps=\"%s\" name=\"appsrc\" ! "
                                       "audioconvert ! audioresample ! autoaudiosink name=\"audiosink\"", audio_caps);
        }
        SPICE_DEBUG("audio pipeline: %s", pipeline);
        p->playback.pipe = gst_parse_launch(pipeline, &error);
        if (error != NULL) {
            g_warning("Failed to create pipeline: %s", error->message);
            goto cleanup;
        }
        p->playback.src = gst_bin_get_by_name(GST_BIN(p->playback.pipe), "appsrc");
        p->playback.sink = gst_bin_get_by_name(GST_BIN(p->playback.pipe), "audiosink");
//        p->playback.queue = gst_bin_get_by_name(GST_BIN(p->playback.pipe), "queue");
        p->playback.queue = NULL;
        p->playback.rate = frequency;
        p->playback.channels = channels;

cleanup:
        if (error != NULL)
            g_clear_pointer(&p->playback.pipe, gst_object_unref);
        g_clear_error(&error);
        g_free(audio_caps);
        g_free(pipeline);
    }

    if (p->playback.pipe)
        gst_element_set_state(p->playback.pipe, GST_STATE_PLAYING);

    if (!p->playback.fake && p->mmtime_id == 0) {
        update_mmtime_timeout_cb(gstaudio);
        p->mmtime_id = g_timeout_add_seconds(1, update_mmtime_timeout_cb, gstaudio);
    }
}