コード例 #1
0
ファイル: channelmap.c プロジェクト: KimT/pulseaudio_kt
pa_channel_map* pa_channel_map_init_extend(pa_channel_map *m, unsigned channels, pa_channel_map_def_t def) {
    unsigned c;

    pa_assert(m);
    pa_assert(channels > 0);
    pa_assert(channels <= PA_CHANNELS_MAX);
    pa_assert(def < PA_CHANNEL_MAP_DEF_MAX);

    pa_channel_map_init(m);

    for (c = channels; c > 0; c--) {

        if (pa_channel_map_init_auto(m, c, def)) {
            unsigned i = 0;

            for (; c < channels; c++) {

                m->map[c] = PA_CHANNEL_POSITION_AUX0 + i;
                i++;
            }

            m->channels = (uint8_t) channels;

            return m;
        }
    }

    return NULL;
}
コード例 #2
0
ファイル: backend.c プロジェクト: chrippa/xmms2
static pa_channel_map *
xmms_pulse_backend_default_channel_map (pa_channel_map *m, int channels)
{
	assert(m);
	assert(channels > 0);
	assert(channels <= PA_CHANNELS_MAX);

	pa_channel_map_init(m);

	m->channels = (uint8_t) channels;
	switch (channels) {
		case 4:
			m->map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
			m->map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
			m->map[2] = PA_CHANNEL_POSITION_REAR_LEFT;
			m->map[3] = PA_CHANNEL_POSITION_REAR_RIGHT;
			return m;
		case 5:
			m->map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
			m->map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
			m->map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
			m->map[3] = PA_CHANNEL_POSITION_REAR_LEFT;
			m->map[4] = PA_CHANNEL_POSITION_REAR_RIGHT;
			return m;
		case 7:
			m->map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
			m->map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
			m->map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
			m->map[3] = PA_CHANNEL_POSITION_LFE;
			m->map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
			m->map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
			m->map[6] = PA_CHANNEL_POSITION_REAR_CENTER;
			return m;
		case 8:
			m->map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
			m->map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
			m->map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
			m->map[3] = PA_CHANNEL_POSITION_LFE;
			m->map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
			m->map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
			m->map[6] = PA_CHANNEL_POSITION_SIDE_LEFT;
			m->map[7] = PA_CHANNEL_POSITION_SIDE_RIGHT;
			return m;
		default:
			return pa_channel_map_init_auto (m, channels,
			                                 PA_CHANNEL_MAP_WAVEEX);
	}
}
コード例 #3
0
ファイル: pulseaudio.c プロジェクト: siana/2p-openal
static ALCboolean pulse_reset_playback(ALCdevice *device) //{{{
{
    pulse_data *data = device->ExtraData;
    pa_stream_flags_t flags = 0;
    pa_channel_map chanmap;

    pa_threaded_mainloop_lock(data->loop);

    if(!(device->Flags&DEVICE_CHANNELS_REQUEST))
    {
        pa_operation *o;
        o = pa_context_get_sink_info_by_name(data->context, data->device_name, sink_info_callback, device);
        while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
            pa_threaded_mainloop_wait(data->loop);
        pa_operation_unref(o);
    }
    if(!(device->Flags&DEVICE_FREQUENCY_REQUEST))
        flags |= PA_STREAM_FIX_RATE;

    data->frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
    data->attr.prebuf = -1;
    data->attr.fragsize = -1;
    data->attr.minreq = device->UpdateSize * data->frame_size;
    data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
    data->attr.maxlength = -1;
    flags |= PA_STREAM_EARLY_REQUESTS;
    flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE;

    switch(device->FmtType)
    {
        case DevFmtByte:
            device->FmtType = DevFmtUByte;
            /* fall-through */
        case DevFmtUByte:
            data->spec.format = PA_SAMPLE_U8;
            break;
        case DevFmtUShort:
            device->FmtType = DevFmtShort;
            /* fall-through */
        case DevFmtShort:
            data->spec.format = PA_SAMPLE_S16NE;
            break;
        case DevFmtFloat:
            data->spec.format = PA_SAMPLE_FLOAT32NE;
            break;
    }
    data->spec.rate = device->Frequency;
    data->spec.channels = ChannelsFromDevFmt(device->FmtChans);

    if(pa_sample_spec_valid(&data->spec) == 0)
    {
        ERR("Invalid sample format\n");
        pa_threaded_mainloop_unlock(data->loop);
        return ALC_FALSE;
    }

    if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
    {
        ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
        pa_threaded_mainloop_unlock(data->loop);
        return ALC_FALSE;
    }
    SetDefaultWFXChannelOrder(device);

    data->stream = connect_playback_stream(device, flags, &data->attr, &data->spec, &chanmap);
    if(!data->stream)
    {
        pa_threaded_mainloop_unlock(data->loop);
        return ALC_FALSE;
    }

    pa_stream_set_state_callback(data->stream, stream_state_callback2, device);

    data->spec = *(pa_stream_get_sample_spec(data->stream));
    if(device->Frequency != data->spec.rate)
    {
        pa_operation *o;

        if((device->Flags&DEVICE_FREQUENCY_REQUEST))
            ERR("Failed to set frequency %dhz, got %dhz instead\n", device->Frequency, data->spec.rate);
        device->Flags &= ~DEVICE_FREQUENCY_REQUEST;

        /* Server updated our playback rate, so modify the buffer attribs
         * accordingly. */
        data->attr.minreq = (ALuint64)(data->attr.minreq/data->frame_size) *
                            data->spec.rate / device->Frequency * data->frame_size;
        data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);

        o = pa_stream_set_buffer_attr(data->stream, &data->attr,
                                      stream_success_callback, device);
        while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
            pa_threaded_mainloop_wait(data->loop);
        pa_operation_unref(o);

        device->Frequency = data->spec.rate;
    }

#if PA_CHECK_VERSION(0,9,15)
    if(pa_stream_set_buffer_attr_callback)
        pa_stream_set_buffer_attr_callback(data->stream, stream_buffer_attr_callback, device);
#endif
    pa_stream_set_moved_callback(data->stream, stream_device_callback, device);
    pa_stream_set_write_callback(data->stream, stream_write_callback, device);
    pa_stream_set_underflow_callback(data->stream, stream_signal_callback, device);

    data->attr = *(pa_stream_get_buffer_attr(data->stream));
    ERR("PulseAudio returned minreq=%d, tlength=%d\n", data->attr.minreq, data->attr.tlength);
    device->UpdateSize = data->attr.minreq / data->frame_size;
    device->NumUpdates = (data->attr.tlength/data->frame_size) / device->UpdateSize;
    while(device->NumUpdates <= 2)
    {
        pa_operation *o;

        ERR("minreq too high - expect lag or break up\n");

        /* Server gave a comparatively large minreq, so modify the tlength. */
        device->NumUpdates = 2;
        data->attr.tlength = data->attr.minreq * device->NumUpdates;

        o = pa_stream_set_buffer_attr(data->stream, &data->attr,
                                      stream_success_callback, device);

        while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
            pa_threaded_mainloop_wait(data->loop);
        pa_operation_unref(o);

        data->attr = *(pa_stream_get_buffer_attr(data->stream));
        ERR("PulseAudio returned minreq=%d, tlength=%d", data->attr.minreq, data->attr.tlength);
        device->UpdateSize = data->attr.minreq / data->frame_size;
        device->NumUpdates = (data->attr.tlength/data->frame_size) / device->UpdateSize;
    }


    data->thread = StartThread(PulseProc, device);
    if(!data->thread)
    {
#if PA_CHECK_VERSION(0,9,15)
        if(pa_stream_set_buffer_attr_callback)
            pa_stream_set_buffer_attr_callback(data->stream, NULL, NULL);
#endif
        pa_stream_set_moved_callback(data->stream, NULL, NULL);
        pa_stream_set_write_callback(data->stream, NULL, NULL);
        pa_stream_set_underflow_callback(data->stream, NULL, NULL);
        pa_stream_disconnect(data->stream);
        pa_stream_unref(data->stream);
        data->stream = NULL;

        pa_threaded_mainloop_unlock(data->loop);
        return ALC_FALSE;
    }

    pa_threaded_mainloop_unlock(data->loop);
    return ALC_TRUE;
} //}}}
コード例 #4
0
ファイル: pulseaudio.c プロジェクト: siana/2p-openal
static ALCenum pulse_open_capture(ALCdevice *device, const ALCchar *device_name) //{{{
{
    char *pulse_name = NULL;
    pulse_data *data;
    pa_stream_flags_t flags = 0;
    pa_stream_state_t state;
    pa_channel_map chanmap;

    if(!allCaptureDevNameMap)
        probe_devices(AL_TRUE);

    if(!device_name)
        device_name = pulse_device;
    else if(strcmp(device_name, pulse_device) != 0)
    {
        ALuint i;

        for(i = 0;i < numCaptureDevNames;i++)
        {
            if(strcmp(device_name, allCaptureDevNameMap[i].name) == 0)
            {
                pulse_name = allCaptureDevNameMap[i].device_name;
                break;
            }
        }
        if(i == numCaptureDevNames)
            return ALC_INVALID_VALUE;
    }

    if(pulse_open(device, device_name) == ALC_FALSE)
        return ALC_INVALID_VALUE;

    data = device->ExtraData;
    pa_threaded_mainloop_lock(data->loop);

    data->samples = device->UpdateSize * device->NumUpdates;
    data->frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
    data->samples = maxu(data->samples, 100 * device->Frequency / 1000);

    if(!(data->ring = CreateRingBuffer(data->frame_size, data->samples)))
    {
        pa_threaded_mainloop_unlock(data->loop);
        goto fail;
    }

    data->attr.minreq = -1;
    data->attr.prebuf = -1;
    data->attr.maxlength = data->samples * data->frame_size;
    data->attr.tlength = -1;
    data->attr.fragsize = minu(data->samples, 50*device->Frequency/1000) *
                          data->frame_size;

    data->spec.rate = device->Frequency;
    data->spec.channels = ChannelsFromDevFmt(device->FmtChans);

    switch(device->FmtType)
    {
        case DevFmtUByte:
            data->spec.format = PA_SAMPLE_U8;
            break;
        case DevFmtShort:
            data->spec.format = PA_SAMPLE_S16NE;
            break;
        case DevFmtFloat:
            data->spec.format = PA_SAMPLE_FLOAT32NE;
            break;
        case DevFmtByte:
        case DevFmtUShort:
            ERR("Capture format type %#x capture not supported on PulseAudio\n", device->FmtType);
            pa_threaded_mainloop_unlock(data->loop);
            goto fail;
    }

    if(pa_sample_spec_valid(&data->spec) == 0)
    {
        ERR("Invalid sample format\n");
        pa_threaded_mainloop_unlock(data->loop);
        goto fail;
    }

    if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
    {
        ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
        pa_threaded_mainloop_unlock(data->loop);
        goto fail;
    }

    data->stream = pa_stream_new(data->context, "Capture Stream", &data->spec, &chanmap);
    if(!data->stream)
    {
        ERR("pa_stream_new() failed: %s\n",
            pa_strerror(pa_context_errno(data->context)));

        pa_threaded_mainloop_unlock(data->loop);
        goto fail;
    }

    pa_stream_set_state_callback(data->stream, stream_state_callback, data->loop);

    flags |= PA_STREAM_START_CORKED|PA_STREAM_ADJUST_LATENCY;
    if(pa_stream_connect_record(data->stream, pulse_name, &data->attr, flags) < 0)
    {
        ERR("Stream did not connect: %s\n",
            pa_strerror(pa_context_errno(data->context)));

        pa_stream_unref(data->stream);
        data->stream = NULL;

        pa_threaded_mainloop_unlock(data->loop);
        goto fail;
    }

    while((state=pa_stream_get_state(data->stream)) != PA_STREAM_READY)
    {
        if(!PA_STREAM_IS_GOOD(state))
        {
            ERR("Stream did not get ready: %s\n",
                pa_strerror(pa_context_errno(data->context)));

            pa_stream_unref(data->stream);
            data->stream = NULL;

            pa_threaded_mainloop_unlock(data->loop);
            goto fail;
        }

        pa_threaded_mainloop_wait(data->loop);
    }
    pa_stream_set_state_callback(data->stream, stream_state_callback2, device);

    pa_threaded_mainloop_unlock(data->loop);
    return ALC_NO_ERROR;

fail:
    pulse_close(device);
    return ALC_INVALID_VALUE;
} //}}}
コード例 #5
0
ファイル: audiooutputpulse.cpp プロジェクト: microe/mythtv
bool AudioOutputPulseAudio::OpenDevice()
{
    QString fn_log_tag = "OpenDevice, ";
    if (channels > PULSE_MAX_CHANNELS )
    {
        VBERROR(fn_log_tag + QString("audio channel limit %1, but %2 requested")
                .arg(PULSE_MAX_CHANNELS).arg(channels));
        return false;
    }

    sample_spec.rate = samplerate;
    sample_spec.channels = volume_control.channels = channels;
    switch (output_format)
    {
    case FORMAT_U8:
        sample_spec.format = PA_SAMPLE_U8;
        break;
    case FORMAT_S16:
        sample_spec.format = PA_SAMPLE_S16NE;
        break;
// define from PA 0.9.15 only
#ifdef PA_MAJOR
    case FORMAT_S24LSB:
        sample_spec.format = PA_SAMPLE_S24_32NE;
        break;
#endif
    case FORMAT_S32:
        sample_spec.format = PA_SAMPLE_S32NE;
        break;
    case FORMAT_FLT:
        sample_spec.format = PA_SAMPLE_FLOAT32NE;
        break;
        break;
    default:
        VBERROR(fn_log_tag + QString("unsupported sample format %1")
                .arg(output_format));
        return false;
    }

    if (!pa_sample_spec_valid(&sample_spec))
    {
        VBERROR(fn_log_tag + "invalid sample spec");
        return false;
    }
    else
    {
        char spec[PA_SAMPLE_SPEC_SNPRINT_MAX];
        pa_sample_spec_snprint(spec, sizeof(spec), &sample_spec);
        VBAUDIO(fn_log_tag + QString("using sample spec %1").arg(spec));
    }

    pa_channel_map *pmap = NULL;

    if(!(pmap = pa_channel_map_init_auto(&channel_map, channels,
                                         PA_CHANNEL_MAP_WAVEEX)) < 0)
    {
        VBERROR(fn_log_tag + "failed to init channel map");
        return false;
    }

    channel_map = *pmap;

    mainloop = pa_threaded_mainloop_new();
    if (!mainloop)
    {
        VBERROR(fn_log_tag + "failed to get new threaded mainloop");
        return false;
    }

    pa_threaded_mainloop_start(mainloop);
    pa_threaded_mainloop_lock(mainloop);

    if (!ContextConnect())
    {
        pa_threaded_mainloop_unlock(mainloop);
        pa_threaded_mainloop_stop(mainloop);
        return false;
    }

    if (!ConnectPlaybackStream())
    {
        pa_threaded_mainloop_unlock(mainloop);
        pa_threaded_mainloop_stop(mainloop);
        return false;
    }

    pa_threaded_mainloop_unlock(mainloop);
    return true;
}
コード例 #6
0
ファイル: module-rtp-send.c プロジェクト: jprvita/pulseaudio
int pa__init(pa_module*m) {
    struct userdata *u;
    pa_modargs *ma = NULL;
    const char *dst_addr;
    const char *src_addr;
    uint32_t port = DEFAULT_PORT, mtu;
    uint32_t ttl = DEFAULT_TTL;
    sa_family_t af;
    int fd = -1, sap_fd = -1;
    pa_source *s;
    pa_sample_spec ss;
    pa_channel_map cm;
    struct sockaddr_in dst_sa4, dst_sap_sa4, src_sa4, src_sap_sa4;
#ifdef HAVE_IPV6
    struct sockaddr_in6 dst_sa6, dst_sap_sa6, src_sa6, src_sap_sa6;
#endif
    struct sockaddr_storage sa_dst;
    pa_source_output *o = NULL;
    uint8_t payload;
    char *p;
    int r, j;
    socklen_t k;
    char hn[128], *n;
    bool loop = false;
    enum inhibit_auto_suspend inhibit_auto_suspend = INHIBIT_AUTO_SUSPEND_ONLY_WITH_NON_MONITOR_SOURCES;
    const char *inhibit_auto_suspend_str;
    pa_source_output_new_data data;

    pa_assert(m);

    if (!(ma = pa_modargs_new(m->argument, valid_modargs))) {
        pa_log("Failed to parse module arguments");
        goto fail;
    }

    if (!(s = pa_namereg_get(m->core, pa_modargs_get_value(ma, "source", NULL), PA_NAMEREG_SOURCE))) {
        pa_log("Source does not exist.");
        goto fail;
    }

    if (pa_modargs_get_value_boolean(ma, "loop", &loop) < 0) {
        pa_log("Failed to parse \"loop\" parameter.");
        goto fail;
    }

    if ((inhibit_auto_suspend_str = pa_modargs_get_value(ma, "inhibit_auto_suspend", NULL))) {
        if (pa_streq(inhibit_auto_suspend_str, "always"))
            inhibit_auto_suspend = INHIBIT_AUTO_SUSPEND_ALWAYS;
        else if (pa_streq(inhibit_auto_suspend_str, "never"))
            inhibit_auto_suspend = INHIBIT_AUTO_SUSPEND_NEVER;
        else if (pa_streq(inhibit_auto_suspend_str, "only_with_non_monitor_sources"))
            inhibit_auto_suspend = INHIBIT_AUTO_SUSPEND_ONLY_WITH_NON_MONITOR_SOURCES;
        else {
            pa_log("Failed to parse the \"inhibit_auto_suspend\" parameter.");
            goto fail;
        }
    }

    ss = s->sample_spec;
    pa_rtp_sample_spec_fixup(&ss);
    cm = s->channel_map;
    if (pa_modargs_get_sample_spec(ma, &ss) < 0) {
        pa_log("Failed to parse sample specification");
        goto fail;
    }

    if (!pa_rtp_sample_spec_valid(&ss)) {
        pa_log("Specified sample type not compatible with RTP");
        goto fail;
    }

    if (ss.channels != cm.channels)
        pa_channel_map_init_auto(&cm, ss.channels, PA_CHANNEL_MAP_AIFF);

    payload = pa_rtp_payload_from_sample_spec(&ss);

    mtu = (uint32_t) pa_frame_align(DEFAULT_MTU, &ss);

    if (pa_modargs_get_value_u32(ma, "mtu", &mtu) < 0 || mtu < 1 || mtu % pa_frame_size(&ss) != 0) {
        pa_log("Invalid MTU.");
        goto fail;
    }

    port = DEFAULT_PORT + ((uint32_t) (rand() % 512) << 1);
    if (pa_modargs_get_value_u32(ma, "port", &port) < 0 || port < 1 || port > 0xFFFF) {
        pa_log("port= expects a numerical argument between 1 and 65535.");
        goto fail;
    }

    if (port & 1)
        pa_log_warn("Port number not even as suggested in RFC3550!");

    if (pa_modargs_get_value_u32(ma, "ttl", &ttl) < 0 || ttl < 1 || ttl > 0xFF) {
        pa_log("ttl= expects a numerical argument between 1 and 255.");
        goto fail;
    }

    src_addr = pa_modargs_get_value(ma, "source_ip", DEFAULT_SOURCE_IP);

    if (inet_pton(AF_INET, src_addr, &src_sa4.sin_addr) > 0) {
        src_sa4.sin_family = af = AF_INET;
        src_sa4.sin_port = htons(0);
        memset(&src_sa4.sin_zero, 0, sizeof(src_sa4.sin_zero));
        src_sap_sa4 = src_sa4;
#ifdef HAVE_IPV6
    } else if (inet_pton(AF_INET6, src_addr, &src_sa6.sin6_addr) > 0) {
        src_sa6.sin6_family = af = AF_INET6;
        src_sa6.sin6_port = htons(0);
        src_sa6.sin6_flowinfo = 0;
        src_sa6.sin6_scope_id = 0;
        src_sap_sa6 = src_sa6;
#endif
    } else {
        pa_log("Invalid source address '%s'", src_addr);
        goto fail;
    }

    dst_addr = pa_modargs_get_value(ma, "destination", NULL);
    if (dst_addr == NULL)
        dst_addr = pa_modargs_get_value(ma, "destination_ip", DEFAULT_DESTINATION_IP);

    if (inet_pton(AF_INET, dst_addr, &dst_sa4.sin_addr) > 0) {
        dst_sa4.sin_family = af = AF_INET;
        dst_sa4.sin_port = htons((uint16_t) port);
        memset(&dst_sa4.sin_zero, 0, sizeof(dst_sa4.sin_zero));
        dst_sap_sa4 = dst_sa4;
        dst_sap_sa4.sin_port = htons(SAP_PORT);
#ifdef HAVE_IPV6
    } else if (inet_pton(AF_INET6, dst_addr, &dst_sa6.sin6_addr) > 0) {
        dst_sa6.sin6_family = af = AF_INET6;
        dst_sa6.sin6_port = htons((uint16_t) port);
        dst_sa6.sin6_flowinfo = 0;
        dst_sa6.sin6_scope_id = 0;
        dst_sap_sa6 = dst_sa6;
        dst_sap_sa6.sin6_port = htons(SAP_PORT);
#endif
    } else {
        pa_log("Invalid destination '%s'", dst_addr);
        goto fail;
    }

    if ((fd = pa_socket_cloexec(af, SOCK_DGRAM, 0)) < 0) {
        pa_log("socket() failed: %s", pa_cstrerror(errno));
        goto fail;
    }

    if (af == AF_INET && bind(fd, (struct sockaddr*) &src_sa4, sizeof(src_sa4)) < 0) {
        pa_log("bind() failed: %s", pa_cstrerror(errno));
        goto fail;
#ifdef HAVE_IPV6
    } else if (af == AF_INET6 && bind(fd, (struct sockaddr*) &src_sa6, sizeof(src_sa6)) < 0) {
        pa_log("bind() failed: %s", pa_cstrerror(errno));
        goto fail;
#endif
    }

    if (af == AF_INET && connect(fd, (struct sockaddr*) &dst_sa4, sizeof(dst_sa4)) < 0) {
        pa_log("connect() failed: %s", pa_cstrerror(errno));
        goto fail;
#ifdef HAVE_IPV6
    } else if (af == AF_INET6 && connect(fd, (struct sockaddr*) &dst_sa6, sizeof(dst_sa6)) < 0) {
        pa_log("connect() failed: %s", pa_cstrerror(errno));
        goto fail;
#endif
    }

    if ((sap_fd = pa_socket_cloexec(af, SOCK_DGRAM, 0)) < 0) {
        pa_log("socket() failed: %s", pa_cstrerror(errno));
        goto fail;
    }

    if (af == AF_INET && bind(sap_fd, (struct sockaddr*) &src_sap_sa4, sizeof(src_sap_sa4)) < 0) {
        pa_log("bind() failed: %s", pa_cstrerror(errno));
        goto fail;
#ifdef HAVE_IPV6
    } else if (af == AF_INET6 && bind(sap_fd, (struct sockaddr*) &src_sap_sa6, sizeof(src_sap_sa6)) < 0) {
        pa_log("bind() failed: %s", pa_cstrerror(errno));
        goto fail;
#endif
    }

    if (af == AF_INET && connect(sap_fd, (struct sockaddr*) &dst_sap_sa4, sizeof(dst_sap_sa4)) < 0) {
        pa_log("connect() failed: %s", pa_cstrerror(errno));
        goto fail;
#ifdef HAVE_IPV6
    } else if (af == AF_INET6 && connect(sap_fd, (struct sockaddr*) &dst_sap_sa6, sizeof(dst_sap_sa6)) < 0) {
        pa_log("connect() failed: %s", pa_cstrerror(errno));
        goto fail;
#endif
    }

    j = loop;
    if (setsockopt(fd, IPPROTO_IP, IP_MULTICAST_LOOP, &j, sizeof(j)) < 0 ||
        setsockopt(sap_fd, IPPROTO_IP, IP_MULTICAST_LOOP, &j, sizeof(j)) < 0) {
        pa_log("IP_MULTICAST_LOOP failed: %s", pa_cstrerror(errno));
        goto fail;
    }

    if (ttl != DEFAULT_TTL) {
        int _ttl = (int) ttl;

        if (setsockopt(fd, IPPROTO_IP, IP_MULTICAST_TTL, &_ttl, sizeof(_ttl)) < 0) {
            pa_log("IP_MULTICAST_TTL failed: %s", pa_cstrerror(errno));
            goto fail;
        }

        if (setsockopt(sap_fd, IPPROTO_IP, IP_MULTICAST_TTL, &_ttl, sizeof(_ttl)) < 0) {
            pa_log("IP_MULTICAST_TTL (sap) failed: %s", pa_cstrerror(errno));
            goto fail;
        }
    }

    /* If the socket queue is full, let's drop packets */
    pa_make_fd_nonblock(fd);
    pa_make_udp_socket_low_delay(fd);

    pa_source_output_new_data_init(&data);
    pa_proplist_sets(data.proplist, PA_PROP_MEDIA_NAME, "RTP Monitor Stream");
    pa_proplist_sets(data.proplist, "rtp.source", src_addr);
    pa_proplist_sets(data.proplist, "rtp.destination", dst_addr);
    pa_proplist_setf(data.proplist, "rtp.mtu", "%lu", (unsigned long) mtu);
    pa_proplist_setf(data.proplist, "rtp.port", "%lu", (unsigned long) port);
    pa_proplist_setf(data.proplist, "rtp.ttl", "%lu", (unsigned long) ttl);
    data.driver = __FILE__;
    data.module = m;
    pa_source_output_new_data_set_source(&data, s, false, true);
    pa_source_output_new_data_set_sample_spec(&data, &ss);
    pa_source_output_new_data_set_channel_map(&data, &cm);
    data.flags |= get_dont_inhibit_auto_suspend_flag(s, inhibit_auto_suspend);

    pa_source_output_new(&o, m->core, &data);
    pa_source_output_new_data_done(&data);

    if (!o) {
        pa_log("failed to create source output.");
        goto fail;
    }

    o->parent.process_msg = source_output_process_msg;
    o->push = source_output_push_cb;
    o->moving = source_output_moving_cb;
    o->kill = source_output_kill_cb;

    pa_log_info("Configured source latency of %llu ms.",
                (unsigned long long) pa_source_output_set_requested_latency(o, pa_bytes_to_usec(mtu, &o->sample_spec)) / PA_USEC_PER_MSEC);

    m->userdata = o->userdata = u = pa_xnew(struct userdata, 1);
    u->module = m;
    u->source_output = o;

    u->memblockq = pa_memblockq_new(
            "module-rtp-send memblockq",
            0,
            MEMBLOCKQ_MAXLENGTH,
            MEMBLOCKQ_MAXLENGTH,
            &ss,
            1,
            0,
            0,
            NULL);

    u->mtu = mtu;

    k = sizeof(sa_dst);
    pa_assert_se((r = getsockname(fd, (struct sockaddr*) &sa_dst, &k)) >= 0);

    n = pa_xstrdup(pa_modargs_get_value(ma, "stream_name", NULL));
    if (n == NULL)
        n = pa_sprintf_malloc("PulseAudio RTP Stream on %s", pa_get_fqdn(hn, sizeof(hn)));

    if (af == AF_INET) {
        p = pa_sdp_build(af,
                     (void*) &((struct sockaddr_in*) &sa_dst)->sin_addr,
                     (void*) &dst_sa4.sin_addr,
                     n, (uint16_t) port, payload, &ss);
#ifdef HAVE_IPV6
    } else {
        p = pa_sdp_build(af,
                     (void*) &((struct sockaddr_in6*) &sa_dst)->sin6_addr,
                     (void*) &dst_sa6.sin6_addr,
                     n, (uint16_t) port, payload, &ss);
#endif
    }

    pa_xfree(n);

    pa_rtp_context_init_send(&u->rtp_context, fd, m->core->cookie, payload, pa_frame_size(&ss));
    pa_sap_context_init_send(&u->sap_context, sap_fd, p);

    pa_log_info("RTP stream initialized with mtu %u on %s:%u from %s ttl=%u, SSRC=0x%08x, payload=%u, initial sequence #%u", mtu, dst_addr, port, src_addr, ttl, u->rtp_context.ssrc, payload, u->rtp_context.sequence);
    pa_log_info("SDP-Data:\n%s\nEOF", p);

    pa_sap_send(&u->sap_context, 0);

    u->sap_event = pa_core_rttime_new(m->core, pa_rtclock_now() + SAP_INTERVAL, sap_event_cb, u);
    u->inhibit_auto_suspend = inhibit_auto_suspend;

    pa_source_output_put(u->source_output);

    pa_modargs_free(ma);

    return 0;

fail:
    if (ma)
        pa_modargs_free(ma);

    if (fd >= 0)
        pa_close(fd);

    if (sap_fd >= 0)
        pa_close(sap_fd);

    return -1;
}
コード例 #7
0
ファイル: pulse.c プロジェクト: Kafay/vlc
/*****************************************************************************
 * Open: open the audio device
 *****************************************************************************/
static int Open ( vlc_object_t *p_this )
{
    aout_instance_t *p_aout = (aout_instance_t *)p_this;
    struct aout_sys_t * p_sys;
    struct pa_sample_spec ss;
    const struct pa_buffer_attr *buffer_attr;
    struct pa_buffer_attr a;
    struct pa_channel_map map;

    /* Allocate structures */
    p_aout->output.p_sys = p_sys = calloc( 1, sizeof( aout_sys_t ) );
    if( p_sys == NULL )
        return VLC_ENOMEM;

    PULSE_DEBUG( "Pulse start initialization");

    ss.channels = aout_FormatNbChannels( &p_aout->output.output ); /* Get the input stream channel count */

    /* Setup the pulse audio stream based on the input stream count */
    switch(ss.channels)
    {
        case 8:
            p_aout->output.output.i_physical_channels
                = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
                | AOUT_CHAN_MIDDLELEFT | AOUT_CHAN_MIDDLERIGHT
                | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT
                | AOUT_CHAN_LFE;
            break;
        case 6:
            p_aout->output.output.i_physical_channels
                = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
                | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT
                | AOUT_CHAN_LFE;
            break;

        case 4:
            p_aout->output.output.i_physical_channels
                = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT
                | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT;
            break;

        case 2:
            p_aout->output.output.i_physical_channels
                = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
            break;

        case 1:
            p_aout->output.output.i_physical_channels = AOUT_CHAN_CENTER;
            break;

        default:
            msg_Err(p_aout,"Invalid number of channels");
        goto fail;
    }

    /* Add a quick command line info message */
    msg_Info(p_aout, "No. of Audio Channels: %d", ss.channels);

    ss.rate = p_aout->output.output.i_rate;
    ss.format = PA_SAMPLE_FLOAT32NE;
    p_aout->output.output.i_format = VLC_CODEC_FL32;

    if (!pa_sample_spec_valid(&ss)) {
        msg_Err(p_aout,"Invalid sample spec");
        goto fail;
    }

    /* Reduce overall latency to 200mS to reduce audible clicks
     * Also pulse minreq and internal buffers are now 20mS which reduces resampling
     */
    a.tlength = pa_bytes_per_second(&ss)/5;
    a.maxlength = a.tlength * 2;
    a.prebuf = a.tlength / 2;
    a.minreq = a.tlength / 10;

    /* Buffer size is 20mS */
    p_sys->buffer_size = a.minreq;

    /* Initialise the speaker map setup above */
    pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_ALSA);

    if (!(p_sys->mainloop = pa_threaded_mainloop_new())) {
        msg_Err(p_aout, "Failed to allocate main loop");
        goto fail;
    }

    if (!(p_sys->context = pa_context_new(pa_threaded_mainloop_get_api(p_sys->mainloop), _( PULSE_CLIENT_NAME )))) {
        msg_Err(p_aout, "Failed to allocate context");
        goto fail;
    }

    pa_context_set_state_callback(p_sys->context, context_state_cb, p_aout);

    PULSE_DEBUG( "Pulse before context connect");

    if (pa_context_connect(p_sys->context, NULL, 0, NULL) < 0) {
        msg_Err(p_aout, "Failed to connect to server: %s", pa_strerror(pa_context_errno(p_sys->context)));
        goto fail;
    }

    PULSE_DEBUG( "Pulse after context connect");

    pa_threaded_mainloop_lock(p_sys->mainloop);

    if (pa_threaded_mainloop_start(p_sys->mainloop) < 0) {
        msg_Err(p_aout, "Failed to start main loop");
        goto unlock_and_fail;
    }

    msg_Dbg(p_aout, "Pulse mainloop started");

    /* Wait until the context is ready */
    pa_threaded_mainloop_wait(p_sys->mainloop);

    if (pa_context_get_state(p_sys->context) != PA_CONTEXT_READY) {
        msg_Err(p_aout, "Failed to connect to server: %s", pa_strerror(pa_context_errno(p_sys->context)));
        goto unlock_and_fail;
    }

    if (!(p_sys->stream = pa_stream_new(p_sys->context, "audio stream", &ss, &map))) {
        msg_Err(p_aout, "Failed to create stream: %s", pa_strerror(pa_context_errno(p_sys->context)));
        goto unlock_and_fail;
    }

    PULSE_DEBUG( "Pulse after new stream");

    pa_stream_set_state_callback(p_sys->stream, stream_state_cb, p_aout);
    pa_stream_set_write_callback(p_sys->stream, stream_request_cb, p_aout);
    pa_stream_set_latency_update_callback(p_sys->stream, stream_latency_update_cb, p_aout);

    if (pa_stream_connect_playback(p_sys->stream, NULL, &a, PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE|PA_STREAM_ADJUST_LATENCY, NULL, NULL) < 0) {
        msg_Err(p_aout, "Failed to connect stream: %s", pa_strerror(pa_context_errno(p_sys->context)));
        goto unlock_and_fail;
    }

     PULSE_DEBUG("Pulse stream connect");

    /* Wait until the stream is ready */
    pa_threaded_mainloop_wait(p_sys->mainloop);

    msg_Dbg(p_aout,"Pulse stream connected");

    if (pa_stream_get_state(p_sys->stream) != PA_STREAM_READY) {
        msg_Err(p_aout, "Failed to connect to server: %s", pa_strerror(pa_context_errno(p_sys->context)));
        goto unlock_and_fail;
    }


    PULSE_DEBUG("Pulse after stream get status");

    pa_threaded_mainloop_unlock(p_sys->mainloop);

    buffer_attr = pa_stream_get_buffer_attr(p_sys->stream);
    p_aout->output.i_nb_samples = buffer_attr->minreq / pa_frame_size(&ss);
    p_aout->output.pf_play = Play;
    aout_VolumeSoftInit(p_aout);
    msg_Dbg(p_aout, "Pulse initialized successfully");
    {
        char cmt[PA_CHANNEL_MAP_SNPRINT_MAX], sst[PA_SAMPLE_SPEC_SNPRINT_MAX];

        msg_Dbg(p_aout, "Buffer metrics: maxlength=%u, tlength=%u, prebuf=%u, minreq=%u", buffer_attr->maxlength, buffer_attr->tlength, buffer_attr->prebuf, buffer_attr->minreq);
        msg_Dbg(p_aout, "Using sample spec '%s', channel map '%s'.",
                pa_sample_spec_snprint(sst, sizeof(sst), pa_stream_get_sample_spec(p_sys->stream)),
                pa_channel_map_snprint(cmt, sizeof(cmt), pa_stream_get_channel_map(p_sys->stream)));

            msg_Dbg(p_aout, "Connected to device %s (%u, %ssuspended).",
                        pa_stream_get_device_name(p_sys->stream),
                        pa_stream_get_device_index(p_sys->stream),
                        pa_stream_is_suspended(p_sys->stream) ? "" : "not ");
    }

    return VLC_SUCCESS;

unlock_and_fail:
    msg_Dbg(p_aout, "Pulse initialization unlock and fail");

    if (p_sys->mainloop)
        pa_threaded_mainloop_unlock(p_sys->mainloop);
fail:
    msg_Err(p_aout, "Pulse initialization failed");
    uninit(p_aout);
    return VLC_EGENERIC;
}
コード例 #8
0
ファイル: source.c プロジェクト: felfert/pulseaudio
/* Called from main context */
pa_source* pa_source_new(
        pa_core *core,
        pa_source_new_data *data,
        pa_source_flags_t flags) {

    pa_source *s;
    const char *name;
    char st[PA_SAMPLE_SPEC_SNPRINT_MAX], cm[PA_CHANNEL_MAP_SNPRINT_MAX];
    char *pt;

    pa_assert(core);
    pa_assert(data);
    pa_assert(data->name);
    pa_assert_ctl_context();

    s = pa_msgobject_new(pa_source);

    if (!(name = pa_namereg_register(core, data->name, PA_NAMEREG_SOURCE, s, data->namereg_fail))) {
        pa_log_debug("Failed to register name %s.", data->name);
        pa_xfree(s);
        return NULL;
    }

    pa_source_new_data_set_name(data, name);

    if (pa_hook_fire(&core->hooks[PA_CORE_HOOK_SOURCE_NEW], data) < 0) {
        pa_xfree(s);
        pa_namereg_unregister(core, name);
        return NULL;
    }

    /* FIXME, need to free s here on failure */

    pa_return_null_if_fail(!data->driver || pa_utf8_valid(data->driver));
    pa_return_null_if_fail(data->name && pa_utf8_valid(data->name) && data->name[0]);

    pa_return_null_if_fail(data->sample_spec_is_set && pa_sample_spec_valid(&data->sample_spec));

    if (!data->channel_map_is_set)
        pa_return_null_if_fail(pa_channel_map_init_auto(&data->channel_map, data->sample_spec.channels, PA_CHANNEL_MAP_DEFAULT));

    pa_return_null_if_fail(pa_channel_map_valid(&data->channel_map));
    pa_return_null_if_fail(data->channel_map.channels == data->sample_spec.channels);

    if (!data->volume_is_set)
        pa_cvolume_reset(&data->volume, data->sample_spec.channels);

    pa_return_null_if_fail(pa_cvolume_valid(&data->volume));
    pa_return_null_if_fail(pa_cvolume_compatible(&data->volume, &data->sample_spec));

    if (!data->muted_is_set)
        data->muted = FALSE;

    if (data->card)
        pa_proplist_update(data->proplist, PA_UPDATE_MERGE, data->card->proplist);

    pa_device_init_description(data->proplist);
    pa_device_init_icon(data->proplist, FALSE);
    pa_device_init_intended_roles(data->proplist);

    if (pa_hook_fire(&core->hooks[PA_CORE_HOOK_SOURCE_FIXATE], data) < 0) {
        pa_xfree(s);
        pa_namereg_unregister(core, name);
        return NULL;
    }

    s->parent.parent.free = source_free;
    s->parent.process_msg = pa_source_process_msg;

    s->core = core;
    s->state = PA_SOURCE_INIT;
    s->flags = flags;
    s->priority = 0;
    s->suspend_cause = 0;
    s->name = pa_xstrdup(name);
    s->proplist = pa_proplist_copy(data->proplist);
    s->driver = pa_xstrdup(pa_path_get_filename(data->driver));
    s->module = data->module;
    s->card = data->card;

    s->priority = pa_device_init_priority(s->proplist);

    s->sample_spec = data->sample_spec;
    s->channel_map = data->channel_map;

    s->outputs = pa_idxset_new(NULL, NULL);
    s->n_corked = 0;
    s->monitor_of = NULL;
    s->output_from_master = NULL;

    s->volume = data->volume;
    pa_cvolume_reset(&s->soft_volume, s->sample_spec.channels);
    s->base_volume = PA_VOLUME_NORM;
    s->n_volume_steps = PA_VOLUME_NORM+1;
    s->muted = data->muted;
    s->refresh_volume = s->refresh_muted = FALSE;

    reset_callbacks(s);
    s->userdata = NULL;

    s->asyncmsgq = NULL;

    /* As a minor optimization we just steal the list instead of
     * copying it here */
    s->ports = data->ports;
    data->ports = NULL;

    s->active_port = NULL;
    s->save_port = FALSE;

    if (data->active_port && s->ports)
        if ((s->active_port = pa_hashmap_get(s->ports, data->active_port)))
            s->save_port = data->save_port;

    if (!s->active_port && s->ports) {
        void *state;
        pa_device_port *p;

        PA_HASHMAP_FOREACH(p, s->ports, state)
            if (!s->active_port || p->priority > s->active_port->priority)
                s->active_port = p;
    }
コード例 #9
0
static int init(int rate_hz, int channels, int format, int flags) {
    struct pa_sample_spec ss;
    struct pa_channel_map map;
    const struct format_map_s *fmt_map;
    char *devarg = NULL;
    char *host = NULL;
    char *sink = NULL;
    char *version = pa_get_library_version();

    if (ao_subdevice) {
        devarg = strdup(ao_subdevice);
        sink = strchr(devarg, ':');
        if (sink) *sink++ = 0;
        if (devarg[0]) host = devarg;
    }

    broken_pause = 0;
    // not sure which versions are affected, assume 0.9.11* to 0.9.14*
    // known bad: 0.9.14, 0.9.13
    // known good: 0.9.9, 0.9.10, 0.9.15
    // to test: pause, wait ca. 5 seconds framestep and see if MPlayer hangs somewhen
    if (strncmp(version, "0.9.1", 5) == 0 && version[5] >= '1' && version[5] <= '4') {
        mp_msg(MSGT_AO, MSGL_WARN, "[pulse] working around probably broken pause functionality,\n"
                                   "        see http://www.pulseaudio.org/ticket/440\n");
        broken_pause = 1;
    }

    ss.channels = channels;
    ss.rate = rate_hz;

    ao_data.samplerate = rate_hz;
    ao_data.channels = channels;

    fmt_map = format_maps;
    while (fmt_map->mp_format != format) {
        if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) {
            mp_msg(MSGT_AO, MSGL_V, "AO: [pulse] Unsupported format, using default\n");
            fmt_map = format_maps;
            break;
        }
        fmt_map++;
    }
    ao_data.format = fmt_map->mp_format;
    ss.format = fmt_map->pa_format;

    if (!pa_sample_spec_valid(&ss)) {
        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Invalid sample spec\n");
        goto fail;
    }

    pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_ALSA);
    ao_data.bps = pa_bytes_per_second(&ss);

    pa_cvolume_reset(&volume, ss.channels);

    if (!(mainloop = pa_threaded_mainloop_new())) {
        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate main loop\n");
        goto fail;
    }

    if (!(context = pa_context_new(pa_threaded_mainloop_get_api(mainloop), PULSE_CLIENT_NAME))) {
        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate context\n");
        goto fail;
    }

    pa_context_set_state_callback(context, context_state_cb, NULL);

    if (pa_context_connect(context, host, 0, NULL) < 0)
        goto fail;

    pa_threaded_mainloop_lock(mainloop);

    if (pa_threaded_mainloop_start(mainloop) < 0)
        goto unlock_and_fail;

    /* Wait until the context is ready */
    pa_threaded_mainloop_wait(mainloop);

    if (pa_context_get_state(context) != PA_CONTEXT_READY)
        goto unlock_and_fail;

    if (!(stream = pa_stream_new(context, "audio stream", &ss, &map)))
        goto unlock_and_fail;

    pa_stream_set_state_callback(stream, stream_state_cb, NULL);
    pa_stream_set_write_callback(stream, stream_request_cb, NULL);
    pa_stream_set_latency_update_callback(stream, stream_latency_update_cb, NULL);

    if (pa_stream_connect_playback(stream, sink, NULL, PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE, &volume, NULL) < 0)
        goto unlock_and_fail;

    /* Wait until the stream is ready */
    pa_threaded_mainloop_wait(mainloop);

    if (pa_stream_get_state(stream) != PA_STREAM_READY)
        goto unlock_and_fail;

    pa_threaded_mainloop_unlock(mainloop);

    free(devarg);
    return 1;

unlock_and_fail:

    if (mainloop)
        pa_threaded_mainloop_unlock(mainloop);

fail:
    if (context)
        GENERIC_ERR_MSG(context, "Init failed");
    free(devarg);
    uninit(1);
    return 0;
}
コード例 #10
0
ファイル: pulseaudio.c プロジェクト: 24BitGames/LoomSDK
static ALCboolean pulse_reset_playback(ALCdevice *device)
{
    pulse_data *data = device->ExtraData;
    pa_stream_flags_t flags = 0;
    pa_channel_map chanmap;

    pa_threaded_mainloop_lock(data->loop);

    if(data->stream)
    {
#if PA_CHECK_VERSION(0,9,15)
        if(pa_stream_set_buffer_attr_callback)
            pa_stream_set_buffer_attr_callback(data->stream, NULL, NULL);
#endif
        pa_stream_disconnect(data->stream);
        pa_stream_unref(data->stream);
        data->stream = NULL;
    }

    if(!(device->Flags&DEVICE_CHANNELS_REQUEST))
    {
        pa_operation *o;
        o = pa_context_get_sink_info_by_name(data->context, data->device_name, sink_info_callback, device);
        WAIT_FOR_OPERATION(o, data->loop);
    }
    if(!(device->Flags&DEVICE_FREQUENCY_REQUEST))
        flags |= PA_STREAM_FIX_RATE;

    flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE;
    flags |= PA_STREAM_ADJUST_LATENCY;
    flags |= PA_STREAM_START_CORKED;
    flags |= PA_STREAM_DONT_MOVE;

    switch(device->FmtType)
    {
        case DevFmtByte:
            device->FmtType = DevFmtUByte;
            /* fall-through */
        case DevFmtUByte:
            data->spec.format = PA_SAMPLE_U8;
            break;
        case DevFmtUShort:
            device->FmtType = DevFmtShort;
            /* fall-through */
        case DevFmtShort:
            data->spec.format = PA_SAMPLE_S16NE;
            break;
        case DevFmtUInt:
            device->FmtType = DevFmtInt;
            /* fall-through */
        case DevFmtInt:
            data->spec.format = PA_SAMPLE_S32NE;
            break;
        case DevFmtFloat:
            data->spec.format = PA_SAMPLE_FLOAT32NE;
            break;
    }
    data->spec.rate = device->Frequency;
    data->spec.channels = ChannelsFromDevFmt(device->FmtChans);

    if(pa_sample_spec_valid(&data->spec) == 0)
    {
        ERR("Invalid sample format\n");
        pa_threaded_mainloop_unlock(data->loop);
        return ALC_FALSE;
    }

    if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
    {
        ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
        pa_threaded_mainloop_unlock(data->loop);
        return ALC_FALSE;
    }
    SetDefaultWFXChannelOrder(device);

    data->attr.fragsize = -1;
    data->attr.prebuf = 0;
    data->attr.minreq = device->UpdateSize * pa_frame_size(&data->spec);
    data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
    data->attr.maxlength = -1;

    data->stream = connect_playback_stream(data->device_name, data->loop,
                                           data->context, flags, &data->attr,
                                           &data->spec, &chanmap);
    if(!data->stream)
    {
        pa_threaded_mainloop_unlock(data->loop);
        return ALC_FALSE;
    }
    pa_stream_set_state_callback(data->stream, stream_state_callback2, device);

    data->spec = *(pa_stream_get_sample_spec(data->stream));
    if(device->Frequency != data->spec.rate)
    {
        pa_operation *o;

        /* Server updated our playback rate, so modify the buffer attribs
         * accordingly. */
        data->attr.minreq = (ALuint64)device->UpdateSize * data->spec.rate /
                            device->Frequency * pa_frame_size(&data->spec);
        data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
        data->attr.prebuf = 0;

        o = pa_stream_set_buffer_attr(data->stream, &data->attr,
                                      stream_success_callback, device);
        WAIT_FOR_OPERATION(o, data->loop);

        device->Frequency = data->spec.rate;
    }

#if PA_CHECK_VERSION(0,9,15)
    if(pa_stream_set_buffer_attr_callback)
        pa_stream_set_buffer_attr_callback(data->stream, stream_buffer_attr_callback, device);
#endif
    stream_buffer_attr_callback(data->stream, device);

    device->NumUpdates = device->UpdateSize*device->NumUpdates /
                         (data->attr.minreq/pa_frame_size(&data->spec));
    device->NumUpdates = maxu(device->NumUpdates, 2);
    device->UpdateSize = data->attr.minreq / pa_frame_size(&data->spec);

    pa_threaded_mainloop_unlock(data->loop);
    return ALC_TRUE;
}
コード例 #11
0
ファイル: pulseaudio.c プロジェクト: 24BitGames/LoomSDK
static ALCenum pulse_open_capture(ALCdevice *device, const ALCchar *device_name)
{
    const char *pulse_name = NULL;
    pa_stream_flags_t flags = 0;
    pa_channel_map chanmap;
    pulse_data *data;
    pa_operation *o;
    ALuint samples;

    if(device_name)
    {
        ALuint i;

        if(!allCaptureDevNameMap)
            probe_devices(AL_TRUE);

        for(i = 0;i < numCaptureDevNames;i++)
        {
            if(strcmp(device_name, allCaptureDevNameMap[i].name) == 0)
            {
                pulse_name = allCaptureDevNameMap[i].device_name;
                break;
            }
        }
        if(i == numCaptureDevNames)
            return ALC_INVALID_VALUE;
    }

    if(pulse_open(device) == ALC_FALSE)
        return ALC_INVALID_VALUE;

    data = device->ExtraData;
    pa_threaded_mainloop_lock(data->loop);

    data->spec.rate = device->Frequency;
    data->spec.channels = ChannelsFromDevFmt(device->FmtChans);

    switch(device->FmtType)
    {
        case DevFmtUByte:
            data->spec.format = PA_SAMPLE_U8;
            break;
        case DevFmtShort:
            data->spec.format = PA_SAMPLE_S16NE;
            break;
        case DevFmtInt:
            data->spec.format = PA_SAMPLE_S32NE;
            break;
        case DevFmtFloat:
            data->spec.format = PA_SAMPLE_FLOAT32NE;
            break;
        case DevFmtByte:
        case DevFmtUShort:
        case DevFmtUInt:
            ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
            pa_threaded_mainloop_unlock(data->loop);
            goto fail;
    }

    if(pa_sample_spec_valid(&data->spec) == 0)
    {
        ERR("Invalid sample format\n");
        pa_threaded_mainloop_unlock(data->loop);
        goto fail;
    }

    if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
    {
        ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
        pa_threaded_mainloop_unlock(data->loop);
        goto fail;
    }

    samples = device->UpdateSize * device->NumUpdates;
    samples = maxu(samples, 100 * device->Frequency / 1000);

    data->attr.minreq = -1;
    data->attr.prebuf = -1;
    data->attr.maxlength = samples * pa_frame_size(&data->spec);
    data->attr.tlength = -1;
    data->attr.fragsize = minu(samples, 50*device->Frequency/1000) *
                          pa_frame_size(&data->spec);

    flags |= PA_STREAM_DONT_MOVE;
    flags |= PA_STREAM_START_CORKED|PA_STREAM_ADJUST_LATENCY;
    data->stream = connect_record_stream(pulse_name, data->loop, data->context,
                                         flags, &data->attr, &data->spec,
                                         &chanmap);
    if(!data->stream)
    {
        pa_threaded_mainloop_unlock(data->loop);
        goto fail;
    }
    pa_stream_set_state_callback(data->stream, stream_state_callback2, device);

    data->device_name = strdup(pa_stream_get_device_name(data->stream));
    o = pa_context_get_source_info_by_name(data->context, data->device_name,
                                           source_name_callback, device);
    WAIT_FOR_OPERATION(o, data->loop);

    pa_threaded_mainloop_unlock(data->loop);
    return ALC_NO_ERROR;

fail:
    pulse_close(device);
    return ALC_INVALID_VALUE;
}
コード例 #12
0
ファイル: cubeb_pulse.c プロジェクト: rillian/cubeb
int
cubeb_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name,
                  cubeb_stream_params stream_params, unsigned int latency,
                  cubeb_data_callback data_callback, cubeb_state_callback state_callback,
                  void * user_ptr)
{
  pa_sample_spec ss;
  cubeb_stream * stm;
  pa_operation * o;
  pa_buffer_attr battr;
  pa_channel_map map;

  if (stream_params.rate < 1 || stream_params.rate > 192000 ||
      stream_params.channels < 1 || stream_params.channels > 32 ||
      latency < 1 || latency > 2000) {
    return CUBEB_ERROR_INVALID_FORMAT;
  }

  switch (stream_params.format) {
  case CUBEB_SAMPLE_S16LE:
    ss.format = PA_SAMPLE_S16LE;
    break;
  case CUBEB_SAMPLE_S16BE:
    ss.format = PA_SAMPLE_S16BE;
    break;
  case CUBEB_SAMPLE_FLOAT32LE:
    ss.format = PA_SAMPLE_FLOAT32LE;
    break;
  case CUBEB_SAMPLE_FLOAT32BE:
    ss.format = PA_SAMPLE_FLOAT32BE;
    break;
  default:
    return CUBEB_ERROR_INVALID_FORMAT;
  }

  ss.rate = stream_params.rate;
  ss.channels = stream_params.channels;

  /* XXX check that this does the right thing for Vorbis and WaveEx */
  pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_DEFAULT);

  stm = calloc(1, sizeof(*stm));
  assert(stm);

  stm->context = context;
  assert(stm->context);

  stm->data_callback = data_callback;
  stm->state_callback = state_callback;
  stm->user_ptr = user_ptr;

  stm->sample_spec = ss;

  battr.maxlength = -1;
  battr.tlength = pa_usec_to_bytes(latency * 1000, &stm->sample_spec);
  battr.prebuf = -1;
  battr.minreq = battr.tlength / 2;
  battr.fragsize = -1;

  pa_threaded_mainloop_lock(stm->context->mainloop);
  stm->stream = pa_stream_new(stm->context->context, stream_name, &ss, &map);
  pa_stream_set_state_callback(stm->stream, stream_state_callback, stm->context->mainloop);
  pa_stream_set_write_callback(stm->stream, stream_request_callback, stm);
  pa_stream_connect_playback(stm->stream, NULL, &battr,
                             PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_INTERPOLATE_TIMING |
                             PA_STREAM_START_CORKED,
                             NULL, NULL);
  pa_threaded_mainloop_unlock(stm->context->mainloop);

  stream_state_wait(stm, PA_STREAM_READY);

  /* force a timing update now, otherwise timing info does not become valid
     until some point after initialization has completed. */
  pa_threaded_mainloop_lock(stm->context->mainloop);
  o = pa_stream_update_timing_info(stm->stream, stream_success_callback, stm->context->mainloop);
  operation_wait(stm->context, o);
  pa_operation_unref(o);
  pa_threaded_mainloop_unlock(stm->context->mainloop);

  *stream = stm;

  return CUBEB_OK;
}
コード例 #13
0
void PulseAudioPlayer::OpenStream()
{
	if (open) CloseStream();

	// Initialise a mainloop
	//printf("Initialising threaded main loop\n");
	mainloop = pa_threaded_mainloop_new();
	if (!mainloop) {
		throw agi::AudioPlayerOpenError("Failed to initialise PulseAudio threaded mainloop object", 0);
	}
	//printf("Starting main loop\n");
	pa_threaded_mainloop_start(mainloop);

	// Create context
	//printf("Creating context\n");
	context = pa_context_new(pa_threaded_mainloop_get_api(mainloop), "Aegisub");
	if (!context) {
		pa_threaded_mainloop_free(mainloop);
		throw agi::AudioPlayerOpenError("Failed to create PulseAudio context", 0);
	}
	pa_context_set_state_callback(context, (pa_context_notify_cb_t)pa_context_notify, this);

	// Connect the context
	//printf("Connecting context\n");
	pa_context_connect(context, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL);
	// Wait for connection
	while (true) {
		context_notify.Wait();
		if (cstate == PA_CONTEXT_READY) {
			break;
		} else if (cstate == PA_CONTEXT_FAILED) {
			// eww
			paerror = pa_context_errno(context);
			pa_context_unref(context);
			pa_threaded_mainloop_stop(mainloop);
			pa_threaded_mainloop_free(mainloop);
			throw agi::AudioPlayerOpenError(std::string("PulseAudio reported error: ") + pa_strerror(paerror), 0);
		}
		// otherwise loop once more
	}
	//printf("Context connected\n");

	// Set up stream
	bpf = provider->GetChannels() * provider->GetBytesPerSample();
	pa_sample_spec ss;
	ss.format = PA_SAMPLE_S16LE; // FIXME
	ss.rate = provider->GetSampleRate();
	ss.channels = provider->GetChannels();
	pa_channel_map map;
	pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_DEFAULT);
	//printf("Creating stream\n");
	stream = pa_stream_new(context, "Sound", &ss, &map);
	if (!stream) {
		// argh!
		pa_context_disconnect(context);
		pa_context_unref(context);
		pa_threaded_mainloop_stop(mainloop);
		pa_threaded_mainloop_free(mainloop);
		throw agi::AudioPlayerOpenError("PulseAudio could not create stream", 0);
	}
	pa_stream_set_state_callback(stream, (pa_stream_notify_cb_t)pa_stream_notify, this);
	pa_stream_set_write_callback(stream, (pa_stream_request_cb_t)pa_stream_write, this);

	// Connect stream
	//printf("Connecting playback stream\n");
	paerror = pa_stream_connect_playback(stream, NULL, NULL, (pa_stream_flags_t)(PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_NOT_MONOTONOUS|PA_STREAM_AUTO_TIMING_UPDATE), NULL, NULL);
	if (paerror) {
		printf("PulseAudio reported error: %s (%d)\n", pa_strerror(paerror), paerror);
		throw agi::AudioPlayerOpenError(std::string("PulseAudio reported error: ") + pa_strerror(paerror), 0);
	}
	while (true) {
		stream_notify.Wait();
		if (sstate == PA_STREAM_READY) {
			break;
		} else if (sstate == PA_STREAM_FAILED) {
			paerror = pa_context_errno(context);
			printf("PulseAudio player: Stream connection failed: %s (%d)\n", pa_strerror(paerror), paerror);
			throw agi::AudioPlayerOpenError("PulseAudio player: Something went wrong connecting the stream", 0);
		}
	}
	//printf("Connected playback stream, now playing\n\n");

	// Hopefully this marks success
	//printf("Finished opening PulseAudio\n\n");
	open = true;
}
コード例 #14
0
ファイル: oss.cpp プロジェクト: dreiss/M1-Android
INT16 m1sdr_Init(int sample_rate)
{	
	int format, stereo, rate, fsize, err, state;
	unsigned int nfreq, periodtime;
	snd_pcm_hw_params_t *hwparams;
	#ifdef USE_SDL
	SDL_AudioSpec aspec;
	#endif
	pa_channel_map chanmap;
	pa_buffer_attr my_pa_attr;

	hw_present = 0;

	m1sdr_Callback = NULL;

	nDSoundSegLen = sample_rate / 60;

	switch (lnxdrv_apimode)
	{
	case 0: // SDL
		#ifdef USE_SDL
		SDL_InitSubSystem(SDL_INIT_AUDIO);

	 	m1sdr_SetSamplesPerTick(sample_rate/60);

		playbuf = 0;
		writebuf = 1;

		aspec.freq = sample_rate;
		aspec.format = AUDIO_S16SYS;	// keep endian independant 
		aspec.channels = 2;
		aspec.samples = 512;		// has to be a power of 2, and we want it smaller than our buffer size
		aspec.callback = sdl_callback;
		aspec.userdata = 0;

		if (SDL_OpenAudio(&aspec, NULL) < 0)
		{
			printf("ERROR: can't open SDL audio\n");
			return 0;
		}

		// make sure we don't start yet
		SDL_PauseAudio(1);
		#endif
		break;
				
	case 1:	// ALSA
		// Try to open audio device
		if ((err = snd_pcm_open(&pHandle, "default", SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
			fprintf(stderr, "ALSA: Could not open soundcard (%s)\n", snd_strerror(err));
			hw_present = 0;
			return 0;
		}

		if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0) {
			fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
				 snd_strerror(err));
			return 0;
		}

		// Init hwparams with full configuration space
		if ((err = snd_pcm_hw_params_any(pHandle, hwparams)) < 0) {
			fprintf(stderr, "ALSA: couldn't set hw params (%s)\n", snd_strerror(err));
			hw_present = 0;
			return 0;
		}

		// Set access type
		if ((err = snd_pcm_hw_params_set_access(pHandle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
			fprintf(stderr, "ALSA: can't set access (%s)\n", snd_strerror(err));
			return 0;
		}

		// Set sample format
		if ((err = snd_pcm_hw_params_set_format(pHandle, hwparams, SND_PCM_FORMAT_S16)) < 0) {
			fprintf(stderr, "ALSA: can't set format (%s)\n", snd_strerror(err));
			return 0;
		}

		// Set sample rate (nearest possible)
		nfreq = sample_rate;
		if ((err = snd_pcm_hw_params_set_rate_near(pHandle, hwparams, &nfreq, 0)) < 0) {
			fprintf(stderr, "ALSA: can't set sample rate (%s)\n", snd_strerror(err));
			return 0;
		}

		// Set number of channels
		if ((err = snd_pcm_hw_params_set_channels(pHandle, hwparams, 2)) < 0) {
			fprintf(stderr, "ALSA: can't set stereo (%s)\n", snd_strerror(err));
			return 0;
		}

		// Set period time (nearest possible)
		periodtime = 16;
		if ((err = snd_pcm_hw_params_set_period_time_near(pHandle, hwparams, &periodtime, 0)) < 0) {
			fprintf(stderr, "ALSA: can't set period time (%s)\n", snd_strerror(err));
			return 0;
		}

		// Apply HW parameter settings to PCM device and prepare device
		if ((err = snd_pcm_hw_params(pHandle, hwparams)) < 0) {
			fprintf(stderr, "ALSA: unable to install hw_params (%s)\n", snd_strerror(err));
			snd_pcm_hw_params_free(hwparams);
			return 0;
		}

		snd_pcm_hw_params_free(hwparams);

		if ((err = snd_pcm_prepare(pHandle)) < 0) {
			fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror(err));
			return 0;
		}
		break;	

	case 2:	// OSS
		audiofd = open("/dev/dsp", O_WRONLY, 0);
		if (audiofd == -1)
		{
			audiofd = open("/dev/dsp1", O_WRONLY, 0);

			if (audiofd == -1)
			{
				perror("/dev/dsp1");
				return(0);
			}
		}

		// reset things
		ioctl(audiofd, SNDCTL_DSP_RESET, 0);

		is_broken_driver = 0;
		num_frags = NUM_FRAGS_NORMAL;

		// set the buffer size we want
		fsize = OSS_FRAGMENT;
		if (ioctl(audiofd, SNDCTL_DSP_SETFRAGMENT, &fsize) == - 1)
		{
			perror("SNDCTL_DSP_SETFRAGMENT");
			return(0);
		}

		// set 16-bit output
		format = AFMT_S16_NE;	// 16 bit signed "native"-endian
		if (ioctl(audiofd, SNDCTL_DSP_SETFMT, &format) == - 1)
		{
			perror("SNDCTL_DSP_SETFMT");
			return(0);
		}

		// now set stereo
		stereo = 1;
		if (ioctl(audiofd, SNDCTL_DSP_STEREO, &stereo) == - 1)
		{
			perror("SNDCTL_DSP_STEREO");
			return(0);
		}

		// and the sample rate
		rate = sample_rate;
		if (ioctl(audiofd, SNDCTL_DSP_SPEED, &rate) == - 1)
		{
			perror("SNDCTL_DSP_SPEED");
			return(0);
		}

		// and make sure that did what we wanted
		ioctl(audiofd, SNDCTL_DSP_GETBLKSIZE, &fsize);
		break;

	case 3: // PulseAudio
		sample_spec.format = PA_SAMPLE_S16NE;
		sample_spec.rate = sample_rate;
		sample_spec.channels = 2;

		my_pa_context = NULL;
		my_pa_stream = NULL;
		my_pa_mainloop = NULL;
		my_pa_mainloop_api = NULL;

		#if !PULSE_USE_SIMPLE
		// get default channel mapping
		pa_channel_map_init_auto(&chanmap, sample_spec.channels, PA_CHANNEL_MAP_WAVEEX);


		if (!(my_pa_mainloop = pa_mainloop_new()))
		{
			fprintf(stderr, "pa_mainloop_new() failed\n");
			return 0;
		}

		my_pa_mainloop_api = pa_mainloop_get_api(my_pa_mainloop);

/*		if (pa_signal_init(my_pa_mainloop_api) != 0)
		{
			fprintf(stderr, "pa_signal_init() failed\n");
			return 0;
		}*/

		/* Create a new connection context */
		if (!(my_pa_context = pa_context_new(my_pa_mainloop_api, "Audio Overload"))) 
		{
			fprintf(stderr, "pa_context_new() failed\n");
			return 0;
		}

		/* set the context state CB */
//		pa_context_set_state_callback(my_pa_context, context_state_callback, NULL);

		/* Connect the context */
		if (pa_context_connect(my_pa_context, NULL, (pa_context_flags_t)0, NULL) < 0)
		{
			fprintf(stderr, "pa_context_connect() failed: %s", pa_strerror(pa_context_errno(my_pa_context)));
			return 0;
		}
		
		do
		{
			pa_mainloop_iterate(my_pa_mainloop, 1, NULL);
			state = pa_context_get_state(my_pa_context);
			if (!PA_CONTEXT_IS_GOOD((pa_context_state_t)state))
			{
				printf("PA CONTEXT NOT GOOD\n");
				hw_present = 0;
				return 0;
			}
		} while (state != PA_CONTEXT_READY);

		if (!(my_pa_stream = pa_stream_new(my_pa_context, "Audio Overload", &sample_spec, &chanmap))) 
		{
			fprintf(stderr, "pa_stream_new() failed: %s\n", pa_strerror(pa_context_errno(my_pa_context)));
			return 0;
		}

		memset(&my_pa_attr, 0, sizeof(my_pa_attr));
		my_pa_attr.tlength = nDSoundSegLen * 4 * 4;
		my_pa_attr.prebuf = -1;
		my_pa_attr.maxlength = -1;
		my_pa_attr.minreq = nDSoundSegLen * 4 * 2;

		if ((err = pa_stream_connect_playback(my_pa_stream, NULL, &my_pa_attr, PA_STREAM_ADJUST_LATENCY, NULL, NULL)) < 0)
		{
			fprintf(stderr, "pa_stream_connect_playback() failed: %s\n", pa_strerror(pa_context_errno(my_pa_context)));
			return 0;
		}

		do
		{
			pa_mainloop_iterate(my_pa_mainloop, 1, NULL);
			state = pa_stream_get_state(my_pa_stream);
			if (!PA_STREAM_IS_GOOD((pa_stream_state_t)state))
			{
				printf("PA STREAM NOT GOOD\n");
				hw_present = 0;
				return 0;
			}
		} while (state != PA_STREAM_READY);

//		printf("PulseAudio setup OK so far, len %d\n", nDSoundSegLen*4);
		#else
		my_simple = NULL;
		#endif
		break;
	}

	hw_present = 1;

	return (1);
}
コード例 #15
0
ファイル: mime-type.c プロジェクト: Distrotech/pulseaudio
void pa_sample_spec_mimefy(pa_sample_spec *ss, pa_channel_map *cm) {

    pa_assert(pa_channel_map_compatible(cm, ss));

    /* Turns the sample type passed in into the next 'better' one that
     * can be encoded for HTTP. If there is no 'better' one we pick
     * the 'best' one that is 'worse'. */

    if (ss->channels > 2)
        ss->channels = 2;

    if (ss->rate > 44100)
        ss->rate = 48000;
    else if (ss->rate > 32000)
        ss->rate = 44100;
    else if (ss->rate > 24000)
        ss->rate = 32000;
    else if (ss->rate > 22050)
        ss->rate = 24000;
    else if (ss->rate > 16000)
        ss->rate = 22050;
    else if (ss->rate > 11025)
        ss->rate = 16000;
    else if (ss->rate > 8000)
        ss->rate = 11025;
    else
        ss->rate = 8000;

    switch (ss->format) {
        case PA_SAMPLE_S24BE:
        case PA_SAMPLE_S24LE:
        case PA_SAMPLE_S24_32LE:
        case PA_SAMPLE_S24_32BE:
        case PA_SAMPLE_S32LE:
        case PA_SAMPLE_S32BE:
        case PA_SAMPLE_FLOAT32LE:
        case PA_SAMPLE_FLOAT32BE:
            ss->format = PA_SAMPLE_S24BE;
            break;

        case PA_SAMPLE_S16BE:
        case PA_SAMPLE_S16LE:
            ss->format = PA_SAMPLE_S16BE;
            break;

        case PA_SAMPLE_ULAW:
        case PA_SAMPLE_ALAW:

            if (ss->rate == 8000 && ss->channels == 1)
                ss->format = PA_SAMPLE_ULAW;
            else
                ss->format = PA_SAMPLE_S16BE;
            break;

        case PA_SAMPLE_U8:
            ss->format = PA_SAMPLE_U8;
            break;

        case PA_SAMPLE_MAX:
        case PA_SAMPLE_INVALID:
            pa_assert_not_reached();
    }

    pa_channel_map_init_auto(cm, ss->channels, PA_CHANNEL_MAP_DEFAULT);

    pa_assert(pa_sample_spec_is_mime(ss, cm));
}
コード例 #16
0
ファイル: ao_pulse.c プロジェクト: Newbleeto/mplayer2
static int init(struct ao *ao, char *params)
{
    struct pa_sample_spec ss;
    struct pa_channel_map map;
    char *devarg = NULL;
    char *host = NULL;
    char *sink = NULL;
    const char *version = pa_get_library_version();

    struct priv *priv = talloc_zero(ao, struct priv);
    ao->priv = priv;

    if (params) {
        devarg = strdup(params);
        sink = strchr(devarg, ':');
        if (sink)
            *sink++ = 0;
        if (devarg[0])
            host = devarg;
    }

    priv->broken_pause = false;
    /* not sure which versions are affected, assume 0.9.11* to 0.9.14*
     * known bad: 0.9.14, 0.9.13
     * known good: 0.9.9, 0.9.10, 0.9.15
     * To test: pause, wait ca. 5 seconds, framestep and see if MPlayer
     * hangs somewhen. */
    if (strncmp(version, "0.9.1", 5) == 0 && version[5] >= '1'
        && version[5] <= '4') {
        mp_msg(MSGT_AO, MSGL_WARN,
               "[pulse] working around probably broken pause functionality,\n"
               "        see http://www.pulseaudio.org/ticket/440\n");
        priv->broken_pause = true;
    }

    ss.channels = ao->channels;
    ss.rate = ao->samplerate;

    const struct format_map *fmt_map = format_maps;
    while (fmt_map->mp_format != ao->format) {
        if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) {
            mp_msg(MSGT_AO, MSGL_V,
                   "AO: [pulse] Unsupported format, using default\n");
            fmt_map = format_maps;
            break;
        }
        fmt_map++;
    }
    ao->format = fmt_map->mp_format;
    ss.format = fmt_map->pa_format;

    if (!pa_sample_spec_valid(&ss)) {
        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Invalid sample spec\n");
        goto fail;
    }

    pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_ALSA);
    ao->bps = pa_bytes_per_second(&ss);

    if (!(priv->mainloop = pa_threaded_mainloop_new())) {
        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate main loop\n");
        goto fail;
    }

    if (!(priv->context = pa_context_new(pa_threaded_mainloop_get_api(
                                 priv->mainloop), PULSE_CLIENT_NAME))) {
        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate context\n");
        goto fail;
    }

    pa_context_set_state_callback(priv->context, context_state_cb, ao);

    if (pa_context_connect(priv->context, host, 0, NULL) < 0)
        goto fail;

    pa_threaded_mainloop_lock(priv->mainloop);

    if (pa_threaded_mainloop_start(priv->mainloop) < 0)
        goto unlock_and_fail;

    /* Wait until the context is ready */
    pa_threaded_mainloop_wait(priv->mainloop);

    if (pa_context_get_state(priv->context) != PA_CONTEXT_READY)
        goto unlock_and_fail;

    if (!(priv->stream = pa_stream_new(priv->context, "audio stream", &ss,
                                       &map)))
        goto unlock_and_fail;

    pa_stream_set_state_callback(priv->stream, stream_state_cb, ao);
    pa_stream_set_write_callback(priv->stream, stream_request_cb, ao);
    pa_stream_set_latency_update_callback(priv->stream,
                                          stream_latency_update_cb, ao);
    pa_buffer_attr bufattr = {
        .maxlength = -1,
        .tlength = pa_usec_to_bytes(1000000, &ss),
        .prebuf = -1,
        .minreq = -1,
        .fragsize = -1,
    };
    if (pa_stream_connect_playback(priv->stream, sink, &bufattr,
                                   PA_STREAM_INTERPOLATE_TIMING
                                   | PA_STREAM_AUTO_TIMING_UPDATE, NULL,
                                   NULL) < 0)
        goto unlock_and_fail;

    /* Wait until the stream is ready */
    pa_threaded_mainloop_wait(priv->mainloop);

    if (pa_stream_get_state(priv->stream) != PA_STREAM_READY)
        goto unlock_and_fail;

    pa_threaded_mainloop_unlock(priv->mainloop);

    free(devarg);
    return 0;

unlock_and_fail:

    if (priv->mainloop)
        pa_threaded_mainloop_unlock(priv->mainloop);

fail:
    if (priv->context)
        GENERIC_ERR_MSG(priv->context, "Init failed");
    free(devarg);
    uninit(ao, true);
    return -1;
}

static void cork(struct ao *ao, bool pause)
{
    struct priv *priv = ao->priv;
    pa_threaded_mainloop_lock(priv->mainloop);
    priv->retval = 0;
    if (!waitop(priv, pa_stream_cork(priv->stream, pause, success_cb, ao)) ||
        !priv->retval)
        GENERIC_ERR_MSG(priv->context, "pa_stream_cork() failed");
}

// Play the specified data to the pulseaudio server
static int play(struct ao *ao, void *data, int len, int flags)
{
    struct priv *priv = ao->priv;
    /* For some reason Pulseaudio behaves worse if this is done after
     * the write - rapidly repeated seeks result in bogus increasing
     * reported latency. */
    if (priv->did_reset)
        cork(ao, false);
    pa_threaded_mainloop_lock(priv->mainloop);
    if (pa_stream_write(priv->stream, data, len, NULL, 0,
                        PA_SEEK_RELATIVE) < 0) {
        GENERIC_ERR_MSG(priv->context, "pa_stream_write() failed");
        len = -1;
    }
    if (priv->did_reset) {
        priv->did_reset = false;
        if (!waitop(priv, pa_stream_update_timing_info(priv->stream,
                                                       success_cb, ao))
            || !priv->retval)
            GENERIC_ERR_MSG(priv->context, "pa_stream_UPP() failed");
    } else
        pa_threaded_mainloop_unlock(priv->mainloop);
    return len;
}

// Reset the audio stream, i.e. flush the playback buffer on the server side
static void reset(struct ao *ao)
{
    // pa_stream_flush() works badly if not corked
    cork(ao, true);
    struct priv *priv = ao->priv;
    pa_threaded_mainloop_lock(priv->mainloop);
    priv->retval = 0;
    if (!waitop(priv, pa_stream_flush(priv->stream, success_cb, ao)) ||
        !priv->retval)
        GENERIC_ERR_MSG(priv->context, "pa_stream_flush() failed");
    priv->did_reset = true;
}
コード例 #17
0
static void resolver_cb(
        AvahiServiceResolver *r,
        AvahiIfIndex interface, AvahiProtocol protocol,
        AvahiResolverEvent event,
        const char *name, const char *type, const char *domain,
        const char *host_name, const AvahiAddress *a, uint16_t port,
        AvahiStringList *txt,
        AvahiLookupResultFlags flags,
        void *userdata) {

    struct userdata *u = userdata;
    struct tunnel *tnl;

    pa_assert(u);

    tnl = tunnel_new(interface, protocol, name, type, domain);

    if (event != AVAHI_RESOLVER_FOUND)
        pa_log("Resolving of '%s' failed: %s", name, avahi_strerror(avahi_client_errno(u->client)));
    else {
        char *device = NULL, *dname, *module_name, *args;
        const char *t;
        char at[AVAHI_ADDRESS_STR_MAX], cmt[PA_CHANNEL_MAP_SNPRINT_MAX];
        pa_sample_spec ss;
        pa_channel_map cm;
        AvahiStringList *l;
        pa_bool_t channel_map_set = FALSE;
        pa_module *m;

        ss = u->core->default_sample_spec;
        pa_assert_se(pa_channel_map_init_auto(&cm, ss.channels, PA_CHANNEL_MAP_AUX));
        pa_channel_map_init_auto(&cm, ss.channels, PA_CHANNEL_MAP_DEFAULT);

        for (l = txt; l; l = l->next) {
            char *key, *value;
            pa_assert_se(avahi_string_list_get_pair(l, &key, &value, NULL) == 0);

            if (strcmp(key, "device") == 0) {
                pa_xfree(device);
                device = value;
                value = NULL;
            } else if (strcmp(key, "rate") == 0)
                ss.rate = atoi(value);
            else if (strcmp(key, "channels") == 0)
                ss.channels = atoi(value);
            else if (strcmp(key, "format") == 0)
                ss.format = pa_parse_sample_format(value);
            else if (strcmp(key, "channel_map") == 0) {
                pa_channel_map_parse(&cm, value);
                channel_map_set = TRUE;
            }

            avahi_free(key);
            avahi_free(value);
        }

        if (!channel_map_set && cm.channels != ss.channels) {
            pa_assert_se(pa_channel_map_init_auto(&cm, ss.channels, PA_CHANNEL_MAP_AUX));
            pa_channel_map_init_auto(&cm, ss.channels, PA_CHANNEL_MAP_DEFAULT);
        }

        if (!pa_sample_spec_valid(&ss)) {
            pa_log("Service '%s' contains an invalid sample specification.", name);
            avahi_free(device);
            goto finish;
        }

        if (!pa_channel_map_valid(&cm) || cm.channels != ss.channels) {
            pa_log("Service '%s' contains an invalid channel map.", name);
            avahi_free(device);
            goto finish;
        }

        if (device)
            dname = pa_sprintf_malloc("tunnel.%s.%s", host_name, device);
        else
            dname = pa_sprintf_malloc("tunnel.%s", host_name);

        if (!pa_namereg_is_valid_name(dname)) {
            pa_log("Cannot construct valid device name from credentials of service '%s'.", dname);
            avahi_free(device);
            pa_xfree(dname);
            goto finish;
        }

        t = strstr(type, "sink") ? "sink" : "source";

        module_name = pa_sprintf_malloc("module-tunnel-%s", t);
        args = pa_sprintf_malloc("server=[%s]:%u "
                                 "%s=%s "
                                 "format=%s "
                                 "channels=%u "
                                 "rate=%u "
                                 "%s_name=%s "
                                 "channel_map=%s",
                                 avahi_address_snprint(at, sizeof(at), a), port,
                                 t, device,
                                 pa_sample_format_to_string(ss.format),
                                 ss.channels,
                                 ss.rate,
                                 t, dname,
                                 pa_channel_map_snprint(cmt, sizeof(cmt), &cm));

        pa_log_debug("Loading module-tunnel-%s with arguments '%s'", module_name, args);

        if ((m = pa_module_load(u->core, module_name, args))) {
            tnl->module_index = m->index;
            pa_hashmap_put(u->tunnels, tnl, tnl);
            tnl = NULL;
        }

        pa_xfree(module_name);
        pa_xfree(dname);
        pa_xfree(args);
        avahi_free(device);
    }

finish:

    avahi_service_resolver_free(r);

    if (tnl)
        tunnel_free(tnl);
}