本文整理汇总了C++中GST_AUDIO_DECODER函数的典型用法代码示例。如果您正苦于以下问题：C++ GST_AUDIO_DECODER函数的具体用法？C++ GST_AUDIO_DECODER怎么用？C++ GST_AUDIO_DECODER使用的例子？那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了GST_AUDIO_DECODER函数的20个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于我们的系统推荐出更棒的C++代码示例。
示例1: gst_amc_audio_dec_init
static void
gst_amc_audio_dec_init (GstAmcAudioDec * self)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE);
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
g_mutex_init (&self->drain_lock);
g_cond_init (&self->drain_cond);
}
开发者ID:iainlane，项目名称:gstreamer，代码行数:9，代码来源:gstamcaudiodechybris.c
示例2: gst_ffmpegauddec_init
static void
gst_ffmpegauddec_init (GstFFMpegAudDec * ffmpegdec)
{
GstFFMpegAudDecClass *klass =
(GstFFMpegAudDecClass *) G_OBJECT_GET_CLASS (ffmpegdec);
/* some ffmpeg data */
ffmpegdec->context = avcodec_alloc_context3 (klass->in_plugin);
ffmpegdec->opened = FALSE;
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (ffmpegdec), TRUE);
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (ffmpegdec), TRUE);
}
开发者ID:cablelabs，项目名称:gst-libav，代码行数:13，代码来源:gstavauddec.c
示例3: gst_ffmpegauddec_drain
static void
gst_ffmpegauddec_drain (GstFFMpegAudDec * ffmpegdec)
{
GstFFMpegAudDecClass *oclass;
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
if (oclass->in_plugin->capabilities & CODEC_CAP_DELAY) {
gint have_data, len;
GST_LOG_OBJECT (ffmpegdec,
"codec has delay capabilities, calling until libav has drained everything");
do {
GstFlowReturn ret;
len = gst_ffmpegauddec_frame (ffmpegdec, NULL, 0, &have_data, &ret);
} while (len >= 0 && have_data == 1);
avcodec_flush_buffers (ffmpegdec->context);
}
if (ffmpegdec->outbuf)
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (ffmpegdec),
ffmpegdec->outbuf, 1);
ffmpegdec->outbuf = NULL;
}
开发者ID:GStreamer，项目名称:gst-libav，代码行数:27，代码来源:gstavauddec.c
示例4: gst_dtsdec_renegotiate
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
gint channels;
gboolean result = FALSE;
GstAudioChannelPosition from[6], to[6];
GstAudioInfo info;
channels = gst_dtsdec_channels (dts->using_channels, from);
if (!channels)
goto done;
GST_INFO_OBJECT (dts, "dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
memcpy (to, from, sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (to, channels);
gst_audio_get_channel_reorder_map (channels, from, to,
dts->channel_reorder_map);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
SAMPLE_TYPE, dts->sample_rate, channels, (channels > 1 ? to : NULL));
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dts), &info))
goto done;
result = TRUE;
done:
return result;
}
开发者ID:lubing521，项目名称:gst-embedded-builder，代码行数:34，代码来源:gstdtsdec.c
示例5: gst_mulawdec_init
static void
gst_mulawdec_init (GstMuLawDec * mulawdec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (mulawdec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(mulawdec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (mulawdec));
}
开发者ID:DylanZA，项目名称:gst-plugins-good，代码行数:8，代码来源:mulaw-decode.c
示例6: gst_droidadec_init
static void
gst_droidadec_init (GstDroidADec * dec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (dec), TRUE);
dec->codec = NULL;
dec->codec_type = NULL;
dec->downstream_flow_ret = GST_FLOW_OK;
dec->eos = FALSE;
dec->codec_data = NULL;
dec->channels = 0;
dec->rate = 0;
g_mutex_init (&dec->eos_lock);
g_cond_init (&dec->eos_cond);
dec->adapter = gst_adapter_new ();
}
开发者ID:rss351，项目名称:gst-droid，代码行数:18，代码来源:gstdroidadec.c
示例7: gst_wavpack_dec_init
static void
gst_wavpack_dec_init (GstWavpackDec * dec)
{
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_wavpack_dec_reset (dec);
}
开发者ID:Lachann，项目名称:gst-plugins-good，代码行数:10，代码来源:gstwavpackdec.c
示例8: gst_opus_dec_init
static void
gst_opus_dec_init (GstOpusDec * dec)
{
dec->use_inband_fec = FALSE;
dec->apply_gain = DEFAULT_APPLY_GAIN;
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_opus_dec_reset (dec);
}
开发者ID:Distrotech，项目名称:gst-plugins-bad，代码行数:10，代码来源:gstopusdec.c
示例9: gst_sbc_dec_init
static void
gst_sbc_dec_init (GstSbcDec * dec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
dec->samples_per_frame = 0;
dec->frame_len = 0;
}
开发者ID:0p1pp1，项目名称:gst-plugins-bad，代码行数:11，代码来源:gstsbcdec.c
示例10: gst_ffmpegauddec_init
static void
gst_ffmpegauddec_init (GstFFMpegAudDec * ffmpegdec)
{
GstFFMpegAudDecClass *klass =
(GstFFMpegAudDecClass *) G_OBJECT_GET_CLASS (ffmpegdec);
/* some ffmpeg data */
ffmpegdec->context = avcodec_alloc_context3 (klass->in_plugin);
ffmpegdec->context->opaque = ffmpegdec;
ffmpegdec->opened = FALSE;
ffmpegdec->frame = av_frame_alloc ();
GST_PAD_SET_ACCEPT_TEMPLATE (GST_VIDEO_DECODER_SINK_PAD (ffmpegdec));
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(ffmpegdec), TRUE);
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (ffmpegdec), TRUE);
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (ffmpegdec), TRUE);
}
开发者ID:GStreamer，项目名称:gst-libav，代码行数:20，代码来源:gstavauddec.c
示例11: gst_ffmpegauddec_get_buffer
/* called when ffmpeg wants us to allocate a buffer to write the decoded frame
* into. We try to give it memory from our pool */
static int
gst_ffmpegauddec_get_buffer (AVCodecContext * context, AVFrame * frame)
{
GstFFMpegAudDec *ffmpegdec;
GstAudioInfo *info;
BufferInfo *buffer_info;
ffmpegdec = (GstFFMpegAudDec *) context->opaque;
if (G_UNLIKELY (!gst_ffmpegauddec_negotiate (ffmpegdec, FALSE)))
goto negotiate_failed;
/* Always use the default allocator for planar audio formats because
* we will have to copy and deinterleave later anyway */
if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt))
goto fallback;
info = gst_audio_decoder_get_audio_info (GST_AUDIO_DECODER (ffmpegdec));
buffer_info = g_slice_new (BufferInfo);
buffer_info->buffer =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (ffmpegdec),
frame->nb_samples * info->bpf);
gst_buffer_map (buffer_info->buffer, &buffer_info->map, GST_MAP_WRITE);
frame->opaque = buffer_info;
frame->data[0] = buffer_info->map.data;
frame->extended_data = frame->data;
frame->linesize[0] = buffer_info->map.size;
frame->type = FF_BUFFER_TYPE_USER;
return 0;
/* fallbacks */
negotiate_failed:
{
GST_DEBUG_OBJECT (ffmpegdec, "negotiate failed");
goto fallback;
}
fallback:
{
return avcodec_default_get_buffer (context, frame);
}
}
开发者ID:cablelabs，项目名称:gst-libav，代码行数:43，代码来源:gstavauddec.c
示例12: gst_speex_dec_init
static void
gst_speex_dec_init (GstSpeexDec * dec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
dec->enh = DEFAULT_ENH;
gst_speex_dec_reset (dec);
}
开发者ID:pexip，项目名称:gst-plugins-good，代码行数:12，代码来源:gstspeexdec.c
示例13: gst_opus_dec_init
static void
gst_opus_dec_init (GstOpusDec * dec)
{
dec->use_inband_fec = FALSE;
dec->apply_gain = DEFAULT_APPLY_GAIN;
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
gst_opus_dec_reset (dec);
}
开发者ID:GrokImageCompression，项目名称:gst-plugins-base，代码行数:13，代码来源:gstopusdec.c
示例14: gst_wavpack_dec_init
static void
gst_wavpack_dec_init (GstWavpackDec * dec)
{
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
gst_wavpack_dec_reset (dec);
}
开发者ID:DylanZA，项目名称:gst-plugins-good，代码行数:13，代码来源:gstwavpackdec.c
示例15: gst_a52dec_update_streaminfo
static void
gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
{
GstTagList *taglist;
taglist = gst_tag_list_new_empty ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
(guint) a52dec->bit_rate, NULL);
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (a52dec), taglist,
GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (taglist);
}
开发者ID:lubing521，项目名称:gst-embedded-builder，代码行数:13，代码来源:gsta52dec.c
示例16: gst_dtsdec_update_streaminfo
static void
gst_dtsdec_update_streaminfo (GstDtsDec * dts)
{
GstTagList *taglist;
if (dts->bit_rate > 3) {
taglist = gst_tag_list_new_empty ();
/* 1 => open bitrate, 2 => variable bitrate, 3 => lossless */
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
(guint) dts->bit_rate, NULL);
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dts), taglist,
GST_TAG_MERGE_REPLACE);
}
}
开发者ID:lubing521，项目名称:gst-embedded-builder，代码行数:14，代码来源:gstdtsdec.c
示例17: gst_imx_audio_uniaudio_dec_init
void gst_imx_audio_uniaudio_dec_init(GstImxAudioUniaudioDec *imx_decoder)
{
GstAudioDecoder *base = GST_AUDIO_DECODER(imx_decoder);
gst_audio_decoder_set_drainable(base, TRUE);
gst_audio_decoder_set_plc_aware(base, FALSE);
imx_decoder->codec = NULL;
imx_decoder->handle = NULL;
imx_decoder->original_channel_positions = NULL;
imx_decoder->reordered_channel_positions = NULL;
imx_decoder->out_adapter = gst_adapter_new();
imx_decoder->skip_header_counter = 0;
imx_decoder->codec_data = NULL;
}
开发者ID:Freescale，项目名称:gstreamer-imx，代码行数:14，代码来源:uniaudio_decoder.c
示例18: gst_amc_audio_dec_set_src_caps
static gboolean
gst_amc_audio_dec_set_src_caps (GstAmcAudioDec * self, GstAmcFormat * format)
{
gint rate, channels;
guint32 channel_mask = 0;
GstAudioChannelPosition to[64];
GError *err = NULL;
if (!gst_amc_format_get_int (format, "sample-rate", &rate, &err) ||
!gst_amc_format_get_int (format, "channel-count", &channels, &err)) {
GST_ERROR_OBJECT (self, "Failed to get output format metadata: %s",
err->message);
g_clear_error (&err);
return FALSE;
}
if (rate == 0 || channels == 0) {
GST_ERROR_OBJECT (self, "Rate or channels not set");
return FALSE;
}
/* Not always present */
if (gst_amc_format_contains_key (format, "channel-mask", NULL))
gst_amc_format_get_int (format, "channel-mask", (gint *) & channel_mask,
NULL);
gst_amc_audio_channel_mask_to_positions (channel_mask, channels,
self->positions);
memcpy (to, self->positions, sizeof (to));
gst_audio_channel_positions_to_valid_order (to, channels);
self->needs_reorder =
(memcmp (self->positions, to,
sizeof (GstAudioChannelPosition) * channels) != 0);
if (self->needs_reorder)
gst_audio_get_channel_reorder_map (channels, self->positions, to,
self->reorder_map);
gst_audio_info_init (&self->info);
gst_audio_info_set_format (&self->info, GST_AUDIO_FORMAT_S16, rate, channels,
to);
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self),
&self->info))
return FALSE;
self->input_caps_changed = FALSE;
return TRUE;
}
开发者ID:Distrotech，项目名称:gst-plugins-bad，代码行数:49，代码来源:gstamcaudiodec.c
示例19: gst_droidadec_finalize
static void
gst_droidadec_finalize (GObject * object)
{
GstDroidADec *dec = GST_DROIDADEC (object);
GST_DEBUG_OBJECT (dec, "finalize");
gst_droidadec_stop (GST_AUDIO_DECODER (dec));
g_mutex_clear (&dec->eos_lock);
g_cond_clear (&dec->eos_cond);
gst_object_unref (dec->adapter);
dec->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
开发者ID:rss351，项目名称:gst-droid，代码行数:17，代码来源:gstdroidadec.c
示例20: vorbis_handle_identification_packet
static GstFlowReturn
vorbis_handle_identification_packet (GstVorbisDec * vd)
{
GstAudioInfo info;
switch (vd->vi.channels) {
case 1:
case 2:
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
{
const GstAudioChannelPosition *pos;
pos = gst_vorbis_default_channel_positions[vd->vi.channels - 1];
gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate,
vd->vi.channels, pos);
break;
}
default:{
GstAudioChannelPosition position[64];
gint i, max_pos = MAX (vd->vi.channels, 64);
GST_ELEMENT_WARNING (vd, STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < max_pos; i++)
position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate,
vd->vi.channels, position);
break;
}
}
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (vd), &info);
vd->info = info;
/* select a copy_samples function, this way we can have specialized versions
* for mono/stereo and avoid the depth switch in tremor case */
vd->copy_samples = gst_vorbis_get_copy_sample_func (info.channels);
return GST_FLOW_OK;
}
开发者ID:Lachann，项目名称:gst-plugins-base，代码行数:45，代码来源:gstvorbisdec.c
注：本文中的GST_AUDIO_DECODER函数示例由纯净天空整理自Github/MSDocs等源码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。 |
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