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Merge branch 'master' into 0.11

Browse source 
Conflicts:
	ext/ffmpeg/gstffmpegcodecmap.c
This commit is contained in:
Wim Taymans 2011-10-27 16:31:30 +02:00
commit 1fffbe9876
2 changed files with 77 additions and 69 deletions
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144
ext/ffmpeg/gstffmpegcodecmap.c
View file

@ -374,79 +374,88 @@ gst_ff_aud_caps_new (AVCodecContext * context, enum CodecID codec_id,
const gint *rates = NULL;
gint n_rates = 0;
if (context) {
/* so we must be after restricted caps in this particular case */
switch (codec_id) {
case CODEC_ID_MP2:
{
const static gint l_rates[] =
{ 48000, 44100, 32000, 24000, 22050, 16000 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_EAC3:
case CODEC_ID_AC3:
{
const static gint l_rates[] = { 48000, 44100, 32000 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_ADPCM_SWF:
{
const static gint l_rates[] = { 11025, 22050, 44100 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_ROQ_DPCM:
{
const static gint l_rates[] = { 22050 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_ADPCM_G726:
maxchannels = 1;
break;
case CODEC_ID_AMR_NB:
{
const static gint l_rates[] = { 8000 };
maxchannels = 1;
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_AMR_WB:
{
const static gint l_rates[] = { 16000 };
maxchannels = 1;
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
default:
break;
}
/* TODO: handle context->channel_layouts here to set
* the list of channel layouts supported by the encoder.
* Unfortunately no encoder uses this yet....
*/
}
#if 0
/* regardless of encode/decode, open up channels if applicable */
/* Until decoders/encoders expose the maximum number of channels
* they support, we whitelist them here. */
/* so we must be after restricted caps in this case */
switch (codec_id) {
case CODEC_ID_AC3:
case CODEC_ID_EAC3:
case CODEC_ID_AAC:
case CODEC_ID_AAC_LATM:
case CODEC_ID_DTS:
maxchannels = 6;
break;
case CODEC_ID_MP2:
{
const static gint l_rates[] =
{ 48000, 44100, 32000, 24000, 22050, 16000 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_EAC3:
case CODEC_ID_AC3:
{
const static gint l_rates[] = { 48000, 44100, 32000 };
maxchannels = 6;
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_ADPCM_G722:
{
const static gint l_rates[] = { 16000 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
maxchannels = 1;
break;
}
case CODEC_ID_ADPCM_G726:
{
const static gint l_rates[] = { 8000 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
maxchannels = 1;
break;
}
case CODEC_ID_ADPCM_SWF:
{
const static gint l_rates[] = { 11025, 22050, 44100 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_ROQ_DPCM:
{
const static gint l_rates[] = { 22050 };
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_AMR_NB:
{
const static gint l_rates[] = { 8000 };
maxchannels = 1;
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
case CODEC_ID_AMR_WB:
{
const static gint l_rates[] = { 16000 };
maxchannels = 1;
n_rates = G_N_ELEMENTS (l_rates);
rates = l_rates;
break;
}
default:
break;
}
/* TODO: handle context->channel_layouts here to set
* the list of channel layouts supported by the encoder.
* Unfortunately no encoder uses this yet....
*/
/* regardless of encode/decode, open up channels if applicable */
/* Until decoders/encoders expose the maximum number of channels
* they support, we whitelist them here. */
switch (codec_id) {
case CODEC_ID_WMAPRO:
case CODEC_ID_TRUEHD:
maxchannels = 8;
@ -454,7 +463,6 @@ gst_ff_aud_caps_new (AVCodecContext * context, enum CodecID codec_id,
default:
break;
}
#endif
if (maxchannels == 1)
caps = gst_caps_new_simple (mimetype,

2
ext/ffmpeg/gstffmpegenc.c
View file

@ -896,7 +896,7 @@ gst_ffmpegenc_encode_audio (GstFFMpegEnc * ffmpegenc, guint8 * audio_in,
if (ffmpegenc->buffer_size != max_size)
ffmpegenc->buffer_size = max_size;
res = avcodec_encode_audio (ctx, audio_out, in_size, (short *) audio_in);
res = avcodec_encode_audio (ctx, audio_out, max_size, (short *) audio_in);
if (res < 0) {
gst_buffer_unmap (outbuf, audio_out, 0);