* [gentoo-commits] gentoo-x86 commit in media-plugins/gst-plugins-ffmpeg/files: 0.10.13_p201211-ffmpeg2.patch 0.10.13_p201211-planaraudio.patch
@ 2013-08-06 15:34 Alexis Ballier (aballier)
0 siblings, 0 replies; only message in thread
From: Alexis Ballier (aballier) @ 2013-08-06 15:34 UTC (permalink / raw
To: gentoo-commits
aballier 13/08/06 15:34:36
Added: 0.10.13_p201211-ffmpeg2.patch
0.10.13_p201211-planaraudio.patch
Log:
Attempt to fix planar audio support. Fix build with FFmpeg 2.0, bug #476528.
(Portage version: 2.2.0_alpha194/cvs/Linux x86_64, signed Manifest commit with key 160F534A)
Revision Changes Path
1.1 media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-ffmpeg2.patch
file : http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-ffmpeg2.patch?rev=1.1&view=markup
plain: http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-ffmpeg2.patch?rev=1.1&content-type=text/plain
Index: 0.10.13_p201211-ffmpeg2.patch
===================================================================
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcfg.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegcfg.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcfg.c
@@ -171,13 +171,10 @@ gst_ffmpeg_idct_algo_get_type (void)
{FF_IDCT_INT, "JPEG reference Integer", "int"},
{FF_IDCT_SIMPLE, "Simple", "simple"},
{FF_IDCT_SIMPLEMMX, "Simple MMX", "simplemmx"},
- {FF_IDCT_LIBMPEG2MMX, "LIBMPEG2MMX", "libmpeg2mmx"},
{FF_IDCT_ARM, "ARM", "arm"},
{FF_IDCT_ALTIVEC, "ALTIVEC", "altivec"},
{FF_IDCT_SH4, "SH4", "sh4"},
{FF_IDCT_SIMPLEARM, "SIMPLEARM", "simplearm"},
- {FF_IDCT_H264, "H264", "h264"},
- {FF_IDCT_VP3, "VP3", "vp3"},
{FF_IDCT_IPP, "IPP", "ipp"},
{FF_IDCT_XVIDMMX, "XVIDMMX", "xvidmmx"},
{0, NULL, NULL},
@@ -274,9 +271,6 @@ gst_ffmpeg_flags_get_type (void)
"global-headers"},
{CODEC_FLAG_AC_PRED, "H263 Advanced Intra Coding / MPEG4 AC prediction",
"aic"},
- {CODEC_FLAG_CBP_RD, "Rate Distoration Optimization for CBP", "cbp-rd"},
- {CODEC_FLAG_QP_RD, "Rate Distoration Optimization for QP selection",
- "qp-rd"},
{CODEC_FLAG_CLOSED_GOP, "Closed GOP", "closedgop"},
{0, NULL, NULL},
};
@@ -580,18 +574,6 @@ gst_ffmpeg_cfg_init (void)
-100, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
gst_ffmpeg_add_pspec (pspec, max_key_interval, FALSE, mpeg, NULL);
- pspec = g_param_spec_int ("luma-elim-threshold",
- "Luma Elimination Threshold",
- "Luma Single Coefficient Elimination Threshold",
- -99, 99, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
- gst_ffmpeg_add_pspec (pspec, config.luma_elim_threshold, FALSE, mpeg, NULL);
-
- pspec = g_param_spec_int ("chroma-elim-threshold",
- "Chroma Elimination Threshold",
- "Chroma Single Coefficient Elimination Threshold",
- -99, 99, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
- gst_ffmpeg_add_pspec (pspec, config.chroma_elim_threshold, FALSE, mpeg, NULL);
-
pspec = g_param_spec_float ("lumi-masking", "Luminance Masking",
"Luminance Masking", -1.0f, 1.0f, 0.0f,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcodecmap.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegcodecmap.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcodecmap.c
@@ -572,13 +572,11 @@ gst_ffmpeg_codecid_to_caps (enum AVCodec
break;
}
- /* FIXME: context->sub_id must be filled in during decoding */
caps =
gst_ff_vid_caps_new (context, codec_id, encode,
"video/x-pn-realvideo", "systemstream", G_TYPE_BOOLEAN, FALSE,
"rmversion", G_TYPE_INT, version, NULL);
if (context) {
- gst_caps_set_simple (caps, "format", G_TYPE_INT, context->sub_id, NULL);
if (context->extradata_size >= 8) {
gst_caps_set_simple (caps,
"subformat", G_TYPE_INT, GST_READ_UINT32_BE (context->extradata),
@@ -2559,18 +2557,6 @@ gst_ffmpeg_caps_with_codecid (enum AVCod
}
break;
- case AV_CODEC_ID_RV10:
- case AV_CODEC_ID_RV20:
- case AV_CODEC_ID_RV30:
- case AV_CODEC_ID_RV40:
- {
- gint format;
-
- if (gst_structure_get_int (str, "format", &format))
- context->sub_id = format;
-
- break;
- }
case AV_CODEC_ID_COOK:
case AV_CODEC_ID_RA_288:
case AV_CODEC_ID_RA_144:
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegenc.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegenc.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegenc.c
@@ -572,7 +572,6 @@ gst_ffmpegenc_setcaps (GstPad * pad, Gst
ffmpegenc->context->coder_type = 0;
ffmpegenc->context->context_model = 0;
ffmpegenc->context->scenechange_threshold = 0;
- ffmpegenc->context->inter_threshold = 0;
/* and last but not least the pass; CBR, 2-pass, etc */
ffmpegenc->context->flags |= ffmpegenc->pass;
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegdec.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegdec.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegdec.c
@@ -2082,7 +2082,7 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec
const GstTSInfo * dec_info, GstBuffer ** outbuf, GstFlowReturn * ret)
{
gint len = -1, got_frame;
- gint have_data = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ gint have_data = 0;
GstClockTime out_timestamp, out_duration;
gint64 out_offset;
AVPacket packet;
@@ -2101,21 +2101,22 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec
goto beach;
}
- *outbuf =
- new_aligned_buffer (AVCODEC_MAX_AUDIO_FRAME_SIZE,
- GST_PAD_CAPS (ffmpegdec->srcpad));
-
gst_avpacket_init (&packet, data, size);
len = avcodec_decode_audio4 (ffmpegdec->context, frame, &got_frame, &packet);
GST_DEBUG_OBJECT (ffmpegdec,
"Decode audio: ret=%d, got_frame=%d", len, got_frame);
if (!got_frame) {
- gst_buffer_unref (*outbuf);
- *outbuf = NULL;
len = -1;
goto beach;
}
- if (len >= 0) have_data = copy_samples(ffmpegdec->context, frame, GST_BUFFER_DATA (*outbuf), AVCODEC_MAX_AUDIO_FRAME_SIZE);
+
+ int obuf_size = av_samples_get_buffer_size(NULL, ffmpegdec->context->channels, frame->nb_samples, frame->format, 0);
+
+ *outbuf=
+ new_aligned_buffer (obuf_size,
+ GST_PAD_CAPS (ffmpegdec->srcpad));
+
+ if (len >= 0) have_data = copy_samples(ffmpegdec->context, frame, GST_BUFFER_DATA (*outbuf), obuf_size);
if (len >= 0 && have_data > 0) {
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
1.1 media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-planaraudio.patch
file : http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-planaraudio.patch?rev=1.1&view=markup
plain: http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-planaraudio.patch?rev=1.1&content-type=text/plain
Index: 0.10.13_p201211-planaraudio.patch
===================================================================
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcodecmap.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegcodecmap.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcodecmap.c
@@ -1925,6 +1925,10 @@ gst_ffmpeg_smpfmt_to_caps (enum AVSample
gboolean integer = TRUE;
gboolean signedness = FALSE;
+#if LIBAVUTIL_VERSION_INT > AV_VERSION_INT(51,46,0)
+ sample_fmt = av_get_packed_sample_fmt (sample_fmt);
+#endif
+
switch (sample_fmt) {
case AV_SAMPLE_FMT_S16:
signedness = TRUE;
@@ -2009,7 +2013,7 @@ gst_ffmpeg_codectype_to_audio_caps (AVCo
ctx.channels = -1;
caps = gst_caps_new_empty ();
- for (i = 0; i <= AV_SAMPLE_FMT_DBL; i++) {
+ for (i = 0; i < AV_SAMPLE_FMT_NB; i++) {
temp =
gst_ffmpeg_smpfmt_to_caps (i, encode ? &ctx : NULL, codec_id, encode);
if (temp != NULL) {
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegutils.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegutils.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegutils.c
@@ -47,6 +47,9 @@ gint
av_smp_format_depth (enum AVSampleFormat smp_fmt)
{
gint depth = -1;
+#if LIBAVUTIL_VERSION_INT > AV_VERSION_INT(51,46,0)
+ smp_fmt = av_get_packed_sample_fmt (smp_fmt);
+#endif
switch (smp_fmt) {
case AV_SAMPLE_FMT_U8:
depth = 1;
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegdec.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegdec.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegdec.c
@@ -2044,16 +2044,49 @@ out_of_segment:
}
}
+static void copy_samples_planar(unsigned bps,
+ unsigned nb_samples,
+ unsigned nb_channels,
+ unsigned char *dst,
+ unsigned char **src)
+{
+ unsigned s, c, o = 0;
+
+ for (s = 0; s < nb_samples; s++) {
+ for (c = 0; c < nb_channels; c++) {
+ memcpy(dst, src[c] + o, bps);
+ dst += bps;
+ }
+ o += bps;
+ }
+}
+
+static int copy_samples(AVCodecContext *avc, AVFrame *frame,
+ unsigned char *buf, int max_size)
+{
+ int channels = avc->channels;
+ int sample_size = av_get_bytes_per_sample(avc->sample_fmt);
+ int size = channels * sample_size * frame->nb_samples;
+ if (size > max_size) {
+ return -1;
+ }
+ if (av_sample_fmt_is_planar(avc->sample_fmt))
+ copy_samples_planar(sample_size, frame->nb_samples, channels, buf, frame->extended_data);
+ else memcpy(buf, frame->data[0], size);
+ return size;
+}
+
static gint
gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
AVCodec * in_plugin, guint8 * data, guint size,
const GstTSInfo * dec_info, GstBuffer ** outbuf, GstFlowReturn * ret)
{
- gint len = -1;
+ gint len = -1, got_frame;
gint have_data = AVCODEC_MAX_AUDIO_FRAME_SIZE;
GstClockTime out_timestamp, out_duration;
gint64 out_offset;
AVPacket packet;
+ AVFrame *frame;
GST_DEBUG_OBJECT (ffmpegdec,
"size:%d, offset:%" G_GINT64_FORMAT ", ts:%" GST_TIME_FORMAT ", dur:%"
@@ -2061,15 +2094,28 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec
dec_info->offset, GST_TIME_ARGS (dec_info->timestamp),
GST_TIME_ARGS (dec_info->duration), GST_TIME_ARGS (ffmpegdec->next_out));
+ frame = avcodec_alloc_frame();
+ if (!frame) {
+ *outbuf = NULL;
+ len = -1;
+ goto beach;
+ }
+
*outbuf =
new_aligned_buffer (AVCODEC_MAX_AUDIO_FRAME_SIZE,
GST_PAD_CAPS (ffmpegdec->srcpad));
gst_avpacket_init (&packet, data, size);
- len = avcodec_decode_audio3 (ffmpegdec->context,
- (int16_t *) GST_BUFFER_DATA (*outbuf), &have_data, &packet);
+ len = avcodec_decode_audio4 (ffmpegdec->context, frame, &got_frame, &packet);
GST_DEBUG_OBJECT (ffmpegdec,
- "Decode audio: len=%d, have_data=%d", len, have_data);
+ "Decode audio: ret=%d, got_frame=%d", len, got_frame);
+ if (!got_frame) {
+ gst_buffer_unref (*outbuf);
+ *outbuf = NULL;
+ len = -1;
+ goto beach;
+ }
+ if (len >= 0) have_data = copy_samples(ffmpegdec->context, frame, GST_BUFFER_DATA (*outbuf), AVCODEC_MAX_AUDIO_FRAME_SIZE);
if (len >= 0 && have_data > 0) {
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
@@ -2145,6 +2191,7 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec
}
beach:
+ av_free(frame);
GST_DEBUG_OBJECT (ffmpegdec, "return flow %d, out %p, len %d",
*ret, *outbuf, len);
return len;
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegenc.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegenc.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegenc.c
@@ -843,12 +843,30 @@ gst_ffmpegenc_chain_video (GstPad * pad,
return gst_pad_push (ffmpegenc->srcpad, outbuf);
}
+static void copy_samples_to_planar(unsigned bps,
+ unsigned nb_samples,
+ unsigned nb_channels,
+ unsigned char *dst,
+ unsigned char *src)
+{
+ unsigned s, c, o = 0;
+
+ for (s = 0; s < nb_samples; s++) {
+ for (c = 0; c < nb_channels; c++) {
+ memcpy(dst + nb_samples * c, src + o, bps);
+ o += bps;
+ }
+ dst += bps;
+ }
+}
+
static GstFlowReturn
gst_ffmpegenc_encode_audio (GstFFMpegEnc * ffmpegenc, guint8 * audio_in,
guint in_size, guint max_size, GstClockTime timestamp,
GstClockTime duration, gboolean discont)
{
GstBuffer *outbuf;
+ GstBuffer *inbuf2;
AVCodecContext *ctx;
guint8 *audio_out;
gint res;
@@ -864,7 +882,18 @@ gst_ffmpegenc_encode_audio (GstFFMpegEnc
if (ffmpegenc->buffer_size != max_size)
ffmpegenc->buffer_size = max_size;
+ if (av_sample_fmt_is_planar(ctx->sample_fmt)) {
+ guint8 * audio_in2;
+ inbuf2 = gst_buffer_new_and_alloc (in_size + FF_MIN_BUFFER_SIZE);
+ audio_in2 = GST_BUFFER_DATA (inbuf2);
+ copy_samples_to_planar(av_get_bytes_per_sample(ctx->sample_fmt), in_size / (av_get_bytes_per_sample(ctx->sample_fmt) * ctx->channels),
+ ctx->channels, audio_in2, audio_in);
+ audio_in = audio_in2;
+ }
res = avcodec_encode_audio (ctx, audio_out, max_size, (short *) audio_in);
+ if (av_sample_fmt_is_planar(ctx->sample_fmt)) {
+ gst_buffer_unref (inbuf2);
+ }
if (res < 0) {
GST_ERROR_OBJECT (ffmpegenc, "Failed to encode buffer: %d", res);
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2013-08-06 15:34 [gentoo-commits] gentoo-x86 commit in media-plugins/gst-plugins-ffmpeg/files: 0.10.13_p201211-ffmpeg2.patch 0.10.13_p201211-planaraudio.patch Alexis Ballier (aballier)
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