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ext/ffmpeg/gstffmpegcodecmap.*: Add mapping for EAC3 and QCELP audio codecs.
Original commit message from CVS: * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ff_aud_caps_new), (gst_ffmpeg_codecid_to_caps), (gst_ffmpeg_smpfmt_to_caps), (gst_ffmpeg_codectype_to_caps), (gst_ffmpeg_caps_to_smpfmt), (gst_ffmpeg_caps_to_codecid), (av_smp_format_depth): * ext/ffmpeg/gstffmpegcodecmap.h: Add mapping for EAC3 and QCELP audio codecs. Add conversion functions for all available audo SampleFormat. * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_open), (gst_ffmpegdec_setcaps), (gst_ffmpegdec_negotiate), (clip_audio_buffer), (gst_ffmpegdec_audio_frame): Remove assumptions that we can only handle stereo 16bit signed integer audio, and store the depth locally.
This commit is contained in:
Edward Hervey
parent
75694feb3b
commit
3d089f4862
4 changed files with 125 additions and 20 deletions
15
ChangeLog
15
ChangeLog
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@ -1,3 +1,18 @@
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2008-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
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* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ff_aud_caps_new),
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(gst_ffmpeg_codecid_to_caps), (gst_ffmpeg_smpfmt_to_caps),
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(gst_ffmpeg_codectype_to_caps), (gst_ffmpeg_caps_to_smpfmt),
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(gst_ffmpeg_caps_to_codecid), (av_smp_format_depth):
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* ext/ffmpeg/gstffmpegcodecmap.h:
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Add mapping for EAC3 and QCELP audio codecs.
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Add conversion functions for all available audo SampleFormat.
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* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_open),
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(gst_ffmpegdec_setcaps), (gst_ffmpegdec_negotiate),
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(clip_audio_buffer), (gst_ffmpegdec_audio_frame):
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Remove assumptions that we can only handle stereo 16bit signed integer
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audio, and store the depth locally.
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2008-12-16 Stefan Kost <ensonic@users.sf.net>
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* configure.ac:
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@ -322,6 +322,7 @@ gst_ff_aud_caps_new (AVCodecContext * context, enum CodecID codec_id,
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rates = l_rates;
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break;
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}
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case CODEC_ID_EAC3:
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case CODEC_ID_AC3:
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{
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const static gint l_rates[] = { 48000, 44100, 32000 };
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@ -361,6 +362,7 @@ gst_ff_aud_caps_new (AVCodecContext * context, enum CodecID codec_id,
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* they support, we whitelist them here. */
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switch (codec_id) {
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case CODEC_ID_AC3:
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case CODEC_ID_EAC3:
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case CODEC_ID_AAC:
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case CODEC_ID_DTS:
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maxchannels = 6;
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@ -560,6 +562,11 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
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caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-ac3", NULL);
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break;
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case CODEC_ID_EAC3:
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/* FIXME: bitrate */
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caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-eac3", NULL);
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break;
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case CODEC_ID_ATRAC3:
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caps = gst_ff_aud_caps_new (context, codec_id, "audio/atrac3", NULL);
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break;
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@ -986,6 +993,10 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
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gst_ff_aud_caps_new (context, codec_id, "audio/x-truespeech", NULL);
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break;
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case CODEC_ID_QCELP:
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caps = gst_ff_aud_caps_new (context, codec_id, "audio/qcelp", NULL);
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break;
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case CODEC_ID_WS_VQA:
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case CODEC_ID_IDCIN:
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case CODEC_ID_8BPS:
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@ -1529,6 +1540,7 @@ gst_ffmpeg_smpfmt_to_caps (enum SampleFormat sample_fmt,
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GstCaps *caps = NULL;
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int bpp = 0;
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gboolean integer = TRUE;
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gboolean signedness = FALSE;
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switch (sample_fmt) {
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@ -1537,16 +1549,36 @@ gst_ffmpeg_smpfmt_to_caps (enum SampleFormat sample_fmt,
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bpp = 16;
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break;
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case SAMPLE_FMT_S32:
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signedness = TRUE;
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bpp = 32;
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break;
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case SAMPLE_FMT_FLT:
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integer = FALSE;
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bpp = 32;
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break;
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case SAMPLE_FMT_DBL:
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integer = FALSE;
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bpp = 64;
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break;
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default:
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/* .. */
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break;
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}
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if (bpp) {
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if (integer) {
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caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-raw-int",
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"signed", G_TYPE_BOOLEAN, signedness,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, bpp, "depth", G_TYPE_INT, bpp, NULL);
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} else {
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caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-raw-float",
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, bpp, NULL);
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}
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}
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if (caps != NULL) {
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@ -1603,7 +1635,7 @@ gst_ffmpeg_codectype_to_caps (enum CodecType codec_type,
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ctx.channels = -1;
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caps = gst_caps_new_empty ();
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for (i = 0; i <= SAMPLE_FMT_S16; i++) {
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for (i = 0; i <= SAMPLE_FMT_DBL; i++) {
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temp = gst_ffmpeg_smpfmt_to_caps (i, encode ? &ctx : NULL, codec_id);
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if (temp != NULL) {
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gst_caps_append (caps, temp);
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@ -1646,13 +1678,29 @@ gst_ffmpeg_caps_to_smpfmt (const GstCaps * caps,
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if (!raw)
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return;
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if (!strcmp (gst_structure_get_name (structure), "audio/x-raw-float")) {
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/* FLOAT */
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if (gst_structure_get_int (structure, "width", &width) &&
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gst_structure_get_int (structure, "endianness", &endianness)) {
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if (endianness == G_BYTE_ORDER) {
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if (width == 32)
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context->sample_fmt = SAMPLE_FMT_FLT;
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else if (width == 64)
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context->sample_fmt = SAMPLE_FMT_DBL;
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}
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}
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} else {
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/* INT */
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if (gst_structure_get_int (structure, "width", &width) &&
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gst_structure_get_int (structure, "depth", &depth) &&
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gst_structure_get_boolean (structure, "signed", &signedness) &&
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gst_structure_get_int (structure, "endianness", &endianness)) {
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if (width == 16 && depth == 16 &&
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endianness == G_BYTE_ORDER && signedness == TRUE) {
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if ((endianness == G_BYTE_ORDER) && (signedness == TRUE)) {
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if ((width == 16) && (depth == 16))
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context->sample_fmt = SAMPLE_FMT_S16;
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else if ((width == 32) && (depth == 32))
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context->sample_fmt = SAMPLE_FMT_S32;
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}
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}
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}
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}
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@ -2550,6 +2598,9 @@ gst_ffmpeg_caps_to_codecid (const GstCaps * caps, AVCodecContext * context)
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} else if (!strcmp (mimetype, "audio/x-ac3")) {
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id = CODEC_ID_AC3;
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audio = TRUE;
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} else if (!strcmp (mimetype, "audio/x-eac3")) {
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id = CODEC_ID_EAC3;
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audio = TRUE;
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} else if (!strcmp (mimetype, "audio/atrac3")) {
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id = CODEC_ID_ATRAC3;
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audio = TRUE;
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@ -2821,6 +2872,9 @@ gst_ffmpeg_caps_to_codecid (const GstCaps * caps, AVCodecContext * context)
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} else if (!strcmp (mimetype, "audio/AMR-WB")) {
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id = CODEC_ID_AMR_WB;
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audio = TRUE;
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} else if (!strcmp (mimetype, "audio/qcelp")) {
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id = CODEC_ID_QCELP;
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audio = TRUE;
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} else if (!strcmp (mimetype, "video/x-h264")) {
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id = CODEC_ID_H264;
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video = TRUE;
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@ -3236,3 +3290,28 @@ gst_ffmpeg_avpicture_fill (AVPicture * picture,
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return 0;
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}
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gint
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av_smp_format_depth (enum SampleFormat smp_fmt)
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{
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gint depth = -1;
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switch (smp_fmt) {
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case SAMPLE_FMT_U8:
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depth = 1;
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break;
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case SAMPLE_FMT_S16:
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depth = 2;
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break;
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case SAMPLE_FMT_S32:
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case SAMPLE_FMT_FLT:
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depth = 4;
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break;
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case SAMPLE_FMT_DBL:
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depth = 8;
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break;
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default:
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GST_ERROR ("UNHANDLED SAMPLE FORMAT !");
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break;
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}
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return depth;
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}
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@ -163,4 +163,7 @@ gst_ffmpeg_time_gst_to_ff (guint64 time, AVRational base)
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void
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gst_ffmpeg_init_pix_fmt_info();
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gint
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av_smp_format_depth(enum SampleFormat smp_fmt);
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#endif /* __GST_FFMPEG_CODECMAP_H__ */
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@ -67,6 +67,7 @@ struct _GstFFMpegDec
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{
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gint channels;
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gint samplerate;
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gint depth;
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} audio;
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} format;
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gboolean waiting_for_key;
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@ -587,6 +588,7 @@ gst_ffmpegdec_open (GstFFMpegDec * ffmpegdec)
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case CODEC_TYPE_AUDIO:
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ffmpegdec->format.audio.samplerate = 0;
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ffmpegdec->format.audio.channels = 0;
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ffmpegdec->format.audio.depth = 0;
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break;
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default:
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break;
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@ -707,8 +709,7 @@ gst_ffmpegdec_setcaps (GstPad * pad, GstCaps * caps)
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GST_DEBUG_OBJECT (ffmpegdec, "enabled direct rendering");
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ffmpegdec->current_dr = TRUE;
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}
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}
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else {
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} else {
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GST_DEBUG_OBJECT (ffmpegdec, "direct rendering not supported");
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}
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}
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@ -1075,16 +1076,22 @@ gst_ffmpegdec_negotiate (GstFFMpegDec * ffmpegdec)
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ffmpegdec->format.video.pix_fmt = ffmpegdec->context->pix_fmt;
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break;
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case CODEC_TYPE_AUDIO:
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{
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gint depth = av_smp_format_depth (ffmpegdec->context->sample_fmt);
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if (ffmpegdec->format.audio.samplerate ==
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ffmpegdec->context->sample_rate &&
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ffmpegdec->format.audio.channels == ffmpegdec->context->channels)
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ffmpegdec->format.audio.channels == ffmpegdec->context->channels &&
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ffmpegdec->format.audio.depth == depth)
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return TRUE;
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GST_DEBUG_OBJECT (ffmpegdec,
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"Renegotiating audio from %dHz@%dchannels to %dHz@%dchannels",
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"Renegotiating audio from %dHz@%dchannels (%d) to %dHz@%dchannels (%d)",
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ffmpegdec->format.audio.samplerate, ffmpegdec->format.audio.channels,
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ffmpegdec->context->sample_rate, ffmpegdec->context->channels);
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ffmpegdec->format.audio.depth,
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ffmpegdec->context->sample_rate, ffmpegdec->context->channels, depth);
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ffmpegdec->format.audio.samplerate = ffmpegdec->context->sample_rate;
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ffmpegdec->format.audio.channels = ffmpegdec->context->channels;
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ffmpegdec->format.audio.depth = depth;
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}
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break;
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default:
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break;
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@ -1738,7 +1745,7 @@ clip_audio_buffer (GstFFMpegDec * dec, GstBuffer * buf, GstClockTime in_ts,
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/* bring clipped time to bytes */
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diff =
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gst_util_uint64_scale_int (diff, dec->format.audio.samplerate,
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GST_SECOND) * (2 * dec->format.audio.channels);
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GST_SECOND) * (dec->format.audio.depth * dec->format.audio.channels);
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GST_DEBUG_OBJECT (dec, "clipping start to %" GST_TIME_FORMAT " %"
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G_GINT64_FORMAT " bytes", GST_TIME_ARGS (ctime), diff);
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@ -1750,7 +1757,7 @@ clip_audio_buffer (GstFFMpegDec * dec, GstBuffer * buf, GstClockTime in_ts,
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/* bring clipped time to bytes */
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diff =
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gst_util_uint64_scale_int (diff, dec->format.audio.samplerate,
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GST_SECOND) * (2 * dec->format.audio.channels);
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GST_SECOND) * (dec->format.audio.depth * dec->format.audio.channels);
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GST_DEBUG_OBJECT (dec, "clipping stop to %" GST_TIME_FORMAT " %"
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G_GINT64_FORMAT " bytes", GST_TIME_ARGS (cstop), diff);
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@ -1824,7 +1831,8 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
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* 1) calculate based on number of samples
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*/
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in_duration = gst_util_uint64_scale_int (have_data, GST_SECOND,
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2 * ffmpegdec->context->channels * ffmpegdec->context->sample_rate);
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ffmpegdec->format.audio.depth * ffmpegdec->format.audio.channels *
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ffmpegdec->format.audio.samplerate);
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GST_DEBUG_OBJECT (ffmpegdec,
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"Buffer created. Size:%d , timestamp:%" GST_TIME_FORMAT " , duration:%"
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