Attachment #396300 for bug #539914




      "libavutil",
  ]
  if os.path.exists("./ffmpeg"):
    FFMPEG_LIBS.append('libswresample')
    pkg_path = list(set(map(os.path.dirname, glob('./ffmpeg/*/*.pc'))))
    pkg_path.append(os.environ.get('PKG_CONFIG_PATH', ''))
    os.environ['PKG_CONFIG_PATH'] = ':'.join(pkg_path)

    env.Append(CCFLAGS=[
      '-Iffmpeg'
    ])
 
  if conf.CheckPKG('libavresample'):
    FFMPEG_LIBS.append('libavresample')
  else:
    FFMPEG_LIBS.append('libswresample')
    env.Append(CCFLAGS=[
      '-DUSE_SWRESAMPLE'
    ])

  if not conf.CheckPKG(' '.join(FFMPEG_LIBS)): 
    print """

Lines 33-38 Link Here

(-)a/src/ffmpeg2theora.c (-49 / +103 lines)
33	#include "libswscale/swscale.h"	33	#include "libswscale/swscale.h"
34	#include "libpostproc/postprocess.h"	34	#include "libpostproc/postprocess.h"
35		35
		36	#include "libavutil/opt.h"
		37	#include "libavutil/channel_layout.h"
		38	#include "libavutil/samplefmt.h"
		39	#include "libswresample_compat.h"
		40
36	#include "theora/theoraenc.h"	41	#include "theora/theoraenc.h"
37	#include "vorbis/codec.h"	42	#include "vorbis/codec.h"
38	#include "vorbis/vorbisenc.h"	43	#include "vorbis/vorbisenc.h"
Lines 537-542 void ff2theora_output(ff2theora this) { Link Here
537	int synced = this->start_time == 0.0;	542	int synced = this->start_time == 0.0;
538	AVRational display_aspect_ratio, sample_aspect_ratio;	543	AVRational display_aspect_ratio, sample_aspect_ratio;
539		544
		545	struct SwrContext *swr_ctx;
		546	uint8_t **dst_audio_data = NULL;
		547	int dst_linesize;
		548	int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
		549
540	if (this->audiostream >= 0 && this->context->nb_streams > this->audiostream) {	550	if (this->audiostream >= 0 && this->context->nb_streams > this->audiostream) {
541	AVCodecContext *enc = this->context->streams[this->audiostream]->codec;	551	AVCodecContext *enc = this->context->streams[this->audiostream]->codec;
542	if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {	552	if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {
Lines 962-983 void ff2theora_output(ff2theora this) { Link Here
962	if (acodec != NULL && avcodec_open2 (aenc, acodec, NULL) >= 0) {	972	if (acodec != NULL && avcodec_open2 (aenc, acodec, NULL) >= 0) {
963	if (this->sample_rate != sample_rate	973	if (this->sample_rate != sample_rate
964	\|\| this->channels != aenc->channels	974	\|\| this->channels != aenc->channels
965	\|\| aenc->sample_fmt != AV_SAMPLE_FMT_S16) {	975	\|\| aenc->sample_fmt != AV_SAMPLE_FMT_FLTP) {
966	// values take from libavcodec/resample.c	976	swr_ctx = swr_alloc();
967	this->audio_resample_ctx = av_audio_resample_init(this->channels, aenc->channels,	977	/* set options */
968	this->sample_rate, sample_rate,	978	if (aenc->channel_layout) {
969	AV_SAMPLE_FMT_S16, aenc->sample_fmt,	979	av_opt_set_int(swr_ctx, "in_channel_layout", aenc->channel_layout, 0);
970	16, 10, 0, 0.8);	980	} else {
971	if (!this->audio_resample_ctx) {	981	av_opt_set_int(swr_ctx, "in_channel_layout", av_get_default_channel_layout(aenc->channels), 0);
972	this->channels = aenc->channels;	982	}
		983	av_opt_set_int(swr_ctx, "in_sample_rate", aenc->sample_rate, 0);
		984	av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", aenc->sample_fmt, 0);
		985
		986	av_opt_set_int(swr_ctx, "out_channel_layout", av_get_default_channel_layout(this->channels), 0);
		987	av_opt_set_int(swr_ctx, "out_sample_rate", this->sample_rate, 0);
		988	av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
		989
		990	/* initialize the resampling context */
		991	if (swr_init(swr_ctx) < 0) {
		992	fprintf(stderr, "Failed to initialize the resampling context\n");
		993	exit(1);
973	}	994	}
		995
		996	max_dst_nb_samples = dst_nb_samples =
		997	av_rescale_rnd(src_nb_samples, this->sample_rate, sample_rate, AV_ROUND_UP);
		998
		999	if (av_samples_alloc_array_and_samples(&dst_audio_data, &dst_linesize, this->channels,
		1000	dst_nb_samples, AV_SAMPLE_FMT_FLTP, 0) < 0) {
		1001	fprintf(stderr, "Could not allocate destination samples\n");
		1002	exit(1);
		1003	}
		1004
974	if (!info.frontend && this->sample_rate!=sample_rate)	1005	if (!info.frontend && this->sample_rate!=sample_rate)
975	fprintf(stderr, " Resample: %dHz => %dHz\n", sample_rate,this->sample_rate);	1006	fprintf(stderr, " Resample: %dHz => %dHz\n", sample_rate,this->sample_rate);
976	if (!info.frontend && this->channels!=aenc->channels)	1007	if (!info.frontend && this->channels!=aenc->channels)
977	fprintf(stderr, " Channels: %d => %d\n",aenc->channels,this->channels);	1008	fprintf(stderr, " Channels: %d => %d\n",aenc->channels,this->channels);
978	}	1009	}
979	else{	1010	else{
980	this->audio_resample_ctx=NULL;	1011	swr_ctx = NULL;
981	}	1012	}
982	}	1013	}
983	else{	1014	else{
Lines 1068-1080 void ff2theora_output(ff2theora this) { Link Here
1068	AVPacket pkt;	1099	AVPacket pkt;
1069	AVPacket avpkt;	1100	AVPacket avpkt;
1070	int len1;	1101	int len1;
1071	int got_picture;	1102	int got_frame;
1072	int first = 1;	1103	int first = 1;
1073	int audio_eos = 0, video_eos = 0, audio_done = 0, video_done = 0;	1104	int audio_eos = 0, video_eos = 0, audio_done = 0, video_done = 0;
1074	int ret;	1105	int ret;
1075	int16_t audio_buf=av_malloc(4MAX_AUDIO_FRAME_SIZE);	1106	AVFrame *audio_frame = NULL;
1076	int16_t resampled=av_malloc(4MAX_AUDIO_FRAME_SIZE);	1107	uint8_t **audio_p = NULL;
1077	int16_t *audio_p=NULL;
1078	int no_frames;	1108	int no_frames;
1079	int no_samples;	1109	int no_samples;
1080		1110
Lines 1370-1376 void ff2theora_output(ff2theora this) { Link Here
1370	first frame decodec in case its not a keyframe	1400	first frame decodec in case its not a keyframe
1371	*/	1401	*/
1372	if (pkt.stream_index == this->video_index) {	1402	if (pkt.stream_index == this->video_index) {
1373	avcodec_decode_video2(venc, frame, &got_picture, &pkt);	1403	avcodec_decode_video2(venc, frame, &got_frame, &pkt);
1374	}	1404	}
1375	av_free_packet (&pkt);	1405	av_free_packet (&pkt);
1376	continue;	1406	continue;
Lines 1389-1397 void ff2theora_output(ff2theora this) { Link Here
1389	while(video_eos \|\| avpkt.size > 0) {	1419	while(video_eos \|\| avpkt.size > 0) {
1390	int dups = 0;	1420	int dups = 0;
1391	static th_ycbcr_buffer ycbcr;	1421	static th_ycbcr_buffer ycbcr;
1392	len1 = avcodec_decode_video2(venc, frame, &got_picture, &avpkt);	1422	len1 = avcodec_decode_video2(venc, frame, &got_frame, &avpkt);
1393	if (len1>=0) {	1423	if (len1>=0) {
1394	if (got_picture) {	1424	if (got_frame) {
1395	// this is disabled by default since it does not work	1425	// this is disabled by default since it does not work
1396	// for all input formats the way it should.	1426	// for all input formats the way it should.
1397	if (this->sync == 1 && pkt.dts != AV_NOPTS_VALUE) {	1427	if (this->sync == 1 && pkt.dts != AV_NOPTS_VALUE) {
Lines 1428-1434 void ff2theora_output(ff2theora this) { Link Here
1428		1458
1429	if (venc_pix_fmt != this->pix_fmt) {	1459	if (venc_pix_fmt != this->pix_fmt) {
1430	sws_scale(this->sws_colorspace_ctx,	1460	sws_scale(this->sws_colorspace_ctx,
1431	frame->data, frame->linesize, 0, display_height,	1461	(const uint8_t * const*)frame->data, frame->linesize, 0, display_height,
1432	output_tmp->data, output_tmp->linesize);	1462	output_tmp->data, output_tmp->linesize);
1433	}	1463	}
1434	else{	1464	else{
Lines 1472-1478 void ff2theora_output(ff2theora this) { Link Here
1472	}	1502	}
1473	if (this->sws_scale_ctx) {	1503	if (this->sws_scale_ctx) {
1474	sws_scale(this->sws_scale_ctx,	1504	sws_scale(this->sws_scale_ctx,
1475	output_cropped->data,	1505	(const uint8_t * const*)output_cropped->data,
1476	output_cropped->linesize, 0,	1506	output_cropped->linesize, 0,
1477	display_height - (this->frame_topBand + this->frame_bottomBand),	1507	display_height - (this->frame_topBand + this->frame_bottomBand),
1478	output_resized->data,	1508	output_resized->data,
Lines 1500-1506 void ff2theora_output(ff2theora this) { Link Here
1500	//now output_resized	1530	//now output_resized
1501		1531
1502	if (!first) {	1532	if (!first) {
1503	if (got_picture \|\| video_eos) {	1533	if (got_frame \|\| video_eos) {
1504	prepare_ycbcr_buffer(this, ycbcr, output_buffered);	1534	prepare_ycbcr_buffer(this, ycbcr, output_buffered);
1505	if(dups>0) {	1535	if(dups>0) {
1506	//this only works if dups < keyint,	1536	//this only works if dups < keyint,
Lines 1520-1530 void ff2theora_output(ff2theora this) { Link Here
1520	info.videotime = this->frame_count / av_q2d(this->framerate);	1550	info.videotime = this->frame_count / av_q2d(this->framerate);
1521	}	1551	}
1522	}	1552	}
1523	if (got_picture) {	1553	if (got_frame) {
1524	first=0;	1554	first=0;
1525	av_picture_copy((AVPicture )output_buffered, (AVPicture )output_padded, this->pix_fmt, this->frame_width, this->frame_height);	1555	av_picture_copy((AVPicture )output_buffered, (AVPicture )output_padded, this->pix_fmt, this->frame_width, this->frame_height);
1526	}	1556	}
1527	if (!got_picture) {	1557	if (!got_frame) {
1528	break;	1558	break;
1529	}	1559	}
1530	}	1560	}
Lines 1532-1573 void ff2theora_output(ff2theora this) { Link Here
1532	if (info.passno!=1)	1562	if (info.passno!=1)
1533	if ((audio_eos && !audio_done) \|\| (ret >= 0 && pkt.stream_index == this->audio_index)) {	1563	if ((audio_eos && !audio_done) \|\| (ret >= 0 && pkt.stream_index == this->audio_index)) {
1534	while((audio_eos && !audio_done) \|\| avpkt.size > 0 ) {	1564	while((audio_eos && !audio_done) \|\| avpkt.size > 0 ) {
1535	int samples=0;
1536	int samples_out=0;
1537	int data_size = 4*MAX_AUDIO_FRAME_SIZE;
1538	int bytes_per_sample = av_get_bytes_per_sample(aenc->sample_fmt);	1565	int bytes_per_sample = av_get_bytes_per_sample(aenc->sample_fmt);
1539		1566
1540	if (avpkt.size > 0) {	1567	if (avpkt.size > 0) {
1541	len1 = avcodec_decode_audio3(astream->codec, audio_buf, &data_size, &avpkt);	1568	if (!audio_frame && !(audio_frame = avcodec_alloc_frame())) {
		1569	fprintf(stderr, "Failed to allocate memory\n");
		1570	exit(1);
		1571	}
		1572	len1 = avcodec_decode_audio4(astream->codec, audio_frame, &got_frame, &avpkt);
1542	if (len1 < 0) {	1573	if (len1 < 0) {
1543	/* if error, we skip the frame */	1574	/* if error, we skip the frame */
1544	break;	1575	break;
1545	}	1576	}
1546	avpkt.size -= len1;	1577	/* Some audio decoders decode only part of the packet, and have to be
1547	avpkt.data += len1;	1578	* called again with the remainder of the packet data.
1548	if (data_size >0) {	1579	* Sample: http://fate-suite.libav.org/lossless-audio/luckynight-partial.shn
1549	samples = data_size / (aenc->channels * bytes_per_sample);	1580	* Also, some decoders might over-read the packet. */
1550	samples_out = samples;	1581	len1 = FFMIN(len1, avpkt.size);
1551	if (this->audio_resample_ctx) {	1582	if (got_frame) {
1552	samples_out = audio_resample(this->audio_resample_ctx, resampled, audio_buf, samples);	1583	dst_nb_samples = audio_frame->nb_samples;
1553	audio_p = resampled;	1584	if (swr_ctx) {
		1585	dst_nb_samples = av_rescale_rnd(audio_frame->nb_samples,
		1586	this->sample_rate, aenc->sample_rate, AV_ROUND_UP);
		1587	if (dst_nb_samples > max_dst_nb_samples) {
		1588	av_free(dst_audio_data[0]);
		1589	if (av_samples_alloc(dst_audio_data, &dst_linesize, this->channels,
		1590	dst_nb_samples, AV_SAMPLE_FMT_FLTP, 1) < 0) {
		1591	fprintf(stderr, "Error while converting audio\n");
		1592	exit(1);
		1593	}
		1594	max_dst_nb_samples = dst_nb_samples;
		1595	}
		1596	if (swr_convert(swr_ctx, dst_audio_data, dst_nb_samples,
		1597	(const uint8_t**)audio_frame->extended_data, audio_frame->nb_samples) < 0) {
		1598	fprintf(stderr, "Error while converting audio\n");
		1599	exit(1);
		1600	}
		1601	audio_p = dst_audio_data;
		1602	} else {
		1603	audio_p = audio_frame->extended_data;
1554	}	1604	}
1555	else
1556	audio_p = audio_buf;
1557	}	1605	}
		1606	avpkt.size -= len1;
		1607	avpkt.data += len1;
1558	}	1608	}
1559		1609	if(got_frame \|\| audio_eos) {
1560	if (no_samples > 0 && this->sample_count + samples_out > no_samples) {	1610	if (no_samples > 0 && this->sample_count + dst_nb_samples > no_samples) {
1561	audio_eos = 1;	1611	audio_eos = 1;
1562	samples_out = no_samples - this->sample_count;	1612	dst_nb_samples = no_samples - this->sample_count;
1563	if (samples_out <= 0) {	1613	if (dst_nb_samples <= 0) {
1564	break;	1614	break;
		1615	}
1565	}	1616	}
		1617	oggmux_add_audio(&info, audio_p, dst_nb_samples, audio_eos);
		1618	avcodec_free_frame(&audio_frame);
		1619	this->sample_count += dst_nb_samples;
1566	}	1620	}
1567
1568	oggmux_add_audio(&info, audio_p,
1569	samples_out * (this->channels), samples_out, audio_eos);
1570	this->sample_count += samples_out;
1571	if(audio_eos) {	1621	if(audio_eos) {
1572	audio_done = 1;	1622	audio_done = 1;
1573	}	1623	}
Lines 1752-1759 void ff2theora_output(ff2theora this) { Link Here
1752	avcodec_close(venc);	1802	avcodec_close(venc);
1753	}	1803	}
1754	if (this->audio_index >= 0) {	1804	if (this->audio_index >= 0) {
1755	if (this->audio_resample_ctx)	1805	if (swr_ctx)
1756	audio_resample_close(this->audio_resample_ctx);	1806	swr_free(&swr_ctx);
1757	avcodec_close(aenc);	1807	avcodec_close(aenc);
1758	}	1808	}
1759		1809
Lines 1774-1781 void ff2theora_output(ff2theora this) { Link Here
1774	frame_dealloc(output_cropped_p);	1824	frame_dealloc(output_cropped_p);
1775	frame_dealloc(output_padded_p);	1825	frame_dealloc(output_padded_p);
1776	}	1826	}
1777	av_free(audio_buf);	1827	if (dst_audio_data)
1778	av_free(resampled);	1828	av_freep(&dst_audio_data[0]);
		1829	av_freep(&dst_audio_data);
		1830	if(swr_ctx) {
		1831	swr_close(swr_ctx);
		1832	}
1779	}	1833	}
1780	else{	1834	else{
1781	fprintf(stderr, "No video or audio stream found.\n");	1835	fprintf(stderr, "No video or audio stream found.\n");

Lines 62-68 typedef struct ff2theora{ Link Here

(-)a/src/ffmpeg2theora.h (-1 lines)
62	double fps;	62	double fps;
63	struct SwsContext sws_colorspace_ctx; / for image resampling/resizing */	63	struct SwsContext sws_colorspace_ctx; / for image resampling/resizing */
64	struct SwsContext sws_scale_ctx; / for image resampling/resizing */	64	struct SwsContext sws_scale_ctx; / for image resampling/resizing */
65	ReSampleContext *audio_resample_ctx;
66	ogg_int32_t aspect_numerator;	65	ogg_int32_t aspect_numerator;
67	ogg_int32_t aspect_denominator;	66	ogg_int32_t aspect_denominator;
68	int colorspace;	67	int colorspace;




// This header serves to smooth out the differences in FFmpeg and LibAV.

#ifdef USE_SWRESAMPLE

    #include <libswresample/swresample.h>

    //swr does not have the equivalent so this does nothing
    void swr_close(SwrContext *ctx) {};

#else

    #include <libavresample/avresample.h>

    #define SwrContext AVAudioResampleContext
    #define swr_init(ctx) avresample_open(ctx)
    #define swr_close(ctx) avresample_close(ctx)
    #define swr_free(ctx) avresample_free(ctx)
    #define swr_alloc() avresample_alloc_context()
    #define swr_get_delay(ctx, ...) avresample_get_delay(ctx)
    #define swr_convert(ctx, out, out_count, in, in_count) \
       avresample_convert(ctx, out, 0, out_count, (uint8_t **)in, 0, in_count)

#endif




/**
 * adds audio samples to encoding sink
 * @param buffer pointer to buffer
 * @param bytes bytes in buffer
 * @param samples samples in buffer
 * @param e_o_s 1 indicates end of stream.
 */
void oggmux_add_audio (oggmux_info *info, uint8_t **buffer, int samples, int e_o_s) {
    ogg_packet op;

    int i, j, k, count = 0;
    float **vorbis_buffer;

    if (samples <= 0) {
        /* end of audio stream */
        if (e_o_s)
            vorbis_analysis_wrote (&info->vd, 0);

                        default: k = j;
                    }
                }
                vorbis_buffer[k][i] = ((const float  *)buffer[j])[i];
            }
        }
        vorbis_analysis_wrote (&info->vd, samples);

                if (op.packetno != 4) {
                    /* We only expect negative start granule in the first content
                       packet, not any of the others... */
                    fprintf(stderr, "WARNING: vorbis packet %" PRId64 " has calculated start"
                            " granule of %" PRId64 ", but it should be non-negative!",
                            op.packetno, start_granule);
                }
                start_granule = 0;

                   allowed by the specification in the last packet only, and the
                   trailing samples should be discarded and not played/indexed. */
                if (!op.e_o_s) {
                    fprintf(stderr, "WARNING: vorbis packet %" PRId64 " (granulepos %" PRId64 ") starts before"
                            " the end of the preceeding packet!", op.packetno, op.granulepos);
                }
                start_granule = info->vorbis_granulepos;

Lines 168-174 void init_info(oggmux_info *info); Link Here

(-)a/src/theorautils.h (-2 / +1 lines)
168	extern void oggmux_setup_kate_streams(oggmux_info *info, int n_kate_streams);	168	extern void oggmux_setup_kate_streams(oggmux_info *info, int n_kate_streams);
169	extern void oggmux_init (oggmux_info *info);	169	extern void oggmux_init (oggmux_info *info);
170	extern void oggmux_add_video (oggmux_info *info, th_ycbcr_buffer ycbcr, int e_o_s);	170	extern void oggmux_add_video (oggmux_info *info, th_ycbcr_buffer ycbcr, int e_o_s);
171	extern void oggmux_add_audio (oggmux_info info, int16_t readbuffer, int bytesread, int samplesread,int e_o_s);	171	extern void oggmux_add_audio (oggmux_info info, uint8_t *buffer, int samples,int e_o_s);
172	#ifdef HAVE_KATE	172	#ifdef HAVE_KATE
173	extern void oggmux_add_kate_text (oggmux_info info, int idx, double t0, double t1, const char text, size_t len, int x1, int x2, int y1, int y2);	173	extern void oggmux_add_kate_text (oggmux_info info, int idx, double t0, double t1, const char text, size_t len, int x1, int x2, int y1, int y2);
174	extern void oggmux_add_kate_image (oggmux_info info, int idx, double t0, double t1, const kate_region kr, const kate_palette kp, const kate_bitmap kb);	174	extern void oggmux_add_kate_image (oggmux_info info, int idx, double t0, double t1, const kate_region kr, const kate_palette kp, const kate_bitmap kb);
175	-

Return to bug 539914

Lines 1219-1235 vorbis_time(vorbis_dsp_state * dsp, ogg_int64_t granulepos) { Link Here

(-)a/src/theorautils.c (-7 / +6 lines)
1219	/**	1219	/**
1220	* adds audio samples to encoding sink	1220	* adds audio samples to encoding sink
1221	* @param buffer pointer to buffer	1221	* @param buffer pointer to buffer
1222	* @param bytes bytes in buffer
1223	* @param samples samples in buffer	1222	* @param samples samples in buffer
1224	* @param e_o_s 1 indicates end of stream.	1223	* @param e_o_s 1 indicates end of stream.
1225	*/	1224	*/
1226	void oggmux_add_audio (oggmux_info info, int16_t buffer, int bytes, int samples, int e_o_s) {	1225	void oggmux_add_audio (oggmux_info info, uint8_t *buffer, int samples, int e_o_s) {
1227	ogg_packet op;	1226	ogg_packet op;
1228		1227
1229	int i, j, k, count = 0;	1228	int i, j, k, count = 0;
1230	float **vorbis_buffer;	1229	float **vorbis_buffer;
1231		1230
1232	if (bytes <= 0 && samples <= 0) {	1231	if (samples <= 0) {
1233	/* end of audio stream */	1232	/* end of audio stream */
1234	if (e_o_s)	1233	if (e_o_s)
1235	vorbis_analysis_wrote (&info->vd, 0);	1234	vorbis_analysis_wrote (&info->vd, 0);
Lines 1252-1258 void oggmux_add_audio (oggmux_info info, int16_t buffer, int bytes, int sampl Link Here
1252	default: k = j;	1251	default: k = j;
1253	}	1252	}
1254	}	1253	}
1255	vorbis_buffer[k][i] = buffer[count++] / 32768.f;	1254	vorbis_buffer[k][i] = ((const float *)buffer[j])[i];
1256	}	1255	}
1257	}	1256	}
1258	vorbis_analysis_wrote (&info->vd, samples);	1257	vorbis_analysis_wrote (&info->vd, samples);
Lines 1291-1298 void oggmux_add_audio (oggmux_info info, int16_t buffer, int bytes, int sampl Link Here
1291	if (op.packetno != 4) {	1290	if (op.packetno != 4) {
1292	/* We only expect negative start granule in the first content	1291	/* We only expect negative start granule in the first content
1293	packet, not any of the others... */	1292	packet, not any of the others... */
1294	fprintf(stderr, "WARNING: vorbis packet %lld has calculated start"	1293	fprintf(stderr, "WARNING: vorbis packet %" PRId64 " has calculated start"
1295	" granule of %lld, but it should be non-negative!",	1294	" granule of %" PRId64 ", but it should be non-negative!",
1296	op.packetno, start_granule);	1295	op.packetno, start_granule);
1297	}	1296	}
1298	start_granule = 0;	1297	start_granule = 0;
Lines 1302-1308 void oggmux_add_audio (oggmux_info info, int16_t buffer, int bytes, int sampl Link Here
1302	allowed by the specification in the last packet only, and the	1301	allowed by the specification in the last packet only, and the
1303	trailing samples should be discarded and not played/indexed. */	1302	trailing samples should be discarded and not played/indexed. */
1304	if (!op.e_o_s) {	1303	if (!op.e_o_s) {
1305	fprintf(stderr, "WARNING: vorbis packet %lld (granulepos %lld) starts before"	1304	fprintf(stderr, "WARNING: vorbis packet %" PRId64 " (granulepos %" PRId64 ") starts before"
1306	" the end of the preceeding packet!", op.packetno, op.granulepos);	1305	" the end of the preceeding packet!", op.packetno, op.granulepos);
1307	}	1306	}
1308	start_granule = info->vorbis_granulepos;	1307	start_granule = info->vorbis_granulepos;

Lines 155-161 if not env.GetOption('clean'): Link Here

(-)a/SConstruct (-1 / +8 lines)
155	"libavutil",	155	"libavutil",
156	]	156	]
157	if os.path.exists("./ffmpeg"):	157	if os.path.exists("./ffmpeg"):
158	FFMPEG_LIBS.append('libswresample')
159	pkg_path = list(set(map(os.path.dirname, glob('./ffmpeg//.pc'))))	158	pkg_path = list(set(map(os.path.dirname, glob('./ffmpeg//.pc'))))
160	pkg_path.append(os.environ.get('PKG_CONFIG_PATH', ''))	159	pkg_path.append(os.environ.get('PKG_CONFIG_PATH', ''))
161	os.environ['PKG_CONFIG_PATH'] = ':'.join(pkg_path)	160	os.environ['PKG_CONFIG_PATH'] = ':'.join(pkg_path)
Lines 168-173 if not env.GetOption('clean'): Link Here
168	env.Append(CCFLAGS=[	167	env.Append(CCFLAGS=[
169	'-Iffmpeg'	168	'-Iffmpeg'
170	])	169	])
		170
		171	if conf.CheckPKG('libavresample'):
		172	FFMPEG_LIBS.append('libavresample')
		173	else:
		174	FFMPEG_LIBS.append('libswresample')
		175	env.Append(CCFLAGS=[
		176	'-DUSE_SWRESAMPLE'
		177	])
171		178
172	if not conf.CheckPKG(' '.join(FFMPEG_LIBS)):	179	if not conf.CheckPKG(' '.join(FFMPEG_LIBS)):
173	print """	180	print """

Line 0 Link Here

(-)a/src/libswresample_compat.h (+23 lines)
	1	// This header serves to smooth out the differences in FFmpeg and LibAV.
	2
	3	#ifdef USE_SWRESAMPLE
	4
	5	#include <libswresample/swresample.h>
	6
	7	//swr does not have the equivalent so this does nothing
	8	void swr_close(SwrContext *ctx) {};
	9
	10	#else
	11
	12	#include <libavresample/avresample.h>
	13
	14	#define SwrContext AVAudioResampleContext
	15	#define swr_init(ctx) avresample_open(ctx)
	16	#define swr_close(ctx) avresample_close(ctx)
	17	#define swr_free(ctx) avresample_free(ctx)
	18	#define swr_alloc() avresample_alloc_context()
	19	#define swr_get_delay(ctx, ...) avresample_get_delay(ctx)
	20	#define swr_convert(ctx, out, out_count, in, in_count) \
	21	avresample_convert(ctx, out, 0, out_count, (uint8_t **)in, 0, in_count)
	22
	23	#endif