diff options
Diffstat (limited to 'multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff')
| -rw-r--r-- | multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff | 405 |
1 files changed, 0 insertions, 405 deletions
diff --git a/multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff b/multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff deleted file mode 100644 index 2a50e2d317..0000000000 --- a/multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff +++ /dev/null @@ -1,405 +0,0 @@ -diff -Naur ffmpeg2theora-0.29/SConstruct ffmpeg2theora-0.29.patched/SConstruct ---- ffmpeg2theora-0.29/SConstruct 2012-06-25 13:15:16.000000000 -0400 -+++ ffmpeg2theora-0.29.patched/SConstruct 2014-05-14 15:02:17.000000000 -0400 -@@ -162,6 +162,14 @@ - '-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 """ - Could not find %s. -diff -Naur ffmpeg2theora-0.29/src/ffmpeg2theora.c ffmpeg2theora-0.29.patched/src/ffmpeg2theora.c ---- ffmpeg2theora-0.29/src/ffmpeg2theora.c 2014-05-14 14:57:30.000000000 -0400 -+++ ffmpeg2theora-0.29.patched/src/ffmpeg2theora.c 2014-05-14 14:59:43.000000000 -0400 -@@ -33,6 +33,11 @@ - #include "libswscale/swscale.h" - #include "libpostproc/postprocess.h" - -+#include "libavutil/opt.h" -+#include "libavutil/channel_layout.h" -+#include "libavutil/samplefmt.h" -+#include "libswresample_compat.h" -+ - #include "theora/theoraenc.h" - #include "vorbis/codec.h" - #include "vorbis/vorbisenc.h" -@@ -537,6 +542,11 @@ - int synced = this->start_time == 0.0; - AVRational display_aspect_ratio, sample_aspect_ratio; - -+ struct SwrContext *swr_ctx; -+ uint8_t **dst_audio_data = NULL; -+ int dst_linesize; -+ int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples; -+ - if (this->audiostream >= 0 && this->context->nb_streams > this->audiostream) { - AVCodecContext *enc = this->context->streams[this->audiostream]->codec; - if (enc->codec_type == AVMEDIA_TYPE_AUDIO) { -@@ -961,22 +971,43 @@ - if (acodec != NULL && avcodec_open2 (aenc, acodec, NULL) >= 0) { - if (this->sample_rate != sample_rate - || this->channels != aenc->channels -- || aenc->sample_fmt != AV_SAMPLE_FMT_S16) { -- // values take from libavcodec/resample.c -- this->audio_resample_ctx = av_audio_resample_init(this->channels, aenc->channels, -- this->sample_rate, sample_rate, -- AV_SAMPLE_FMT_S16, aenc->sample_fmt, -- 16, 10, 0, 0.8); -- if (!this->audio_resample_ctx) { -- this->channels = aenc->channels; -+ || aenc->sample_fmt != AV_SAMPLE_FMT_FLTP) { -+ swr_ctx = swr_alloc(); -+ /* set options */ -+ if (aenc->channel_layout) { -+ av_opt_set_int(swr_ctx, "in_channel_layout", aenc->channel_layout, 0); -+ } else { -+ av_opt_set_int(swr_ctx, "in_channel_layout", av_get_default_channel_layout(aenc->channels), 0); -+ } -+ av_opt_set_int(swr_ctx, "in_sample_rate", aenc->sample_rate, 0); -+ av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", aenc->sample_fmt, 0); -+ -+ av_opt_set_int(swr_ctx, "out_channel_layout", av_get_default_channel_layout(this->channels), 0); -+ av_opt_set_int(swr_ctx, "out_sample_rate", this->sample_rate, 0); -+ av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); -+ -+ /* initialize the resampling context */ -+ if (swr_init(swr_ctx) < 0) { -+ fprintf(stderr, "Failed to initialize the resampling context\n"); -+ exit(1); - } -+ -+ max_dst_nb_samples = dst_nb_samples = -+ av_rescale_rnd(src_nb_samples, this->sample_rate, sample_rate, AV_ROUND_UP); -+ -+ if (av_samples_alloc_array_and_samples(&dst_audio_data, &dst_linesize, this->channels, -+ dst_nb_samples, AV_SAMPLE_FMT_FLTP, 0) < 0) { -+ fprintf(stderr, "Could not allocate destination samples\n"); -+ exit(1); -+ } -+ - if (!info.frontend && this->sample_rate!=sample_rate) - fprintf(stderr, " Resample: %dHz => %dHz\n", sample_rate,this->sample_rate); - if (!info.frontend && this->channels!=aenc->channels) - fprintf(stderr, " Channels: %d => %d\n",aenc->channels,this->channels); - } - else{ -- this->audio_resample_ctx=NULL; -+ swr_ctx = NULL; - } - } - else{ -@@ -1067,13 +1098,12 @@ - AVPacket pkt; - AVPacket avpkt; - int len1; -- int got_picture; -+ int got_frame; - int first = 1; - int audio_eos = 0, video_eos = 0, audio_done = 0, video_done = 0; - int ret; -- int16_t *audio_buf=av_malloc(4*MAX_AUDIO_FRAME_SIZE); -- int16_t *resampled=av_malloc(4*MAX_AUDIO_FRAME_SIZE); -- int16_t *audio_p=NULL; -+ AVFrame *audio_frame = NULL; -+ uint8_t **audio_p = NULL; - int no_frames; - int no_samples; - -@@ -1369,7 +1399,7 @@ - first frame decodec in case its not a keyframe - */ - if (pkt.stream_index == this->video_index) { -- avcodec_decode_video2(venc, frame, &got_picture, &pkt); -+ avcodec_decode_video2(venc, frame, &got_frame, &pkt); - } - av_free_packet (&pkt); - continue; -@@ -1388,9 +1418,9 @@ - while(video_eos || avpkt.size > 0) { - int dups = 0; - static th_ycbcr_buffer ycbcr; -- len1 = avcodec_decode_video2(venc, frame, &got_picture, &avpkt); -+ len1 = avcodec_decode_video2(venc, frame, &got_frame, &avpkt); - if (len1>=0) { -- if (got_picture) { -+ if (got_frame) { - // this is disabled by default since it does not work - // for all input formats the way it should. - if (this->sync == 1 && pkt.dts != AV_NOPTS_VALUE) { -@@ -1427,7 +1457,7 @@ - - if (venc_pix_fmt != this->pix_fmt) { - sws_scale(this->sws_colorspace_ctx, -- frame->data, frame->linesize, 0, display_height, -+ (const uint8_t * const*)frame->data, frame->linesize, 0, display_height, - output_tmp->data, output_tmp->linesize); - } - else{ -@@ -1471,7 +1501,7 @@ - } - if (this->sws_scale_ctx) { - sws_scale(this->sws_scale_ctx, -- output_cropped->data, -+ (const uint8_t * const*)output_cropped->data, - output_cropped->linesize, 0, - display_height - (this->frame_topBand + this->frame_bottomBand), - output_resized->data, -@@ -1499,7 +1529,7 @@ - //now output_resized - - if (!first) { -- if (got_picture || video_eos) { -+ if (got_frame || video_eos) { - prepare_ycbcr_buffer(this, ycbcr, output_buffered); - if(dups>0) { - //this only works if dups < keyint, -@@ -1519,11 +1549,11 @@ - info.videotime = this->frame_count / av_q2d(this->framerate); - } - } -- if (got_picture) { -+ if (got_frame) { - first=0; - av_picture_copy((AVPicture *)output_buffered, (AVPicture *)output_padded, this->pix_fmt, this->frame_width, this->frame_height); - } -- if (!got_picture) { -+ if (!got_frame) { - break; - } - } -@@ -1531,42 +1561,62 @@ - if (info.passno!=1) - if ((audio_eos && !audio_done) || (ret >= 0 && pkt.stream_index == this->audio_index)) { - while((audio_eos && !audio_done) || avpkt.size > 0 ) { -- int samples=0; -- int samples_out=0; -- int data_size = 4*MAX_AUDIO_FRAME_SIZE; - int bytes_per_sample = av_get_bytes_per_sample(aenc->sample_fmt); - - if (avpkt.size > 0) { -- len1 = avcodec_decode_audio3(astream->codec, audio_buf, &data_size, &avpkt); -+ if (!audio_frame && !(audio_frame = avcodec_alloc_frame())) { -+ fprintf(stderr, "Failed to allocate memory\n"); -+ exit(1); -+ } -+ len1 = avcodec_decode_audio4(astream->codec, audio_frame, &got_frame, &avpkt); - if (len1 < 0) { - /* if error, we skip the frame */ - break; - } -- avpkt.size -= len1; -- avpkt.data += len1; -- if (data_size >0) { -- samples = data_size / (aenc->channels * bytes_per_sample); -- samples_out = samples; -- if (this->audio_resample_ctx) { -- samples_out = audio_resample(this->audio_resample_ctx, resampled, audio_buf, samples); -- audio_p = resampled; -+ /* Some audio decoders decode only part of the packet, and have to be -+ * called again with the remainder of the packet data. -+ * Sample: http://fate-suite.libav.org/lossless-audio/luckynight-partial.shn -+ * Also, some decoders might over-read the packet. */ -+ len1 = FFMIN(len1, avpkt.size); -+ if (got_frame) { -+ dst_nb_samples = audio_frame->nb_samples; -+ if (swr_ctx) { -+ dst_nb_samples = av_rescale_rnd(audio_frame->nb_samples, -+ this->sample_rate, aenc->sample_rate, AV_ROUND_UP); -+ if (dst_nb_samples > max_dst_nb_samples) { -+ av_free(dst_audio_data[0]); -+ if (av_samples_alloc(dst_audio_data, &dst_linesize, this->channels, -+ dst_nb_samples, AV_SAMPLE_FMT_FLTP, 1) < 0) { -+ fprintf(stderr, "Error while converting audio\n"); -+ exit(1); -+ } -+ max_dst_nb_samples = dst_nb_samples; -+ } -+ if (swr_convert(swr_ctx, dst_audio_data, dst_nb_samples, -+ (const uint8_t**)audio_frame->extended_data, audio_frame->nb_samples) < 0) { -+ fprintf(stderr, "Error while converting audio\n"); -+ exit(1); -+ } -+ audio_p = dst_audio_data; -+ } else { -+ audio_p = audio_frame->extended_data; - } -- else -- audio_p = audio_buf; - } -+ avpkt.size -= len1; -+ avpkt.data += len1; - } -- -- if (no_samples > 0 && this->sample_count + samples_out > no_samples) { -- audio_eos = 1; -- samples_out = no_samples - this->sample_count; -- if (samples_out <= 0) { -- break; -+ if(got_frame || audio_eos) { -+ if (no_samples > 0 && this->sample_count + dst_nb_samples > no_samples) { -+ audio_eos = 1; -+ dst_nb_samples = no_samples - this->sample_count; -+ if (dst_nb_samples <= 0) { -+ break; -+ } - } -+ oggmux_add_audio(&info, audio_p, dst_nb_samples, audio_eos); -+ avcodec_free_frame(&audio_frame); -+ this->sample_count += dst_nb_samples; - } -- -- oggmux_add_audio(&info, audio_p, -- samples_out * (this->channels), samples_out, audio_eos); -- this->sample_count += samples_out; - if(audio_eos) { - audio_done = 1; - } -@@ -1751,8 +1801,8 @@ - avcodec_close(venc); - } - if (this->audio_index >= 0) { -- if (this->audio_resample_ctx) -- audio_resample_close(this->audio_resample_ctx); -+ if (swr_ctx) -+ swr_free(&swr_ctx); - avcodec_close(aenc); - } - -@@ -1773,8 +1823,12 @@ - frame_dealloc(output_cropped_p); - frame_dealloc(output_padded_p); - } -- av_free(audio_buf); -- av_free(resampled); -+ if (dst_audio_data) -+ av_freep(&dst_audio_data[0]); -+ av_freep(&dst_audio_data); -+ if(swr_ctx) { -+ swr_close(swr_ctx); -+ } - } - else{ - fprintf(stderr, "No video or audio stream found.\n"); -diff -Naur ffmpeg2theora-0.29/src/ffmpeg2theora.c.orig ffmpeg2theora-0.29.patched/src/ffmpeg2theora.c.orig ---- ffmpeg2theora-0.29/src/ffmpeg2theora.c.orig 2014-05-14 14:57:25.000000000 -0400 -+++ ffmpeg2theora-0.29.patched/src/ffmpeg2theora.c.orig 2014-05-14 14:57:30.000000000 -0400 -@@ -2772,6 +2772,9 @@ - outputfile_set=1; - } - optind++; -+ } else { -+ fprintf(stderr, "ERROR: no input specified\n"); -+ exit(1); - } - if(optind<argc) { - fprintf(stderr, "WARNING: Only one input file supported, others will be ignored\n"); -diff -Naur ffmpeg2theora-0.29/src/ffmpeg2theora.h ffmpeg2theora-0.29.patched/src/ffmpeg2theora.h ---- ffmpeg2theora-0.29/src/ffmpeg2theora.h 2010-10-10 10:56:00.000000000 -0400 -+++ ffmpeg2theora-0.29.patched/src/ffmpeg2theora.h 2014-05-14 14:59:43.000000000 -0400 -@@ -62,7 +62,6 @@ - double fps; - struct SwsContext *sws_colorspace_ctx; /* for image resampling/resizing */ - struct SwsContext *sws_scale_ctx; /* for image resampling/resizing */ -- ReSampleContext *audio_resample_ctx; - ogg_int32_t aspect_numerator; - ogg_int32_t aspect_denominator; - int colorspace; -diff -Naur ffmpeg2theora-0.29/src/libswresample_compat.h ffmpeg2theora-0.29.patched/src/libswresample_compat.h ---- ffmpeg2theora-0.29/src/libswresample_compat.h 1969-12-31 19:00:00.000000000 -0500 -+++ ffmpeg2theora-0.29.patched/src/libswresample_compat.h 2014-05-14 14:59:43.000000000 -0400 -@@ -0,0 +1,23 @@ -+// 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 -diff -Naur ffmpeg2theora-0.29/src/theorautils.c ffmpeg2theora-0.29.patched/src/theorautils.c ---- ffmpeg2theora-0.29/src/theorautils.c 2012-06-21 17:36:01.000000000 -0400 -+++ ffmpeg2theora-0.29.patched/src/theorautils.c 2014-05-14 14:59:43.000000000 -0400 -@@ -1219,17 +1219,16 @@ - /** - * 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, int16_t * buffer, int bytes, int samples, int e_o_s) { -+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 (bytes <= 0 && samples <= 0) { -+ if (samples <= 0) { - /* end of audio stream */ - if (e_o_s) - vorbis_analysis_wrote (&info->vd, 0); -@@ -1252,7 +1251,7 @@ - default: k = j; - } - } -- vorbis_buffer[k][i] = buffer[count++] / 32768.f; -+ vorbis_buffer[k][i] = ((const float *)buffer[j])[i]; - } - } - vorbis_analysis_wrote (&info->vd, samples); -@@ -1291,8 +1290,8 @@ - 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 %lld has calculated start" -- " granule of %lld, but it should be non-negative!", -+ 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; -@@ -1302,7 +1301,7 @@ - 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 %lld (granulepos %lld) starts before" -+ 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; -diff -Naur ffmpeg2theora-0.29/src/theorautils.h ffmpeg2theora-0.29.patched/src/theorautils.h ---- ffmpeg2theora-0.29/src/theorautils.h 2011-09-15 16:20:46.000000000 -0400 -+++ ffmpeg2theora-0.29.patched/src/theorautils.h 2014-05-14 14:59:43.000000000 -0400 -@@ -168,7 +168,7 @@ - extern void oggmux_setup_kate_streams(oggmux_info *info, int n_kate_streams); - extern void oggmux_init (oggmux_info *info); - extern void oggmux_add_video (oggmux_info *info, th_ycbcr_buffer ycbcr, int e_o_s); --extern void oggmux_add_audio (oggmux_info *info, int16_t * readbuffer, int bytesread, int samplesread,int e_o_s); -+extern void oggmux_add_audio (oggmux_info *info, uint8_t **buffer, int samples,int e_o_s); - #ifdef HAVE_KATE - 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); - 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); 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