Commit 82d2d6a6 authored by Jan Gerber's avatar Jan Gerber
Browse files

avcodec_decode_audio4 and libswresample

add libavresample fallback that should work with libav
parent bff0c7c4
......@@ -155,7 +155,6 @@ if not env.GetOption('clean'):
"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)
......@@ -168,6 +167,13 @@ if not env.GetOption('clean'):
'-Lffmpeg/' + lib
])
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 """
......
......@@ -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 @@ void ff2theora_output(ff2theora this) {
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) {
......@@ -962,22 +972,43 @@ void ff2theora_output(ff2theora this) {
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{
......@@ -1068,13 +1099,12 @@ void ff2theora_output(ff2theora this) {
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;
......@@ -1370,7 +1400,7 @@ void ff2theora_output(ff2theora this) {
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;
......@@ -1389,9 +1419,9 @@ void ff2theora_output(ff2theora this) {
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) {
......@@ -1428,7 +1458,7 @@ void ff2theora_output(ff2theora this) {
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{
......@@ -1472,7 +1502,7 @@ void ff2theora_output(ff2theora this) {
}
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,
......@@ -1500,7 +1530,7 @@ void ff2theora_output(ff2theora this) {
//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,
......@@ -1520,11 +1550,11 @@ void ff2theora_output(ff2theora this) {
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;
}
}
......@@ -1532,42 +1562,62 @@ void ff2theora_output(ff2theora this) {
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;
}
......@@ -1752,8 +1802,8 @@ void ff2theora_output(ff2theora this) {
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);
}
......@@ -1774,8 +1824,12 @@ void ff2theora_output(ff2theora this) {
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");
......
......@@ -62,7 +62,6 @@ typedef struct ff2theora{
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;
......
// 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
......@@ -1219,17 +1219,16 @@ vorbis_time(vorbis_dsp_state * dsp, ogg_int64_t granulepos) {
/**
* 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 @@ void oggmux_add_audio (oggmux_info *info, int16_t * buffer, int bytes, int sampl
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 @@ void oggmux_add_audio (oggmux_info *info, int16_t * buffer, int bytes, int sampl
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 @@ void oggmux_add_audio (oggmux_info *info, int16_t * buffer, int bytes, int sampl
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;
......
......@@ -168,7 +168,7 @@ void init_info(oggmux_info *info);
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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