[Bf-blender-cvs] SVN commit: /data/svn/bf-blender [56175] trunk/blender: Bring back support of FFmpeg >= 0.7
Sergey Sharybin
sergey.vfx at gmail.com
Fri Apr 19 19:23:08 CEST 2013
Revision: 56175
http://projects.blender.org/scm/viewvc.php?view=rev&root=bf-blender&revision=56175
Author: nazgul
Date: 2013-04-19 17:23:08 +0000 (Fri, 19 Apr 2013)
Log Message:
-----------
Bring back support of FFmpeg >= 0.7
After planar codecs support minimal FFmpeg was
bumped to 0.10 which was not so much nice because
it was only released only later last year.
Didn't find a way to make compatibility code local
in ffmpeg_compat, so there're some ifdefs in
audaspace and writeffmpeg.
Not entirely happy, but having a bit of ifdefs in
code better than lots of real PITA for platform
maintainers.
Modified Paths:
--------------
trunk/blender/intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp
trunk/blender/intern/ffmpeg/ffmpeg_compat.h
trunk/blender/source/blender/blenkernel/intern/writeffmpeg.c
Modified: trunk/blender/intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp
===================================================================
--- trunk/blender/intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp 2013-04-19 16:23:02 UTC (rev 56174)
+++ trunk/blender/intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp 2013-04-19 17:23:08 UTC (rev 56175)
@@ -43,6 +43,7 @@
int AUD_FFMPEGReader::decode(AVPacket& packet, AUD_Buffer& buffer)
{
+#ifdef FFMPEG_HAVE_DECODE_AUDIO4
AVFrame* frame = NULL;
int got_frame;
int read_length;
@@ -101,6 +102,54 @@
av_free(frame);
return buf_pos;
+#else
+ // save packet parameters
+ uint8_t *audio_pkg_data = packet.data;
+ int audio_pkg_size = packet.size;
+
+ int buf_size = buffer.getSize();
+ int buf_pos = 0;
+
+ int read_length, data_size;
+
+ AVPacket tmp_pkt;
+
+ av_init_packet(&tmp_pkt);
+
+ // as long as there is still data in the package
+ while(audio_pkg_size > 0)
+ {
+ // resize buffer if needed
+ if(buf_size - buf_pos < AVCODEC_MAX_AUDIO_FRAME_SIZE)
+ {
+ buffer.resize(buf_size + AVCODEC_MAX_AUDIO_FRAME_SIZE, true);
+ buf_size += AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ }
+
+ // read samples from the packet
+ data_size = buf_size - buf_pos;
+
+ tmp_pkt.data = audio_pkg_data;
+ tmp_pkt.size = audio_pkg_size;
+
+ read_length = avcodec_decode_audio3(
+ m_codecCtx,
+ (int16_t*)(((data_t*)buffer.getBuffer()) + buf_pos),
+ &data_size, &tmp_pkt);
+
+ // read error, next packet!
+ if(read_length < 0)
+ break;
+
+ buf_pos += data_size;
+
+ // move packet parameters
+ audio_pkg_data += read_length;
+ audio_pkg_size -= read_length;
+ }
+
+ return buf_pos;
+#endif
}
static const char* streaminfo_error = "AUD_FFMPEGReader: Stream info couldn't "
Modified: trunk/blender/intern/ffmpeg/ffmpeg_compat.h
===================================================================
--- trunk/blender/intern/ffmpeg/ffmpeg_compat.h 2013-04-19 16:23:02 UTC (rev 56174)
+++ trunk/blender/intern/ffmpeg/ffmpeg_compat.h 2013-04-19 17:23:08 UTC (rev 56175)
@@ -76,9 +76,26 @@
#define FFMPEG_HAVE_CANON_H264_RESOLUTION_FIX
#endif
+#if ((LIBAVCODEC_VERSION_MAJOR > 53) || (LIBAVCODEC_VERSION_MAJOR >= 53) && (LIBAVCODEC_VERSION_MINOR >= 60))
+#define FFMPEG_HAVE_ENCODE_AUDIO2
+#endif
+
+#if ((LIBAVCODEC_VERSION_MAJOR > 53) || (LIBAVCODEC_VERSION_MAJOR >= 53) && (LIBAVCODEC_VERSION_MINOR >= 42))
+#define FFMPEG_HAVE_DECODE_AUDIO4
+#endif
+
#if ((LIBAVUTIL_VERSION_MAJOR > 51) || (LIBAVUTIL_VERSION_MAJOR == 51) && (LIBAVUTIL_VERSION_MINOR >= 32))
#define FFMPEG_FFV1_ALPHA_SUPPORTED
#define FFMPEG_SAMPLE_FMT_S16P_SUPPORTED
+#else
+
+static inline
+int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
+{
+ /* no planar formats in FFmpeg < 0.9 */
+ return 0;
+}
+
#endif
static inline
Modified: trunk/blender/source/blender/blenkernel/intern/writeffmpeg.c
===================================================================
--- trunk/blender/source/blender/blenkernel/intern/writeffmpeg.c 2013-04-19 16:23:02 UTC (rev 56174)
+++ trunk/blender/source/blender/blenkernel/intern/writeffmpeg.c 2013-04-19 17:23:08 UTC (rev 56175)
@@ -90,6 +90,10 @@
static uint8_t *audio_input_buffer = 0;
static uint8_t *audio_deinterleave_buffer = 0;
static int audio_input_samples = 0;
+#ifndef FFMPEG_HAVE_ENCODE_AUDIO2
+static uint8_t *audio_output_buffer = 0;
+static int audio_outbuf_size = 0;
+#endif
static double audio_time = 0.0f;
static bool audio_deinterleave = false;
static int audio_sample_size = 0;
@@ -123,7 +127,7 @@
{
AVCodecContext *c = NULL;
AVPacket pkt;
- AVFrame *frame;
+ AVFrame *frame = NULL;
int got_output = 0;
c = audio_stream->codec;
@@ -132,6 +136,10 @@
pkt.size = 0;
pkt.data = NULL;
+ AUD_readDevice(audio_mixdown_device, audio_input_buffer, audio_input_samples);
+ audio_time += (double) audio_input_samples / (double) c->sample_rate;
+
+#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
frame = avcodec_alloc_frame();
frame->nb_samples = audio_input_samples;
frame->format = c->sample_fmt;
@@ -139,9 +147,6 @@
frame->channel_layout = c->channel_layout;
#endif
- AUD_readDevice(audio_mixdown_device, audio_input_buffer, audio_input_samples);
- audio_time += (double) audio_input_samples / (double) c->sample_rate;
-
if (audio_deinterleave) {
int channel, i;
uint8_t *temp;
@@ -166,6 +171,22 @@
return -1;
}
+ if (!got_output) {
+ avcodec_free_frame(&frame);
+ return 0;
+ }
+#else
+ pkt.size = avcodec_encode_audio(c, audio_output_buffer, audio_outbuf_size, (short *) audio_input_buffer);
+
+ if (pkt.size < 0) {
+ // XXX error("Error writing audio packet");
+ return -1;
+ }
+
+ pkt.data = audio_output_buffer;
+ got_output = 1;
+#endif
+
if (got_output) {
if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE) {
pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, audio_stream->time_base);
@@ -178,13 +199,16 @@
if (av_interleaved_write_frame(outfile, &pkt) != 0) {
fprintf(stderr, "Error writing audio packet!\n");
+ if (frame)
+ avcodec_free_frame(&frame);
return -1;
}
av_free_packet(&pkt);
}
- avcodec_free_frame(&frame);
+ if (frame)
+ avcodec_free_frame(&frame);
return 0;
}
@@ -718,6 +742,10 @@
st->codec->time_base.num = 1;
st->codec->time_base.den = st->codec->sample_rate;
+#ifndef FFMPEG_HAVE_ENCODE_AUDIO2
+ audio_outbuf_size = FF_MIN_BUFFER_SIZE;
+#endif
+
if (c->frame_size == 0)
// used to be if((c->codec_id >= CODEC_ID_PCM_S16LE) && (c->codec_id <= CODEC_ID_PCM_DVD))
// not sure if that is needed anymore, so let's try out if there are any
@@ -725,6 +753,10 @@
audio_input_samples = FF_MIN_BUFFER_SIZE * 8 / c->bits_per_coded_sample / c->channels;
else {
audio_input_samples = c->frame_size;
+#ifndef FFMPEG_HAVE_ENCODE_AUDIO2
+ if (c->frame_size * c->channels * sizeof(int16_t) * 4 > audio_outbuf_size)
+ audio_outbuf_size = c->frame_size * c->channels * sizeof(int16_t) * 4;
+#endif
}
audio_deinterleave = av_sample_fmt_is_planar(c->sample_fmt);
@@ -732,6 +764,9 @@
audio_sample_size = av_get_bytes_per_sample(c->sample_fmt);
audio_input_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * audio_sample_size);
+#ifndef FFMPEG_HAVE_ENCODE_AUDIO2
+ audio_output_buffer = (uint8_t *) av_malloc(audio_outbuf_size);
+#endif
if (audio_deinterleave)
audio_deinterleave_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * audio_sample_size);
@@ -1182,6 +1217,12 @@
av_free(audio_input_buffer);
audio_input_buffer = 0;
}
+#ifndef FFMPEG_HAVE_ENCODE_AUDIO2
+ if (audio_output_buffer) {
+ av_free(audio_output_buffer);
+ audio_output_buffer = 0;
+ }
+#endif
if (audio_deinterleave_buffer) {
av_free(audio_deinterleave_buffer);
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