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.

intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp

@@ -43,6 +43,7 @@ extern "C" {
 
 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 @@ int AUD_FFMPEGReader::decode(AVPacket& packet, AUD_Buffer& buffer)
 	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 "

intern/ffmpeg/ffmpeg_compat.h

@@ -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

source/blender/blenkernel/intern/writeffmpeg.c

@@ -90,6 +90,10 @@ static int video_buffersize = 0;
 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 @@ static int write_audio_frame(void)
 {
 	AVCodecContext *c = NULL;
 	AVPacket pkt;
-	AVFrame *frame;
+	AVFrame *frame = NULL;
 	int got_output = 0;
 
 	c = audio_stream->codec;
@@ -132,6 +136,10 @@ static int write_audio_frame(void)
 	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 @@ static int write_audio_frame(void)
 	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 @@ static int write_audio_frame(void)
 		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 @@ static int write_audio_frame(void)
 
 		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 @@ static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
 	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 @@ static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
 		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 @@ static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
 	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 @@ static void end_ffmpeg_impl(int is_autosplit)
 		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);