Flac encoding now working. Thanks peter schlaile for the help!

diff --git a/source/blender/blenkernel/intern/writeffmpeg.c b/source/blender/blenkernel/intern/writeffmpeg.c
index d71836ab1817fd9eaf749614a11732806b061e98..917ecd044ea6fb4b4f8dff30d8aaa77a33bc4a29 100644 (file)
--- a/source/blender/blenkernel/intern/writeffmpeg.c
+++ b/source/blender/blenkernel/intern/writeffmpeg.c
@@ -94,7 +94,7 @@ static uint8_t* video_buffer = 0;
 static int video_buffersize = 0;
 
 static uint8_t* audio_input_buffer = 0;
-static int audio_input_frame_size = 0;
+static int audio_input_samples = 0;
 static uint8_t* audio_output_buffer = 0;
 static int audio_outbuf_size = 0;
 static double audio_time = 0.0f;
@@ -135,14 +135,14 @@ static int write_audio_frame(void)
        av_init_packet(&pkt);
        pkt.size = 0;
 
-       AUD_readDevice(audio_mixdown_device, audio_input_buffer, audio_input_frame_size);
-       audio_time += (double) audio_input_frame_size / (double) c->sample_rate;
+       AUD_readDevice(audio_mixdown_device, audio_input_buffer, audio_input_samples);
+       audio_time += (double) audio_input_samples / (double) c->sample_rate;
 
        pkt.size = avcodec_encode_audio(c, audio_output_buffer,
                                        audio_outbuf_size,
                                        (short*) audio_input_buffer);
 
-       if(pkt.size <= 0)
+       if(pkt.size < 0)
        {
                // XXX error("Error writing audio packet");
                return -1;
@@ -599,16 +599,20 @@ static AVStream* alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
 
        audio_outbuf_size = FF_MIN_BUFFER_SIZE;
 
-       audio_output_buffer = (uint8_t*)MEM_mallocN(
-               audio_outbuf_size, "FFMPEG audio encoder input buffer");
-
        if((c->codec_id >= CODEC_ID_PCM_S16LE) && (c->codec_id <= CODEC_ID_PCM_DVD))
-               audio_input_frame_size = audio_outbuf_size * 8 / c->bits_per_coded_sample / c->channels;
+               audio_input_samples = audio_outbuf_size * 8 / c->bits_per_coded_sample / c->channels;
        else
-               audio_input_frame_size = c->frame_size;
+       {
+               audio_input_samples = c->frame_size;
+               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;
+       }
+
+       audio_output_buffer = (uint8_t*)MEM_mallocN(
+               audio_outbuf_size, "FFMPEG audio encoder input buffer");
 
        audio_input_buffer = (uint8_t*)MEM_mallocN(
-               audio_input_frame_size * c->channels * sizeof(int16_t),
+               audio_input_samples * c->channels * sizeof(int16_t),
                "FFMPEG audio encoder output buffer");
 
        audio_time = 0.0f;