Fix build with very recent copies of FFmpeg

This is a backport of the 1.2 code that was commited to utilize the new API
provided by FFmpeg for awhile now but this is especially important because
the old API has been eliminated all together from said copies of FFmpeg.

1 files changed, 29 insertions, 1 deletions

diff --git a/src/combined/ffmpeg/ff_audio_decoder.c b/src/combined/ffmpeg/ff_audio_decoder.c
index 125eff12c..60544ad0c 100644
--- a/src/combined/ffmpeg/ff_audio_decoder.c
+++ b/src/combined/ffmpeg/ff_audio_decoder.c
@@ -46,6 +46,12 @@
 
 #define AUDIOBUFSIZE (64 * 1024)
 
+#if LIBAVCODEC_VERSION_MAJOR >= 53 || (LIBAVCODEC_VERSION_MAJOR == 52 && LIBAVCODEC_VERSION_MINOR >= 32)
+#  define AVAUDIO 3
+#else
+#  define AVAUDIO 2
+#endif
+
 typedef struct {
   audio_decoder_class_t   decoder_class;
 } ff_audio_class_t;
@@ -255,6 +261,9 @@ static void ff_audio_decode_data (audio_decoder_t *this_gen, buf_element_t *buf)
       buf->decoder_info[2]);
 
   } else if (!(buf->decoder_flags & BUF_FLAG_SPECIAL)) {
+#if AVAUDIO > 2
+    AVPacket avpkt;
+#endif
 
     if( !this->decoder_ok ) {
       if ( ! this->context || ! this->codec ) {
@@ -286,11 +295,21 @@ static void ff_audio_decode_data (audio_decoder_t *this_gen, buf_element_t *buf)
     if (!this->output_open) {
       if (!this->audio_bits || !this->audio_sample_rate || !this->audio_channels) {
         decode_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+#if AVAUDIO > 2
+	av_init_packet (&avpkt);
+	avpkt.data = (uint8_t *)&this->buf[0];
+	avpkt.size = this->size;
+	avpkt.flags = AV_PKT_FLAG_KEY;
+	avcodec_decode_audio3 (this->context,
+			       (int16_t *)this->decode_buffer,
+			       &decode_buffer_size, &avpkt);
+#else
         avcodec_decode_audio2 (this->context,
                               (int16_t *)this->decode_buffer,
                               &decode_buffer_size,
                               &this->buf[0],
                               this->size);
+#endif
 	this->audio_bits = this->context->bits_per_sample;
 	this->audio_sample_rate = this->context->sample_rate;
 	this->audio_channels = this->context->channels;
@@ -311,12 +330,21 @@ static void ff_audio_decode_data (audio_decoder_t *this_gen, buf_element_t *buf)
       offset = 0;
       while (this->size>0) {
         decode_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+#if AVAUDIO > 2
+	av_init_packet (&avpkt);
+	avpkt.data = (uint8_t *)&this->buf[offset];
+	avpkt.size = this->size;
+	avpkt.flags = AV_PKT_FLAG_KEY;
+	bytes_consumed = avcodec_decode_audio3 (this->context,
+						(int16_t *)this->decode_buffer,
+						&decode_buffer_size, &avpkt);
+#else
         bytes_consumed = avcodec_decode_audio2 (this->context,
                                                (int16_t *)this->decode_buffer,
                                                &decode_buffer_size,
                                                &this->buf[offset],
                                                this->size);
-
+#endif
         if (bytes_consumed<0) {
           xprintf (this->stream->xine, XINE_VERBOSITY_DEBUG,
                    "ffmpeg_audio_dec: error decompressing audio frame\n");