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-rw-r--r--NEWS2
-rw-r--r--src/decoder/ffmpeg_decoder_plugin.c59
2 files changed, 59 insertions, 2 deletions
diff --git a/NEWS b/NEWS
index 4c36ab798..2f5efe8d7 100644
--- a/NEWS
+++ b/NEWS
@@ -2,7 +2,7 @@ ver 0.16.7 (2011/??/??)
* input:
- ffmpeg: support libavformat 0.7
* decoder:
- - ffmpeg: support libavformat 0.7
+ - ffmpeg: support libavformat 0.7, libavcodec 0.8
* output:
- httpd: fix excessive buffering
- openal: force 16 bit playback, as 8 bit doesn't work
diff --git a/src/decoder/ffmpeg_decoder_plugin.c b/src/decoder/ffmpeg_decoder_plugin.c
index 83628c5dd..1be26d9a0 100644
--- a/src/decoder/ffmpeg_decoder_plugin.c
+++ b/src/decoder/ffmpeg_decoder_plugin.c
@@ -208,6 +208,7 @@ ffmpeg_find_audio_stream(const AVFormatContext *format_context)
return -1;
}
+#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53,25,0)
/**
* On some platforms, libavcodec wants the output buffer aligned to 16
* bytes (because it uses SSE/Altivec internally). This function
@@ -222,6 +223,7 @@ align16(void *p, size_t *length_p)
*length_p -= add;
return (char *)p + add;
}
+#endif
G_GNUC_CONST
static double
@@ -241,6 +243,40 @@ time_to_ffmpeg(double t, const AVRational time_base)
time_base);
}
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
+/**
+ * Copy PCM data from a AVFrame to an interleaved buffer.
+ */
+static int
+copy_interleave_frame(const AVCodecContext *codec_context,
+ const AVFrame *frame,
+ uint8_t *buffer, size_t buffer_size)
+{
+ int plane_size;
+ const int data_size =
+ av_samples_get_buffer_size(&plane_size,
+ codec_context->channels,
+ frame->nb_samples,
+ codec_context->sample_fmt, 1);
+ if (buffer_size < (size_t)data_size)
+ /* buffer is too small - shouldn't happen */
+ return AVERROR(EINVAL);
+
+ if (av_sample_fmt_is_planar(codec_context->sample_fmt) &&
+ codec_context->channels > 1) {
+ for (int i = 0, channels = codec_context->channels;
+ i < channels; i++) {
+ memcpy(buffer, frame->extended_data[i], plane_size);
+ buffer += plane_size;
+ }
+ } else {
+ memcpy(buffer, frame->extended_data[0], data_size);
+ }
+
+ return data_size;
+}
+#endif
+
static enum decoder_command
ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
const AVPacket *packet,
@@ -258,9 +294,15 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
int packet_size = packet->size;
#endif
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
+ uint8_t aligned_buffer[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
+ const size_t buffer_size = sizeof(aligned_buffer);
+#else
+ /* libavcodec < 0.8 needs an aligned buffer */
uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
size_t buffer_size = sizeof(audio_buf);
int16_t *aligned_buffer = align16(audio_buf, &buffer_size);
+#endif
enum decoder_command cmd = DECODE_COMMAND_NONE;
while (
@@ -271,7 +313,22 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
#endif
cmd == DECODE_COMMAND_NONE) {
int audio_size = buffer_size;
-#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
+ AVFrame frame;
+ int got_frame = 0;
+ int len = avcodec_decode_audio4(codec_context,
+ &frame, &got_frame,
+ &packet2);
+ if (len >= 0 && got_frame) {
+ audio_size = copy_interleave_frame(codec_context,
+ &frame,
+ aligned_buffer,
+ buffer_size);
+ if (audio_size < 0)
+ len = audio_size;
+ } else if (len >= 0)
+ len = -1;
+#elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
int len = avcodec_decode_audio3(codec_context,
aligned_buffer, &audio_size,
&packet2);