Merge branch 'v0.16.x'

Conflicts:
	src/decoder/ffmpeg_decoder_plugin.c
	src/input/ffmpeg_input_plugin.c

3 files changed, 119 insertions, 6 deletions

diff --git a/NEWS b/NEWS
index ea9bfe6d7..f5b8fb203 100644
--- a/NEWS
+++ b/NEWS
@@ -8,11 +8,9 @@ ver 0.17 (2011/??/??)
   - curl: enable CURLOPT_NETRC
   - curl: non-blocking I/O
   - soup: new input plugin based on libsoup
-  - ffmpeg: support libavformat 0.7
 * decoder:
   - mpg123: implement seeking
   - ffmpeg: drop support for pre-0.5 ffmpeg
-  - ffmpeg: support libavformat 0.7
   - oggflac: delete this obsolete plugin
   - dsdiff: new decoder plugin
 * output:
@@ -33,6 +31,10 @@ ver 0.17 (2011/??/??)
 
 
 ver 0.16.7 (2011/??/??)
+* input:
+  - ffmpeg: support libavformat 0.7
+* decoder:
+  - ffmpeg: support libavformat 0.8, 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 5dc95ccf8..05a59c1e0 100644
--- a/src/decoder/ffmpeg_decoder_plugin.c
+++ b/src/decoder/ffmpeg_decoder_plugin.c
@@ -36,6 +36,7 @@
 #include <libavformat/avformat.h>
 #include <libavformat/avio.h>
 #include <libavutil/log.h>
+#include <libavutil/mathematics.h>
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "ffmpeg"
@@ -196,6 +197,7 @@ 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
@@ -210,6 +212,7 @@ align16(void *p, size_t *length_p)
 	*length_p -= add;
 	return (char *)p + add;
 }
+#endif
 
 G_GNUC_CONST
 static double
@@ -229,6 +232,40 @@ 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,
@@ -246,9 +283,15 @@ 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 (
@@ -259,7 +302,22 @@ 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);
@@ -370,9 +428,19 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 		return;
 	}
 
-	if (av_find_stream_info(format_context)<0) {
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,2,0)
+	const int find_result =
+		avformat_find_stream_info(format_context, NULL);
+#else
+	const int find_result = av_find_stream_info(format_context);
+#endif
+	if (find_result < 0) {
 		g_warning("Couldn't find stream info\n");
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
+		avformat_close_input(&format_context);
+#else
 		av_close_input_stream(format_context);
+#endif
 		mpd_ffmpeg_stream_close(stream);
 		return;
 	}
@@ -380,7 +448,11 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 	int audio_stream = ffmpeg_find_audio_stream(format_context);
 	if (audio_stream == -1) {
 		g_warning("No audio stream inside\n");
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
+		avformat_close_input(&format_context);
+#else
 		av_close_input_stream(format_context);
+#endif
 		mpd_ffmpeg_stream_close(stream);
 		return;
 	}
@@ -395,7 +467,11 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 
 	if (!codec) {
 		g_warning("Unsupported audio codec\n");
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
+		avformat_close_input(&format_context);
+#else
 		av_close_input_stream(format_context);
+#endif
 		mpd_ffmpeg_stream_close(stream);
 		return;
 	}
@@ -408,7 +484,11 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 				       codec_context->channels, &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
+		avformat_close_input(&format_context);
+#else
 		av_close_input_stream(format_context);
+#endif
 		mpd_ffmpeg_stream_close(stream);
 		return;
 	}
@@ -418,9 +498,18 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 	   values into AVCodecContext.channels - a change that will be
 	   reverted later by avcodec_decode_audio3() */
 
-	if (avcodec_open(codec_context, codec)<0) {
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,6,0)
+	const int open_result = avcodec_open2(codec_context, codec, NULL);
+#else
+	const int open_result = avcodec_open(codec_context, codec);
+#endif
+	if (open_result < 0) {
 		g_warning("Could not open codec\n");
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
+		avformat_close_input(&format_context);
+#else
 		av_close_input_stream(format_context);
+#endif
 		mpd_ffmpeg_stream_close(stream);
 		return;
 	}
@@ -469,7 +558,11 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 	} while (cmd != DECODE_COMMAND_STOP);
 
 	avcodec_close(codec_context);
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
+	avformat_close_input(&format_context);
+#else
 	av_close_input_stream(format_context);
+#endif
 	mpd_ffmpeg_stream_close(stream);
 }
 
@@ -542,8 +635,18 @@ ffmpeg_stream_tag(struct input_stream *is)
 		return NULL;
 	}
 
-	if (av_find_stream_info(f) < 0) {
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,2,0)
+	const int find_result =
+		avformat_find_stream_info(f, NULL);
+#else
+	const int find_result = av_find_stream_info(f);
+#endif
+	if (find_result < 0) {
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
+		avformat_close_input(&f);
+#else
 		av_close_input_stream(f);
+#endif
 		mpd_ffmpeg_stream_close(stream);
 		return NULL;
 	}
@@ -565,7 +668,11 @@ ffmpeg_stream_tag(struct input_stream *is)
 			ffmpeg_copy_metadata(tag, f->streams[idx]->metadata, ffmpeg_tag_maps[i]);
 	}
 
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
+	avformat_close_input(&f);
+#else
 	av_close_input_stream(f);
+#endif
 	mpd_ffmpeg_stream_close(stream);
 
 	return tag;
diff --git a/src/input/ffmpeg_input_plugin.c b/src/input/ffmpeg_input_plugin.c
index 1c64b52c1..d71b3d4c0 100644
--- a/src/input/ffmpeg_input_plugin.c
+++ b/src/input/ffmpeg_input_plugin.c
@@ -28,6 +28,10 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "input_ffmpeg"
 
+#ifndef AV_VERSION_INT
+#define AV_VERSION_INT(a, b, c) (a<<16 | b<<8 | c)
+#endif
+
 struct input_ffmpeg {
 	struct input_stream base;