spek

commit d683b008973e88b80ce743ec5f3a592a6a16fd43
parent 166d31e69398dcfb423980aa408b8f2f6cb53832
Author: Alexander Kojevnikov <alexander@kojevnikov.com>
Date:   Sat, 10 Jul 2010 12:10:15 +1000

Use av_malloc to allocate audio buffers

libavcodec crashes on Windows when using the usual malloc,
av_malloc makes sure the allocated buffer is properly aligned.

Diffstat:
M
src/spek-audio.c
 | 
10
+++++++---
M
src/spek-audio.h
 | 
7
+++----
M
src/spek-pipeline.vala
 | 
6
++----
M
vapi/spek-audio.vapi
 | 
4
++--

4 files changed, 14 insertions(+), 13 deletions(-)
diff --git a/src/spek-audio.c b/src/spek-audio.c
@@ -77,7 +77,6 @@ SpekAudioContext * spek_audio_open (const char *file_name) {
 		cx->error = _("No audio channels");
 		return cx;
 	}
-	cx->buffer_size = (AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2;
 	if (avcodec_open (cx->codec_context, cx->codec) < 0) {
 		cx->error = _("Cannot open decoder");
 		return cx;
@@ -103,6 +102,8 @@ SpekAudioContext * spek_audio_open (const char *file_name) {
 		cx->error = _("Unsupported sample format");
 		return cx;
 	}
+	cx->buffer_size = (AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2;
+	cx->buffer = av_malloc (cx->buffer_size);
 	av_init_packet (&cx->packet);
 	cx->offset = 0;
 	return cx;
@@ -116,7 +117,7 @@ void spek_audio_start (SpekAudioContext *cx, gint samples) {
 	cx->error_per_interval = (cx->stream->duration * rate) % cx->error_base;
 }
 
-gint spek_audio_read (SpekAudioContext *cx, guint8 *buffer) {
+gint spek_audio_read (SpekAudioContext *cx) {
 	gint buffer_size;
 	gint len;
 	gint res;
@@ -129,7 +130,7 @@ gint spek_audio_read (SpekAudioContext *cx, guint8 *buffer) {
 		while (cx->packet.size > 0) {
 			buffer_size = cx->buffer_size;
 			len = avcodec_decode_audio3 (
-				cx->codec_context, (int16_t *) buffer, &buffer_size, &cx->packet);
+				cx->codec_context, (int16_t *) cx->buffer, &buffer_size, &cx->packet);
 			if (len < 0) {
 				/* Error, skip the frame. */
 				cx->packet.size = 0;
@@ -171,6 +172,9 @@ void spek_audio_close (SpekAudioContext *cx) {
 	if (cx->codec_name != NULL) {
 		g_free (cx->codec_name);
 	}
+	if (cx->buffer) {
+		av_free (cx->buffer);
+	}
 	if (cx->packet.data) {
 		cx->packet.data -= cx->offset;
 		cx->packet.size += cx->offset;
diff --git a/src/spek-audio.h b/src/spek-audio.h
@@ -30,6 +30,7 @@ typedef struct {
 	AVCodecContext *codec_context;
 	AVStream *stream;
 	AVCodec *codec;
+	gint buffer_size;
 	AVPacket packet;
 	gint offset;
 
@@ -44,7 +45,7 @@ typedef struct {
 	gboolean fp; /* floating-point sample representation */
 	gint channels;
 	gdouble duration;
-	gint buffer_size; /* minimum buffer size for spek_audio_read() */
+	guint8 *buffer;
 	gint64 frames_per_interval;
 	gint64 error_per_interval;
 	gint64 error_base;
@@ -64,10 +65,8 @@ void spek_audio_start (SpekAudioContext *cx, gint samples);
 /* Read and decode the opened audio stream.
  * Returns -1 on error, 0 if there's nothing left to read
  * or the number of bytes decoded into the buffer.
- * The buffer must be allocated (and later freed) by the caller,
- * minimum size is `buffer_size`.
  */
-gint spek_audio_read (SpekAudioContext *cx, guint8 *buffer);
+gint spek_audio_read (SpekAudioContext *cx);
 
 /* Closes the file opened with spek_audio_open,
  * frees all allocated buffers and the context
diff --git a/src/spek-pipeline.vala b/src/spek-pipeline.vala
@@ -36,7 +36,6 @@ namespace Spek {
 		private int threshold;
 		private Callback cb;
 
-		private uint8[] buffer;
 		private Fft.Plan fft;
 		private int nfft; // Size of the FFT transform.
 		private const int NFFT = 64; // Number of FFTs to pre-fetch.
@@ -89,7 +88,6 @@ namespace Spek {
 			}
 
 			this.sample_rate = cx.sample_rate;
-			this.buffer = new uint8[cx.buffer_size];
 			this.nfft = 2 * bands - 2;
 			this.fft = new Fft.Plan (nfft, threshold);
 			this.input_size = nfft * (NFFT * 2 + 1);
@@ -140,10 +138,10 @@ namespace Spek {
 				return null;
 			}
 
-			while ((size = cx.read (this.buffer)) > 0) {
+			while ((size = cx.read ()) > 0) {
 				lock (quit) if (quit) break;
 
-				uint8 *buffer = (uint8 *) this.buffer;
+				uint8 *buffer = (uint8 *) cx.buffer;
 				while (size >= block_size) {
 					input[pos] = average_input (buffer);
 					buffer += block_size;
diff --git a/vapi/spek-audio.vapi b/vapi/spek-audio.vapi
@@ -13,7 +13,7 @@ namespace Spek.Audio {
 		public bool fp;
 		public int channels;
 		public double duration;
-		public int buffer_size;
+		public uint8 *buffer;
 		public int64 frames_per_interval;
 		public int64 error_per_interval;
 		public int64 error_base;
@@ -23,7 +23,7 @@ namespace Spek.Audio {
 		[CCode (cname = "spek_audio_start")]
 		public int start (int samples);
 		[CCode (cname = "spek_audio_read")]
-		public int read ([CCode (array_length = false)] uint8[] buffer);
+		public int read ();
 	}
 	public static void init ();
 }