[FFmpeg-devel] [PATCH v2] lavc/audiotoolboxenc: fix noise in encoded audio

[zhangjiejun1992 at gmail.com]

From: Jiejun Zhang <zhangjiejun1992 at gmail.com>

This fixes #6940

Although undocumented, AudioToolbox seems to require the data supplied
by the callback (i.e. ffat_encode_callback) being unchanged until the
next time the callback is called. In the old implementation, the
AVBuffer backing the frame is recycled after the frame is freed, and
somebody else (maybe the decoder) will write into the AVBuffer and
change the data. AudioToolbox then encodes some wrong data and noise
is produced. Copying the data to a separate buffer solves this
problem.
---
 libavcodec/audiotoolboxenc.c | 32 +++++++++++++++++++++++++++-----
 1 file changed, 27 insertions(+), 5 deletions(-)

diff --git a/libavcodec/audiotoolboxenc.c b/libavcodec/audiotoolboxenc.c
index 71885d1530..dcac88cdde 100644
--- a/libavcodec/audiotoolboxenc.c
+++ b/libavcodec/audiotoolboxenc.c
@@ -48,6 +48,9 @@ typedef struct ATDecodeContext {
     AudioFrameQueue afq;
     int eof;
     int frame_size;
+
+    uint8_t* audio_data_buf;
+    uint32_t audio_data_buf_size;
 } ATDecodeContext;
 
 static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile)
@@ -442,6 +445,9 @@ static av_cold int ffat_init_encoder(AVCodecContext *avctx)
 
     ff_af_queue_init(avctx, &at->afq);
 
+    at->audio_data_buf_size = 0;
+    at->audio_data_buf = NULL;
+
     return 0;
 }
 
@@ -465,13 +471,27 @@ static OSStatus ffat_encode_callback(AudioConverterRef converter, UInt32 *nb_pac
     }
 
     frame = ff_bufqueue_get(&at->frame_queue);
-
+    int audio_data_size = frame->nb_samples *
+                          av_get_bytes_per_sample(avctx->sample_fmt) *
+                          avctx->channels;
+    if (at->audio_data_buf_size < audio_data_size) {
+        av_log(avctx, AV_LOG_INFO, "Increasing audio data buffer size to %d\n",
+               audio_data_size);
+        av_free(at->audio_data_buf);
+        at->audio_data_buf_size = audio_data_size;
+        at->audio_data_buf = av_malloc(at->audio_data_buf_size);
+        if (!at->audio_data_buf) {
+            at->audio_data_buf_size = 0;
+            data->mNumberBuffers = 0;
+            *nb_packets = 0;
+            return AVERROR(ENOMEM);
+        }
+    }
     data->mNumberBuffers              = 1;
     data->mBuffers[0].mNumberChannels = avctx->channels;
-    data->mBuffers[0].mDataByteSize   = frame->nb_samples *
-                                        av_get_bytes_per_sample(avctx->sample_fmt) *
-                                        avctx->channels;
-    data->mBuffers[0].mData           = frame->data[0];
+    data->mBuffers[0].mDataByteSize   = audio_data_size;
+    data->mBuffers[0].mData           = at->audio_data_buf;
+    memcpy(at->audio_data_buf, frame->data[0], data->mBuffers[0].mDataByteSize);
     if (*nb_packets > frame->nb_samples)
         *nb_packets = frame->nb_samples;
 
@@ -565,6 +585,8 @@ static av_cold int ffat_close_encoder(AVCodecContext *avctx)
     ff_bufqueue_discard_all(&at->frame_queue);
     ff_bufqueue_discard_all(&at->used_frame_queue);
     ff_af_queue_close(&at->afq);
+    at->audio_data_buf_size = 0;
+    av_freep(&at->audio_data_buf);
     return 0;
 }
 
-- 
2.14.3 (Apple Git-98)