The example transcode_aac.c uses a global pts for counting. For libavcodec
novices this can cause them to overlook this and result with incorrect "start"
times of output files if called multiple times (see user error resulting in bug
report https://trac.ffmpeg.org/ticket/9228)
From 52cbed063ee54e667905ca243e8ee4a811a108dc Mon Sep 17 00:00:00 2001
From: whatdoineed2do/Ray <whatdoineed...@gmail.com>
Date: Tue, 11 May 2021 13:16:48 +0100
Subject: [PATCH] Do not use global pts for frame pts counting
Signed-off-by: whatdoineed2do/Ray <whatdoineed...@gmail.com>
---
doc/examples/transcode_aac.c | 18 ++++++++++--------
1 file changed, 10 insertions(+), 8 deletions(-)
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
index c9badaa561..7f6148fa15 100644
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -648,8 +648,6 @@ static int init_output_frame(AVFrame **frame,
return 0;
}
-/* Global timestamp for the audio frames. */
-static int64_t pts = 0;
/**
* Encode one frame worth of audio to the output file.
@@ -663,6 +661,7 @@ static int64_t pts = 0;
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
+ int64_t* pts,
int *data_present)
{
/* Packet used for temporary storage. */
@@ -675,8 +674,8 @@ static int encode_audio_frame(AVFrame *frame,
/* Set a timestamp based on the sample rate for the container. */
if (frame) {
- frame->pts = pts;
- pts += frame->nb_samples;
+ frame->pts = *pts;
+ *pts += frame->nb_samples;
}
/* Send the audio frame stored in the temporary packet to the encoder.
@@ -734,7 +733,8 @@ cleanup:
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
- AVCodecContext *output_codec_context)
+ AVCodecContext *output_codec_context,
+ int64_t* pts)
{
/* Temporary storage of the output samples of the frame written to the
file. */
AVFrame *output_frame;
@@ -759,7 +759,7 @@ static int load_encode_and_write(AVAudioFifo *fifo,
/* Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
- output_codec_context, &data_written)) {
+ output_codec_context, pts, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
@@ -790,6 +790,8 @@ int main(int argc, char **argv)
SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
+ /* timestamp for the audio frames. */
+ int64_t pts = 0;
if (argc != 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -850,7 +852,7 @@ int main(int argc, char **argv)
/* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file. */
if (load_encode_and_write(fifo, output_format_context,
- output_codec_context))
+ output_codec_context, &pts))
goto cleanup;
/* If we are at the end of the input file and have encoded
@@ -861,7 +863,7 @@ int main(int argc, char **argv)
do {
data_written = 0;
if (encode_audio_frame(NULL, output_format_context,
- output_codec_context, &data_written))
+ output_codec_context, &pts,
&data_written))
goto cleanup;
} while (data_written);
break;
--
2.26.3
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