[FFmpeg-cvslog] examples/transcode_aac: properly select the output sample format

[Andreas Unterweger]

ffmpeg | branch: master | Andreas Unterweger <dustsigns at gmail.com> | Tue Jan 27 08:58:47 2015 +0100| [749a89d1b8bb73b4d4f14c48f33259a1300c1761] | committer: Anton Khirnov

examples/transcode_aac: properly select the output sample format

Makes the example work with all the supported AAC encoders.

Signed-off-by: Anton Khirnov <anton at khirnov.net>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=749a89d1b8bb73b4d4f14c48f33259a1300c1761
---

 doc/examples/transcode_aac.c |   16 +++++++++-------
 1 file changed, 9 insertions(+), 7 deletions(-)

diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
index 6206afe..75dd1e2 100644
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -40,11 +40,9 @@
 #include "libavresample/avresample.h"
 
 /** The output bit rate in kbit/s */
-#define OUTPUT_BIT_RATE 48000
+#define OUTPUT_BIT_RATE 96000
 /** The number of output channels */
 #define OUTPUT_CHANNELS 2
-/** The audio sample output format */
-#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
 
 /**
  * Convert an error code into a text message.
@@ -178,9 +176,12 @@ static int open_output_file(const char *filename,
     (*output_codec_context)->channels       = OUTPUT_CHANNELS;
     (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
     (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
-    (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
+    (*output_codec_context)->sample_fmt     = output_codec->sample_fmts[0];
     (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
 
+    /** Allow the use of the experimental AAC encoder */
+    (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+
     /**
      * Some container formats (like MP4) require global headers to be present
      * Mark the encoder so that it behaves accordingly.
@@ -276,10 +277,11 @@ static int init_resampler(AVCodecContext *input_codec_context,
 }
 
 /** Initialize a FIFO buffer for the audio samples to be encoded. */
-static int init_fifo(AVAudioFifo **fifo)
+static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
 {
     /** Create the FIFO buffer based on the specified output sample format. */
-    if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
+    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
+                                      output_codec_context->channels, 1))) {
         fprintf(stderr, "Could not allocate FIFO\n");
         return AVERROR(ENOMEM);
     }
@@ -673,7 +675,7 @@ int main(int argc, char **argv)
                        &resample_context))
         goto cleanup;
     /** Initialize the FIFO buffer to store audio samples to be encoded. */
-    if (init_fifo(&fifo))
+    if (init_fifo(&fifo, output_codec_context))
         goto cleanup;
     /** Write the header of the output file container. */
     if (write_output_file_header(output_format_context))