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authorAndreas Unterweger <dustsigns@gmail.com>2015-01-27 08:58:47 +0100
committerAnton Khirnov <anton@khirnov.net>2015-01-27 08:58:47 +0100
commit749a89d1b8bb73b4d4f14c48f33259a1300c1761 (patch)
treed25e859c1863f0e3fdf63bb9e84ef17702df695d /doc/examples/transcode_aac.c
parent443b71928b2f36362e805c037751e6c3c79ea4e8 (diff)
downloadffmpeg-749a89d1b8bb73b4d4f14c48f33259a1300c1761.tar.gz
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@khirnov.net>
Diffstat (limited to 'doc/examples/transcode_aac.c')
-rw-r--r--doc/examples/transcode_aac.c16
1 files changed, 9 insertions, 7 deletions
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
index 6206afe2e4..75dd1e2738 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))