1 files changed, 43 insertions, 60 deletions

diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
index 60698f719f..339d65c71c 100644
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -1,18 +1,18 @@
 /*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -21,7 +21,7 @@
  * simple audio converter
  *
  * @example transcode_aac.c
- * Convert an input audio file to AAC in an MP4 container using Libav.
+ * Convert an input audio file to AAC in an MP4 container using FFmpeg.
  * @author Andreas Unterweger (dustsigns@gmail.com)
  */
 
@@ -33,11 +33,12 @@
 #include "libavcodec/avcodec.h"
 
 #include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/frame.h"
 #include "libavutil/opt.h"
 
-#include "libavresample/avresample.h"
+#include "libswresample/swresample.h"
 
 /** The output bit rate in kbit/s */
 #define OUTPUT_BIT_RATE 96000
@@ -49,7 +50,7 @@
  * @param error Error code to be converted
  * @return Corresponding error text (not thread-safe)
  */
-static char *const get_error_text(const int error)
+static const char *get_error_text(const int error)
 {
     static char error_buffer[255];
     av_strerror(error, error_buffer, sizeof(error_buffer));
@@ -203,7 +204,7 @@ static int open_output_file(const char *filename,
     return 0;
 
 cleanup:
-    avio_close((*output_format_context)->pb);
+    avio_closep(&(*output_format_context)->pb);
     avformat_free_context(*output_format_context);
     *output_format_context = NULL;
     return error < 0 ? error : AVERROR_EXIT;
@@ -231,52 +232,46 @@ static int init_input_frame(AVFrame **frame)
 /**
  * Initialize the audio resampler based on the input and output codec settings.
  * If the input and output sample formats differ, a conversion is required
- * libavresample takes care of this, but requires initialization.
+ * libswresample takes care of this, but requires initialization.
  */
 static int init_resampler(AVCodecContext *input_codec_context,
                           AVCodecContext *output_codec_context,
-                          AVAudioResampleContext **resample_context)
+                          SwrContext **resample_context)
 {
-    /**
-     * Only initialize the resampler if it is necessary, i.e.,
-     * if and only if the sample formats differ.
-     */
-    if (input_codec_context->sample_fmt != output_codec_context->sample_fmt ||
-        input_codec_context->channels != output_codec_context->channels) {
         int error;
 
-        /** Create a resampler context for the conversion. */
-        if (!(*resample_context = avresample_alloc_context())) {
-            fprintf(stderr, "Could not allocate resample context\n");
-            return AVERROR(ENOMEM);
-        }
-
         /**
+         * Create a resampler context for the conversion.
          * Set the conversion parameters.
          * Default channel layouts based on the number of channels
          * are assumed for simplicity (they are sometimes not detected
          * properly by the demuxer and/or decoder).
          */
-        av_opt_set_int(*resample_context, "in_channel_layout",
-                       av_get_default_channel_layout(input_codec_context->channels), 0);
-        av_opt_set_int(*resample_context, "out_channel_layout",
-                       av_get_default_channel_layout(output_codec_context->channels), 0);
-        av_opt_set_int(*resample_context, "in_sample_rate",
-                       input_codec_context->sample_rate, 0);
-        av_opt_set_int(*resample_context, "out_sample_rate",
-                       output_codec_context->sample_rate, 0);
-        av_opt_set_int(*resample_context, "in_sample_fmt",
-                       input_codec_context->sample_fmt, 0);
-        av_opt_set_int(*resample_context, "out_sample_fmt",
-                       output_codec_context->sample_fmt, 0);
+        *resample_context = swr_alloc_set_opts(NULL,
+                                              av_get_default_channel_layout(output_codec_context->channels),
+                                              output_codec_context->sample_fmt,
+                                              output_codec_context->sample_rate,
+                                              av_get_default_channel_layout(input_codec_context->channels),
+                                              input_codec_context->sample_fmt,
+                                              input_codec_context->sample_rate,
+                                              0, NULL);
+        if (!*resample_context) {
+            fprintf(stderr, "Could not allocate resample context\n");
+            return AVERROR(ENOMEM);
+        }
+        /**
+        * Perform a sanity check so that the number of converted samples is
+        * not greater than the number of samples to be converted.
+        * If the sample rates differ, this case has to be handled differently
+        */
+        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
 
         /** Open the resampler with the specified parameters. */
-        if ((error = avresample_open(*resample_context)) < 0) {
+        if ((error = swr_init(*resample_context)) < 0) {
             fprintf(stderr, "Could not open resample context\n");
-            avresample_free(resample_context);
+            swr_free(resample_context);
             return error;
         }
-    }
     return 0;
 }
 
@@ -317,7 +312,7 @@ static int decode_audio_frame(AVFrame *frame,
 
     /** Read one audio frame from the input file into a temporary packet. */
     if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
-        /** If we are the the end of the file, flush the decoder below. */
+        /** If we are at the end of the file, flush the decoder below. */
         if (error == AVERROR_EOF)
             *finished = 1;
         else {
@@ -396,30 +391,21 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
  * The conversion happens on a per-frame basis, the size of which is specified
  * by frame_size.
  */
-static int convert_samples(uint8_t **input_data,
+static int convert_samples(const uint8_t **input_data,
                            uint8_t **converted_data, const int frame_size,
-                           AVAudioResampleContext *resample_context)
+                           SwrContext *resample_context)
 {
     int error;
 
     /** Convert the samples using the resampler. */
-    if ((error = avresample_convert(resample_context, converted_data, 0,
-                                    frame_size, input_data, 0, frame_size)) < 0) {
+    if ((error = swr_convert(resample_context,
+                             converted_data, frame_size,
+                             input_data    , frame_size)) < 0) {
         fprintf(stderr, "Could not convert input samples (error '%s')\n",
                 get_error_text(error));
         return error;
     }
 
-    /**
-     * Perform a sanity check so that the number of converted samples is
-     * not greater than the number of samples to be converted.
-     * If the sample rates differ, this case has to be handled differently
-     */
-    if (avresample_available(resample_context)) {
-        fprintf(stderr, "Converted samples left over\n");
-        return AVERROR_EXIT;
-    }
-
     return 0;
 }
 
@@ -456,7 +442,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
                                          AVFormatContext *input_format_context,
                                          AVCodecContext *input_codec_context,
                                          AVCodecContext *output_codec_context,
-                                         AVAudioResampleContext *resampler_context,
+                                         SwrContext *resampler_context,
                                          int *finished)
 {
     /** Temporary storage of the input samples of the frame read from the file. */
@@ -493,7 +479,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
          * Convert the input samples to the desired output sample format.
          * This requires a temporary storage provided by converted_input_samples.
          */
-        if (convert_samples(input_frame->extended_data, converted_input_samples,
+        if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
                             input_frame->nb_samples, resampler_context))
             goto cleanup;
 
@@ -664,7 +650,7 @@ int main(int argc, char **argv)
 {
     AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
     AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
-    AVAudioResampleContext *resample_context = NULL;
+    SwrContext *resample_context = NULL;
     AVAudioFifo *fifo = NULL;
     int ret = AVERROR_EXIT;
 
@@ -768,14 +754,11 @@ int main(int argc, char **argv)
 cleanup:
     if (fifo)
         av_audio_fifo_free(fifo);
-    if (resample_context) {
-        avresample_close(resample_context);
-        avresample_free(&resample_context);
-    }
+    swr_free(&resample_context);
     if (output_codec_context)
         avcodec_close(output_codec_context);
     if (output_format_context) {
-        avio_close(output_format_context->pb);
+        avio_closep(&output_format_context->pb);
         avformat_free_context(output_format_context);
     }
     if (input_codec_context)