[FFmpeg-devel] New API usage example (encode_raw_audio_file_to_aac)

Submitted by Paolo Prete on March 31, 2017, 1:11 p.m.

Details

Message ID 20170331131111.25457-1-p4olo_prete@yahoo.it
State New
Headers show

Commit Message

Paolo Prete March 31, 2017, 1:11 p.m.
---
 configure                                   |   2 +
 doc/Makefile                                |  41 ++--
 doc/examples/.gitignore                     |   1 +
 doc/examples/Makefile                       |   1 +
 doc/examples/encode_raw_audio_file_to_aac.c | 338 ++++++++++++++++++++++++++++
 5 files changed, 363 insertions(+), 20 deletions(-)
 create mode 100644 doc/examples/encode_raw_audio_file_to_aac.c

Comments

Ivan Kalvachev May 6, 2017, 8:49 a.m.
On 3/31/17, Paolo Prete <p4olo_prete-at-yahoo.it@ffmpeg.org> wrote:
> ---
>  configure                                   |   2 +
>  doc/Makefile                                |  41 ++--
>  doc/examples/.gitignore                     |   1 +
>  doc/examples/Makefile                       |   1 +
>  doc/examples/encode_raw_audio_file_to_aac.c | 338
> ++++++++++++++++++++++++++++
>  5 files changed, 363 insertions(+), 20 deletions(-)
>  create mode 100644 doc/examples/encode_raw_audio_file_to_aac.c

What is the status of this patch?
It's been quite a while and there have been no objections.

Maybe it should be committed.

Patch hide | download patch | download mbox

diff --git a/configure b/configure
index 6d76cf7..1069f9f 100755
--- a/configure
+++ b/configure
@@ -1466,6 +1466,7 @@  EXAMPLE_LIST="
     decode_video_example
     demuxing_decoding_example
     encode_audio_example
+    encode_raw_audio_file_to_aac_example
     encode_video_example
     extract_mvs_example
     filter_audio_example
@@ -3175,6 +3176,7 @@  decode_audio_example_deps="avcodec avutil"
 decode_video_example_deps="avcodec avutil"
 demuxing_decoding_example_deps="avcodec avformat avutil"
 encode_audio_example_deps="avcodec avutil"
+encode_raw_audio_file_to_aac_example_deps="avcodec avformat avutil swresample"
 encode_video_example_deps="avcodec avutil"
 extract_mvs_example_deps="avcodec avformat avutil"
 filter_audio_example_deps="avfilter avutil"
diff --git a/doc/Makefile b/doc/Makefile
index c193fc3..a7c349b 100644
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -36,26 +36,27 @@  DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 
-DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
-DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
-DOC_EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE)      += decode_audio
-DOC_EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE)      += decode_video
-DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE)      += encode_audio
-DOC_EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE)      += encode_video
-DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
-DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
-DOC_EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE)  += http_multiclient
-DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
-DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
-DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
-DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
-DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
-DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
-DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
+DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)                      += avio_dir_cmd
+DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)                      += avio_reading
+DOC_EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE)                      += decode_audio
+DOC_EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE)                      += decode_video
+DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE)                 += demuxing_decoding
+DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE)                      += encode_audio
+DOC_EXAMPLES-$(CONFIG_ENCODE_RAW_AUDIO_FILE_TO_AAC_EXAMPLE)      += encode_raw_audio_file_to_aac
+DOC_EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE)                      += encode_video
+DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)                       += extract_mvs
+DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)                      += filter_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)                   += filtering_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)                   += filtering_video
+DOC_EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE)                  += http_multiclient
+DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)                          += metadata
+DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)                            += muxing
+DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)                            += qsvdec
+DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)                          += remuxing
+DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)                  += resampling_audio
+DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)                     += scaling_video
+DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)                     += transcode_aac
+DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)                       += transcoding
 ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
 
 DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
diff --git a/doc/examples/.gitignore b/doc/examples/.gitignore
index 6bd9dc1..7b25718 100644
--- a/doc/examples/.gitignore
+++ b/doc/examples/.gitignore
@@ -4,6 +4,7 @@ 
 /decode_video
 /demuxing_decoding
 /encode_audio
+/encode_raw_audio_file_to_aac
 /encode_video
 /extract_mvs
 /filter_audio
diff --git a/doc/examples/Makefile b/doc/examples/Makefile
index 2d0a306..24929e3 100644
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -17,6 +17,7 @@  EXAMPLES=       avio_dir_cmd                       \
                 decode_video                       \
                 demuxing_decoding                  \
                 encode_audio                       \
+                encode_raw_audio_file_to_aac       \
                 encode_video                       \
                 extract_mvs                        \
                 filtering_video                    \
diff --git a/doc/examples/encode_raw_audio_file_to_aac.c b/doc/examples/encode_raw_audio_file_to_aac.c
new file mode 100644
index 0000000..5e6df5c
--- /dev/null
+++ b/doc/examples/encode_raw_audio_file_to_aac.c
@@ -0,0 +1,338 @@ 
+/*
+ * Copyright (c) 2017 Paolo Prete (p4olo_prete@yahoo.it)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * API example for adts-aac encoding raw audio files. 
+ * This example reads a raw audio input file, converts it to float-planar format, performs 
+ * aac encoding and puts the encoded frames into an ADTS container. 
+ * The encoded stream is written to a file named "out.aac"
+ * It can be adapted, with few changes, to a custom raw audio source (i.e: a live one). 
+ * It uses a custom I/O write callback (write_adts_muxed_data()) in order to show to the user 
+ * how to access muxed packets written in memory, before they are written to the output file.
+ * The raw input audio file can be created with: 
+ * 
+ * ffmpeg -i some_audio_file -f f32le -acodec pcm_f32le -ac 2 -ar 16000 raw_audio_file.raw
+ * 
+ * @example encode_raw_audio_file_to_aac.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavutil/timestamp.h>
+#include <libswresample/swresample.h>
+
+#define ENCODER_BITRATE 64000
+#define SAMPLE_RATE 16000
+#define INPUT_SAMPLE_FMT AV_SAMPLE_FMT_FLT
+#define CHANNELS 2
+
+static int encoded_pkt_counter = 1;
+
+static int write_adts_muxed_data(void *opaque, uint8_t *adts_data, int size)
+{
+    FILE *encoded_audio_file = (FILE *)opaque;
+    fwrite(adts_data, 1, size, encoded_audio_file); //(f)
+    return size;
+}
+
+static int mux_aac_packet_to_adts (AVPacket *encoded_audio_packet, AVFormatContext *adts_container_ctx)
+{
+    int ret_val;
+    if ((ret_val = av_write_frame(adts_container_ctx, encoded_audio_packet)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error calling av_write_frame() (error '%s')\n", av_err2str(ret_val));
+    }
+    else {
+        av_log(NULL, AV_LOG_INFO, "Encoded AAC packet %d, size=%d, pts_time=%s\n", 
+               encoded_pkt_counter, encoded_audio_packet->size, 
+               av_ts2timestr(encoded_audio_packet->pts, &adts_container_ctx->streams[0]->time_base));
+    }
+    return ret_val;
+}
+
+static int check_if_samplerate_is_supported(AVCodec *audio_codec, int samplerate)
+{
+    const int *samplerates_list = audio_codec->supported_samplerates;
+    while (*samplerates_list) {
+        if (*samplerates_list == samplerate)
+            return 0;
+        samplerates_list++;      
+    }
+    return 1;
+}
+
+int main(int argc, char **argv)
+{
+    FILE *input_audio_file = NULL, *encoded_audio_file = NULL;
+    AVCodec *audio_codec = NULL;
+    AVCodecContext *audio_encoder_ctx = NULL;
+    AVFrame *input_audio_frame = NULL, *converted_audio_frame = NULL;
+    SwrContext *audio_convert_context = NULL;
+    AVOutputFormat *adts_container = NULL;
+    AVFormatContext *adts_container_ctx = NULL;
+    uint8_t *adts_container_buffer = NULL;
+    size_t adts_container_buffer_size = 4096;
+    AVIOContext *adts_avio_ctx = NULL;
+    AVStream *adts_stream = NULL;  
+    AVPacket *encoded_audio_packet = NULL;
+    int ret_val = 0;
+    int audio_bytes_to_encode;
+    int64_t curr_pts;
+    
+    if (argc != 2) {
+        printf("Usage: %s <raw audio input file (CHANNELS, INPUT_SAMPLE_FMT, SAMPLE_RATE)>\n", argv[0]);
+        return 1;
+    }    
+    
+    input_audio_file = fopen(argv[1], "rb");
+    if (!input_audio_file) {
+        av_log(NULL, AV_LOG_ERROR, "Could not open input audio file\n");
+        return AVERROR_EXIT;
+    }
+    
+    encoded_audio_file = fopen("out.aac", "wb");  
+    if (!encoded_audio_file) {
+        av_log(NULL, AV_LOG_ERROR, "Could not open output audio file\n");
+        fclose(input_audio_file);        
+        return AVERROR_EXIT;
+    }
+
+    av_register_all();
+
+    /**
+    * Allocate the encoder's context and open the encoder
+    */
+    audio_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
+    if (!audio_codec) {
+        av_log(NULL, AV_LOG_ERROR, "Could not find aac codec\n");
+        ret_val = AVERROR_EXIT;
+        goto end;
+    }
+    if ((ret_val = check_if_samplerate_is_supported(audio_codec, SAMPLE_RATE)) != 0) {
+        av_log(NULL, AV_LOG_ERROR, "Audio codec doesn't support input samplerate %d\n", SAMPLE_RATE);
+        goto end;
+    }    
+    audio_encoder_ctx = avcodec_alloc_context3(audio_codec);
+    if (!audio_codec) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate the encoding context\n");
+        ret_val = AVERROR_EXIT;
+        goto end;
+    }
+    audio_encoder_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+    audio_encoder_ctx->bit_rate = ENCODER_BITRATE;
+    audio_encoder_ctx->sample_rate = SAMPLE_RATE;
+    audio_encoder_ctx->channels = CHANNELS;
+    audio_encoder_ctx->channel_layout = av_get_default_channel_layout(CHANNELS);
+    audio_encoder_ctx->time_base = (AVRational){1, SAMPLE_RATE};
+    audio_encoder_ctx->codec_type = AVMEDIA_TYPE_AUDIO ;
+    if ((ret_val = avcodec_open2(audio_encoder_ctx, audio_codec, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Could not open input codec (error '%s')\n", av_err2str(ret_val));
+        goto end;
+    }
+    
+    /**
+    * Allocate an AVFrame which will be filled with the input file's data. 
+    */
+    if (!(input_audio_frame = av_frame_alloc())) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate input frame\n");
+        ret_val = AVERROR(ENOMEM);
+        goto end;
+    }    
+    input_audio_frame->nb_samples     = audio_encoder_ctx->frame_size;
+    input_audio_frame->format         = INPUT_SAMPLE_FMT;
+    input_audio_frame->channels       = CHANNELS;
+    input_audio_frame->sample_rate    = SAMPLE_RATE;
+    input_audio_frame->channel_layout = av_get_default_channel_layout(CHANNELS);
+    // Allocate the frame's data buffer 
+    if ((ret_val = av_frame_get_buffer(input_audio_frame, 0)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, 
+               "Could not allocate container for input frame samples (error '%s')\n", av_err2str(ret_val));
+        ret_val = AVERROR(ENOMEM);
+        goto end;
+    }    
+    
+    /**
+    * Input data must be converted to float-planar format, which is the format required by the AAC encoder. 
+    * We allocate a SwrContext and an AVFrame (which will contain the converted samples) for this task. 
+    * The AVFrame will feed the encoding function (avcodec_send_frame())
+    */
+    audio_convert_context = swr_alloc_set_opts(NULL, 
+                                               av_get_default_channel_layout(CHANNELS), 
+                                               AV_SAMPLE_FMT_FLTP,
+                                               SAMPLE_RATE,
+                                               av_get_default_channel_layout(CHANNELS),
+                                               INPUT_SAMPLE_FMT,
+                                               SAMPLE_RATE, 
+                                               0, 
+                                               NULL);
+    if (!audio_convert_context) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate resample context\n");                 
+        ret_val = AVERROR(ENOMEM);
+        goto end;
+    }
+    if (!(converted_audio_frame = av_frame_alloc())) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate resampled frame\n");
+        ret_val = AVERROR(ENOMEM);
+        goto end;
+    }
+    converted_audio_frame->nb_samples     = audio_encoder_ctx->frame_size;
+    converted_audio_frame->format         = audio_encoder_ctx->sample_fmt;
+    converted_audio_frame->channels       = audio_encoder_ctx->channels;
+    converted_audio_frame->channel_layout = audio_encoder_ctx->channel_layout;
+    converted_audio_frame->sample_rate    = SAMPLE_RATE;     
+    if ((ret_val = av_frame_get_buffer(converted_audio_frame, 0)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, 
+               "Could not allocate a buffer for resampled frame samples (error '%s')\n", av_err2str(ret_val));
+        goto end;
+    }    
+    
+    /**
+    * Create the ADTS container for the encoded frames
+    */
+    adts_container = av_guess_format("adts", NULL, NULL);
+    if (!adts_container) {
+        av_log(NULL, AV_LOG_ERROR, "Could not find adts output format\n");       
+        ret_val = AVERROR_EXIT;
+        goto end;
+    }     
+    if ((ret_val = avformat_alloc_output_context2(&adts_container_ctx, adts_container, "", NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Could not create output context (error '%s')\n", av_err2str(ret_val));
+        goto end;
+    }
+    if (!(adts_container_buffer = av_malloc(adts_container_buffer_size))) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate a buffer for the I/O output context\n");       
+        ret_val = AVERROR(ENOMEM);
+        goto end; 
+    }
+    /** 
+    * Create an I/O context for the adts container with a write callback (write_adts_muxed_data()), 
+    * so that muxed data will be accessed through this function and can be managed by the user.
+    */
+    if (!(adts_avio_ctx = avio_alloc_context(adts_container_buffer, adts_container_buffer_size,
+                                             1, encoded_audio_file, NULL,
+                                             &write_adts_muxed_data, NULL))) {
+        av_log(NULL, AV_LOG_ERROR, "Could not create I/O output context\n");
+        ret_val = AVERROR_EXIT;
+        goto end;
+    }
+    // Link the container's context to the previous I/O context
+    adts_container_ctx->pb = adts_avio_ctx;
+    if (!(adts_stream = avformat_new_stream(adts_container_ctx, NULL))) {
+        av_log(NULL, AV_LOG_ERROR, "Could not create new stream\n");       
+        ret_val = AVERROR(ENOMEM);
+        goto end;        
+    }    
+    adts_stream->id = adts_container_ctx->nb_streams-1;
+    // Copy the encoder's parameters 
+    avcodec_parameters_from_context(adts_stream->codecpar, audio_encoder_ctx);    
+    // Allocate the stream private data and write the stream header
+    if (avformat_write_header(adts_container_ctx, NULL) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "avformat_write_header() error\n");
+        ret_val = AVERROR_EXIT;
+        goto end;
+    }
+    
+    /**
+    * Fill the input frame's data buffer with input file data (a), 
+    * Convert the input frame to float-planar format (b), 
+    * Send the converted frame to the encoder (c), 
+    * Get the encoded packet (d),
+    * Send the encoded packet to the adts muxer (e). 
+    * Muxed data is caught in write_adts_muxed_data() callback and it is written
+    * to the output audio file ( (f) : see above)
+    */
+    encoded_audio_packet = av_packet_alloc();
+    while (1) {
+        
+        audio_bytes_to_encode = fread(input_audio_frame->data[0], 1, 
+                                      input_audio_frame->linesize[0], input_audio_file); //(a)
+        if (audio_bytes_to_encode != input_audio_frame->linesize[0]) {            
+            break;
+        }
+        else {
+            if (av_frame_make_writable(converted_audio_frame) < 0) {
+                av_log(NULL, AV_LOG_ERROR, "av_frame_make_writable() error\n");
+                ret_val = AVERROR_EXIT;
+                goto end;                
+            }
+            
+            if ((ret_val = swr_convert_frame(audio_convert_context, 
+                                             converted_audio_frame, 
+                                             (const AVFrame *)input_audio_frame)) != 0) { //(b)
+                av_log(NULL, AV_LOG_ERROR, 
+                       "Error resampling input audio frame (error '%s')\n", av_err2str(ret_val));
+                goto end;
+            }            
+            
+            if ((ret_val = avcodec_send_frame(audio_encoder_ctx, converted_audio_frame)) == 0)  //(c)
+                ret_val = avcodec_receive_packet(audio_encoder_ctx, encoded_audio_packet); //(d)
+            else {
+                av_log(NULL, AV_LOG_ERROR, 
+                       "Error encoding frame (error '%s')\n", av_err2str(ret_val));
+                goto end;
+            }
+            
+            if (ret_val == 0) {                
+                curr_pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1);
+                encoded_audio_packet->pts = encoded_audio_packet->dts = curr_pts;           
+                if ((ret_val = mux_aac_packet_to_adts(encoded_audio_packet, adts_container_ctx)) < 0) //(e)
+                    goto end;
+                encoded_pkt_counter++;
+            }
+            else if (ret_val != AVERROR(EAGAIN)) {
+                av_log(NULL, AV_LOG_ERROR, 
+                       "Error receiving encoded packet (error '%s')\n", av_err2str(ret_val));
+                goto end;                
+            }
+        }            
+    }
+    // Flush cached packets    
+    if ((ret_val = avcodec_send_frame(audio_encoder_ctx, NULL)) == 0)   
+        do {
+            ret_val = avcodec_receive_packet(audio_encoder_ctx, encoded_audio_packet);
+            if (ret_val == 0) {
+                curr_pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1);
+                encoded_audio_packet->pts = encoded_audio_packet->dts = curr_pts;           
+                if ((ret_val = mux_aac_packet_to_adts(encoded_audio_packet, adts_container_ctx)) < 0)
+                    goto end;
+                encoded_pkt_counter++;
+            }        
+        } while (ret_val == 0);
+
+    av_write_trailer(adts_container_ctx);  
+
+end:
+
+    fclose(input_audio_file);
+    fclose(encoded_audio_file);    
+    avcodec_free_context(&audio_encoder_ctx);     
+    av_frame_free(&input_audio_frame);    
+    swr_free(&audio_convert_context);       
+    av_frame_free(&converted_audio_frame); 
+    avformat_free_context(adts_container_ctx);
+    av_freep(&adts_avio_ctx);  
+    av_freep(&adts_container_buffer); 
+    av_packet_free(&encoded_audio_packet); 
+    
+    return ret_val;
+    
+}