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I want to insert an audio at a certain time on the video using moviepy.editor
28 novembre 2022, par Khong Phai HexI want to insert an audio at a certain time on the video using moviepy.editor, how do I do it ?, or do you have another way ? please show me.


Examples like this :enter image description here


I was expecting someone to answer me, because I searched for a long time but couldn't.


-
ffmpeg "Time code of first frame" not making sense [closed]
23 janvier, par Kevin VaskoI can't seem to make sense of where ffmpeg is coming up with the "Time code of first frame" value.


My original file that I am using.


This video has the general timestamps of the original encoding date (aka the start time of the video the way I understanding it) and a duration of 8 min and 52s


$ mediainfo GL0100002_output.mp4 
General
Complete name : GL0100002_output.mp4
Format : MPEG-4
Format profile : Base Media
Codec ID : isom (isom/iso2/avc1/mp41)
File size : 93.7 MiB
Duration : 8 min 52 s
Overall bit rate : 1 476 kb/s
Encoded date : UTC 2022-02-23 15:18:42
Tagged date : UTC 2022-02-23 15:18:42
Writing application : Lavf58.29.100

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L3
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, Reference frames : 4 frames
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 8 min 52 s
Bit rate : 1 342 kb/s
Width : 848 pixels
Height : 480 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 23.976 (24000/1001) FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.138
Stream size : 85.2 MiB (91%)
Title : GoPro AVC 
Writing library : x264 core 155 r2917 0a84d98
Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=7 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=12 / lookahead_threads=2 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=1 / weightb=1 / open_gop=0 / weightp=2 / keyint=250 / keyint_min=23 / scenecut=40 / intra_refresh=0 / rc_lookahead=40 / rc=crf / mbtree=1 / crf=23.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:1.00
Language : English
Encoded date : UTC 2022-02-23 15:18:42
Tagged date : UTC 2022-02-23 15:18:42
Color range : Full
Codec configuration box : avcC

Audio
ID : 2
Format : AAC LC
Format/Info : Advanced Audio Codec Low Complexity
Codec ID : mp4a-40-2
Duration : 8 min 52 s
Bit rate mode : Constant
Bit rate : 129 kb/s
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48.0 kHz
Frame rate : 46.875 FPS (1024 SPF)
Compression mode : Lossy
Stream size : 8.16 MiB (9%)
Title : GoPro AAC 
Language : English
Default : Yes
Alternate group : 1
Encoded date : UTC 2022-02-23 15:18:42
Tagged date : UTC 2022-02-23 15:18:42

Other
ID : 3
Type : Time code
Format : QuickTime TC
Duration : 8 min 52 s
Frame rate : 23.976 (24000/1001) FPS
Time code of first frame : 15:57:41:15
Time code, striped : Yes
Title : GoPro AVC 
Language : English
Default : No
Encoded date : UTC 2022-02-23 15:18:42
Tagged date : UTC 2022-02-23 15:18:42


Now if I run


ffmpeg -y -nostdin -v error -ss 00:00:00.042 -i GL0100002_output.mp4 -t 00:00:25.192 -map 0 -map -0:d -c:v libx264 -preset fast -crf 21 -c:a aac -sn GL0100002_test.DSME_0001.mp4

I get the following mediainfo


$ mediainfo GL0100002_test.DSME_0001.mp4 
General
Complete name : GL0100002_test.DSME_0001.mp4
Format : MPEG-4
Format profile : Base Media
Codec ID : isom (isom/iso2/avc1/mp41)
File size : 4.54 MiB
Duration : 25 s 234 ms
Overall bit rate : 1 510 kb/s
Writing application : Lavf58.29.100

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L3
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, Reference frames : 4 frames
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 25 s 234 ms
Bit rate : 1 376 kb/s
Width : 848 pixels
Height : 480 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 23.976 (24000/1001) FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.141
Stream size : 4.14 MiB (91%)
Title : GoPro AVC 
Writing library : x264 core 155 r2917 0a84d98
Encoding settings : cabac=1 / ref=2 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=6 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=12 / lookahead_threads=2 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=1 / weightb=1 / open_gop=0 / weightp=1 / keyint=250 / keyint_min=23 / scenecut=40 / intra_refresh=0 / rc_lookahead=30 / rc=crf / mbtree=1 / crf=21.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:1.00
Language : English
Color range : Full
Codec configuration box : avcC

Audio
ID : 2
Format : AAC LC
Format/Info : Advanced Audio Codec Low Complexity
Codec ID : mp4a-40-2
Duration : 25 s 214 ms
Duration_LastFrame : -3 ms
Bit rate mode : Constant
Bit rate : 129 kb/s
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48.0 kHz
Frame rate : 46.875 FPS (1024 SPF)
Compression mode : Lossy
Stream size : 398 KiB (9%)
Title : GoPro AAC 
Language : English
Default : Yes
Alternate group : 1

Other
ID : 3
Type : Time code
Format : QuickTime TC
Duration : 25 s 234 ms
Frame rate : 23.976 (24000/1001) FPS
Time code of first frame : 15:57:41:15
Time code, striped : Yes
Title : GoPro AVC 
Language : English
Default : No


But what I don't understand is where
Time code of first frameis coming from. I would expect it would be00:00:00.042or evenUTC 2022-02-23 15:18:42+ .042 but appear to be.

Time code of first frame : 15:57:41:15


Is this potentially clobbering together something to get this time that isn't obvious ?


-
ffmpeg : libavformat/libswresample to transcode and resample at same time
21 février 2024, par whatdoidoI want to transcode and down/re-sample the audio for output using ffmpeg's libav*/libswresample - I am using ffmpeg's (4.x) transcode_aac.c and resample_audio.c as reference - but the code produces audio with glitches that is clearly not what ffmpeg itself would produce (ie ffmpeg -i foo.wav -ar 22050 foo.m4a)


Based on the ffmpeg examples, to resample audio it appears that I need to set the output AVAudioContext and SwrContext sample_rate to what I desire and ensure the swr_convert() is provided with the correct number of output samples based av_rescale_rnd( swr_delay(), ...) once I have an decoded input audio. I've taken care to ensure all the relevant calculations of samples for output are taken into account in the merged code (below) :


- 

- open_output_file() - AVCodecContext.sample_rate (avctx variable) set to our target (down sampled) sample_rate
- read_decode_convert_and_store() is where the work happens : input audio is decoded to an AVFrame and this input frame is converted before being encoded.

- 

- init_converted_samples() and av_samples_alloc() uses the input frame's nb_samples
- ADDED : calc the number of output samples via av_rescale_rnd() and swr_delay()
- UPDATED : convert_samples() and swr_convert() uses the input frame's samples and our calculated output samples as parameters














However the resulting audio file is produced with audio glitches. Does the community know of any references for how transcode AND resample should be done or what is missing in this example ?


/* compile and run:
 gcc -I/usr/include/ffmpeg transcode-swr-aac.c -lavformat -lavutil -lavcodec -lswresample -lm
 ./a.out foo.wav foo.m4a
 */

/*
 * Copyright (c) 2013-2018 Andreas Unterweger
 * 
 * This file is part of FFmpeg. 
 ... ...
 * 
 * @example transcode_aac.c 
 * Convert an input audio file to AAC in an MP4 container using FFmpeg. 
 * Formats other than MP4 are supported based on the output file extension. 
 * @author Andreas Unterweger (xxxx@xxxxx.com)
 */ 
 #include 
 

 #include "libavformat/avformat.h"
 #include "libavformat/avio.h"
 
 #include "libavcodec/avcodec.h"
 
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/frame.h"
 #include "libavutil/opt.h"
 
 #include "libswresample/swresample.h"
 
 #define OUTPUT_BIT_RATE 128000
 #define OUTPUT_CHANNELS 2
 
 static int open_input_file(const char *filename,
 AVFormatContext **input_format_context,
 AVCodecContext **input_codec_context)
 {
 AVCodecContext *avctx;
 const AVCodec *input_codec;
 const AVStream *stream;
 int error;
 
 if ((error = avformat_open_input(input_format_context, filename, NULL,
 NULL)) < 0) {
 fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
 filename, av_err2str(error));
 *input_format_context = NULL;
 return error;
 }
 

 if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
 fprintf(stderr, "Could not open find stream info (error '%s')\n",
 av_err2str(error));
 avformat_close_input(input_format_context);
 return error;
 }
 
 if ((*input_format_context)->nb_streams != 1) {
 fprintf(stderr, "Expected one audio input stream, but found %d\n",
 (*input_format_context)->nb_streams);
 avformat_close_input(input_format_context);
 return AVERROR_EXIT;
 }
 
 stream = (*input_format_context)->streams[0];
 
 if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
 fprintf(stderr, "Could not find input codec\n");
 avformat_close_input(input_format_context);
 return AVERROR_EXIT;
 }
 
 avctx = avcodec_alloc_context3(input_codec);
 if (!avctx) {
 fprintf(stderr, "Could not allocate a decoding context\n");
 avformat_close_input(input_format_context);
 return AVERROR(ENOMEM);
 }
 
 /* Initialize the stream parameters with demuxer information. */
 error = avcodec_parameters_to_context(avctx, stream->codecpar);
 if (error < 0) {
 avformat_close_input(input_format_context);
 avcodec_free_context(&avctx);
 return error;
 }
 
 /* Open the decoder for the audio stream to use it later. */
 if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
 fprintf(stderr, "Could not open input codec (error '%s')\n",
 av_err2str(error));
 avcodec_free_context(&avctx);
 avformat_close_input(input_format_context);
 return error;
 }
 
 /* Set the packet timebase for the decoder. */
 avctx->pkt_timebase = stream->time_base;
 
 /* Save the decoder context for easier access later. */
 *input_codec_context = avctx;
 
 return 0;
 }
 
 static int open_output_file(const char *filename,
 AVCodecContext *input_codec_context,
 AVFormatContext **output_format_context,
 AVCodecContext **output_codec_context)
 {
 AVCodecContext *avctx = NULL;
 AVIOContext *output_io_context = NULL;
 AVStream *stream = NULL;
 const AVCodec *output_codec = NULL;
 int error;
 

 if ((error = avio_open(&output_io_context, filename,
 AVIO_FLAG_WRITE)) < 0) {
 fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
 filename, av_err2str(error));
 return error;
 }
 

 if (!(*output_format_context = avformat_alloc_context())) {
 fprintf(stderr, "Could not allocate output format context\n");
 return AVERROR(ENOMEM);
 }
 

 (*output_format_context)->pb = output_io_context;
 

 if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
 NULL))) {
 fprintf(stderr, "Could not find output file format\n");
 goto cleanup;
 }
 
 if (!((*output_format_context)->url = av_strdup(filename))) {
 fprintf(stderr, "Could not allocate url.\n");
 error = AVERROR(ENOMEM);
 goto cleanup;
 }
 

 if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
 fprintf(stderr, "Could not find an AAC encoder.\n");
 goto cleanup;
 }
 
 /* Create a new audio stream in the output file container. */
 if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
 fprintf(stderr, "Could not create new stream\n");
 error = AVERROR(ENOMEM);
 goto cleanup;
 }
 
 avctx = avcodec_alloc_context3(output_codec);
 if (!avctx) {
 fprintf(stderr, "Could not allocate an encoding context\n");
 error = AVERROR(ENOMEM);
 goto cleanup;
 }
 
 /* Set the basic encoder parameters.
 * SET OUR DESIRED output sample_rate here
 */
 avctx->channels = OUTPUT_CHANNELS;
 avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
 // avctx->sample_rate = input_codec_context->sample_rate;
 avctx->sample_rate = 22050;
 avctx->sample_fmt = output_codec->sample_fmts[0];
 avctx->bit_rate = OUTPUT_BIT_RATE;
 
 avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
 
 /* Set the sample rate for the container. */
 stream->time_base.den = avctx->sample_rate;
 stream->time_base.num = 1;
 
 if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
 avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 
 if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
 fprintf(stderr, "Could not open output codec (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 
 error = avcodec_parameters_from_context(stream->codecpar, avctx);
 if (error < 0) {
 fprintf(stderr, "Could not initialize stream parameters\n");
 goto cleanup;
 }
 
 /* Save the encoder context for easier access later. */
 *output_codec_context = avctx;
 
 return 0;
 
 cleanup:
 avcodec_free_context(&avctx);
 avio_closep(&(*output_format_context)->pb);
 avformat_free_context(*output_format_context);
 *output_format_context = NULL;
 return error < 0 ? error : AVERROR_EXIT;
 }
 
 /**
 * Initialize one data packet for reading or writing.
 */
 static int init_packet(AVPacket **packet)
 {
 if (!(*packet = av_packet_alloc())) {
 fprintf(stderr, "Could not allocate packet\n");
 return AVERROR(ENOMEM);
 }
 return 0;
 }
 
 static int init_input_frame(AVFrame **frame)
 {
 if (!(*frame = av_frame_alloc())) {
 fprintf(stderr, "Could not allocate input frame\n");
 return AVERROR(ENOMEM);
 }
 return 0;
 }
 
 static int init_resampler(AVCodecContext *input_codec_context,
 AVCodecContext *output_codec_context,
 SwrContext **resample_context)
 {
 int error;

 /**
 * create the resample, including ref to the desired output sample rate
 */
 *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 < 0) {
 fprintf(stderr, "Could not allocate resample context\n");
 return AVERROR(ENOMEM);
 }
 
 if ((error = swr_init(*resample_context)) < 0) {
 fprintf(stderr, "Could not open resample context\n");
 swr_free(resample_context);
 return error;
 }
 return 0;
 }
 
 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
 {
 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);
 }
 return 0;
 }
 
 static int write_output_file_header(AVFormatContext *output_format_context)
 {
 int error;
 if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
 fprintf(stderr, "Could not write output file header (error '%s')\n",
 av_err2str(error));
 return error;
 }
 return 0;
 }
 
 static int decode_audio_frame(AVFrame *frame,
 AVFormatContext *input_format_context,
 AVCodecContext *input_codec_context,
 int *data_present, int *finished)
 {
 AVPacket *input_packet;
 int error;
 
 error = init_packet(&input_packet);
 if (error < 0)
 return error;
 
 *data_present = 0;
 *finished = 0;

 if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
 if (error == AVERROR_EOF)
 *finished = 1;
 else {
 fprintf(stderr, "Could not read frame (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 }
 
 if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
 fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 
 error = avcodec_receive_frame(input_codec_context, frame);
 if (error == AVERROR(EAGAIN)) {
 error = 0;
 goto cleanup;
 } else if (error == AVERROR_EOF) {
 *finished = 1;
 error = 0;
 goto cleanup;
 } else if (error < 0) {
 fprintf(stderr, "Could not decode frame (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 } else {
 *data_present = 1;
 goto cleanup;
 }
 
 cleanup:
 av_packet_free(&input_packet);
 return error;
 }
 
 static int init_converted_samples(uint8_t ***converted_input_samples,
 AVCodecContext *output_codec_context,
 int frame_size)
 {
 int error;
 
 if (!(*converted_input_samples = calloc(output_codec_context->channels,
 sizeof(**converted_input_samples)))) {
 fprintf(stderr, "Could not allocate converted input sample pointers\n");
 return AVERROR(ENOMEM);
 }
 

 if ((error = av_samples_alloc(*converted_input_samples, NULL,
 output_codec_context->channels,
 frame_size,
 output_codec_context->sample_fmt, 0)) < 0) {
 fprintf(stderr,
 "Could not allocate converted input samples (error '%s')\n",
 av_err2str(error));
 av_freep(&(*converted_input_samples)[0]);
 free(*converted_input_samples);
 return error;
 }
 return 0;
 }
 
 static int convert_samples(const uint8_t **input_data, const int input_nb_samples,
 uint8_t **converted_data, const int output_nb_samples,
 SwrContext *resample_context)
 {
 int error;
 
 if ((error = swr_convert(resample_context,
 converted_data, output_nb_samples,
 input_data , input_nb_samples)) < 0) {
 fprintf(stderr, "Could not convert input samples (error '%s')\n",
 av_err2str(error));
 return error;
 }
 
 return 0;
 }
 
 static int add_samples_to_fifo(AVAudioFifo *fifo,
 uint8_t **converted_input_samples,
 const int frame_size)
 {
 int error;
 
 if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
 fprintf(stderr, "Could not reallocate FIFO\n");
 return error;
 }
 
 if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
 frame_size) < frame_size) {
 fprintf(stderr, "Could not write data to FIFO\n");
 return AVERROR_EXIT;
 }
 return 0;
 }
 
 static int read_decode_convert_and_store(AVAudioFifo *fifo,
 AVFormatContext *input_format_context,
 AVCodecContext *input_codec_context,
 AVCodecContext *output_codec_context,
 SwrContext *resampler_context,
 int *finished)
 {
 AVFrame *input_frame = NULL;
 uint8_t **converted_input_samples = NULL;
 int data_present;
 int ret = AVERROR_EXIT;
 

 if (init_input_frame(&input_frame))
 goto cleanup;

 if (decode_audio_frame(input_frame, input_format_context,
 input_codec_context, &data_present, finished))
 goto cleanup;

 if (*finished) {
 ret = 0;
 goto cleanup;
 }

 if (data_present) {
 /* Initialize the temporary storage for the converted input samples. */
 if (init_converted_samples(&converted_input_samples, output_codec_context,
 input_frame->nb_samples))
 goto cleanup;
 
 /* figure out how many samples are required for target sample_rate incl
 * any items left in the swr buffer
 */ 
 int output_nb_samples = av_rescale_rnd(
 swr_get_delay(resampler_context, input_codec_context->sample_rate) + input_frame->nb_samples,
 output_codec_context->sample_rate, 
 input_codec_context->sample_rate,
 AV_ROUND_UP);
 
 /* ignore, just to ensure we've got enough buffer alloc'd for conversion buffer */
 av_assert1(input_frame->nb_samples > output_nb_samples);
 
 /* Convert the input samples to the desired output sample format, via swr_convert().
 */
 if (convert_samples((const uint8_t**)input_frame->extended_data, input_frame->nb_samples,
 converted_input_samples, output_nb_samples,
 resampler_context))
 goto cleanup;
 
 /* Add the converted input samples to the FIFO buffer for later processing. */
 if (add_samples_to_fifo(fifo, converted_input_samples,
 output_nb_samples))
 goto cleanup;
 ret = 0;
 }
 ret = 0;
 
 cleanup:
 if (converted_input_samples) {
 av_freep(&converted_input_samples[0]);
 free(converted_input_samples);
 }
 av_frame_free(&input_frame);
 
 return ret;
 }
 
 static int init_output_frame(AVFrame **frame,
 AVCodecContext *output_codec_context,
 int frame_size)
 {
 int error;
 
 if (!(*frame = av_frame_alloc())) {
 fprintf(stderr, "Could not allocate output frame\n");
 return AVERROR_EXIT;
 }
 
 /* Set the frame's parameters, especially its size and format.
 * av_frame_get_buffer needs this to allocate memory for the
 * audio samples of the frame.
 * Default channel layouts based on the number of channels
 * are assumed for simplicity. */
 (*frame)->nb_samples = frame_size;
 (*frame)->channel_layout = output_codec_context->channel_layout;
 (*frame)->format = output_codec_context->sample_fmt;
 (*frame)->sample_rate = output_codec_context->sample_rate;
 
 /* Allocate the samples of the created frame. This call will make
 * sure that the audio frame can hold as many samples as specified. */
 if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
 fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
 av_err2str(error));
 av_frame_free(frame);
 return error;
 }
 
 return 0;
 }
 
 /* Global timestamp for the audio frames. */
 static int64_t pts = 0;
 
 /**
 * Encode one frame worth of audio to the output file.
 */
 static int encode_audio_frame(AVFrame *frame,
 AVFormatContext *output_format_context,
 AVCodecContext *output_codec_context,
 int *data_present)
 {
 AVPacket *output_packet;
 int error;
 
 error = init_packet(&output_packet);
 if (error < 0)
 return error;
 
 /* Set a timestamp based on the sample rate for the container. */
 if (frame) {
 frame->pts = pts;
 pts += frame->nb_samples;
 }
 
 *data_present = 0;
 error = avcodec_send_frame(output_codec_context, frame);
 if (error < 0 && error != AVERROR_EOF) {
 fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 

 error = avcodec_receive_packet(output_codec_context, output_packet);
 if (error == AVERROR(EAGAIN)) {
 error = 0;
 goto cleanup;
 } else if (error == AVERROR_EOF) {
 error = 0;
 goto cleanup;
 } else if (error < 0) {
 fprintf(stderr, "Could not encode frame (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 } else {
 *data_present = 1;
 }
 
 /* Write one audio frame from the temporary packet to the output file. */
 if (*data_present &&
 (error = av_write_frame(output_format_context, output_packet)) < 0) {
 fprintf(stderr, "Could not write frame (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 
 cleanup:
 av_packet_free(&output_packet);
 return error;
 }
 
 /**
 * Load one audio frame from the FIFO buffer, encode and write it to the
 * output file.
 */
 static int load_encode_and_write(AVAudioFifo *fifo,
 AVFormatContext *output_format_context,
 AVCodecContext *output_codec_context)
 {
 AVFrame *output_frame;
 /* Use the maximum number of possible samples per frame.
 * If there is less than the maximum possible frame size in the FIFO
 * buffer use this number. Otherwise, use the maximum possible frame size. */
 const int frame_size = FFMIN(av_audio_fifo_size(fifo),
 output_codec_context->frame_size);
 int data_written;
 
 if (init_output_frame(&output_frame, output_codec_context, frame_size))
 return AVERROR_EXIT;
 
 /* Read as many samples from the FIFO buffer as required to fill the frame.
 * The samples are stored in the frame temporarily. */
 if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
 fprintf(stderr, "Could not read data from FIFO\n");
 av_frame_free(&output_frame);
 return AVERROR_EXIT;
 }
 
 /* Encode one frame worth of audio samples. */
 if (encode_audio_frame(output_frame, output_format_context,
 output_codec_context, &data_written)) {
 av_frame_free(&output_frame);
 return AVERROR_EXIT;
 }
 av_frame_free(&output_frame);
 return 0;
 }
 
 /**
 * Write the trailer of the output file container.
 */
 static int write_output_file_trailer(AVFormatContext *output_format_context)
 {
 int error;
 if ((error = av_write_trailer(output_format_context)) < 0) {
 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
 av_err2str(error));
 return error;
 }
 return 0;
 }
 
 int main(int argc, char **argv)
 {
 AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
 AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
 SwrContext *resample_context = NULL;
 AVAudioFifo *fifo = NULL;
 int ret = AVERROR_EXIT;
 
 if (argc != 3) {
 fprintf(stderr, "Usage: %s <input file="file" /> <output file="file">\n", argv[0]);
 exit(1);
 }
 

 if (open_input_file(argv[1], &input_format_context,
 &input_codec_context))
 goto cleanup;

 if (open_output_file(argv[2], input_codec_context,
 &output_format_context, &output_codec_context))
 goto cleanup;

 if (init_resampler(input_codec_context, output_codec_context,
 &resample_context))
 goto cleanup;

 if (init_fifo(&fifo, output_codec_context))
 goto cleanup;

 if (write_output_file_header(output_format_context))
 goto cleanup;
 
 while (1) {
 /* Use the encoder's desired frame size for processing. */
 const int output_frame_size = output_codec_context->frame_size;
 int finished = 0;
 
 while (av_audio_fifo_size(fifo) < output_frame_size) {
 /* Decode one frame worth of audio samples, convert it to the
 * output sample format and put it into the FIFO buffer. */
 if (read_decode_convert_and_store(fifo, input_format_context,
 input_codec_context,
 output_codec_context,
 resample_context, &finished))
 goto cleanup;
 
 if (finished)
 break;
 }
 
 while (av_audio_fifo_size(fifo) >= output_frame_size ||
 (finished && av_audio_fifo_size(fifo) > 0))
 if (load_encode_and_write(fifo, output_format_context,
 output_codec_context))
 goto cleanup;
 
 if (finished) {
 int data_written;
 do {
 if (encode_audio_frame(NULL, output_format_context,
 output_codec_context, &data_written))
 goto cleanup;
 } while (data_written);
 break;
 }
 }
 
 if (write_output_file_trailer(output_format_context))
 goto cleanup;
 ret = 0;
 
 cleanup:
 if (fifo)
 av_audio_fifo_free(fifo);
 swr_free(&resample_context);
 if (output_codec_context)
 avcodec_free_context(&output_codec_context);
 if (output_format_context) {
 avio_closep(&output_format_context->pb);
 avformat_free_context(output_format_context);
 }
 if (input_codec_context)
 avcodec_free_context(&input_codec_context);
 if (input_format_context)
 avformat_close_input(&input_format_context);
 
 return ret;
 }
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