/* Wrapper for receive-side integer-to-float array resampling */
int rx_resmpl_int_flt(float *smpls_out, short *smpls_in, int num_smpls)
{
int num_resmpld, num_chunks;
signalVector *convert_vec, *resamp_vec, *trunc_vec;
if (!rx_lpf || !rx_vec || !rx_hist)
init_resampler(&rx_lpf, &rx_vec, &rx_hist, false);
/* Convert and add samples to the receive buffer */
convert_vec = short_to_sigvec(smpls_in, num_smpls);
rx_vec = concat(rx_vec, convert_vec);
num_chunks = rx_vec->size() / OUTCHUNK;
if (num_chunks < 1)
return 0;
/* Resample */
resamp_vec = resmpl_sigvec(rx_hist, &rx_vec, rx_lpf,
INRATE, OUTRATE, OUTCHUNK);
/* Truncate */
trunc_vec = segment(resamp_vec, INHISTORY,
resamp_vec->size() - INHISTORY);
/* Convert */
num_resmpld = sigvec_to_float(trunc_vec, smpls_out);
return num_resmpld;
}
/* Wrapper for transmit-side float-to-int array resampling */
int tx_resmpl_flt_int(short *smpls_out, float *smpls_in, int num_smpls)
{
int num_resmpl, num_chunks;
signalVector *convert_vec, *resamp_vec;
if (!tx_lpf || !tx_vec || !tx_hist)
init_resampler(&tx_lpf, &tx_vec, &tx_hist, true);
/* Convert and add samples to the transmit buffer */
convert_vec = float_to_sigvec(smpls_in, num_smpls);
tx_vec = concat(tx_vec, convert_vec);
num_chunks = tx_vec->size() / INCHUNK;
if (num_chunks < 1)
return 0;
/* Resample and convert to an integer array */
resamp_vec = resmpl_sigvec(tx_hist, &tx_vec, tx_lpf,
OUTRATE, INRATE, INCHUNK);
num_resmpl = sigvec_to_short(resamp_vec, smpls_out);
return num_resmpl;
}
/** Convert an audio file to an AAC file in an MP4 container. */
int compress(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> <output file>\n", argv[0]);
exit(1);
}
const char *input_filename = argv[1];
const char *output_filename = argv[2];
/** Register all codecs and formats so that they can be used. */
av_register_all();
/** Open the input file for reading. */
if (open_input_file(input_filename, &input_format_context, &input_codec_context))
goto cleanup;
/** Open the output file for writing. */
if (open_output_file(output_filename, input_codec_context, &output_format_context, &output_codec_context))
goto cleanup;
/** Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context, &resample_context))
goto cleanup;
/** Initialize the FIFO buffer to store audio samples to be encoded. */
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))
goto cleanup;
/**
* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither.
*/
while (1) {
/** Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/**
* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples.
*/
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 we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file.
*/
if (finished)
break;
}
/**
* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder.
*/
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/**
* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file.
*/
if (load_encode_and_write(fifo, output_format_context, output_codec_context))
goto cleanup;
/**
* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish.
*/
if (finished) {
int data_written;
/** Flush the encoder as it may have delayed frames. */
do {
if (encode_audio_frame(NULL, output_format_context, output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
/** Write the trailer of the output file container. */
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_close(output_codec_context);
if (output_format_context) {
avio_closep(&output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_close(input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}
/**
*
* @param orig
* @param dest
* @return
*/
int Transcode::transcode(string orig, string dest, TID3Tags *tags)
{
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;
this->pts = 0;
if (orig.empty() || dest.empty()) {
fprintf(stderr, "Files to process not especified\n");
return ret;
}
/* Register all codecs and formats so that they can be used. */
av_register_all();
/* Open the input file for reading. */
if (open_input_file(orig.c_str(), &input_format_context,
&input_codec_context))
goto cleanup;
/* Open the output file for writing. */
if (open_output_file(dest.c_str(), input_codec_context,
&output_format_context, &output_codec_context))
goto cleanup;
/**Copy the metadata if exist*/
copy_metadata(input_format_context, output_format_context, tags);
/* Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/* Initialize the FIFO buffer to store audio samples to be encoded. */
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))
goto cleanup;
/* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither. */
while (1) {
/* Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples. */
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 we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file. */
if (finished)
break;
}
/* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder. */
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file. */
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish. */
if (finished) {
int data_written;
/* Flush the encoder as it may have delayed frames. */
do {
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
/* Write the trailer of the output file container. */
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;
}
int AudioDecoder::initDecoder(AVCodecContext* context, AVCodec* dec_codec)
{
ELOG_DEBUG("initDecoder started");
codec_ = dec_codec;
input_codec_context = context; // ok? ok
if (avcodec_open2(input_codec_context, codec_, NULL) < 0) {
ELOG_DEBUG("AudioDecoder initDecoder Error open2 audio decoder");
return 0;
}
ELOG_DEBUG("input sample_fmts[0] is %s", av_get_sample_fmt_name(codec_->sample_fmts[0]));
ELOG_DEBUG("input sample_fmt is %s", av_get_sample_fmt_name(input_codec_context->sample_fmt));
ELOG_DEBUG("input frame size is %d, bitrate=%d", input_codec_context->frame_size, input_codec_context->bit_rate);
// Init output encoder as well.
AVCodec *output_codec = NULL;
int error;
if (!(output_codec = avcodec_find_encoder(OUTPUT_CODEC_ID))) {
ELOG_DEBUG( "Could not find the encoder.");
return 0;
}
output_codec_context = avcodec_alloc_context3(output_codec);
if (!output_codec_context)
{
ELOG_DEBUG( "Could not allocate an encoding context");
return 0;
}
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
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 = output_codec->sample_fmts[0]; //u8
output_codec_context->bit_rate = 510000;
/** Allow the use of the experimental feature */
output_codec_context->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/** Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(output_codec_context, output_codec, NULL)) < 0) {
ELOG_DEBUG("Could not open output codec %s", get_error_text(error));
return 0;
}
ELOG_DEBUG("output sample_fmts[0] is %s", av_get_sample_fmt_name(output_codec->sample_fmts[0]));
ELOG_DEBUG("output sample_fmt is %s", av_get_sample_fmt_name(output_codec_context->sample_fmt));
ELOG_DEBUG("output frame size is %d", output_codec_context->frame_size);
output_codec_context->frame_size = 960; //20ms. actually no need, as it already is
/** Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context))
{
ELOG_DEBUG(" init resampleer failed !!");
return 0;
}
init_fifo(&fifo);
ELOG_DEBUG("initDecoder end");
return 1;
}
