Feature: move static method together

pull/221/head
xufuji456 2 years ago
parent 71d7dbdb5d
commit 2eee129552
  1. 101
      app/src/main/cpp/ff_audio_resample.cpp
  2. 2
      app/src/main/cpp/ff_audio_resample.h

@ -15,6 +15,54 @@ FFAudioResample::~FFAudioResample() {
delete resample; delete resample;
} }
static int initOutputFrame(AudioResample **pResample) {
AudioResample *ar = *pResample;
AVFrame *frame = av_frame_alloc();
frame->format = ar->outCodecCtx->sample_fmt;
frame->nb_samples = ar->outCodecCtx->frame_size;
frame->sample_rate = ar->outCodecCtx->sample_rate;
frame->channel_layout = ar->outCodecCtx->channel_layout;
int ret = av_frame_get_buffer(frame, 0);
ar->outFrame = frame;
*pResample = ar;
return ret;
}
static int initResample(AudioResample **pResample) {
AudioResample *ar = *pResample;
SwrContext *context = swr_alloc_set_opts(nullptr,
av_get_default_channel_layout(ar->outCodecCtx->channels),
ar->outCodecCtx->sample_fmt,
ar->outCodecCtx->sample_rate,
av_get_default_channel_layout(ar->inCodecCtx->channels),
ar->inCodecCtx->sample_fmt,
ar->inCodecCtx->sample_rate,
0, nullptr);
int ret = swr_init(context);
ar->resampleCtx = context;
*pResample = ar;
return ret;
}
static int initConvertedSamples(AudioResample **pResample, uint8_t ***converted_input_samples, int frame_size) {
int ret;
AudioResample *ar = *pResample;
*converted_input_samples = (uint8_t **) calloc(ar->outCodecCtx->channels, sizeof(**converted_input_samples));
if ((ret = av_samples_alloc(*converted_input_samples, nullptr,
ar->outCodecCtx->channels,
frame_size,
ar->outCodecCtx->sample_fmt, 0)) < 0) {
ALOGE("av_samples_alloc error:%s", av_err2str(ret));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return ret;
}
return 0;
}
int FFAudioResample::openInputFile(const char *filename) { int FFAudioResample::openInputFile(const char *filename) {
int ret; int ret;
const AVCodec *input_codec; const AVCodec *input_codec;
@ -105,22 +153,6 @@ int FFAudioResample::openOutputFile(const char *filename, int sample_rate) {
return 0; return 0;
} }
static int initResample(AudioResample **pResample) {
AudioResample *ar = *pResample;
SwrContext *context = swr_alloc_set_opts(nullptr,
av_get_default_channel_layout(ar->outCodecCtx->channels),
ar->outCodecCtx->sample_fmt,
ar->outCodecCtx->sample_rate,
av_get_default_channel_layout(ar->inCodecCtx->channels),
ar->inCodecCtx->sample_fmt,
ar->inCodecCtx->sample_rate,
0, nullptr);
int ret = swr_init(context);
ar->resampleCtx = context;
*pResample = ar;
return ret;
}
int FFAudioResample::decodeAudioFrame(AVFrame *frame, int *data_present, int *finished) { int FFAudioResample::decodeAudioFrame(AVFrame *frame, int *data_present, int *finished) {
int ret; int ret;
@ -165,26 +197,6 @@ cleanup:
return ret; return ret;
} }
int FFAudioResample::initConvertedSamples(uint8_t ***converted_input_samples, int frame_size) {
int ret;
if (!(*converted_input_samples = (uint8_t **) calloc(resample->outCodecCtx->channels,
sizeof(**converted_input_samples)))) {
ALOGE("Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
if ((ret = av_samples_alloc(*converted_input_samples, nullptr,
resample->outCodecCtx->channels,
frame_size,
resample->outCodecCtx->sample_fmt, 0)) < 0) {
ALOGE("Could not allocate converted input samples (error:%s)\n", av_err2str(ret));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return ret;
}
return 0;
}
/** /**
* Read one audio frame from the input file, decode, convert and store * Read one audio frame from the input file, decode, convert and store
* it in the FIFO buffer. * it in the FIFO buffer.
@ -207,7 +219,7 @@ int FFAudioResample::decodeAndConvert(int *finished) {
int dst_nb_samples = (int) av_rescale_rnd(resample->inFrame->nb_samples, resample->outCodecCtx->sample_rate, int dst_nb_samples = (int) av_rescale_rnd(resample->inFrame->nb_samples, resample->outCodecCtx->sample_rate,
resample->inCodecCtx->sample_rate, AV_ROUND_UP); resample->inCodecCtx->sample_rate, AV_ROUND_UP);
if (initConvertedSamples(&converted_dst_samples, dst_nb_samples)) if (initConvertedSamples(&resample, &converted_dst_samples, dst_nb_samples))
goto cleanup; goto cleanup;
ret = swr_convert(resample->resampleCtx, converted_dst_samples, dst_nb_samples, ret = swr_convert(resample->resampleCtx, converted_dst_samples, dst_nb_samples,
@ -231,21 +243,6 @@ cleanup:
return ret; return ret;
} }
static int initOutputFrame(AudioResample **pResample) {
AudioResample *ar = *pResample;
AVFrame *frame = av_frame_alloc();
frame->format = ar->outCodecCtx->sample_fmt;
frame->nb_samples = ar->outCodecCtx->frame_size;
frame->sample_rate = ar->outCodecCtx->sample_rate;
frame->channel_layout = ar->outCodecCtx->channel_layout;
int ret = av_frame_get_buffer(frame, 0);
ar->outFrame = frame;
*pResample = ar;
return ret;
}
int FFAudioResample::encodeAudioFrame(AVFrame *frame, int *data_present) { int FFAudioResample::encodeAudioFrame(AVFrame *frame, int *data_present) {
int ret; int ret;

@ -54,8 +54,6 @@ private:
int decodeAudioFrame(AVFrame *frame, int *data_present, int *finished); int decodeAudioFrame(AVFrame *frame, int *data_present, int *finished);
int initConvertedSamples(uint8_t ***converted_input_samples, int frame_size);
int decodeAndConvert(int *finished); int decodeAndConvert(int *finished);
int encodeAudioFrame(AVFrame *frame, int *data_present); int encodeAudioFrame(AVFrame *frame, int *data_present);

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