Feature: decode and resample

pull/221/head
xufuji456 2 years ago
parent 0d6f18c021
commit 891bd4afb2
  1. 51
      app/src/main/cpp/audio_resample.cpp

@ -48,18 +48,21 @@ static int get_format_from_sample_fmt(const char **fmt, enum AVSampleFormat samp
}
int init_audio_codec(AVFormatContext *fmt_ctx, AVCodecContext **avcodec_ctx, bool is_encoder) {
AVCodec *codec = is_encoder ? avcodec_find_encoder(fmt_ctx->audio_codec_id)
: avcodec_find_decoder(fmt_ctx->audio_codec_id);
AVCodecContext *codec_ctx = nullptr;
for (int i = 0; i < fmt_ctx->nb_streams; ++i) {
if (fmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
codec_ctx = fmt_ctx->streams[i]->codec;
}
}
AVCodec *codec = is_encoder ? avcodec_find_encoder(codec_ctx->codec_id)
: avcodec_find_decoder(codec_ctx->codec_id);
if (!codec) {
ALOGE("can't found codec id=%d\n", fmt_ctx->audio_codec_id);
ALOGE("can't found codec id=%d\n", codec_ctx->codec_id);
return -1;
}
AVCodecContext *codec_ctx = avcodec_alloc_context3(codec);
if (!codec_ctx) {
ALOGE("avcodec_alloc_context3 fail!\n");
return -2;
}
int ret = avcodec_open2(codec_ctx, codec, nullptr);
if (ret < 0)
ALOGE("avcodec_open2 fail:%d", ret);
*avcodec_ctx = codec_ctx;
return ret;
}
@ -95,10 +98,9 @@ int init_audio_muxer(AVFormatContext **ofmt_ctx, const char* filename) {
int resampling(const char *src_filename, const char *dst_filename, int dst_rate)
{
int src_rate = 0;
int src_nb_samples = 0;
int src_rate;
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_S16;
enum AVSampleFormat src_sample_fmt;
int dst_bufsize;
int dst_linesize;
@ -129,6 +131,10 @@ int resampling(const char *src_filename, const char *dst_filename, int dst_rate)
avformat_find_stream_info(iformat_ctx, nullptr);
frame = av_frame_alloc();
opacket = av_packet_alloc();
init_audio_decoder(iformat_ctx, &icodec_ctx);
src_rate = icodec_ctx->sample_rate;
src_ch_layout = (int64_t) icodec_ctx->channel_layout;
src_sample_fmt = icodec_ctx->sample_fmt;
/* create resample context */
swr_ctx = swr_alloc();
@ -153,12 +159,6 @@ int resampling(const char *src_filename, const char *dst_filename, int dst_rate)
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
(int) av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a raw-audio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
@ -172,13 +172,20 @@ int resampling(const char *src_filename, const char *dst_filename, int dst_rate)
if (ret < 0) {
goto end;
}
init_audio_decoder(iformat_ctx, &icodec_ctx);
init_audio_encoder(oformat_ctx, &ocodec_ctx);
while (av_read_frame(iformat_ctx, &packet) >= 0) {
ret = avcodec_decode_audio4(icodec_ctx, frame, &got_frame_ptr, &packet);
if (ret < 0) {
ALOGE("decode audio error:%d\n", ret);
continue;
}
ALOGE("decode succ, pts=%ld\n", frame->pts);
/* compute destination number of samples */
dst_nb_samples = (int) av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
frame->nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
@ -188,12 +195,6 @@ int resampling(const char *src_filename, const char *dst_filename, int dst_rate)
max_dst_nb_samples = dst_nb_samples;
}
ret = avcodec_decode_audio4(icodec_ctx, frame, &got_frame_ptr, &packet);
if (ret < 0) {
ALOGE("decode audio error:%d\n", ret);
continue;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {

Loading…
Cancel
Save