k
/
FFmpegAndroid
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Feature: adjust samplerate

Browse Source
pull/221/head
xufuji456 2 years ago
parent c1ae76fa6d
commit e2faaabc4a
2 changed files with 48 additions and 30 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Show Stats Download Patch File Download Diff File
  1. 56
      app/src/main/cpp/audio_resample.cpp
  2. 22
      app/src/main/java/com/frank/ffmpeg/activity/MainActivity.kt

56
app/src/main/cpp/audio_resample.cpp
Unescape View File

@ -177,7 +177,7 @@ static int open_output_file(const char *filename,
* The input file's sample rate is used to avoid a sample rate conversion. */
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = sample_rate; //input_codec_context->sample_rate;
avctx->sample_rate = sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
@ -185,7 +185,7 @@ static int open_output_file(const char *filename,
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.den = sample_rate;
stream->time_base.num = 1;
/* Some container formats (like MP4) require global headers to be present.
@ -269,12 +269,6 @@ static int init_resampler(AVCodecContext *input_codec_context,
ALOGE("Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/*
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/* Open the re-sampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
@ -291,7 +285,6 @@ static int init_resampler(AVCodecContext *input_codec_context,
*/
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
/* Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
ALOGE("Could not allocate FIFO\n");
@ -308,8 +301,7 @@ static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
if ((error = avformat_write_header(output_format_context, nullptr)) < 0) {
ALOGE("Could not write output file header (error '%s')\n",
av_err2str(error));
ALOGE("Could not write output file header (error:%s)\n", av_err2str(error));
return error;
}
return 0;
@ -324,9 +316,8 @@ static int decode_audio_frame(AVFrame *frame,
AVCodecContext *input_codec_context,
int *data_present, int *finished)
{
/* Packet used for temporary storage. */
AVPacket input_packet;
int error;
AVPacket input_packet;
init_packet(&input_packet);
/* Read one audio frame from the input file into a temporary packet. */
@ -335,8 +326,7 @@ static int decode_audio_frame(AVFrame *frame,
if (error == AVERROR_EOF)
*finished = 1;
else {
ALOGE("Could not read frame (error:%s)\n",
av_err2str(error));
ALOGE("Could not read frame (error:%s)\n", av_err2str(error));
return error;
}
}
@ -419,11 +409,8 @@ static int convert_samples(const uint8_t **input_data,
SwrContext *resample_context)
{
int error;
/* Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
if ((error = swr_convert(resample_context, converted_data, frame_size,
input_data, frame_size)) < 0) {
ALOGE("Could not convert input samples (error:%s)\n", av_err2str(error));
return error;
}
@ -563,14 +550,14 @@ static int init_output_frame(AVFrame **frame,
* Encode one frame worth of audio to the output file.
*
*/
static int encode_audio_frame(AVFrame *frame,
static int encode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
/* Packet used for temporary storage. */
AVPacket output_packet;
int error;
int stream_index = 0;
AVPacket output_packet;
init_packet(&output_packet);
/* Set a timestamp based on the sample rate for the container. */
@ -605,11 +592,23 @@ static int encode_audio_frame(AVFrame *frame,
} else if (error < 0) {
ALOGE("Could not encode frame (error:%s)\n", av_err2str(error));
goto cleanup;
/* Default case: Return encoded data. */
} else {
*data_present = 1;
}
// rescale pts from input to output
// av_packet_rescale_ts(&output_packet,
// input_format_context->streams[stream_index]->time_base,
// output_format_context->streams[stream_index]->time_base);
output_packet.pts = av_rescale_q_rnd(output_packet.pts,
input_format_context->streams[stream_index]->time_base,
output_format_context->streams[stream_index]->time_base,
static_cast<AVRounding>(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
output_packet.dts = output_packet.pts;
output_packet.duration = av_rescale_q(output_packet.duration,
input_format_context->streams[stream_index]->time_base,
output_format_context->streams[stream_index]->time_base);
/* 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) {
@ -627,7 +626,7 @@ cleanup:
* output file.
*
*/
static int load_encode_and_write(AVAudioFifo *fifo,
static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *input_format_context,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
@ -653,7 +652,7 @@ static int load_encode_and_write(AVAudioFifo *fifo,
}
/* Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
if (encode_audio_frame(output_frame, input_format_context, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
@ -737,8 +736,7 @@ int resampling(const char *src_file, const char *dst_file, int sampleRate)
(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))
if (load_encode_and_write(fifo, input_format_context, output_format_context, output_codec_context))
goto cleanup;
/* If we are at the end of the input file and have encoded
@ -748,7 +746,7 @@ int resampling(const char *src_file, const char *dst_file, int sampleRate)
/* Flush the encoder as it may have delayed frames. */
do {
data_written = 0;
if (encode_audio_frame(nullptr, output_format_context,
if (encode_audio_frame(nullptr, input_format_context, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);

22
app/src/main/java/com/frank/ffmpeg/activity/MainActivity.kt
Unescape View File

@ -2,11 +2,13 @@ package com.frank.ffmpeg.activity
import android.content.Intent
import android.os.Bundle
import android.util.Log
import android.view.View
import androidx.recyclerview.widget.RecyclerView
import androidx.recyclerview.widget.StaggeredGridLayoutManager
import com.frank.ffmpeg.R
import com.frank.ffmpeg.VideoPlayer
import com.frank.ffmpeg.adapter.WaterfallAdapter
import com.frank.ffmpeg.listener.OnItemClickListener
@ -70,7 +72,25 @@ class MainActivity : BaseActivity() {
7 //probe media format
-> intent.setClass(this@MainActivity, ProbeFormatActivity::class.java)
8 //audio effect
-> intent.setClass(this@MainActivity, AudioEffectActivity::class.java)
// -> intent.setClass(this@MainActivity, AudioEffectActivity::class.java)
-> {
val videoPlayer = VideoPlayer()
// val input = "sdcard/what.mp4"
// val output = "sdcard/haha.avi"
// Thread {
// val ret = videoPlayer.executeTranscode(input, output)
// Log.e("Main", "transcode result=$ret")
// }.start()
val input = "sdcard/okok.aac"
val output = "sdcard/haha48000.aac"
Thread {
val ret = videoPlayer.audioResample(input, output, 48000)
Log.e("Main", "transcode result=$ret")
}.start()
return
}
else -> {
}
}

Write Preview
Loading…
Cancel
Save