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Feature: support different sample_rate

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pull/221/head
xufuji456 2 years ago
parent e2faaabc4a
commit f28b9b39cf
2 changed files with 26 additions and 58 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
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  1. 64
      app/src/main/cpp/audio_resample.cpp
  2. 20
      app/src/main/java/com/frank/ffmpeg/activity/MainActivity.kt

64
app/src/main/cpp/audio_resample.cpp
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@ -114,7 +114,6 @@ static int open_input_file(const char *filename,
*/
static int open_output_file(const char *filename,
int sample_rate,
AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
@ -404,13 +403,13 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
* specified by frame_size.
*
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
static int convert_samples(const uint8_t **input_data, const int input_size,
uint8_t **converted_data, const int output_size,
SwrContext *resample_context)
{
int error;
if ((error = swr_convert(resample_context, converted_data, frame_size,
input_data, frame_size)) < 0) {
if ((error = swr_convert(resample_context, converted_data, output_size,
input_data, input_size)) < 0) {
ALOGE("Could not convert input samples (error:%s)\n", av_err2str(error));
return error;
}
@ -459,7 +458,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
/* Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = nullptr;
/* Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = nullptr;
uint8_t **converted_dst_samples = nullptr;
int data_present = 0;
int ret = AVERROR_EXIT;
@ -479,29 +478,32 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
}
/* If there is decoded data, convert and store it. */
if (data_present) {
int dst_nb_samples = (int) av_rescale_rnd(input_frame->nb_samples, output_codec_context->sample_rate,
input_codec_context->sample_rate, AV_ROUND_UP);
/* Initialize the temporary storage for the converted input samples. */
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
if (init_converted_samples(&converted_dst_samples, output_codec_context,
dst_nb_samples))
goto cleanup;
/* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples. */
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
if (convert_samples((const uint8_t**)input_frame->extended_data,
input_frame->nb_samples, converted_dst_samples,
dst_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,
input_frame->nb_samples))
if (add_samples_to_fifo(fifo, converted_dst_samples,
dst_nb_samples))
goto cleanup;
ret = 0;
}
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
if (converted_dst_samples) {
av_freep(&converted_dst_samples[0]);
free(converted_dst_samples);
}
av_frame_free(&input_frame);
@ -550,13 +552,12 @@ static int init_output_frame(AVFrame **frame,
* Encode one frame worth of audio to the output file.
*
*/
static int encode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context,
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
int error;
int stream_index = 0;
AVPacket output_packet;
init_packet(&output_packet);
@ -596,19 +597,6 @@ static int encode_audio_frame(AVFrame *frame, AVFormatContext *input_format_cont
*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) {
@ -626,7 +614,7 @@ cleanup:
* output file.
*
*/
static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *input_format_context,
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
@ -652,7 +640,7 @@ static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *input_forma
}
/* Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, input_format_context, output_format_context,
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
@ -684,12 +672,10 @@ int resampling(const char *src_file, const char *dst_file, int sampleRate)
int ret = AVERROR_EXIT;
/* Open the input file for reading. */
if (open_input_file(src_file, &input_format_context,
&input_codec_context))
if (open_input_file(src_file, &input_format_context, &input_codec_context))
goto cleanup;
/* Open the output file for writing. */
if (open_output_file(dst_file, sampleRate, input_codec_context,
&output_format_context, &output_codec_context))
if (open_output_file(dst_file, sampleRate, &output_format_context, &output_codec_context))
goto cleanup;
/* Initialize the re-sampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
@ -736,7 +722,7 @@ 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, input_format_context, output_format_context, output_codec_context))
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
@ -746,7 +732,7 @@ 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, input_format_context, output_format_context,
if (encode_audio_frame(nullptr, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);

20
app/src/main/java/com/frank/ffmpeg/activity/MainActivity.kt
Unescape View File

@ -72,25 +72,7 @@ class MainActivity : BaseActivity() {
7 //probe media format
-> intent.setClass(this@MainActivity, ProbeFormatActivity::class.java)
8 //audio effect
// -> 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
}
-> intent.setClass(this@MainActivity, AudioEffectActivity::class.java)
else -> {
}
}

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