soundtouch项目官网 https://www.surina.net/soundtouch/
SoundTouch git repository: https://codeberg.org/soundtouch/soundtouch.git
工具:git visiual studio 2019
// 下载源码 当前版本 2.3.1
git clone https://codeberg.org/soundtouch/soundtouch.git
修改soundtouch\include\STTypes.h下的宏定义
注意:默认是使用32bit float samples,需要更改成使用16bit integer samples
#if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_FLOAT_SAMPLES)
/// Choose either 32bit floating point or 16bit integer sampletype
/// by choosing one of the following defines, unless this selection
/// has already been done in some other file.
/// Notes:
/// - In Windows environment, choose the sample format with the
/// following defines.
/// - In GNU environment, the floating point samples are used by
/// default, but integer samples can be chosen by giving the
/// following switch to the configure script:
/// ./configure --enable-integer-samples
/// However, if you still prefer to select the sample format here
/// also in GNU environment, then please #undef the INTEGER_SAMPLE
/// and FLOAT_SAMPLE defines first as in comments above.
#define SOUNDTOUCH_INTEGER_SAMPLES 1 //< 16bit integer samples
//#define SOUNDTOUCH_FLOAT_SAMPLES 1 //< 32bit float samples
#endif
打开vs命令行工具 Developer Command Prompt for VS 2019
切换到源码路径执行命令 make-win.bat
生成的静态库动态库在 lib文件夹下面
vs解决方案在: soundtouch\source\SoundTouch\SoundTouch.sln
#ifndef IJKSOUNDTOUCHWRAP_H
#define IJKSOUNDTOUCHWRAP_H
#include <stdint.h>
void* soundtouch_create();
int soundtouch_translate(void *handle, short* data, float speed, float pitch,
int len, int bytes_per_sample, int n_channel, int n_sampleRate);
void soundtouch_destroy(void *handle);
#endif /* IJKSOUNDTOUCHWRAP_H */
#include "SoundTouch.h"
using namespace std;
using namespace soundtouch;
void* soundtouch_create()
{
SoundTouch *handle_ptr = new SoundTouch();
const char *version = handle_ptr->getVersionString();
return handle_ptr;
}
int soundtouch_translate(void *handle, short* data, float speed, float pitch,
int len, int bytes_per_sample, int n_channel, int n_sampleRate)
{
SoundTouch *handle_ptr = (SoundTouch*)handle;
int put_n_sample = len / n_channel;
int nb = 0;
int pcm_data_size = 0;
if (handle_ptr == NULL)
return 0;
handle_ptr->setPitch(pitch);
handle_ptr->setRate(speed);
handle_ptr->setSampleRate(n_sampleRate);
handle_ptr->setChannels(n_channel);
handle_ptr->putSamples((SAMPLETYPE*)data, put_n_sample);
do {
nb = handle_ptr->receiveSamples((SAMPLETYPE*)data, n_sampleRate / n_channel);
pcm_data_size += nb * n_channel * bytes_per_sample;
} while (nb != 0);
return pcm_data_size;
}
void soundtouch_destroy(void *handle)
{
SoundTouch *handle_ptr = (SoundTouch*)handle;
if (handle_ptr == NULL)
return;
handle_ptr->clear();
delete handle_ptr;
handle_ptr = NULL;
}
#include "soundtouch_wrap.h"
typedef struct VideoState {
void* soundTouchHandle;
short *audio_new_buf; /* soundtouch buf */
unsigned int audio_new_buf_size;
double play_rate; /* 播放速度默认1.0 */
}
static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
{
VideoState *is;
is = av_mallocz(sizeof(VideoState));
if (!is)
return NULL;
is->soundTouchHandle = soundtouch_create();
is->audio_new_buf = NULL;
is->audio_new_buf_size = 0;
id->play_rate = 1.0f;
...
}
if (is->soundTouchHandle)
{
soundtouch_destroy(is->soundTouchHandle);
is->soundTouchHandle = nullptr;
}
if (is->audio_new_buf)
{
av_freep(&is->audio_new_buf);
is->audio_new_buf = NULL;
}
static int audio_decode_frame(VideoState *is)
{
int data_size, resampled_data_size;
int64_t dec_channel_layout;
av_unused double audio_clock0;
int wanted_nb_samples;
Frame *af;
int translate_time = 1;
if (is->paused)
return -1;
reload:
do {
#if defined(_WIN32)
while (frame_queue_nb_remaining(&is->sampq) == 0) {
if ((av_gettime_relative() - is->audio_callback_time) > 1000000LL * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec / 2)
return -1;
av_usleep (1000);
}
#endif
if (!(af = frame_queue_peek_readable(&is->sampq)))
return -1;
frame_queue_next(&is->sampq);
} while (af->serial != is->audioq.serial);
data_size = av_samples_get_buffer_size(NULL, af->frame->channels,
af->frame->nb_samples,
(AVSampleFormat)af->frame->format, 1);
dec_channel_layout =
(af->frame->channel_layout && af->frame->channels == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
af->frame->channel_layout : av_get_default_channel_layout(af->frame->channels);
wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
if (af->frame->format != is->audio_src.fmt ||
dec_channel_layout != is->audio_src.channel_layout ||
af->frame->sample_rate != is->audio_src.freq ||
(wanted_nb_samples != af->frame->nb_samples && !is->swr_ctx)) {
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc_set_opts(NULL,
is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
dec_channel_layout, (AVSampleFormat)af->frame->format, af->frame->sample_rate,
0, NULL);
if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
av_log(NULL, AV_LOG_ERROR,
"Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
af->frame->sample_rate, av_get_sample_fmt_name((AVSampleFormat)af->frame->format), af->frame->channels,
is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
swr_free(&is->swr_ctx);
return -1;
}
is->audio_src.channel_layout = dec_channel_layout;
is->audio_src.channels = af->frame->channels;
is->audio_src.freq = af->frame->sample_rate;
is->audio_src.fmt = (AVSampleFormat)af->frame->format;
}
if (is->swr_ctx) {
const uint8_t **in = (const uint8_t **)af->frame->extended_data;
uint8_t **out = &is->audio_buf1;
int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
int out_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
int len2;
if (out_size < 0) {
av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
return -1;
}
if (wanted_nb_samples != af->frame->nb_samples) {
if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {
av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
return -1;
}
}
av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
if (!is->audio_buf1)
return AVERROR(ENOMEM);
len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);
if (len2 < 0) {
av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");
return -1;
}
if (len2 == out_count) {
av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
if (swr_init(is->swr_ctx) < 0)
swr_free(&is->swr_ctx);
}
is->audio_buf = is->audio_buf1;
resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
//=====================倍速处理 begin==========================
int bytes_per_sample = av_get_bytes_per_sample(is->audio_tgt.fmt);
if (is->soundTouchHandle && is->play_rate != 1.0f && !is->abort_request)
{
av_fast_malloc(&is->audio_new_buf, &is->audio_new_buf_size, out_size * translate_time);
if (!is->audio_new_buf) {
// Allocation failed; buf already freed
return AVERROR(ENOMEM);
}
for (int i = 0; i < (resampled_data_size / 2); i++)
{
is->audio_new_buf[i] = (is->audio_buf1[i * 2] | (is->audio_buf1[i * 2 + 1] << 8));
}
int ret_len = soundtouch_translate(is->soundTouchHandle, is->audio_new_buf, (float)(is->play_rate), (float)(1.0f / is->play_rate),
resampled_data_size / 2, bytes_per_sample, is->audio_tgt.channels, af->frame->sample_rate);
if (ret_len > 0) {
is->audio_buf = (uint8_t*)is->audio_new_buf;
resampled_data_size = ret_len;
}
else {
translate_time++;
goto reload;
}
}
//=====================倍速处理 end======================
} else {
is->audio_buf = af->frame->data[0];
resampled_data_size = data_size;
}
audio_clock0 = is->audio_clock;
/* update the audio clock with the pts */
if (!isnan(af->pts))
is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;
else
is->audio_clock = NAN;
is->audio_clock_serial = af->serial;
#ifdef DEBUG
{
static double last_clock;
printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n",
is->audio_clock - last_clock,
is->audio_clock, audio_clock0);
last_clock = is->audio_clock;
}
#endif
return resampled_data_size;
}
static void event_loop(VideoState *cur_stream)
{
switch (event.type) {
case SDLK_1:
cur_stream->play_rate = 1.0;
break;
case SDLK_2:
cur_stream->play_rate = 1.25;
break;
case SDLK_3:
cur_stream->play_rate = 1.5;
break;
case SDLK_4:
cur_stream->play_rate = 2.0;
break;
case SDLK_5:
cur_stream->play_rate = 0.75;
break;
case SDLK_6:
cur_stream->play_rate = 0.5;
break;
}
}
到此位置,ffplay增加了倍速播放的功能,通过键盘123456按键切换播放速度。
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