mirror of
https://github.com/ggerganov/whisper.cpp.git
synced 2024-12-20 21:23:06 +00:00
300c07b94d
remove call to 'av_register_all()' which does not exist in ffmpeg v5 anymore.
349 lines
9.0 KiB
C++
349 lines
9.0 KiB
C++
/* SPDX-License-Identifier: GPL-2.0 */
|
|
|
|
/*
|
|
* transcode.c - convert audio file to WAVE
|
|
*
|
|
* Copyright (C) 2019 Andrew Clayton <andrew@digital-domain.net>
|
|
* Copyright (C) 2024 William Tambellini <william.tambellini@gmail.com>
|
|
*/
|
|
|
|
// Just for conveninent C++ API
|
|
#include <vector>
|
|
#include <string>
|
|
|
|
// C
|
|
#include <stdio.h>
|
|
#include <stdlib.h>
|
|
#include <string.h>
|
|
#include <stdbool.h>
|
|
#include <stdint.h>
|
|
#include <sys/types.h>
|
|
#include <sys/stat.h>
|
|
#include <fcntl.h>
|
|
#include <unistd.h>
|
|
#include <sys/mman.h>
|
|
|
|
extern "C" {
|
|
#include <libavutil/opt.h>
|
|
#include <libavcodec/avcodec.h>
|
|
#include <libavformat/avformat.h>
|
|
#include <libswresample/swresample.h>
|
|
}
|
|
|
|
typedef uint64_t u64;
|
|
typedef int64_t s64;
|
|
typedef uint32_t u32;
|
|
typedef int32_t s32;
|
|
typedef uint16_t u16;
|
|
typedef int16_t s16;
|
|
typedef uint8_t u8;
|
|
typedef int8_t s8;
|
|
|
|
#define WAVE_SAMPLE_RATE 16000
|
|
#define AVIO_CTX_BUF_SZ 4096
|
|
|
|
static const char* ffmpegLog = getenv("FFMPEG_LOG");
|
|
// Todo: add __FILE__ __LINE__
|
|
#define LOG(...) \
|
|
do { if (ffmpegLog) fprintf(stderr, __VA_ARGS__); } while(0) // C99
|
|
|
|
/*
|
|
* WAVE file header based on definition from
|
|
* https://gist.github.com/Jon-Schneider/8b7c53d27a7a13346a643dac9c19d34f
|
|
*
|
|
* We must ensure this structure doesn't have any holes or
|
|
* padding so we can just map it straight to the WAVE data.
|
|
*/
|
|
struct wave_hdr {
|
|
/* RIFF Header: "RIFF" */
|
|
char riff_header[4];
|
|
/* size of audio data + sizeof(struct wave_hdr) - 8 */
|
|
int wav_size;
|
|
/* "WAVE" */
|
|
char wav_header[4];
|
|
|
|
/* Format Header */
|
|
/* "fmt " (includes trailing space) */
|
|
char fmt_header[4];
|
|
/* Should be 16 for PCM */
|
|
int fmt_chunk_size;
|
|
/* Should be 1 for PCM. 3 for IEEE Float */
|
|
s16 audio_format;
|
|
s16 num_channels;
|
|
int sample_rate;
|
|
/*
|
|
* Number of bytes per second
|
|
* sample_rate * num_channels * bit_depth/8
|
|
*/
|
|
int byte_rate;
|
|
/* num_channels * bytes per sample */
|
|
s16 sample_alignment;
|
|
/* bits per sample */
|
|
s16 bit_depth;
|
|
|
|
/* Data Header */
|
|
/* "data" */
|
|
char data_header[4];
|
|
/*
|
|
* size of audio
|
|
* number of samples * num_channels * bit_depth/8
|
|
*/
|
|
int data_bytes;
|
|
} __attribute__((__packed__));
|
|
|
|
struct audio_buffer {
|
|
u8 *ptr;
|
|
int size; /* size left in the buffer */
|
|
};
|
|
|
|
static void set_wave_hdr(wave_hdr& wh, size_t size) {
|
|
memcpy(&wh.riff_header, "RIFF", 4);
|
|
wh.wav_size = size + sizeof(struct wave_hdr) - 8;
|
|
memcpy(&wh.wav_header, "WAVE", 4);
|
|
memcpy(&wh.fmt_header, "fmt ", 4);
|
|
wh.fmt_chunk_size = 16;
|
|
wh.audio_format = 1;
|
|
wh.num_channels = 1;
|
|
wh.sample_rate = WAVE_SAMPLE_RATE;
|
|
wh.sample_alignment = 2;
|
|
wh.bit_depth = 16;
|
|
wh.byte_rate = wh.sample_rate * wh.sample_alignment;
|
|
memcpy(&wh.data_header, "data", 4);
|
|
wh.data_bytes = size;
|
|
}
|
|
|
|
static void write_wave_hdr(int fd, size_t size) {
|
|
struct wave_hdr wh;
|
|
set_wave_hdr(wh, size);
|
|
write(fd, &wh, sizeof(struct wave_hdr));
|
|
}
|
|
|
|
static int map_file(int fd, u8 **ptr, size_t *size)
|
|
{
|
|
struct stat sb;
|
|
|
|
fstat(fd, &sb);
|
|
*size = sb.st_size;
|
|
|
|
*ptr = (u8*)mmap(NULL, *size, PROT_READ|PROT_WRITE, MAP_PRIVATE, fd, 0);
|
|
if (*ptr == MAP_FAILED) {
|
|
perror("mmap");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int read_packet(void *opaque, u8 *buf, int buf_size)
|
|
{
|
|
struct audio_buffer *audio_buf = (audio_buffer*)opaque;
|
|
|
|
buf_size = FFMIN(buf_size, audio_buf->size);
|
|
|
|
/* copy internal buffer data to buf */
|
|
memcpy(buf, audio_buf->ptr, buf_size);
|
|
audio_buf->ptr += buf_size;
|
|
audio_buf->size -= buf_size;
|
|
|
|
return buf_size;
|
|
}
|
|
|
|
static void convert_frame(struct SwrContext *swr, AVCodecContext *codec,
|
|
AVFrame *frame, s16 **data, int *size, bool flush)
|
|
{
|
|
int nr_samples;
|
|
s64 delay;
|
|
u8 *buffer;
|
|
|
|
delay = swr_get_delay(swr, codec->sample_rate);
|
|
nr_samples = av_rescale_rnd(delay + frame->nb_samples,
|
|
WAVE_SAMPLE_RATE, codec->sample_rate,
|
|
AV_ROUND_UP);
|
|
av_samples_alloc(&buffer, NULL, 1, nr_samples, AV_SAMPLE_FMT_S16, 0);
|
|
|
|
/*
|
|
* !flush is used to check if we are flushing any remaining
|
|
* conversion buffers...
|
|
*/
|
|
nr_samples = swr_convert(swr, &buffer, nr_samples,
|
|
!flush ? (const u8 **)frame->data : NULL,
|
|
!flush ? frame->nb_samples : 0);
|
|
|
|
*data = (s16*)realloc(*data, (*size + nr_samples) * sizeof(s16));
|
|
memcpy(*data + *size, buffer, nr_samples * sizeof(s16));
|
|
*size += nr_samples;
|
|
av_freep(&buffer);
|
|
}
|
|
|
|
static bool is_audio_stream(const AVStream *stream)
|
|
{
|
|
if (stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
|
|
return true;
|
|
|
|
return false;
|
|
}
|
|
|
|
// Return non zero on error, 0 on success
|
|
// audio_buffer: input memory
|
|
// data: decoded output audio data (wav file)
|
|
// size: size of output data
|
|
static int decode_audio(struct audio_buffer *audio_buf, s16 **data, int *size)
|
|
{
|
|
LOG("decode_audio: input size: %d\n", audio_buf->size);
|
|
AVFormatContext *fmt_ctx;
|
|
AVIOContext *avio_ctx;
|
|
AVStream *stream;
|
|
AVCodecContext *codec;
|
|
AVPacket packet;
|
|
AVFrame *frame;
|
|
struct SwrContext *swr;
|
|
u8 *avio_ctx_buffer;
|
|
unsigned int i;
|
|
int stream_index = -1;
|
|
int err;
|
|
const size_t errbuffsize = 1024;
|
|
char errbuff[errbuffsize];
|
|
|
|
fmt_ctx = avformat_alloc_context();
|
|
avio_ctx_buffer = (u8*)av_malloc(AVIO_CTX_BUF_SZ);
|
|
LOG("Creating an avio context: AVIO_CTX_BUF_SZ=%d\n", AVIO_CTX_BUF_SZ);
|
|
avio_ctx = avio_alloc_context(avio_ctx_buffer, AVIO_CTX_BUF_SZ, 0, audio_buf, &read_packet, NULL, NULL);
|
|
fmt_ctx->pb = avio_ctx;
|
|
|
|
// open the input stream and read header
|
|
err = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
|
|
if (err) {
|
|
LOG("Could not read audio buffer: %d: %s\n", err, av_make_error_string(errbuff, errbuffsize, err));
|
|
return err;
|
|
}
|
|
|
|
err = avformat_find_stream_info(fmt_ctx, NULL);
|
|
if (err < 0) {
|
|
LOG("Could not retrieve stream info from audio buffer: %d\n", err);
|
|
return err;
|
|
}
|
|
|
|
for (i = 0; i < fmt_ctx->nb_streams; i++) {
|
|
if (is_audio_stream(fmt_ctx->streams[i])) {
|
|
stream_index = i;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (stream_index == -1) {
|
|
LOG("Could not retrieve audio stream from buffer\n");
|
|
return -1;
|
|
}
|
|
|
|
stream = fmt_ctx->streams[stream_index];
|
|
codec = avcodec_alloc_context3(
|
|
avcodec_find_decoder(stream->codecpar->codec_id));
|
|
avcodec_parameters_to_context(codec, stream->codecpar);
|
|
err = avcodec_open2(codec, avcodec_find_decoder(codec->codec_id),
|
|
NULL);
|
|
if (err) {
|
|
LOG("Failed to open decoder for stream #%d in audio buffer\n", stream_index);
|
|
return err;
|
|
}
|
|
|
|
/* prepare resampler */
|
|
swr = swr_alloc();
|
|
|
|
av_opt_set_int(swr, "in_channel_count", codec->channels, 0);
|
|
av_opt_set_int(swr, "out_channel_count", 1, 0);
|
|
av_opt_set_int(swr, "in_channel_layout", codec->channel_layout, 0);
|
|
av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_MONO, 0);
|
|
av_opt_set_int(swr, "in_sample_rate", codec->sample_rate, 0);
|
|
av_opt_set_int(swr, "out_sample_rate", WAVE_SAMPLE_RATE, 0);
|
|
av_opt_set_sample_fmt(swr, "in_sample_fmt", codec->sample_fmt, 0);
|
|
av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
|
|
|
|
swr_init(swr);
|
|
if (!swr_is_initialized(swr)) {
|
|
LOG("Resampler has not been properly initialized\n");
|
|
return -1;
|
|
}
|
|
|
|
av_init_packet(&packet);
|
|
frame = av_frame_alloc();
|
|
if (!frame) {
|
|
LOG("Error allocating the frame\n");
|
|
return -1;
|
|
}
|
|
|
|
/* iterate through frames */
|
|
*data = NULL;
|
|
*size = 0;
|
|
while (av_read_frame(fmt_ctx, &packet) >= 0) {
|
|
avcodec_send_packet(codec, &packet);
|
|
|
|
err = avcodec_receive_frame(codec, frame);
|
|
if (err == AVERROR(EAGAIN))
|
|
continue;
|
|
|
|
convert_frame(swr, codec, frame, data, size, false);
|
|
}
|
|
/* Flush any remaining conversion buffers... */
|
|
convert_frame(swr, codec, frame, data, size, true);
|
|
|
|
av_frame_free(&frame);
|
|
swr_free(&swr);
|
|
//avio_context_free(); // todo?
|
|
avcodec_close(codec);
|
|
avformat_close_input(&fmt_ctx);
|
|
avformat_free_context(fmt_ctx);
|
|
|
|
if (avio_ctx) {
|
|
av_freep(&avio_ctx->buffer);
|
|
av_freep(&avio_ctx);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
// in mem decoding/conversion/resampling:
|
|
// ifname: input file path
|
|
// owav_data: in mem wav file. Can be forwarded as it to whisper/drwav
|
|
// return 0 on success
|
|
int ffmpeg_decode_audio(const std::string &ifname, std::vector<uint8_t>& owav_data) {
|
|
LOG("ffmpeg_decode_audio: %s\n", ifname.c_str());
|
|
int ifd = open(ifname.c_str(), O_RDONLY);
|
|
if (ifd == -1) {
|
|
fprintf(stderr, "Couldn't open input file %s\n", ifname.c_str());
|
|
return -1;
|
|
}
|
|
u8 *ibuf = NULL;
|
|
size_t ibuf_size;
|
|
int err = map_file(ifd, &ibuf, &ibuf_size);
|
|
if (err) {
|
|
LOG("Couldn't map input file %s\n", ifname.c_str());
|
|
return err;
|
|
}
|
|
LOG("Mapped input file: %s size: %d\n", ibuf, (int) ibuf_size);
|
|
struct audio_buffer inaudio_buf;
|
|
inaudio_buf.ptr = ibuf;
|
|
inaudio_buf.size = ibuf_size;
|
|
|
|
s16 *odata=NULL;
|
|
int osize=0;
|
|
|
|
err = decode_audio(&inaudio_buf, &odata, &osize);
|
|
LOG("decode_audio returned %d \n", err);
|
|
if (err != 0) {
|
|
LOG("decode_audio failed\n");
|
|
return err;
|
|
}
|
|
LOG("decode_audio output size: %d\n", osize);
|
|
|
|
wave_hdr wh;
|
|
const size_t outdatasize = osize * sizeof(s16);
|
|
set_wave_hdr(wh, outdatasize);
|
|
owav_data.resize(sizeof(wave_hdr) + outdatasize);
|
|
// header:
|
|
memcpy(owav_data.data(), &wh, sizeof(wave_hdr));
|
|
// the data:
|
|
memcpy(owav_data.data() + sizeof(wave_hdr), odata, osize* sizeof(s16));
|
|
|
|
return 0;
|
|
}
|