whisper.cpp/examples/ffmpeg-transcode.cpp at master · JustCallMeShiv/whisper.cpp · GitHub

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/* SPDX-License-Identifier: GPL-2.0 */

* transcode.c - convert audio file to WAVE

// 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();

#if LIBAVCODEC_VERSION_MAJOR > 60

AVChannelLayout in_ch_layout = codec->ch_layout;

AVChannelLayout out_ch_layout = AV_CHANNEL_LAYOUT_MONO;

/* Set the source audio layout as-is */

av_opt_set_chlayout(swr, "in_chlayout", &in_ch_layout, 0);

av_opt_set_int(swr, "in_sample_rate", codec->sample_rate, 0);

av_opt_set_sample_fmt(swr, "in_sample_fmt", codec->sample_fmt, 0);

/* Convert it into 16khz Mono */

av_opt_set_chlayout(swr, "out_chlayout", &out_ch_layout, 0);

av_opt_set_int(swr, "out_sample_rate", WAVE_SAMPLE_RATE, 0);

av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);

#else

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);

#endif

swr_init(swr);

if (!swr_is_initialized(swr)) {

LOG("Resampler has not been properly initialized\n");

return -1;

}

packet=av_packet_alloc();

if (!packet) {

LOG("Error allocating the packet\n");

return -1;

}

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_packet_free(&packet);

av_frame_free(&frame);

swr_free(&swr);

//avio_context_free(); // todo?

avcodec_free_context(&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;

}

whisper.cpp/examples/ffmpeg-transcode.cpp at master · JustCallMeShiv/whisper.cpp · GitHub

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