この質問を投稿して以来、生の PCM データから有効な WAV ファイルを自分で書き込もうとしています。FLAC コンバーターを作成することができました (テスト済みで動作します) が、これまでに作成した WAV ファイルをエンコードしません。
何が間違っているのかわかりません。私はインターネットで他の個人の ソース コードを調べ、それを自分のものと比較してきましたが、まだ動作させることができません。
以下はソースコードを簡略化したものです (申し訳ありませんが、まだ少し長いです.wav。自分で記録するには少しコードが必要です)。
// Compile with "g++ test.ccp -o test -lasound"
// Use the newer ALSA API
#define ALSA_PCM_NEW_HW_PARAMS_API
#include <alsa/asoundlib.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <stdint.h>
struct WaveHeader
{
char RIFF_marker[4];
uint32_t file_size;
char filetype_header[4];
char format_marker[4];
uint32_t data_header_length;
uint16_t format_type;
uint16_t number_of_channels;
uint32_t sample_rate;
uint32_t bytes_per_second;
uint16_t bytes_per_frame;
uint16_t bits_per_sample;
};
struct WaveHeader *genericWAVHeader(uint32_t sample_rate, uint16_t bit_depth, uint16_t channels)
{
struct WaveHeader *hdr;
hdr = (WaveHeader*) malloc(sizeof(*hdr));
if (!hdr)
return NULL;
memcpy(&hdr->RIFF_marker, "RIFF", 4);
memcpy(&hdr->filetype_header, "WAVE", 4);
memcpy(&hdr->format_marker, "fmt ", 4);
hdr->data_header_length = 16;
hdr->format_type = 1;
hdr->number_of_channels = channels;
hdr->sample_rate = sample_rate;
hdr->bytes_per_second = sample_rate * channels * bit_depth / 8;
hdr->bytes_per_frame = channels * bit_depth / 8;
hdr->bits_per_sample = bit_depth;
return hdr;
}
int writeWAVHeader(int fd, struct WaveHeader *hdr)
{
if (!hdr)
return -1;
write(fd, &hdr->RIFF_marker, 4);
write(fd, &hdr->file_size, 4);
write(fd, &hdr->filetype_header, 4);
write(fd, &hdr->format_marker, 4);
write(fd, &hdr->data_header_length, 4);
write(fd, &hdr->format_type, 2);
write(fd, &hdr->number_of_channels, 2);
write(fd, &hdr->sample_rate, 4);
write(fd, &hdr->bytes_per_second, 4);
write(fd, &hdr->bytes_per_frame, 2);
write(fd, &hdr->bits_per_sample, 2);
write(fd, "data", 4);
uint32_t data_size = hdr->file_size + 8 - 44;
write(fd, &data_size, 4);
return 0;
}
int recordWAV(const char *fileName, struct WaveHeader *hdr, uint32_t duration)
{
int err;
int size;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int sampleRate = hdr->sample_rate;
int dir;
snd_pcm_uframes_t frames = 32;
char *device = (char*) "plughw:1,0";
char *buffer;
int filedesc;
printf("Capture device is %s\n", device);
/* Open PCM device for recording (capture). */
err = snd_pcm_open(&handle, device, SND_PCM_STREAM_CAPTURE, 0);
if (err)
{
fprintf(stderr, "Unable to open PCM device: %s\n", snd_strerror(err));
return err;
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* ### Set the desired hardware parameters. ### */
/* Interleaved mode */
err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err)
{
fprintf(stderr, "Error setting interleaved mode: %s\n", snd_strerror(err));
snd_pcm_close(handle);
return err;
}
/* Signed 16-bit little-endian format */
if (hdr->bits_per_sample == 16) err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
else err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_U8);
if (err)
{
fprintf(stderr, "Error setting format: %s\n", snd_strerror(err));
snd_pcm_close(handle);
return err;
}
/* Two channels (stereo) */
err = snd_pcm_hw_params_set_channels(handle, params, hdr->number_of_channels);
if (err)
{
fprintf(stderr, "Error setting channels: %s\n", snd_strerror(err));
snd_pcm_close(handle);
return err;
}
/* 44100 bits/second sampling rate (CD quality) */
sampleRate = hdr->sample_rate;
err = snd_pcm_hw_params_set_rate_near(handle, params, &sampleRate, &dir);
if (err)
{
fprintf(stderr, "Error setting sampling rate (%d): %s\n", sampleRate, snd_strerror(err));
snd_pcm_close(handle);
return err;
}
hdr->sample_rate = sampleRate;
/* Set period size*/
err = snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir);
if (err)
{
fprintf(stderr, "Error setting period size: %s\n", snd_strerror(err));
snd_pcm_close(handle);
return err;
}
/* Write the parameters to the driver */
err = snd_pcm_hw_params(handle, params);
if (err < 0)
{
fprintf(stderr, "Unable to set HW parameters: %s\n", snd_strerror(err));
snd_pcm_close(handle);
return err;
}
/* Use a buffer large enough to hold one period */
err = snd_pcm_hw_params_get_period_size(params, &frames, &dir);
if (err)
{
fprintf(stderr, "Error retrieving period size: %s\n", snd_strerror(err));
snd_pcm_close(handle);
return err;
}
size = frames * hdr->bits_per_sample / 8 * hdr->number_of_channels; /* 2 bytes/sample, 2 channels */
buffer = (char *) malloc(size);
if (!buffer)
{
fprintf(stdout, "Buffer error.\n");
snd_pcm_close(handle);
return -1;
}
err = snd_pcm_hw_params_get_period_time(params, &sampleRate, &dir);
if (err)
{
fprintf(stderr, "Error retrieving period time: %s\n", snd_strerror(err));
snd_pcm_close(handle);
free(buffer);
return err;
}
uint32_t pcm_data_size = hdr->sample_rate * hdr->bytes_per_frame * duration / 1000;
hdr->file_size = pcm_data_size + 44 - 8;
filedesc = open(fileName, O_WRONLY | O_CREAT, 0644);
err = writeWAVHeader(filedesc, hdr);
if (err)
{
fprintf(stderr, "Error writing .wav header.");
snd_pcm_close(handle);
free(buffer);
close(filedesc);
return err;
}
fprintf(stdout, "Channels: %d\n", hdr->number_of_channels);
for(int i = duration * 1000 / sampleRate; i > 0; i--)
{
err = snd_pcm_readi(handle, buffer, frames);
if (err == -EPIPE) fprintf(stderr, "Overrun occurred: %d\n", err);
if (err) err = snd_pcm_recover(handle, err, 0);
// Still an error, need to exit.
if (err)
{
fprintf(stderr, "Error occured while recording: %s\n", snd_strerror(err));
snd_pcm_close(handle);
free(buffer);
close(filedesc);
return err;
}
write(filedesc, buffer, size);
}
close(filedesc);
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);
printf("Finished writing to %s\n", fileName);
return 0;
}
int main(int argc, char *argv[]) {
if(argc != 2)
{
fprintf(stderr, "Usage: %s (record duration)\n", argv[0]);
return -1;
}
int err;
struct WaveHeader *hdr;
// Creates a temporary file in /tmp
char wavFile[L_tmpnam + 5];
char *tempFilenameStub = tmpnam(NULL);
sprintf(wavFile, "%s.wav", tempFilenameStub);
hdr = genericWAVHeader(44000, 16, 2);
if (!hdr)
{
fprintf(stderr, "Error allocating WAV header.\n");
return -1;
}
err = recordWAV(wavFile, hdr, 1000 * strtod(argv[1], NULL));
if (err)
{
fprintf(stderr, "Error recording WAV file: %d\n", err);
return err;
}
free(hdr);
return 0;
}
プログラムの実行時に得られる出力:
$ ./capture 5
Capture device is plughw:1,0
Channels: 2
Error occured while recording: Channel number out of range
Error recording WAV file: -44
助言がありますか？私はしばらくこれについて頭を悩ませてきました。