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refactoring

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This commit is contained in:
Maxim Devaev
2022-06-06 16:17:34 +03:00
parent 63fe32ddd9
commit e9a6db02f6
2 changed files with 25 additions and 25 deletions
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48
janus/src/audio.c
View File

@@ -32,15 +32,14 @@
// A number of frames per 1 channel:
// - https://github.com/xiph/opus/blob/7b05f44/src/opus_demo.c#L368
#define BITRATE_TO_FRAMES(_bitrate) (6 * (_bitrate) / 50) // 120ms
#define BITRATE_TO_BUF16(_bitrate) (BITRATE_TO_FRAMES(_bitrate) * 2) // One stereo frame = (16bit L) + (16bit R)
#define BITRATE_TO_BUF8(_bitrate) (BITRATE_TO_BUF16(_bitrate) * sizeof(int16_t))
#define HZ_TO_FRAMES(_hz) (6 * (_hz) / 50) // 120ms
#define HZ_TO_BUF16(_hz) (HZ_TO_FRAMES(_hz) * 2) // One stereo frame = (16bit L) + (16bit R)
#define HZ_TO_BUF8(_hz) (HZ_TO_BUF16(_hz) * sizeof(int16_t))
#define MAX_PCM_BITRATE 192000
#define MAX_BUF16 BITRATE_TO_BUF16(MAX_PCM_BITRATE)
#define MAX_BUF8 BITRATE_TO_BUF8(MAX_PCM_BITRATE)
#define ENCODER_INPUT_BITRATE 48000
#define ENCODER_OUTPUT_BITRATE 48000
#define MAX_PCM_HZ 192000
#define MAX_BUF16 HZ_TO_BUF16(MAX_PCM_HZ)
#define MAX_BUF8 HZ_TO_BUF8(MAX_PCM_HZ)
#define ENCODER_INPUT_HZ 48000
typedef struct {
@@ -61,7 +60,7 @@ static void *_encoder_thread(void *v_audio);
audio_s *audio_init(const char *name) {
audio_s *audio;
A_CALLOC(audio, 1);
audio->pcm_bitrate = 48000;
audio->pcm_hz = 48000;
audio->pcm_queue = queue_init(8);
audio->enc_queue = queue_init(8);
atomic_init(&audio->run, true);
@@ -86,21 +85,21 @@ audio_s *audio_init(const char *name) {
SET_PARAM("Can't set PCM access type", snd_pcm_hw_params_set_access, SND_PCM_ACCESS_RW_INTERLEAVED);
SET_PARAM("Can't set PCM channels numbre", snd_pcm_hw_params_set_channels, 2);
SET_PARAM("Can't set PCM sampling format", snd_pcm_hw_params_set_format, SND_PCM_FORMAT_S16_LE);
SET_PARAM("Can't set PCM sampling rate", snd_pcm_hw_params_set_rate_near, &audio->pcm_bitrate, 0);
if (audio->pcm_bitrate > MAX_PCM_BITRATE) {
JLOG_ERROR("audio", "Unsupported PCM bitrate %u, max=%u", audio->pcm_bitrate, MAX_PCM_BITRATE);
SET_PARAM("Can't set PCM sampling rate", snd_pcm_hw_params_set_rate_near, &audio->pcm_hz, 0);
if (audio->pcm_hz > MAX_PCM_HZ) {
JLOG_ERROR("audio", "Unsupported PCM freq: %u; max=%u", audio->pcm_hz, MAX_PCM_HZ);
goto error;
}
JLOG_INFO("audio", "Using PCM bitrate: %u", audio->pcm_bitrate);
audio->pcm_frames = BITRATE_TO_FRAMES(audio->pcm_bitrate);
audio->pcm_size = BITRATE_TO_BUF8(audio->pcm_bitrate);
JLOG_INFO("audio", "Using PCM freq: %u", audio->pcm_hz);
audio->pcm_frames = HZ_TO_FRAMES(audio->pcm_hz);
audio->pcm_size = HZ_TO_BUF8(audio->pcm_hz);
SET_PARAM("Can't apply PCM params", snd_pcm_hw_params);
# undef SET_PARAM
}
if (audio->pcm_bitrate != ENCODER_INPUT_BITRATE) {
audio->res = speex_resampler_init(2, audio->pcm_bitrate, ENCODER_INPUT_BITRATE, SPEEX_RESAMPLER_QUALITY_DESKTOP, &err);
if (audio->pcm_hz != ENCODER_INPUT_HZ) {
audio->res = speex_resampler_init(2, audio->pcm_hz, ENCODER_INPUT_HZ, SPEEX_RESAMPLER_QUALITY_DESKTOP, &err);
if (err < 0) {
JLOG_PERROR_RES(err, "audio", "Can't create resampler");
goto error;
@@ -109,9 +108,9 @@ audio_s *audio_init(const char *name) {
{
// OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY
audio->enc = opus_encoder_create(ENCODER_INPUT_BITRATE, 2, OPUS_APPLICATION_AUDIO, &err);
audio->enc = opus_encoder_create(ENCODER_INPUT_HZ, 2, OPUS_APPLICATION_AUDIO, &err);
assert(err == 0);
assert(!opus_encoder_ctl(audio->enc, OPUS_SET_BITRATE(ENCODER_OUTPUT_BITRATE)));
assert(!opus_encoder_ctl(audio->enc, OPUS_SET_BITRATE(48000)));
assert(!opus_encoder_ctl(audio->enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)));
assert(!opus_encoder_ctl(audio->enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)));
// OPUS_SET_INBAND_FEC(1), OPUS_SET_PACKET_LOSS_PERC(10): see rtpa.c
@@ -224,19 +223,19 @@ static void *_encoder_thread(void *v_audio) {
if (!queue_get(audio->pcm_queue, (void **)&in, 1)) {
int16_t *in_ptr;
if (audio->res) {
assert(audio->pcm_bitrate != ENCODER_INPUT_BITRATE);
assert(audio->pcm_hz != ENCODER_INPUT_HZ);
uint32_t in_count = audio->pcm_frames;
uint32_t out_count = BITRATE_TO_FRAMES(ENCODER_INPUT_BITRATE);
uint32_t out_count = HZ_TO_FRAMES(ENCODER_INPUT_HZ);
speex_resampler_process_interleaved_int(audio->res, in->data, &in_count, in_res, &out_count);
in_ptr = in_res;
} else {
assert(audio->pcm_bitrate == ENCODER_INPUT_BITRATE);
assert(audio->pcm_hz == ENCODER_INPUT_HZ);
in_ptr = in->data;
}
_enc_buffer_s *out;
A_CALLOC(out, 1);
int size = opus_encode(audio->enc, in_ptr, BITRATE_TO_FRAMES(ENCODER_INPUT_BITRATE), out->data, ARRAY_LEN(out->data));
int size = opus_encode(audio->enc, in_ptr, HZ_TO_FRAMES(ENCODER_INPUT_HZ), out->data, ARRAY_LEN(out->data));
free(in);
if (size < 0) {
JLOG_PERROR_OPUS(size, "audio", "Can't encode PCM frame to OPUS; breaking audio ...");
@@ -245,7 +244,8 @@ static void *_encoder_thread(void *v_audio) {
}
out->used = size;
out->pts = audio->pts;
audio->pts += BITRATE_TO_FRAMES(ENCODER_OUTPUT_BITRATE);
// https://datatracker.ietf.org/doc/html/rfc7587#section-4.2
audio->pts += HZ_TO_FRAMES(ENCODER_INPUT_HZ);
if (queue_get_free(audio->enc_queue)) {
assert(!queue_put(audio->enc_queue, out, 1));

2
janus/src/audio.h
View File

@@ -47,7 +47,7 @@
typedef struct {
snd_pcm_t *pcm;
unsigned pcm_bitrate;
unsigned pcm_hz;
unsigned pcm_frames;
size_t pcm_size;
snd_pcm_hw_params_t *pcm_params;