diff options
Diffstat (limited to 'src/sound.cpp')
| -rw-r--r-- | src/sound.cpp | 292 |
1 files changed, 197 insertions, 95 deletions
diff --git a/src/sound.cpp b/src/sound.cpp index d0b5033..9ca1381 100644 --- a/src/sound.cpp +++ b/src/sound.cpp @@ -20,11 +20,193 @@ #include <stdlib.h> #include <stdio.h> #include <string.h> +#include <time.h> +#include <cmath> -#ifdef PULSEAUDIO -#include <pulse/simple.h> +#include <pulse/pulseaudio.h> +#include <pulse/mainloop.h> +#include <pulse/xmalloc.h> #include <pulse/error.h> +#define CHECK_DEAD_GOTO(p, rerror, label) \ + do { \ + if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \ + !(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \ + if (((p)->context && pa_context_get_state((p)->context) == PA_CONTEXT_FAILED) || \ + ((p)->stream && pa_stream_get_state((p)->stream) == PA_STREAM_FAILED)) { \ + if (rerror) \ + *(rerror) = pa_context_errno((p)->context); \ + } else \ + if (rerror) \ + *(rerror) = PA_ERR_BADSTATE; \ + goto label; \ + } \ + } while(false); + +static double clock_get_monotonic_seconds() { + struct timespec ts; + ts.tv_sec = 0; + ts.tv_nsec = 0; + clock_gettime(CLOCK_MONOTONIC, &ts); + return (double)ts.tv_sec + (double)ts.tv_nsec * 0.000000001; +} + +static int sound_device_index = 0; + +struct pa_handle { + pa_context *context; + pa_stream *stream; + pa_mainloop *mainloop; + + const void *read_data; + size_t read_index, read_length; + + int operation_success; +}; + +static void pa_sound_device_free(pa_handle *s) { + assert(s); + + if (s->stream) + pa_stream_unref(s->stream); + + if (s->context) { + pa_context_disconnect(s->context); + pa_context_unref(s->context); + } + + if (s->mainloop) + pa_mainloop_free(s->mainloop); + + pa_xfree(s); +} + +static pa_handle* pa_sound_device_new(const char *server, + const char *name, + const char *dev, + const char *stream_name, + const pa_sample_spec *ss, + const pa_buffer_attr *attr, + int *rerror) { + pa_handle *p; + int error = PA_ERR_INTERNAL, r; + + p = pa_xnew0(pa_handle, 1); + + if (!(p->mainloop = pa_mainloop_new())) + goto fail; + + if (!(p->context = pa_context_new(pa_mainloop_get_api(p->mainloop), name))) + goto fail; + + if (pa_context_connect(p->context, server, PA_CONTEXT_NOFLAGS, NULL) < 0) { + error = pa_context_errno(p->context); + goto fail; + } + + for (;;) { + pa_context_state_t state = pa_context_get_state(p->context); + + if (state == PA_CONTEXT_READY) + break; + + if (!PA_CONTEXT_IS_GOOD(state)) { + error = pa_context_errno(p->context); + goto fail; + } + + pa_mainloop_iterate(p->mainloop, 1, NULL); + } + + if (!(p->stream = pa_stream_new(p->context, stream_name, ss, NULL))) { + error = pa_context_errno(p->context); + goto fail; + } + + r = pa_stream_connect_record(p->stream, dev, attr, + (pa_stream_flags_t)(PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE)); + + if (r < 0) { + error = pa_context_errno(p->context); + goto fail; + } + + for (;;) { + pa_stream_state_t state = pa_stream_get_state(p->stream); + + if (state == PA_STREAM_READY) + break; + + if (!PA_STREAM_IS_GOOD(state)) { + error = pa_context_errno(p->context); + goto fail; + } + + pa_mainloop_iterate(p->mainloop, 1, NULL); + } + + return p; + +fail: + if (rerror) + *rerror = error; + pa_sound_device_free(p); + return NULL; +} + +// Returns a negative value on failure. Always blocks a time specified matching the sampling rate of the audio. +static int pa_sound_device_read(pa_handle *p, void *data, size_t length) { + assert(p); + + int r = 0; + int *rerror = &r; + bool retry = true; + + pa_mainloop_iterate(p->mainloop, 0, NULL); + const int64_t timeout_ms = std::round((1000.0 / (double)pa_stream_get_sample_spec(p->stream)->rate) * 1000.0); + + CHECK_DEAD_GOTO(p, rerror, fail); + + while(true) { + if(pa_stream_readable_size(p->stream) < length) { + if(!retry) + break; + + retry = false; + + const double start_time = clock_get_monotonic_seconds(); + while((clock_get_monotonic_seconds() - start_time) * 1000.0 < timeout_ms) { + pa_mainloop_prepare(p->mainloop, 1 * 1000); + pa_mainloop_poll(p->mainloop); + pa_mainloop_dispatch(p->mainloop); + } + + continue; + } + + r = pa_stream_peek(p->stream, &p->read_data, &p->read_length); + if(r != 0) { + if(retry) + usleep(timeout_ms * 1000); + return -1; + } + + if(p->read_length < length || !p->read_data) { + pa_stream_drop(p->stream); + if(retry) + usleep(timeout_ms * 1000); + return -1; + } + + memcpy(data, p->read_data, length); + pa_stream_drop(p->stream); + return 0; + } + + fail: + return -1; +} + int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int num_channels, unsigned int period_frame_size) { pa_sample_spec ss; ss.format = PA_SAMPLE_S16LE; @@ -39,8 +221,13 @@ int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int buffer_attr.maxlength = period_frame_size * 2 * num_channels; // 2 bytes/sample, @num_channels channels buffer_attr.fragsize = buffer_attr.maxlength; - pa_simple *pa_handle = pa_simple_new(nullptr, "gpu-screen-recorder", PA_STREAM_RECORD, name, "record", &ss, nullptr, &buffer_attr, &error); - if(!pa_handle) { + // We want a unique stream name for every device which allows each input to be a different box in pipewire graph software + char stream_name[64]; + snprintf(stream_name, sizeof(stream_name), "record-%d", sound_device_index); + ++sound_device_index; + + pa_handle *handle = pa_sound_device_new(nullptr, "gpu-screen-recorder", name, stream_name, &ss, &buffer_attr, &error); + if(!handle) { fprintf(stderr, "pa_simple_new() failed: %s. Audio input device %s might not be valid\n", pa_strerror(error), name); return -1; } @@ -49,13 +236,13 @@ int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int void *buffer = malloc(buffer_size); if(!buffer) { fprintf(stderr, "failed to allocate buffer for audio\n"); - pa_simple_free(pa_handle); + pa_sound_device_free(handle); return -1; } fprintf(stderr, "Using pulseaudio\n"); - device->handle = pa_handle; + device->handle = handle; device->buffer = buffer; device->buffer_size = buffer_size; device->frames = period_frame_size; @@ -63,100 +250,15 @@ int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int } void sound_device_close(SoundDevice *device) { - pa_simple_free((pa_simple*)device->handle); + pa_sound_device_free((pa_handle*)device->handle); free(device->buffer); } int sound_device_read_next_chunk(SoundDevice *device, void **buffer) { - int error = 0; - if(pa_simple_read((pa_simple*)device->handle, device->buffer, device->buffer_size, &error) < 0) { - fprintf(stderr, "pa_simple_read() failed: %s\n", pa_strerror(error)); + if(pa_sound_device_read((pa_handle*)device->handle, device->buffer, device->buffer_size) < 0) { + //fprintf(stderr, "pa_simple_read() failed: %s\n", pa_strerror(error)); return -1; } *buffer = device->buffer; return device->frames; -} -#else -#define ALSA_PCM_NEW_HW_PARAMS_API -#include <alsa/asoundlib.h> - -int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int num_channels, unsigned int period_frame_size) { - int rc; - snd_pcm_t *handle; - - rc = snd_pcm_open(&handle, name, SND_PCM_STREAM_CAPTURE, 0); - if(rc < 0) { - fprintf(stderr, "unable to open pcm device 'default', reason: %s\n", snd_strerror(rc)); - return rc; - } - - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca(¶ms); - // Fill the params with default values - snd_pcm_hw_params_any(handle, params); - // Interleaved mode - snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); - // Signed 16--bit little-endian format - snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); - snd_pcm_hw_params_set_channels(handle, params, num_channels); - - // 48000 bits/second samling rate (DVD quality) - unsigned int val = 48000; - int dir; - snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); - - snd_pcm_uframes_t frames = period_frame_size; - snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir); - - // Write the parmeters to the driver - rc = snd_pcm_hw_params(handle, params); - if(rc < 0) { - fprintf(stderr, "unable to set hw parameters, reason: %s\n", snd_strerror(rc)); - snd_pcm_close(handle); - return rc; - } - - // Use a buffer large enough to hold one period - snd_pcm_hw_params_get_period_size(params, &frames, &dir); - int buffer_size = frames * 2 * num_channels; // 2 bytes/sample, @num_channels channels - void *buffer = malloc(buffer_size); - if(!buffer) { - fprintf(stderr, "failed to allocate buffer for audio\n"); - snd_pcm_close(handle); - return -1; - } - - fprintf(stderr, "Using alsa\n"); - - device->handle = handle; - device->buffer = buffer; - device->buffer_size = buffer_size; - device->frames = frames; - return 0; -} - -void sound_device_close(SoundDevice *device) { - /* TODO: Is this also needed in @sound_device_get_by_name on failure? */ - // TODO: This has been commented out since it causes the thread to block forever. Why? - //snd_pcm_drain((snd_pcm_t*)device->handle); - snd_pcm_close((snd_pcm_t*)device->handle); - free(device->buffer); -} - -int sound_device_read_next_chunk(SoundDevice *device, void **buffer) { - int rc = snd_pcm_readi((snd_pcm_t*)device->handle, device->buffer, device->frames); - if (rc == -EPIPE) { - /* overrun */ - fprintf(stderr, "overrun occured\n"); - snd_pcm_prepare((snd_pcm_t*)device->handle); - return rc; - } else if(rc < 0) { - fprintf(stderr, "failed to read from sound device, reason: %s\n", snd_strerror(rc)); - return rc; - } else if (rc != (int)device->frames) { - fprintf(stderr, "short read, read %d frames\n", rc); - } - *buffer = device->buffer; - return rc; -} -#endif +}
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