X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=nageru%2Faudio_mixer.cpp;h=360689b83406b04e6737cb6b7815d4092aca0a90;hb=25326c82bda01dfa1b86fb4f074d7697705239f8;hp=9e7dd59a0dbc64ab6398e30824c8dae6d676b912;hpb=9b7d691b4cc5db7dbfc18c82e86c1207fcac4722;p=nageru diff --git a/nageru/audio_mixer.cpp b/nageru/audio_mixer.cpp index 9e7dd59..360689b 100644 --- a/nageru/audio_mixer.cpp +++ b/nageru/audio_mixer.cpp @@ -18,11 +18,11 @@ #include #include -#include "db.h" +#include "decibel.h" #include "flags.h" -#include "metrics.h" +#include "shared/metrics.h" #include "state.pb.h" -#include "timebase.h" +#include "shared/timebase.h" using namespace bmusb; using namespace std; @@ -52,6 +52,26 @@ void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_chan } } +void convert_fixed16_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels, + const uint8_t *src, size_t in_channel, size_t in_num_channels, + size_t num_samples) +{ + assert(in_channel < in_num_channels); + assert(out_channel < out_num_channels); + src += in_channel * 2; + dst += out_channel; + + for (size_t i = 0; i < num_samples; ++i) { + uint32_t s = uint32_t(uint16_t(le16toh(*(int16_t *)src))) << 16; + + // Keep the sign bit in place, repeat the other 15 bits as far as they go. + *dst = s | ((s & 0x7fffffff) >> 15) | ((s & 0x7fffffff) >> 30); + + src += 2 * in_num_channels; + dst += out_num_channels; + } +} + void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels, const uint8_t *src, size_t in_channel, size_t in_num_channels, size_t num_samples) @@ -65,8 +85,31 @@ void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_chan uint32_t s1 = src[0]; uint32_t s2 = src[1]; uint32_t s3 = src[2]; - uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24); - *dst = int(s) * (1.0f / 2147483648.0f); + uint32_t s = (s1 << 8) | (s2 << 16) | (s3 << 24); // Note: The bottom eight bits are zero; s3 includes the sign bit. + *dst = int(s) * (1.0f / (256.0f * 8388608.0f)); // 256 for signed down-shift by 8, then 2^23 for the actual conversion. + + src += 3 * in_num_channels; + dst += out_num_channels; + } +} + +void convert_fixed24_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels, + const uint8_t *src, size_t in_channel, size_t in_num_channels, + size_t num_samples) +{ + assert(in_channel < in_num_channels); + assert(out_channel < out_num_channels); + src += in_channel * 3; + dst += out_channel; + + for (size_t i = 0; i < num_samples; ++i) { + uint32_t s1 = src[0]; + uint32_t s2 = src[1]; + uint32_t s3 = src[2]; + uint32_t s = (s1 << 8) | (s2 << 16) | (s3 << 24); + + // Keep the sign bit in place, repeat the other 23 bits as far as they go. + *dst = s | ((s & 0x7fffffff) >> 23); src += 3 * in_num_channels; dst += out_num_channels; @@ -91,6 +134,25 @@ void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_chan } } +// Basically just a reinterleave. +void convert_fixed32_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels, + const uint8_t *src, size_t in_channel, size_t in_num_channels, + size_t num_samples) +{ + assert(in_channel < in_num_channels); + assert(out_channel < out_num_channels); + src += in_channel * 4; + dst += out_channel; + + for (size_t i = 0; i < num_samples; ++i) { + int32_t s = le32toh(*(int32_t *)src); + *dst = s; + + src += 4 * in_num_channels; + dst += out_num_channels; + } +} + float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused)); float find_peak_plain(const float *samples, size_t num_samples) @@ -206,7 +268,7 @@ AudioMixer::AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs) &new_input_mapping)) { fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n", global_flags.input_mapping_filename.c_str()); - exit(1); + abort(); } set_input_mapping(new_input_mapping); } else { @@ -244,7 +306,7 @@ void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec) } } -bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length, steady_clock::time_point frame_time) +bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, steady_clock::time_point frame_time) { AudioDevice *device = find_audio_device(device_spec); @@ -294,7 +356,39 @@ bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned return true; } -bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length) +vector convert_audio_to_fixed32(const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, unsigned num_channels) +{ + vector audio; + + if (num_channels > audio_format.num_channels) { + audio.resize(num_samples * num_channels, 0); + } else { + audio.resize(num_samples * num_channels); + } + for (unsigned channel_index = 0; channel_index < num_channels && channel_index < audio_format.num_channels; ++channel_index) { + switch (audio_format.bits_per_sample) { + case 0: + assert(num_samples == 0); + break; + case 16: + convert_fixed16_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples); + break; + case 24: + convert_fixed24_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples); + break; + case 32: + convert_fixed32_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples); + break; + default: + fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample); + assert(false); + } + } + + return audio; +} + +bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames) { AudioDevice *device = find_audio_device(device_spec);