From f589bdc2584d3d1bee4cdfa6deb287180790854d Mon Sep 17 00:00:00 2001 From: "Steinar H. Gunderson" Date: Mon, 1 Aug 2016 17:50:27 +0200 Subject: [PATCH] Pick out the right channels when resampling. --- audio_mixer.cpp | 94 ++++++++++++++++++++++++++++++------------------- 1 file changed, 57 insertions(+), 37 deletions(-) diff --git a/audio_mixer.cpp b/audio_mixer.cpp index 7baa218..14ad63f 100644 --- a/audio_mixer.cpp +++ b/audio_mixer.cpp @@ -4,6 +4,7 @@ #include #include #include +#include #include #include "db.h" @@ -15,33 +16,45 @@ using namespace std; namespace { -// TODO: If these prove to be a bottleneck, they can be SSSE3-optimized. +// TODO: If these prove to be a bottleneck, they can be SSSE3-optimized +// (usually including multiple channels at a time). -void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples) +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) { - assert(in_channels >= out_channels); + 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) { - for (size_t j = 0; j < out_channels; ++j) { - uint32_t s1 = *src++; - uint32_t s2 = *src++; - uint32_t s3 = *src++; - uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24); - dst[i * out_channels + j] = int(s) * (1.0f / 2147483648.0f); - } - src += 3 * (in_channels - out_channels); + 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); + + src += 3 * in_num_channels; + dst += out_num_channels; } } -void convert_fixed32_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples) +void convert_fixed32_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) { - assert(in_channels >= out_channels); + 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) { - for (size_t j = 0; j < out_channels; ++j) { - int32_t s = le32toh(*(int32_t *)src); - dst[i * out_channels + j] = s * (1.0f / 2147483648.0f); - src += 4; - } - src += 4 * (in_channels - out_channels); + int32_t s = le32toh(*(int32_t *)src); + *dst = s * (1.0f / 2147483648.0f); + + src += 4 * in_num_channels; + dst += out_num_channels; } } @@ -106,22 +119,24 @@ void AudioMixer::add_audio(unsigned card_index, const uint8_t *data, unsigned nu assert(num_channels > 0); // Convert the audio to stereo fp32. - // FIXME: Pick out the right channels; this takes the first ones. vector audio; audio.resize(num_samples * num_channels); - switch (audio_format.bits_per_sample) { - case 0: - assert(num_samples == 0); - break; - case 24: - convert_fixed24_to_fp32(&audio[0], num_channels, data, audio_format.num_channels, num_samples); - break; - case 32: - convert_fixed32_to_fp32(&audio[0], num_channels, data, 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); + unsigned channel_index = 0; + for (auto channel_it = card->interesting_channels.cbegin(); channel_it != card->interesting_channels.end(); ++channel_it, ++channel_index) { + switch (audio_format.bits_per_sample) { + case 0: + assert(num_samples == 0); + break; + case 24: + convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples); + break; + case 32: + convert_fixed32_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, 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); + } } // Now add it. @@ -161,10 +176,15 @@ void AudioMixer::find_sample_src_from_capture_card(const vector *samples_ *stride = 0; return; } - // FIXME: map back through the interesting_channels squeeze map instead of using source_channel - // directly, which will be wrong (and might even overrun). - *srcptr = &samples_card[card_index][source_channel]; - *stride = cards[card_index].interesting_channels.size(); + CaptureCard *card = &cards[card_index]; + unsigned channel_index = 0; + for (int channel : card->interesting_channels) { + if (channel == source_channel) break; + ++channel_index; + } + assert(channel_index < card->interesting_channels.size()); + *srcptr = &samples_card[card_index][channel_index]; + *stride = card->interesting_channels.size(); } vector AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy) -- 2.39.2