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audio resampler: separate impl from iface

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made API for resamplers (mainly to readd speex)
This commit is contained in:
Martin Pulec
2022-10-14 11:45:21 +02:00
parent f224e62872
commit bb9fb011da
2 changed files with 95 additions and 77 deletions
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139
src/audio/types.cpp
View File

@@ -83,24 +83,29 @@ audio_desc::operator string() const
return oss.str();
}
audio_frame2_resampler::~audio_frame2_resampler() {
if (resampler) {
#ifdef HAVE_SOXR
soxr_delete((soxr_t) resampler);
#endif
tuple<bool, audio_frame2> audio_frame2_resampler::resample(audio_frame2 &a, vector<audio_frame2::channel> &out, int new_sample_rate_num, int new_sample_rate_den)
{
if (!impl) {
LOG(LOG_LEVEL_ERROR) << "Audio frame resampler: cannot resample, Soxr/SpeexDSP was not compiled in!\n";
return { false, audio_frame2{} };
}
return impl->resample(a, out, new_sample_rate_num, new_sample_rate_den);
}
/**
* @brief Checks whether the resampler has been set.
*
* @return true The resampler has been initialised.
* @return false The resampler has not been initialised.
*/
bool audio_frame2_resampler::resampler_is_set() {
return this->resampler != nullptr;
}
#ifdef HAVE_SOXR
class soxr_resampler : public audio_frame2_resampler::interface {
public:
tuple<bool, audio_frame2> resample(audio_frame2 &a, vector<audio_frame2::channel> &new_channels, int new_sample_rate_num, int new_sample_rate_den);
private:
bool check_reconfigure(uint32_t original_sample_rate, uint32_t new_sample_rate_num, uint32_t new_sample_rate_den, size_t channel_size, int bps);
soxr_t resampler{nullptr};
uint32_t resample_from{0};
uint32_t resample_to_num{0};
uint32_t resample_to_den{1};
size_t resample_ch_count{0};
};
/**
* @brief This function will create (and destroy) a new resampler.
@@ -114,8 +119,20 @@ bool audio_frame2_resampler::resampler_is_set() {
* @return true Successfully created the resampler
* @return false Initialisation of the resampler failed
*/
bool audio_frame2_resampler::create_resampler(uint32_t original_sample_rate, uint32_t new_sample_rate_num, uint32_t new_sample_rate_den, size_t channel_size, int bps) {
#ifdef HAVE_SOXR
bool soxr_resampler::check_reconfigure(uint32_t original_sample_rate, uint32_t new_sample_rate_num, uint32_t new_sample_rate_den, size_t channel_size, int bps) {
if (resampler != nullptr) {
if (original_sample_rate != resample_from
|| new_sample_rate_num != resample_to_num
|| new_sample_rate_den != resample_to_den) {
// Update the resampler numerator and denomintors
resample_from = original_sample_rate;
resample_to_num = new_sample_rate_num;
resample_to_den = new_sample_rate_den;
soxr_set_io_ratio(resampler, ((double)resample_from / ((double)new_sample_rate_num / (double)new_sample_rate_den)), 0);
}
return true;
}
if (this->resampler) {
soxr_delete((soxr_t)this->resampler);
}
@@ -132,11 +149,10 @@ bool audio_frame2_resampler::create_resampler(uint32_t original_sample_rate, uin
io_spec = soxr_io_spec(SOXR_INT32_S, SOXR_INT32_S);
}
else {
LOG(LOG_LEVEL_ERROR) << "[audio_frame2_resampler] Unsupported BPS of: " << bps << "\n";
LOG(LOG_LEVEL_ERROR) << "[audio_frame2_resampler] Unsupported BPS of: " << bps << "\n";
return false;
}
soxr_error_t error;
/* The ratio of the given input rate and output rates must equate to the
* maximum I/O ratio that will be used. A resample rate of 2 to 1 would be excessive,
@@ -157,10 +173,8 @@ bool audio_frame2_resampler::create_resampler(uint32_t original_sample_rate, uin
this->resample_ch_count = channel_size;
LOG(LOG_LEVEL_DEBUG) << "[audio_frame2] Resampler (re)made at " << new_sample_rate_num / new_sample_rate_den << "\n";
return true;
#endif
UNUSED(original_sample_rate), UNUSED(new_sample_rate_num), UNUSED(new_sample_rate_den), UNUSED(channel_size), UNUSED(bps);
return false;
}
#endif
/**
* @brief Creates empty audio_frame2
@@ -396,7 +410,7 @@ void audio_frame2::change_bps(int new_bps)
return;
}
std::vector<channel> new_channels(channels.size());
vector<channel> new_channels(channels.size());
for (size_t i = 0; i < channels.size(); i++) {
size_t new_size = channels[i].len / bps * new_bps;
@@ -412,57 +426,57 @@ void audio_frame2::change_bps(int new_bps)
channels = std::move(new_channels);
}
tuple<bool, audio_frame2> audio_frame2::resample_fake([[maybe_unused]] audio_frame2_resampler & resampler_state, int new_sample_rate_num, int new_sample_rate_den)
tuple<bool, audio_frame2> audio_frame2::resample_fake(audio_frame2_resampler & resampler_state, int new_sample_rate_num, int new_sample_rate_den)
{
#ifdef HAVE_SOXR
std::chrono::high_resolution_clock::time_point funcBegin = std::chrono::high_resolution_clock::now();
if (!resampler_state.resampler_is_set()) {
bool ret = resampler_state.create_resampler(this->sample_rate, new_sample_rate_num, new_sample_rate_den, this->channels.size(), this->bps);
if (!ret) {
return {false, audio_frame2{}};
}
}
if (sample_rate != resampler_state.resample_from
|| new_sample_rate_num != resampler_state.resample_to_num
|| new_sample_rate_den != resampler_state.resample_to_den) {
// Update the resampler numerator and denomintors
resampler_state.resample_to_num = new_sample_rate_num;
resampler_state.resample_to_den = new_sample_rate_den;
soxr_set_io_ratio((soxr_t)resampler_state.resampler, ((double)this->sample_rate / ((double)new_sample_rate_num / (double)new_sample_rate_den)), 0);
}
// Initialise the new channels that the resampler is going to write into
void * * const obuf_ptrs = (void * *) malloc(sizeof(void *) * this->channels.size());
void * * ibuf_ptrs = (void * *) malloc(sizeof(void *) * this->channels.size());
std::vector<channel> new_channels(channels.size());
vector<channel> new_channels(channels.size());
for (size_t i = 0; i < channels.size(); i++) {
// allocate new storage + 10 ms headroom
size_t new_size = (long long) channels[i].len * new_sample_rate_num / sample_rate / new_sample_rate_den
+ new_sample_rate_num * this->bps / 100 / new_sample_rate_den;
new_channels[i] = {unique_ptr<char []>(new char[new_size]), new_size, new_size, {}};
// Setup the buffers
obuf_ptrs[i] = new_channels[i].data.get();
ibuf_ptrs[i] = this->channels[i].data.get();
}
size_t inlen = this->get_data_len(0) / this->bps;
size_t outlen = new_channels[0].len / this->bps;
auto [ret, remainder] = resampler_state.resample(*this, new_channels, new_sample_rate_num, new_sample_rate_den);
if (!ret) {
return {false, audio_frame2{}};
}
channels = move(new_channels);
return {ret, std::move(remainder)};
}
#ifdef HAVE_SOXR
tuple<bool, audio_frame2> soxr_resampler::resample(audio_frame2 &a, vector<audio_frame2::channel> &new_channels, int new_sample_rate_num, int new_sample_rate_den) {
std::chrono::high_resolution_clock::time_point funcBegin = std::chrono::high_resolution_clock::now();
bool ret = check_reconfigure(a.get_sample_rate(), new_sample_rate_num, new_sample_rate_den, a.get_channel_count(), a.get_bps());
if (!ret) {
return {false, audio_frame2{}};
}
// Initialise the new channels that the resampler is going to write into
void * * const obuf_ptrs = (void * *) malloc(sizeof(void *) * a.get_channel_count());
void * * ibuf_ptrs = (void * *) malloc(sizeof(void *) * a.get_channel_count());
for (size_t i = 0; i < new_channels.size(); i++) {
// Setup the buffers
obuf_ptrs[i] = new_channels[i].data.get();
ibuf_ptrs[i] = a.get_data(i);
}
size_t inlen = a.get_data_len(0) / a.get_bps();
size_t outlen = new_channels[0].len / a.get_bps();
size_t odone = 0;
soxr_error_t error;
error = soxr_process((soxr_t)(resampler_state.resampler), ibuf_ptrs, inlen, NULL, obuf_ptrs, outlen, &odone);
error = soxr_process(resampler, ibuf_ptrs, inlen, NULL, obuf_ptrs, outlen, &odone);
if (error) {
LOG(LOG_LEVEL_ERROR) << "[audio_frame2_resampler] resampler failed: " << soxr_strerror(error) << "\n";
return {false, audio_frame2{}};
}
for (unsigned int i = 0; i < new_channels.size(); i++) {
new_channels[i].len = odone * this->bps;
new_channels[i].len = odone * a.get_bps();
}
channels = std::move(new_channels);
free(obuf_ptrs); free(ibuf_ptrs);
std::chrono::high_resolution_clock::time_point funcEnd = std::chrono::high_resolution_clock::now();
@@ -472,9 +486,12 @@ tuple<bool, audio_frame2> audio_frame2::resample_fake([[maybe_unused]] audio_fra
// Remainders aren't as relevant when using SOXR
audio_frame2 remainder = {};
return {true, std::move(remainder)};
#else
UNUSED(new_sample_rate_num), UNUSED(new_sample_rate_den);
return {false, audio_frame2{}};
}
#endif
audio_frame2_resampler::audio_frame2_resampler() {
#ifdef HAVE_SOXR
impl = unique_ptr<audio_frame2_resampler::interface>(new soxr_resampler());
#endif
}

33
src/audio/types.h
View File

@@ -117,22 +117,7 @@ typedef struct
#include <utility>
#include <vector>
class audio_frame2;
class audio_frame2_resampler {
public:
~audio_frame2_resampler();
private:
bool resampler_is_set();
bool create_resampler(uint32_t original_sample_rate, uint32_t new_sample_rate_num, uint32_t new_sample_rate_den, size_t channel_size, int bps);
void *resampler{nullptr}; // type is (SpeexResamplerState *)
int resample_from{0};
int resample_to_num{0};
int resample_to_den{1};
size_t resample_ch_count{0};
friend class audio_frame2;
};
class audio_frame2_resampler;
/**
* More versatile than audio_frame
@@ -203,6 +188,22 @@ private:
std::vector<channel> channels; /* data should be at least 4B aligned */
audio_codec_t codec;
double duration; ///< for compressed formats where this cannot be directly determined from samples/sample_rate
friend class audio_frame2_resampler;
friend class soxr_resampler;
};
class audio_frame2_resampler {
public:
audio_frame2_resampler();
class interface {
public:
virtual std::tuple<bool, audio_frame2> resample(audio_frame2 &a, std::vector<audio_frame2::channel> &out, int new_sample_rate_num, int new_sample_rate_den) = 0;
virtual ~interface() {}
};
std::tuple<bool, audio_frame2> resample(audio_frame2 &a, std::vector<audio_frame2::channel> &out, int new_sample_rate_num, int new_sample_rate_den);
private:
std::unique_ptr<interface> impl;
};
#endif // __cplusplus