+/*\r
+* copyright (c) 2010 Sveriges Television AB <info@casparcg.com>\r
+*\r
+* This file is part of CasparCG.\r
+*\r
+* CasparCG is free software: you can redistribute it and/or modify\r
+* it under the terms of the GNU General Public License as published by\r
+* the Free Software Foundation, either version 3 of the License, or\r
+* (at your option) any later version.\r
+*\r
+* CasparCG is distributed in the hope that it will be useful,\r
+* but WITHOUT ANY WARRANTY; without even the implied warranty of\r
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the\r
+* GNU General Public License for more details.\r
+\r
+* You should have received a copy of the GNU General Public License\r
+* along with CasparCG. If not, see <http://www.gnu.org/licenses/>.\r
+*\r
+*/\r
#include "../../stdafx.h"\r
\r
#include "audio_mixer.h"\r
\r
+#include <core/mixer/write_frame.h>\r
+#include <core/producer/frame/frame_transform.h>\r
+\r
+#include <tbb/parallel_for.h>\r
+\r
+#include <safeint.h>\r
+\r
+#include <stack>\r
+#include <deque>\r
+\r
namespace caspar { namespace core {\r
- \r
-audio_transform& audio_transform::operator*=(const audio_transform &other) \r
-{\r
- gain *= other.gain;\r
- return *this;\r
-}\r
\r
-const audio_transform audio_transform::operator*(const audio_transform &other) const \r
+struct audio_item\r
{\r
- return audio_transform(*this) *= other;\r
-}\r
-\r
+ const void* tag;\r
+ frame_transform transform;\r
+ audio_buffer audio_data;\r
+};\r
+ \r
struct audio_mixer::implementation\r
{\r
- std::vector<short> audio_data_;\r
- std::stack<audio_transform> transform_stack_;\r
+ std::stack<core::frame_transform> transform_stack_;\r
+ std::map<const void*, core::frame_transform> prev_frame_transforms_;\r
+ const core::video_format_desc format_desc_;\r
+ std::vector<audio_item> items;\r
\r
public:\r
- implementation()\r
+ implementation(const core::video_format_desc& format_desc)\r
+ : format_desc_(format_desc)\r
{\r
- transform_stack_.push(audio_transform());\r
+ transform_stack_.push(core::frame_transform());\r
}\r
-\r
- void begin(const audio_transform& transform)\r
+ \r
+ void begin(core::basic_frame& frame)\r
{\r
- transform_stack_.push(transform_stack_.top()*transform);\r
+ transform_stack_.push(transform_stack_.top()*frame.get_frame_transform());\r
}\r
\r
- void process(const std::vector<short>& audio_data)\r
- { \r
- if(audio_data_.empty())\r
- audio_data_.resize(audio_data.size(), 0);\r
-\r
- tbb::parallel_for\r
- (\r
- tbb::blocked_range<size_t>(0, audio_data.size()),\r
- [&](const tbb::blocked_range<size_t>& r)\r
- {\r
- for(size_t n = r.begin(); n < r.end(); ++n)\r
- {\r
- int sample = static_cast<int>(audio_data[n]);\r
- sample = (static_cast<int>(transform_stack_.top().gain*8192.0)*sample)/8192;\r
- audio_data_[n] = static_cast<short>((static_cast<int>(audio_data_[n]) + sample) & 0xFFFF);\r
- }\r
- }\r
- );\r
+ void visit(core::write_frame& frame)\r
+ {\r
+ // We only care about the last field.\r
+ if(format_desc_.field_mode == field_mode::upper && transform_stack_.top().field_mode == field_mode::upper)\r
+ return;\r
+\r
+ if(format_desc_.field_mode == field_mode::lower && transform_stack_.top().field_mode == field_mode::lower)\r
+ return;\r
+\r
+ // Skip empty audio.\r
+ if(transform_stack_.top().volume < 0.002 || frame.audio_data().empty())\r
+ return;\r
+\r
+ audio_item item;\r
+ item.tag = frame.tag();\r
+ item.transform = transform_stack_.top();\r
+ item.audio_data = std::move(frame.audio_data());\r
+\r
+ items.push_back(item); \r
}\r
\r
+ void begin(const core::frame_transform& transform)\r
+ {\r
+ transform_stack_.push(transform_stack_.top()*transform);\r
+ }\r
+ \r
void end()\r
{\r
transform_stack_.pop();\r
}\r
+ \r
+ audio_buffer mix()\r
+ { \r
+ auto intermediate = std::vector<float, tbb::cache_aligned_allocator<float>>(format_desc_.audio_samples_per_frame+128, 0.0f);\r
\r
- std::vector<short> begin_pass()\r
- {\r
- return std::move(audio_data_);\r
+ std::map<const void*, core::frame_transform> next_frame_transforms;\r
+\r
+ BOOST_FOREACH(auto& item, items)\r
+ { \r
+ const auto next = item.transform;\r
+ auto prev = next;\r
+\r
+ const auto it = prev_frame_transforms_.find(item.tag);\r
+ if(it != prev_frame_transforms_.end())\r
+ prev = it->second;\r
+ \r
+ next_frame_transforms[item.tag] = next; // Store all active tags, inactive tags will be removed at the end.\r
+ \r
+ if(next.volume < 0.001 && prev.volume < 0.001)\r
+ continue;\r
+ \r
+ if(static_cast<size_t>(item.audio_data.size()) != format_desc_.audio_samples_per_frame)\r
+ continue;\r
+\r
+ CASPAR_ASSERT(format_desc_.audio_channels == 2);\r
+ CASPAR_ASSERT(format_desc_.audio_samples_per_frame % 4 == 0);\r
+ \r
+ const float prev_volume = static_cast<float>(prev.volume);\r
+ const float next_volume = static_cast<float>(next.volume);\r
+ const float delta = 1.0f/static_cast<float>(format_desc_.audio_samples_per_frame/2);\r
+ \r
+ tbb::parallel_for\r
+ (\r
+ tbb::blocked_range<size_t>(0, format_desc_.audio_samples_per_frame/4),\r
+ [&](const tbb::blocked_range<size_t>& r)\r
+ {\r
+ for(size_t n = r.begin(); n < r.end(); ++n)\r
+ {\r
+ const float alpha0 = (n*2) * delta;\r
+ const float volume0 = prev_volume * (1.0f - alpha0) + next_volume * alpha0;\r
+ const float volume1 = prev_volume * (1.0f - alpha0 + delta) + next_volume * (alpha0 + delta);\r
+\r
+ auto sample_epi32 = _mm_load_si128(reinterpret_cast<__m128i*>(&item.audio_data[n*4]));\r
+ auto res_sample_ps = _mm_load_ps(&intermediate[n*4]);\r
+\r
+ auto sample_ps = _mm_cvtepi32_ps(sample_epi32); \r
+ sample_ps = _mm_mul_ps(sample_ps, _mm_setr_ps(volume1, volume1, volume0, volume0)); \r
+ res_sample_ps = _mm_add_ps(sample_ps, res_sample_ps); \r
+\r
+ _mm_store_ps(&intermediate[n*4], res_sample_ps);\r
+ }\r
+ }\r
+ );\r
+ }\r
+ \r
+ auto result = audio_buffer(format_desc_.audio_samples_per_frame+128, 0); \r
+\r
+ auto intermediate_128 = reinterpret_cast<__m128i*>(intermediate.data());\r
+ auto result_128 = reinterpret_cast<__m128i*>(result.data());\r
+ for(size_t n = 0; n < format_desc_.audio_samples_per_frame/32; ++n)\r
+ {\r
+ auto xmm0 = _mm_load_ps(reinterpret_cast<float*>(intermediate_128++));\r
+ auto xmm1 = _mm_load_ps(reinterpret_cast<float*>(intermediate_128++));\r
+ auto xmm2 = _mm_load_ps(reinterpret_cast<float*>(intermediate_128++));\r
+ auto xmm3 = _mm_load_ps(reinterpret_cast<float*>(intermediate_128++));\r
+ auto xmm4 = _mm_load_ps(reinterpret_cast<float*>(intermediate_128++));\r
+ auto xmm5 = _mm_load_ps(reinterpret_cast<float*>(intermediate_128++));\r
+ auto xmm6 = _mm_load_ps(reinterpret_cast<float*>(intermediate_128++));\r
+ auto xmm7 = _mm_load_ps(reinterpret_cast<float*>(intermediate_128++));\r
+ \r
+ _mm_stream_si128(result_128++, _mm_cvtps_epi32(xmm0));\r
+ _mm_stream_si128(result_128++, _mm_cvtps_epi32(xmm1));\r
+ _mm_stream_si128(result_128++, _mm_cvtps_epi32(xmm2));\r
+ _mm_stream_si128(result_128++, _mm_cvtps_epi32(xmm3));\r
+ _mm_stream_si128(result_128++, _mm_cvtps_epi32(xmm4));\r
+ _mm_stream_si128(result_128++, _mm_cvtps_epi32(xmm5));\r
+ _mm_stream_si128(result_128++, _mm_cvtps_epi32(xmm6));\r
+ _mm_stream_si128(result_128++, _mm_cvtps_epi32(xmm7));\r
+ }\r
+\r
+ items.clear();\r
+ prev_frame_transforms_ = std::move(next_frame_transforms); \r
+\r
+ result.resize(format_desc_.audio_samples_per_frame);\r
+ return std::move(result);\r
}\r
};\r
\r
-audio_mixer::audio_mixer() : impl_(new implementation()){}\r
-void audio_mixer::begin(const audio_transform& transform){impl_->begin(transform);}\r
-void audio_mixer::process(const std::vector<short>& audio_data){impl_->process(audio_data);}\r
+audio_mixer::audio_mixer(const core::video_format_desc& format_desc) : impl_(new implementation(format_desc)){}\r
+void audio_mixer::begin(core::basic_frame& frame){impl_->begin(frame);}\r
+void audio_mixer::visit(core::write_frame& frame){impl_->visit(frame);}\r
void audio_mixer::end(){impl_->end();}\r
-std::vector<short> audio_mixer::begin_pass(){return impl_->begin_pass();} \r
-void audio_mixer::end_pass(){}\r
+audio_buffer audio_mixer::mix(){return impl_->mix();}\r
+audio_mixer& audio_mixer::operator=(audio_mixer&& other)\r
+{\r
+ impl_ = std::move(other.impl_);\r
+ return *this;\r
+}\r
\r
}}
\ No newline at end of file
projects / casparcg / blobdiff