axmol/cocos/audio/android/audio_utils/format.c

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[big refactoring] Audio latency fix for Android. Support to preload effects on Android now. (#15875) * Audio latency fix for Android. Support preload effects on Android now. Squashed commits: [b6d80fe] log fix [a0a918e] Fixes assetFd didn't be released while PcmData is returned from cache. [4b956ba] Potential crash fix for PcmAudioPlayer while pause / resume. [398ab8c] Updates LOG_TAG position in AudioEngine-inl.cpp [e3634e7] include stdlib.h for posix_memalign [9004074] fixes setVolume logical error. [c96df46] Don't use another thread for mixing, enqueue is in a seperated thread, therefore doing mixing in another thread will waste more time. [0a4c1a8] Adds setLoop, setVolume, setPostion support for Track [c35fb20] Fixed include. [cdd9d32] Do mixing by ourself. (TO BE POLISHED) [6447025] µ -> u since µ could not be shown on some android devices. [97be0c6] Don't send a silence clip. [c1607ed] Make linter.py happy. [0898b54] Puts enqueue & SetPlayState in PcmAudioPlayer::play to thread pool. [b79fc01] Adds getDuration, getPosition support for PcmAudioPlayer [80fa2ab] minor fix of the code position of resetting state to State::INITIALIZED [d9c62f1] underrun fix for PcmAudioPlayer. [9c2212a] UrlAudioPlayer, playOverMutex should be static, and should be used in update method. [1519d2e] static variables [19da936] _pcmAudioPlayer Null pointer check in AudioPlayerProvider. [e6b0d14] Updates audio performance test. [fc01dd4] Registers foreground & background event in AudioEngine-inl.cpp(android), the callback should invoke `provider`'s pause & resume method. [e00a886] TBD: Pause & resume support for PcmAudioPlayerPool. Since OpenSLES audio resources are expensive and device shared, we should delete all unused PcmAudioPlayers in pool while pause and re-create them while resume. But this commit isn't finished yet, I don't find a better way to register pause&resume event in AudioEngine module. [9e42ea3] Interleave mono audio to stereo audio. PcmAudioPlayerPool only contains PcmAudioPlayers with 2 channels. [3f18d05] Adds a strategy for checking small size of different file formats. [753ff49] Adds performance test for AudioEngine. [09d3045] Releases an extra PcmAudioPlayer for UrlAudioPlayer while allocating PcmAudioPlayer fails. [9dd4477] Using std::move for PcmData move constructor & move assignment. [6ca3bcb] some fixes: 1) new -> new (std::nothrow) 2) break if allocate PcmAudioPlayer fails 3) renames 'initForPlayPcmData' to 'init' 4) PcmAudioPlayer destructor deadlock if 'init' failed [54675b6] include path fix. [a1903ca] More refactorings. [19b9498] Makes linter.py happy. :) [923c530] Fixes: 1) Avoid getFileInfo to be invoked twice 2) A critical bug fix for UrlAudioPlayer and adds detailed comments 3) __clang__ compiler option fix for AudioResamplerSinc.cpp. [5ec4faf] minor fix. [faaa0f3] output a log in the destructor of UrlAudioPlayer. [9c20355] NewAudioEngineTest,TestControll crash fix. [f114464] fixes an unused import. [1dc5dab] Better algorithm for allocating PcmAudioPlayer. [331a213] minor fix. [e54084a] null -> nullptr [f9a0389] Support uncache. [89a364f] Removes unused update, and TODO uncache functionality. [1732bf9] Supports AudioEngineImpl::setFinishCallback for android. [43d1596] UrlAudioPlayer::stop fix. [e2ee941] Test case fix in NewAudioEngineTest/AudioIssue11143Test [5c5ba01] More fixes for making cpp-tests/New Audio Engine Test happy. [8b554a3] Adds log while remove player from map. [ed71322] If original file is larger than 30k bytes, consider it's a large audio file. [fb1845a] Updates project.properties [6f3839f] minor log output fix in AudioEngine-inl.cpp [c68bc6c] Don't resample if the sample rate of the decoded pcm data matchs the device's. [43ca45f] PcmAudioPlayers also need to be removed while they play over, but should not be deleted since their lifecycle is managed by PcmAudioPlayerPool. [f5e63c9] Audio latency fix for Android. Support preload effects on Android now. * Supports to loading audio files asynchronously. * Crash fix for stop audio right after play2d. * Minor fix for logic in AudioMixerController.cpp * Adds missing files (CCThreadPool.h/.cpp). * Minor fix for including. * Minor fix for missing include <functional> in Track.h * update license information in audio.h * Don't use std::future/std::promise anymore since ndk counldn't support it well in armeabi arch. * isSmallFile postion updated, fixes large audio file goto the checking logic of cache. * std::atomic<int> isn't supported by ndk-r10e while compiling with `armeabi` arch, using a int with a mutex instead. * fixes __isnanf & posix_memalign doesn't exist on low api (<=16) devices. * namespace updated: cocos2d -> cocos2d::experimental * Removes commented code in AudioMixerController.h/.cpp * Removes unused code again, and fixes a memory leak of `Track` instance. * Oops, namespace changed. * Only outputs log in debug mode. * Uses ALOGV for outputing logs in AudioEngine-inl.cpp * const PcmData& -> PcmData for Track * Fixes a protential crash in NewAudioEngineTest * Adds `COCOS` prefix in header #ifndef COCOS_BALABALA #define COCOS_BALABALA * Uses _ prefix for cocos code style instead of `m` prefix. * Deletes AudioResamplerSinc related files. * Bug fix from @minggo's reply on github. * Don't need to invoke pause after in UrlAudioPlayer::prepare. * Updates ThreadPool class, uses enum class and adds const keyword.
2016-07-18 10:22:40 +08:00
/*
* Copyright (C) 2014 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
/* #define LOG_NDEBUG 0 */
#define LOG_TAG "audio_utils_format"
#include "audio/android/cutils/log.h"
#include "audio/android/audio_utils/include/audio_utils/primitives.h"
#include "audio/android/audio_utils/include/audio_utils/format.h"
#include "audio/android/audio.h"
void memcpy_by_audio_format(void *dst, audio_format_t dst_format,
const void *src, audio_format_t src_format, size_t count)
{
/* default cases for error falls through to fatal log below. */
if (dst_format == src_format) {
switch (dst_format) {
case AUDIO_FORMAT_PCM_16_BIT:
case AUDIO_FORMAT_PCM_FLOAT:
case AUDIO_FORMAT_PCM_8_BIT:
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
case AUDIO_FORMAT_PCM_32_BIT:
case AUDIO_FORMAT_PCM_8_24_BIT:
memcpy(dst, src, count * audio_bytes_per_sample(dst_format));
return;
default:
break;
}
}
switch (dst_format) {
case AUDIO_FORMAT_PCM_16_BIT:
switch (src_format) {
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_i16_from_float((int16_t*)dst, (float*)src, count);
return;
case AUDIO_FORMAT_PCM_8_BIT:
memcpy_to_i16_from_u8((int16_t*)dst, (uint8_t*)src, count);
return;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
memcpy_to_i16_from_p24((int16_t*)dst, (uint8_t*)src, count);
return;
case AUDIO_FORMAT_PCM_32_BIT:
memcpy_to_i16_from_i32((int16_t*)dst, (int32_t*)src, count);
return;
case AUDIO_FORMAT_PCM_8_24_BIT:
memcpy_to_i16_from_q8_23((int16_t*)dst, (int32_t*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_FLOAT:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_float_from_i16((float*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_8_BIT:
memcpy_to_float_from_u8((float*)dst, (uint8_t*)src, count);
return;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
memcpy_to_float_from_p24((float*)dst, (uint8_t*)src, count);
return;
case AUDIO_FORMAT_PCM_32_BIT:
memcpy_to_float_from_i32((float*)dst, (int32_t*)src, count);
return;
case AUDIO_FORMAT_PCM_8_24_BIT:
memcpy_to_float_from_q8_23((float*)dst, (int32_t*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_8_BIT:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_u8_from_i16((uint8_t*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_u8_from_float((uint8_t*)dst, (float*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_p24_from_i16((uint8_t*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_p24_from_float((uint8_t*)dst, (float*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_32_BIT:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_i32_from_i16((int32_t*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_i32_from_float((int32_t*)dst, (float*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_8_24_BIT:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_q8_23_from_i16((int32_t*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_q8_23_from_float_with_clamp((int32_t*)dst, (float*)src, count);
return;
case AUDIO_FORMAT_PCM_24_BIT_PACKED: {
memcpy_to_q8_23_from_p24((int32_t *)dst, (uint8_t *)src, count);
return;
}
default:
break;
}
break;
default:
break;
}
LOG_ALWAYS_FATAL("invalid src format %#x for dst format %#x",
src_format, dst_format);
}
size_t memcpy_by_index_array_initialization_from_channel_mask(int8_t *idxary, size_t arysize,
audio_channel_mask_t dst_channel_mask, audio_channel_mask_t src_channel_mask)
{
const audio_channel_representation_t src_representation =
audio_channel_mask_get_representation(src_channel_mask);
const audio_channel_representation_t dst_representation =
audio_channel_mask_get_representation(dst_channel_mask);
const uint32_t src_bits = audio_channel_mask_get_bits(src_channel_mask);
const uint32_t dst_bits = audio_channel_mask_get_bits(dst_channel_mask);
switch (src_representation) {
case AUDIO_CHANNEL_REPRESENTATION_POSITION:
switch (dst_representation) {
case AUDIO_CHANNEL_REPRESENTATION_POSITION:
return memcpy_by_index_array_initialization(idxary, arysize,
dst_bits, src_bits);
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
return memcpy_by_index_array_initialization_dst_index(idxary, arysize,
dst_bits, src_bits);
default:
return 0;
}
break;
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
switch (dst_representation) {
case AUDIO_CHANNEL_REPRESENTATION_POSITION:
return memcpy_by_index_array_initialization_src_index(idxary, arysize,
dst_bits, src_bits);
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
return memcpy_by_index_array_initialization(idxary, arysize,
dst_bits, src_bits);
default:
return 0;
}
break;
default:
return 0;
}
}