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Usage of resampler for sounds

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Commit fixes #458.

It also loosens the connection
between decoder and player (openAL).

It allows to easier replace
player if there's need.
This commit is contained in:
Filip Gawin 2018-07-03 21:06:49 +02:00
parent 2f8ae7fb0b
commit 7fafd3728e
2 changed files with 115 additions and 23 deletions
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2
cmake/modules/FindFFmpeg.cmake
View File

@ -23,6 +23,7 @@ if (PKG_CONFIG_FOUND)
pkg_check_modules(_FFMPEG_AVCODEC libavcodec QUIET)
pkg_check_modules(_FFMPEG_AVFORMAT libavformat QUIET)
pkg_check_modules(_FFMPEG_AVUTIL libavutil QUIET)
pkg_check_modules(_FFMPEG_SWRESAMPLE libswresample QUIET)
endif (PKG_CONFIG_FOUND)
find_path(FFMPEG_AVCODEC_INCLUDE_DIR
@ -61,6 +62,7 @@ set(FFMPEG_LIBRARIES
${FFMPEG_LIBAVCODEC}
${FFMPEG_LIBAVFORMAT}
${FFMPEG_LIBAVUTIL}
${FFMPEG_SWRESAMPLE}
)
if(CMAKE_SYSTEM_NAME STREQUAL "Windows")

136
rwengine/src/audio/SoundManager.cpp
View File

@ -7,6 +7,8 @@ extern "C" {
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/avutil.h>
#include <libavutil/opt.h>
#include <libswresample/swresample.h>
}
//ab
#include <rw/debug.hpp>
@ -21,6 +23,9 @@ extern "C" {
#define avio_context_free av_freep
#endif
constexpr int kNumOutputChannels = 2;
constexpr AVSampleFormat kOutputFMT = AV_SAMPLE_FMT_S16;
SoundManager::SoundManager() {
initializeOpenAL();
initializeAVCodec();
@ -58,8 +63,6 @@ bool SoundManager::initializeOpenAL() {
}
bool SoundManager::initializeAVCodec() {
av_register_all();
#if RW_DEBUG && RW_VERBOSE_DEBUG_MESSAGES
av_log_set_level(AV_LOG_WARNING);
#else
@ -93,8 +96,7 @@ void SoundManager::SoundSource::loadFromFile(const rwfs::path& filePath) {
}
// Find the audio stream
AVCodec* codec = nullptr;
int streamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
int streamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, nullptr, 0);
if (streamIndex < 0) {
av_frame_free(&frame);
avformat_close_input(&formatContext);
@ -103,6 +105,9 @@ void SoundManager::SoundSource::loadFromFile(const rwfs::path& filePath) {
}
AVStream* audioStream = formatContext->streams[streamIndex];
AVCodec* codec = avcodec_find_decoder(audioStream->codecpar->codec_id);
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(57,5,0)
AVCodecContext* codecContext = audioStream->codec;
codecContext->codec = codec;
@ -113,33 +118,47 @@ void SoundManager::SoundSource::loadFromFile(const rwfs::path& filePath) {
RW_ERROR("Couldn't open the audio codec context");
return;
}
#else
// Initialize codec context for the decoder.
AVCodecContext* codecContext = avcodec_alloc_context3(codec);
if (!codecContext) {
av_frame_free(&frame);
avformat_close_input(&formatContext);
RW_ERROR("Couldn't allocate a decoding context.");
return;
}
// Fill the codecCtx with the parameters of the codec used in the read file.
if (avcodec_parameters_to_context(codecContext, audioStream->codecpar) != 0) {
avcodec_close(codecContext);
avcodec_free_context(&codecContext);
avformat_close_input(&formatContext);
RW_ERROR("Couldn't find parametrs for context");
}
// Initialize the decoder.
if (avcodec_open2(codecContext, codec, nullptr) != 0) {
avcodec_close(codecContext);
avcodec_free_context(&codecContext);
avformat_close_input(&formatContext);
RW_ERROR("Couldn't open the audio codec context");
return;
}
#endif
// Expose audio metadata
channels = codecContext->channels;
sampleRate = codecContext->sample_rate;
channels = kNumOutputChannels;
sampleRate = static_cast<size_t>(codecContext->sample_rate);
// OpenAL only supports mono or stereo, so error on more than 2 channels
if(channels > 2) {
RW_ERROR("Audio has more than two channels");
av_frame_free(&frame);
avcodec_close(codecContext);
avformat_close_input(&formatContext);
return;
}
// Right now we only support signed 16-bit audio
if(codecContext->sample_fmt != AV_SAMPLE_FMT_S16P) {
RW_ERROR("Audio data isn't in a planar signed 16-bit format");
av_frame_free(&frame);
avcodec_close(codecContext);
avformat_close_input(&formatContext);
return;
}
// prepare resampler
SwrContext* swr = nullptr;
// Start reading audio packets
AVPacket readingPacket;
av_init_packet(&readingPacket);
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(57,37,100)
while (av_read_frame(formatContext, &readingPacket) == 0) {
if (readingPacket.stream_index == audioStream->index) {
AVPacket decodingPacket = readingPacket;
@ -171,10 +190,81 @@ void SoundManager::SoundSource::loadFromFile(const rwfs::path& filePath) {
}
av_free_packet(&readingPacket);
}
#else
AVFrame* resampled = nullptr;
while (av_read_frame(formatContext, &readingPacket) == 0) {
if (readingPacket.stream_index == audioStream->index) {
int sendPacket = avcodec_send_packet(codecContext, &readingPacket);
av_packet_unref(&readingPacket);
int receiveFrame = 0;
while ((receiveFrame = avcodec_receive_frame(codecContext, frame)) == 0) {
if(!swr) {
if(frame->channels == 1 || frame->channel_layout == 0) {
frame->channel_layout = av_get_default_channel_layout(1);
}
swr = swr_alloc_set_opts(nullptr,
AV_CH_LAYOUT_STEREO, // output channel layout
kOutputFMT, // output format
frame->sample_rate, // output sample rate
frame->channel_layout, // input channel layout
static_cast<AVSampleFormat>(frame->format), // input format
frame->sample_rate, // input sample rate
0,
nullptr);
if (!swr) {
RW_ERROR("Resampler has not been successfully allocated.");
return;
}
swr_init(swr);
if (!swr_is_initialized(swr)) {
RW_ERROR("Resampler has not been properly initialized.");
return;
}
}
// Decode audio packet
if (receiveFrame == 0 && sendPacket == 0) {
// Write samples to audio buffer
resampled = av_frame_alloc();
resampled->channel_layout = AV_CH_LAYOUT_STEREO;
resampled->sample_rate = frame->sample_rate;
resampled->format = kOutputFMT;
resampled->channels = kNumOutputChannels;
swr_config_frame(swr, resampled, frame);
if (swr_convert_frame(swr, resampled, frame) < 0) {
RW_ERROR("Error resampling "<< filename << '\n');
}
for(size_t i = 0; i < static_cast<size_t>(resampled->nb_samples) * channels; i++) {
data.push_back(reinterpret_cast<int16_t *>(resampled->data[0])[i]);
}
av_frame_unref(resampled);
}
}
}
}
#endif
// Cleanup
/// Free all data used by the frame.
av_frame_free(&frame);
/// Free resampler
swr_free(&swr);
/// Close the context and free all data associated to it, but not the context itself.
avcodec_close(codecContext);
/// Free the context itself.
avcodec_free_context(&codecContext);
/// We are done here. Close the input.
avformat_close_input(&formatContext);
}