mirror of https://github.com/mgba-emu/mgba.git
FFmpeg resampling
This commit is contained in:
Jeffrey Pfau
parent
281f190ae6
commit
e9b26dda08
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@ -81,7 +81,7 @@ add_definitions(-DBINARY_NAME="${BINARY_NAME}" -DPROJECT_NAME="${PROJECT_NAME}"
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# Feature dependencies
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find_feature(USE_CLI_DEBUGGER "libedit")
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find_feature(USE_FFMPEG "libavcodec;libavformat;libavutil")
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find_feature(USE_FFMPEG "libavcodec;libavformat;libavresample;libavutil")
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find_feature(USE_PNG "ZLIB;PNG")
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find_feature(USE_LIBZIP "libzip")
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@ -134,10 +134,10 @@ source_group("ARM debugger" FILES ${DEBUGGER_SRC})
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if(USE_FFMPEG)
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add_definitions(-DUSE_FFMPEG)
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include_directories(AFTER ${LIBAVCODEC_INCLUDE_DIRS} ${LIBAVFORMAT_INCLUDE_DIRS} ${LIBAVUTIL_INCLUDE_DIRS})
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link_directories(${LIBAVCODEC_LIBRARY_DIRS} ${LIBAVFORMAT_LIBRARY_DIRS} ${LIBAVUTIL_LIBRARY_DIRS})
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include_directories(AFTER ${LIBAVCODEC_INCLUDE_DIRS} ${LIBAVFORMAT_INCLUDE_DIRS} ${LIBAVRESAMPLE_INCLUDE_DIRS} ${LIBAVUTIL_INCLUDE_DIRS})
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link_directories(${LIBAVCODEC_LIBRARY_DIRS} ${LIBAVFORMAT_LIBRARY_DIRS} ${LIBAVRESAMPLE_LIBRARY_DIRS} ${LIBAVUTIL_LIBRARY_DIRS})
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list(APPEND UTIL_SRC "${CMAKE_SOURCE_DIR}/src/platform/ffmpeg/ffmpeg-encoder.c")
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list(APPEND DEPENDENCY_LIB ${LIBAVCODEC_LIBRARIES} ${LIBAVFORMAT_LIBRARIES} ${LIBAVUTIL_LIBRARIES})
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list(APPEND DEPENDENCY_LIB ${LIBAVCODEC_LIBRARIES} ${LIBAVFORMAT_LIBRARIES} ${LIBAVRESAMPLE_LIBRARIES} ${LIBAVUTIL_LIBRARIES})
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endif()
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if(USE_PNG)
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@ -3,10 +3,15 @@
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#include "gba-video.h"
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#include <libavutil/imgutils.h>
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#include <libavutil/opt.h>
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static void _ffmpegPostVideoFrame(struct GBAAVStream*, struct GBAVideoRenderer* renderer);
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static void _ffmpegPostAudioFrame(struct GBAAVStream*, int32_t left, int32_t right);
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enum {
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PREFERRED_SAMPLE_RATE = 0x8000
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};
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void FFmpegEncoderInit(struct FFmpegEncoder* encoder) {
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av_register_all();
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@ -26,29 +31,70 @@ void FFmpegEncoderInit(struct FFmpegEncoder* encoder) {
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}
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bool FFmpegEncoderSetAudio(struct FFmpegEncoder* encoder, const char* acodec, unsigned abr) {
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if (!avcodec_find_encoder_by_name(acodec)) {
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AVCodec* codec = avcodec_find_encoder_by_name(acodec);
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if (!codec) {
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return false;
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}
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if (!codec->sample_fmts) {
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return false;
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}
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size_t i;
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encoder->sampleFormat = AV_SAMPLE_FMT_NONE;
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for (i = 0; codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; ++i) {
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if (codec->sample_fmts[i] == AV_SAMPLE_FMT_S16 || codec->sample_fmts[i] == AV_SAMPLE_FMT_S16P) {
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encoder->sampleFormat = codec->sample_fmts[i];
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break;
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}
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}
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if (encoder->sampleFormat == AV_SAMPLE_FMT_NONE) {
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return false;
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}
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encoder->sampleRate = PREFERRED_SAMPLE_RATE;
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if (codec->supported_samplerates) {
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for (i = 0; codec->supported_samplerates[i]; ++i) {
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if (codec->supported_samplerates[i] < PREFERRED_SAMPLE_RATE) {
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continue;
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}
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if (encoder->sampleRate == PREFERRED_SAMPLE_RATE || encoder->sampleRate > codec->supported_samplerates[i]) {
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encoder->sampleRate = codec->supported_samplerates[i];
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}
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}
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} else if (codec->id == AV_CODEC_ID_AAC) {
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// HACK: AAC doesn't support 32768Hz (it rounds to 32000), but libfaac doesn't tell us that
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encoder->sampleRate = 44100;
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}
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encoder->audioCodec = acodec;
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encoder->audioBitrate = abr;
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return true;
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}
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bool FFmpegEncoderSetVideo(struct FFmpegEncoder* encoder, const char* vcodec, unsigned vbr) {
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static struct {
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enum AVPixelFormat format;
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int priority;
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} priorities[] = {
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{ AV_PIX_FMT_RGB24, 0 },
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{ AV_PIX_FMT_BGR0, 1 },
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{ AV_PIX_FMT_YUV422P, 2 },
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{ AV_PIX_FMT_YUV444P, 3 },
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{ AV_PIX_FMT_YUV420P, 4 }
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};
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AVCodec* codec = avcodec_find_encoder_by_name(vcodec);
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if (!codec) {
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return false;
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}
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size_t i;
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size_t j;
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int priority = INT_MAX;
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encoder->pixFormat = AV_PIX_FMT_NONE;
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for (i = 0; codec->pix_fmts[i] != AV_PIX_FMT_NONE; ++i) {
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if (codec->pix_fmts[i] == AV_PIX_FMT_RGB24) {
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encoder->pixFormat = AV_PIX_FMT_RGB24;
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break;
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}
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if (codec->pix_fmts[i] == AV_PIX_FMT_BGR0) {
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encoder->pixFormat = AV_PIX_FMT_BGR0;
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for (j = 0; j < sizeof(priorities) / sizeof(*priorities); ++j) {
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if (codec->pix_fmts[i] == priorities[j].format && priority > priorities[j].priority) {
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priority = priorities[j].priority;
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encoder->pixFormat = codec->pix_fmts[i];
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}
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}
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}
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if (encoder->pixFormat == AV_PIX_FMT_NONE) {
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@ -98,18 +144,34 @@ bool FFmpegEncoderOpen(struct FFmpegEncoder* encoder, const char* outfile) {
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encoder->audioStream = avformat_new_stream(encoder->context, acodec);
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encoder->audio = encoder->audioStream->codec;
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encoder->audio->bit_rate = encoder->audioBitrate;
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encoder->audio->sample_rate = 0x8000;
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encoder->audio->channels = 2;
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encoder->audio->channel_layout = AV_CH_LAYOUT_STEREO;
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encoder->audio->sample_fmt = AV_SAMPLE_FMT_S16;
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encoder->audio->sample_rate = encoder->sampleRate;
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encoder->audio->sample_fmt = encoder->sampleFormat;
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avcodec_open2(encoder->audio, acodec, 0);
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encoder->audioFrame = av_frame_alloc();
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encoder->audioFrame->nb_samples = encoder->audio->frame_size;
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encoder->audioFrame->format = encoder->audio->sample_fmt;
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encoder->audioFrame->pts = 0;
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encoder->audioBufferSize = av_samples_get_buffer_size(0, encoder->audio->channels, encoder->audio->frame_size, encoder->audio->sample_fmt, 0);
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encoder->audioBuffer = av_malloc(encoder->audioBufferSize);
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avcodec_fill_audio_frame(encoder->audioFrame, encoder->audio->channels, encoder->audio->sample_fmt, (const uint8_t*) encoder->audioBuffer, encoder->audioBufferSize, 0);
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if (encoder->sampleRate != PREFERRED_SAMPLE_RATE) {
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encoder->resampleContext = avresample_alloc_context();
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av_opt_set_int(encoder->resampleContext, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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av_opt_set_int(encoder->resampleContext, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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av_opt_set_int(encoder->resampleContext, "in_sample_rate", PREFERRED_SAMPLE_RATE, 0);
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av_opt_set_int(encoder->resampleContext, "out_sample_rate", encoder->sampleRate, 0);
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av_opt_set_int(encoder->resampleContext, "in_sample_fmt", encoder->sampleFormat, 0);
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av_opt_set_int(encoder->resampleContext, "out_sample_fmt", encoder->sampleFormat, 0);
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avresample_open(encoder->resampleContext);
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encoder->audioBufferSize = (encoder->audioFrame->nb_samples * PREFERRED_SAMPLE_RATE / encoder->sampleRate) * 4;
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encoder->audioBuffer = av_malloc(encoder->audioBufferSize);
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} else {
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encoder->resampleContext = 0;
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encoder->audioBufferSize = 0;
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encoder->audioBuffer = 0;
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}
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encoder->postaudioBufferSize = av_samples_get_buffer_size(0, encoder->audio->channels, encoder->audio->frame_size, encoder->audio->sample_fmt, 0);
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encoder->postaudioBuffer = av_malloc(encoder->postaudioBufferSize);
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avcodec_fill_audio_frame(encoder->audioFrame, encoder->audio->channels, encoder->audio->sample_fmt, (const uint8_t*) encoder->postaudioBuffer, encoder->postaudioBufferSize, 0);
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encoder->videoStream = avformat_new_stream(encoder->context, vcodec);
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encoder->video = encoder->videoStream->codec;
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@ -146,14 +208,26 @@ void FFmpegEncoderClose(struct FFmpegEncoder* encoder) {
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av_write_trailer(encoder->context);
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avio_close(encoder->context->pb);
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av_free(encoder->audioBuffer);
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av_free(encoder->postaudioBuffer);
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if (encoder->audioBuffer) {
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av_free(encoder->audioBuffer);
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}
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av_frame_free(&encoder->audioFrame);
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avcodec_close(encoder->audio);
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av_frame_free(&encoder->videoFrame);
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avcodec_close(encoder->video);
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if (encoder->resampleContext) {
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avresample_close(encoder->resampleContext);
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}
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avformat_free_context(encoder->context);
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encoder->context = 0;
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encoder->currentAudioSample = 0;
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encoder->currentAudioFrame = 0;
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encoder->currentVideoFrame = 0;
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}
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void _ffmpegPostAudioFrame(struct GBAAVStream* stream, int32_t left, int32_t right) {
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@ -163,16 +237,56 @@ void _ffmpegPostAudioFrame(struct GBAAVStream* stream, int32_t left, int32_t rig
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}
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av_frame_make_writable(encoder->audioFrame);
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encoder->audioBuffer[encoder->currentAudioSample * 2] = left;
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encoder->audioBuffer[encoder->currentAudioSample * 2 + 1] = right;
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encoder->audioFrame->pts = av_rescale_q(encoder->currentAudioFrame, encoder->audio->time_base, encoder->audioStream->time_base);
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uint16_t* buffers[2];
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int stride;
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bool planar = av_sample_fmt_is_planar(encoder->audio->sample_fmt);
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if (encoder->resampleContext) {
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buffers[0] = (uint16_t*) encoder->audioBuffer;
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if (planar) {
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stride = 1;
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buffers[1] = &buffers[0][encoder->audioBufferSize / 4];
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} else {
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stride = 2;
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buffers[1] = &buffers[0][1];
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}
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} else {
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buffers[0] = (uint16_t*) encoder->postaudioBuffer;
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if (planar) {
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stride = 1;
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buffers[1] = &buffers[0][encoder->postaudioBufferSize / 4];
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} else {
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stride = 2;
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buffers[1] = &buffers[0][1];
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}
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}
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buffers[0][encoder->currentAudioSample * stride] = left;
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buffers[1][encoder->currentAudioSample * stride] = right;
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++encoder->currentAudioFrame;
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++encoder->currentAudioSample;
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if ((encoder->currentAudioSample * 4) < encoder->audioBufferSize) {
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return;
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if (encoder->resampleContext) {
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if ((encoder->currentAudioSample * 4) < encoder->audioBufferSize) {
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return;
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}
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encoder->currentAudioSample = 0;
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avresample_convert(encoder->resampleContext,
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0, 0, encoder->postaudioBufferSize / 4,
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(uint8_t**) buffers, 0, encoder->audioBufferSize / 4);
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if ((ssize_t) avresample_available(encoder->resampleContext) < (ssize_t) encoder->postaudioBufferSize / 4) {
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return;
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}
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avresample_read(encoder->resampleContext, encoder->audioFrame->data, encoder->postaudioBufferSize / 4);
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} else {
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if ((encoder->currentAudioSample * 4) < encoder->postaudioBufferSize) {
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return;
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}
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encoder->currentAudioSample = 0;
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}
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encoder->currentAudioSample = 0;
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AVRational timeBase = { 1, PREFERRED_SAMPLE_RATE };
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encoder->audioFrame->pts = av_rescale_q(encoder->currentAudioFrame, timeBase, encoder->audioStream->time_base);
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AVPacket packet;
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av_init_packet(&packet);
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@ -5,6 +5,7 @@
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libavresample/avresample.h>
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struct FFmpegEncoder {
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struct GBAAVStream d;
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@ -19,11 +20,16 @@ struct FFmpegEncoder {
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const char* containerFormat;
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AVCodecContext* audio;
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enum AVSampleFormat sampleFormat;
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int sampleRate;
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uint16_t* audioBuffer;
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size_t audioBufferSize;
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uint16_t* postaudioBuffer;
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size_t postaudioBufferSize;
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AVFrame* audioFrame;
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size_t currentAudioSample;
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int64_t currentAudioFrame;
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AVAudioResampleContext* resampleContext;
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AVStream* audioStream;
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AVCodecContext* video;
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