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If your program is a subroutine library, you +may consider it more useful to permit linking proprietary applications with +the library. If this is what you want to do, use the GNU Lesser General +Public License instead of this License. But first, please read +. diff --git a/Makefile b/Makefile index 7e6486d..3d721aa 100755 --- a/Makefile +++ b/Makefile @@ -22,7 +22,12 @@ HEADERS = plugin.h audio_encoder.h SRCS = plugin.cpp audio_encoder.cpp wrapper/host_api.cpp OBJS = $(SRCS:%.cpp=$(OBJDIR)/%.o) -CFLAGS += -I$(BASEDIR)/wrapper -I/usr/include/c++/14 -stdlib=libstdc++ -I/usr/include/c++/x86_64-redhat-linux/14 +# FFmpeg headers live in different places per distro (e.g. /usr/include on +# Debian/Ubuntu, /usr/include/ffmpeg on Fedora). Ask pkg-config so the build +# works everywhere. +FFMPEG_CFLAGS := $(shell pkg-config --cflags libavcodec libavutil 2>/dev/null) + +CFLAGS += -I$(BASEDIR)/wrapper $(FFMPEG_CFLAGS) -I/usr/include/c++/14 -stdlib=libstdc++ -I/usr/include/c++/x86_64-redhat-linux/14 all: prereq make-subdirs $(HEADERS) $(SRCS) $(OBJS) $(TARGET) diff --git a/README.md b/README.md index 265489c..258a9eb 100644 --- a/README.md +++ b/README.md @@ -44,6 +44,21 @@ sudo apt update sudo apt install build-essential pkg-config libavcodec-dev libavutil-dev ``` +### Install dependencies (Fedora / RHEL) +The Makefile builds with `clang++` against `libstdc++`, and pulls the FFmpeg +headers via `pkg-config` (on Fedora they live in `/usr/include/ffmpeg`). + +``` +sudo dnf install clang make pkgconf-pkg-config libstdc++-devel ffmpeg-devel +``` + +Notes: +- `ffmpeg-devel` comes from [RPM Fusion](https://rpmfusion.org/). If you don't + want RPM Fusion, Fedora's own packages `libavcodec-free-devel + libavutil-free-devel` also work (AAC-LC is included in the free build). +- On RHEL / Rocky / AlmaLinux, enable **EPEL + RPM Fusion** first, then install + the same packages. + ## Build To build the plugin and package it for DaVinci Resolve: diff --git a/audio_encoder.cpp b/audio_encoder.cpp index 3625d47..cc9bda6 100755 --- a/audio_encoder.cpp +++ b/audio_encoder.cpp @@ -150,24 +150,53 @@ AudioEncoder::~AudioEncoder() StatusCode AudioEncoder::DoInit(HostPropertyCollectionRef* p_pProps) { - uint32_t m_outputBitDepth = 0; - p_pProps->GetUINT32(pIOPropBitDepth, m_outputBitDepth); - if (m_outputBitDepth != 16 && m_outputBitDepth != 24) { - g_Log(logLevelError, "AAC Audio Plugin :: Only 16-bit or 24-bit PCM is supported, got %d", m_outputBitDepth); + // NOTE: Resolve calls DoInit() BEFORE DoOpen(). The actual FFmpeg encoder is + // created lazily in DoOpen() (see InitEncoder), because that is the first + // point where the user's codec settings (bit rate) are guaranteed available. + // Here we only cache the media format coming from the host. + uint32_t outputBitDepth = 0; + p_pProps->GetUINT32(pIOPropBitDepth, outputBitDepth); + if (outputBitDepth != 16 && outputBitDepth != 24) { + g_Log(logLevelError, "AAC Audio Plugin :: Only 16-bit or 24-bit PCM is supported, got %d", outputBitDepth); return errFail; } + m_outputBitDepth_ = outputBitDepth; + uint8_t isFloat = 0; p_pProps->GetUINT8(pIOPropIsFloat, isFloat); - uint32_t samplingRate = 0; - p_pProps->GetUINT32(pIOPropSamplingRate, samplingRate); - uint32_t numChannels = 0; - p_pProps->GetUINT32(pIOPropNumChannels, numChannels); + p_pProps->GetUINT32(pIOPropSamplingRate, m_samplingRate); + p_pProps->GetUINT32(pIOPropNumChannels, m_numChannels); uint32_t trackIdx = 0xFFFFFFFF; p_pProps->GetUINT32(pIOPropTrackIdx, trackIdx); - int bitRate = 128000; // default - if (m_pSettings) bitRate = m_pSettings->GetBitRate() * 1000; - // --- FFMpeg AAC encoder init --- + // The encoder settings may already be present in the init property bag; read + // them here as well so the chosen bit rate is honoured even if DoOpen does + // not receive them. DoOpen will refresh this from the authoritative buffer. + UIAudioSettingsController settings; + settings.Load(p_pProps); + if (settings.GetBitRate() > 0) { + m_bitRate = settings.GetBitRate() * 1000; + } + + g_Log(logLevelWarn, "AAC Audio Plugin :: Init Bit Depth: %d, isFloat: %d, Sampling Rate: %d, Num Channels: %d, Track Index: %d, Bit Rate: %d", m_outputBitDepth_, isFloat, m_samplingRate, m_numChannels, trackIdx, m_bitRate); + + return errNone; +} + +StatusCode AudioEncoder::InitEncoder() +{ + if (m_ffmpegCtx && m_ffmpegCtx->codecCtx) { + return errNone; // already initialised + } + if (m_outputBitDepth_ != 16 && m_outputBitDepth_ != 24) { + g_Log(logLevelError, "AAC Audio Plugin :: Only 16-bit or 24-bit PCM is supported, got %d", m_outputBitDepth_); + return errFail; + } + if (m_samplingRate == 0 || m_numChannels == 0) { + g_Log(logLevelError, "AAC Audio Plugin :: Invalid format (sampleRate=%u, channels=%u)", m_samplingRate, m_numChannels); + return errFail; + } + const AVCodec* codec = avcodec_find_encoder(AV_CODEC_ID_AAC); if (!codec) { g_Log(logLevelError, "AAC encoder not found"); @@ -179,31 +208,46 @@ StatusCode AudioEncoder::DoInit(HostPropertyCollectionRef* p_pProps) g_Log(logLevelError, "Could not allocate AVCodecContext"); return errFail; } - m_ffmpegCtx->codecCtx->bit_rate = bitRate; + m_ffmpegCtx->codecCtx->bit_rate = m_bitRate; m_ffmpegCtx->codecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP; // AAC prefers planar float - m_ffmpegCtx->codecCtx->sample_rate = samplingRate; - av_channel_layout_default(&m_ffmpegCtx->codecCtx->ch_layout, numChannels); + m_ffmpegCtx->codecCtx->sample_rate = m_samplingRate; + av_channel_layout_default(&m_ffmpegCtx->codecCtx->ch_layout, m_numChannels); m_ffmpegCtx->codecCtx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; + // Ask the encoder to emit the AudioSpecificConfig out-of-band (extradata) + // instead of inline, which is what MP4/MOV need for the esds box. + m_ffmpegCtx->codecCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; if (avcodec_open2(m_ffmpegCtx->codecCtx, codec, nullptr) < 0) { g_Log(logLevelError, "Could not open AAC encoder"); return errFail; } - m_ffmpegCtx->frame = av_frame_alloc(); - m_ffmpegCtx->frame->nb_samples = m_ffmpegCtx->codecCtx->frame_size; - m_ffmpegCtx->frame->format = m_ffmpegCtx->codecCtx->sample_fmt; - av_channel_layout_copy(&m_ffmpegCtx->frame->ch_layout, &m_ffmpegCtx->codecCtx->ch_layout); - av_frame_get_buffer(m_ffmpegCtx->frame, 0); + + // Capture the AudioSpecificConfig so DoOpen can publish it as the magic + // cookie (esds). Without it the MP4 reports profile=-1 / mp4a.40.0 and + // browsers play it silently. + if (m_ffmpegCtx->codecCtx->extradata && m_ffmpegCtx->codecCtx->extradata_size > 0) { + m_magicCookie.assign(m_ffmpegCtx->codecCtx->extradata, + m_ffmpegCtx->codecCtx->extradata + m_ffmpegCtx->codecCtx->extradata_size); + } else { + g_Log(logLevelError, "AAC Audio Plugin :: Encoder produced no extradata (AudioSpecificConfig)"); + } + m_ffmpegCtx->pkt = av_packet_alloc(); + if (!m_ffmpegCtx->pkt) { + g_Log(logLevelError, "AAC Audio Plugin :: av_packet_alloc failed"); + return errFail; + } m_ffmpegCtx->frame_size = m_ffmpegCtx->codecCtx->frame_size; m_ffmpegCtx->pts = 0; - g_Log(logLevelWarn, "AAC Audio Plugin :: Init Bit Depth: %d, isFloat: %d, Sampling Rate: %d, Num Channels: %d, Track Index: %d", m_outputBitDepth, isFloat, samplingRate, numChannels, trackIdx); - // Сохраняем желаемый выходной битдпет для использования в DoProcess - m_ffmpegCtx->frame_size = m_ffmpegCtx->codecCtx->frame_size; - m_ffmpegCtx->pts = 0; - m_outputBitDepth_ = m_outputBitDepth; + + if (m_ffmpegCtx->frame_size <= 0) { + g_Log(logLevelError, "AAC Audio Plugin :: Encoder reported invalid frame_size %d", m_ffmpegCtx->frame_size); + return errFail; + } + + g_Log(logLevelWarn, "AAC Audio Plugin :: Encoder opened. Bit Rate: %d, Sampling Rate: %u, Num Channels: %u, ASC bytes: %zu", m_bitRate, m_samplingRate, m_numChannels, m_magicCookie.size()); // --- RINGBUFFER INIT --- - m_channels = numChannels; + m_channels = m_numChannels; m_frameSize = m_ffmpegCtx->frame_size; m_pcmRingBuffer.clear(); m_pcmRingBuffer.resize(m_channels); @@ -229,35 +273,51 @@ StatusCode AudioEncoder::DoOpen(HostBufferRef* p_pBuff) m_pSettings.reset(new UIAudioSettingsController()); m_pSettings->Load(p_pBuff); - // optionally fill bitrate info hint for Resolve + // The settings buffer is the authoritative source for the user-selected bit + // rate. Apply it (this is what makes the UI slider actually take effect). if (m_pSettings->GetBitRate() > 0) { - const uint64_t bitRate = static_cast(m_pSettings->GetBitRate()) * 1000; - StatusCode sts = p_pBuff->SetProperty(pIOPropBitRate, propTypeUInt32, &bitRate, 1); + m_bitRate = m_pSettings->GetBitRate() * 1000; + + // bitrate info hint for Resolve (pIOPropBitRate is uint64 bits/sec) + const uint64_t bitRate = static_cast(m_bitRate); + StatusCode sts = p_pBuff->SetProperty(pIOPropBitRate, propTypeUInt64, &bitRate, 1); if (sts != errNone) { return sts; } } - uint32_t bitDepth = 0; - p_pBuff->GetUINT32(pIOPropBitDepth, bitDepth); + // Refresh the media format from the buffer, falling back to the values + // cached in DoInit if the buffer does not carry them. + uint32_t v = 0; + if (p_pBuff->GetUINT32(pIOPropBitDepth, v) && (v == 16 || v == 24)) m_outputBitDepth_ = v; + v = 0; + if (p_pBuff->GetUINT32(pIOPropSamplingRate, v) && v != 0) m_samplingRate = v; + v = 0; + if (p_pBuff->GetUINT32(pIOPropNumChannels, v) && v != 0) m_numChannels = v; + uint8_t isFloat = 0; p_pBuff->GetUINT8(pIOPropIsFloat, isFloat); - uint32_t samplingRate = 0; - p_pBuff->GetUINT32(pIOPropSamplingRate, samplingRate); - uint32_t numChannels = 0; - p_pBuff->GetUINT32(pIOPropNumChannels, numChannels); uint32_t trackIdx = 0xFFFFFFFF; p_pBuff->GetUINT32(pIOPropTrackIdx, trackIdx); - g_Log(logLevelWarn, "AAC Audio Plugin :: DoOpen params: bitDepth=%u, isFloat=%u, samplingRate=%u, numChannels=%u, trackIdx=%u, bitrate=%d", bitDepth, isFloat, samplingRate, numChannels, trackIdx, m_pSettings->GetBitRate()); + g_Log(logLevelWarn, "AAC Audio Plugin :: DoOpen params: bitDepth=%u, isFloat=%u, samplingRate=%u, numChannels=%u, trackIdx=%u, bitrate(kbps)=%d", m_outputBitDepth_, isFloat, m_samplingRate, m_numChannels, trackIdx, m_pSettings->GetBitRate()); - // fill magic cookie here if needed - std::vector cookie; - if (!cookie.empty()) + // Now that both the media format and the user settings are known, create the + // encoder. This produces the AudioSpecificConfig we need below. + StatusCode sts = InitEncoder(); + if (sts != errNone) { - p_pBuff->SetProperty(pIOPropMagicCookie, propTypeUInt8, &cookie[0], cookie.size()); - uint32_t fourCC = 0; + return sts; + } + + // Publish the AAC AudioSpecificConfig as the container magic cookie so the + // muxer writes a proper esds box (mp4a.40.2 / AAC-LC). Without this the MP4 + // shows profile=-1 / mp4a.40.0 and browsers decode it as silence. + if (!m_magicCookie.empty()) + { + p_pBuff->SetProperty(pIOPropMagicCookie, propTypeUInt8, m_magicCookie.data(), m_magicCookie.size()); + const uint32_t fourCC = 'esds'; p_pBuff->SetProperty(pIOPropMagicCookieType, propTypeUInt32, &fourCC, 1); } @@ -266,183 +326,171 @@ StatusCode AudioEncoder::DoOpen(HostBufferRef* p_pBuff) StatusCode AudioEncoder::DoProcess(HostBufferRef* p_pBuff) { - if (!m_ffmpegCtx || !m_ffmpegCtx->codecCtx) return errFail; + if (!m_ffmpegCtx || !m_ffmpegCtx->codecCtx) { + return errFail; + } + + // p_pBuff == NULL signals end-of-stream: encode any buffered tail samples, + // then flush the encoder's internal delay. + if (p_pBuff == NULL) { + if (m_ringBufferFill > 0) { + PadAndFlushRingBuffer(); + const StatusCode sts = EncodeRingBufferFrame(); + ResetRingBuffer(); + if (sts != errNone) { + return sts; + } + } + const int ret = avcodec_send_frame(m_ffmpegCtx->codecCtx, nullptr); + if (ret < 0 && ret != AVERROR_EOF) { + g_Log(logLevelError, "AAC Audio Plugin :: flush send_frame failed (%d)", ret); + return errFail; + } + return DrainEncoder(); + } + uint32_t inputBitDepth = 0; - if (p_pBuff && !p_pBuff->GetUINT32(pIOPropBitDepth, inputBitDepth)) { + if (!p_pBuff->GetUINT32(pIOPropBitDepth, inputBitDepth) || + (inputBitDepth != 16 && inputBitDepth != 24)) { inputBitDepth = m_outputBitDepth_; } - if (inputBitDepth != 16 && inputBitDepth != 24) { - inputBitDepth = m_outputBitDepth_; + const int numChannels = static_cast(m_channels); + const int frame_size = static_cast(m_frameSize); + + char* pBuf = NULL; + size_t bufSize = 0; + if (!p_pBuff->LockBuffer(&pBuf, &bufSize) || bufSize == 0) { + return errNone; } - int numChannels = m_channels; - int frame_size = m_frameSize; - if (p_pBuff != NULL) - { - char* pBuf = NULL; - size_t bufSize = 0; - if (!p_pBuff->LockBuffer(&pBuf, &bufSize) || bufSize == 0) - { - return errNone; - } - int bytesPerSample = (inputBitDepth == 16) ? 2 : 3; - int totalSamples = bufSize / (numChannels * bytesPerSample); - std::vector> planarPCM(numChannels, std::vector(totalSamples, 0.0f)); - if (inputBitDepth == 16) { - int16_t* src = (int16_t*)pBuf; - for (int i = 0; i < totalSamples; ++i) { - for (int ch = 0; ch < numChannels; ++ch) { - planarPCM[ch][i] = src[i * numChannels + ch] / 32768.0f; - } - } - } else if (inputBitDepth == 24) { - unsigned char* src = (unsigned char*)pBuf; - for (int i = 0; i < totalSamples; ++i) { - for (int ch = 0; ch < numChannels; ++ch) { - int idx = (i * numChannels + ch) * 3; - int32_t sample = (src[idx + 2] << 24) | (src[idx + 1] << 16) | (src[idx] << 8); - sample >>= 8; - planarPCM[ch][i] = sample / 8388608.0f; - } + + const int bytesPerSample = (inputBitDepth == 16) ? 2 : 3; + const int totalSamples = static_cast(bufSize / (numChannels * bytesPerSample)); + std::vector> planarPCM(numChannels, std::vector(totalSamples, 0.0f)); + if (inputBitDepth == 16) { + const int16_t* src = reinterpret_cast(pBuf); + for (int i = 0; i < totalSamples; ++i) { + for (int ch = 0; ch < numChannels; ++ch) { + planarPCM[ch][i] = src[i * numChannels + ch] / 32768.0f; } } - int sampleIdx = 0; - while (sampleIdx < totalSamples) { - int chunk = std::min((int)(frame_size - m_ringBufferFill), totalSamples - sampleIdx); - std::vector chunkPtrs(numChannels); + } else { // 24-bit packed, little-endian signed + const unsigned char* src = reinterpret_cast(pBuf); + for (int i = 0; i < totalSamples; ++i) { for (int ch = 0; ch < numChannels; ++ch) { - chunkPtrs[ch] = &planarPCM[ch][sampleIdx]; - } - AddPCMToRingBuffer(chunkPtrs.data(), chunk); - if (IsRingBufferFull()) { - AVFrame* tempFrame = av_frame_alloc(); - tempFrame->format = m_ffmpegCtx->codecCtx->sample_fmt; - tempFrame->nb_samples = frame_size; - av_channel_layout_copy(&tempFrame->ch_layout, &m_ffmpegCtx->codecCtx->ch_layout); - av_frame_get_buffer(tempFrame, 0); - float** dst = (float**)tempFrame->extended_data; - GetFrameFromRingBuffer(dst, frame_size); - tempFrame->pts = m_ffmpegCtx->pts; - m_ffmpegCtx->pts += frame_size; - int ret = avcodec_send_frame(m_ffmpegCtx->codecCtx, tempFrame); - av_frame_free(&tempFrame); - if (ret < 0) { - return errFail; - } - do { - ret = avcodec_receive_packet(m_ffmpegCtx->codecCtx, m_ffmpegCtx->pkt); - if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) - break; - else if (ret < 0) { - break; - } - HostBufferRef outBuf; - outBuf.Resize(m_ffmpegCtx->pkt->size); - char* outData = nullptr; - size_t outSize = 0; - if (outBuf.LockBuffer(&outData, &outSize) && outSize >= (size_t)m_ffmpegCtx->pkt->size) { - memcpy(outData, m_ffmpegCtx->pkt->data, m_ffmpegCtx->pkt->size); - outBuf.UnlockBuffer(); - outBuf.SetProperty(pIOPropBitDepth, propTypeUInt32, &m_outputBitDepth_, 1); - outBuf.SetProperty(pIOPropSamplingRate, propTypeUInt32, &m_ffmpegCtx->codecCtx->sample_rate, 1); - uint32_t numChannels_ = m_ffmpegCtx->codecCtx->ch_layout.nb_channels; - outBuf.SetProperty(pIOPropNumChannels, propTypeUInt32, &numChannels_, 1); - uint8_t isKey = 0; - outBuf.SetProperty(pIOPropIsKeyFrame, propTypeUInt8, &isKey, 1); - int64_t pkt_pts = m_ffmpegCtx->pkt->pts; - outBuf.SetProperty(pIOPropPTS, propTypeInt64, &pkt_pts, 1); - int64_t pkt_dur = m_ffmpegCtx->pkt->duration; - outBuf.SetProperty(pIOPropDuration, propTypeInt64, &pkt_dur, 1); - IPluginCodecRef::DoProcess(&outBuf); - } - av_packet_unref(m_ffmpegCtx->pkt); - } while (ret >= 0); - ResetRingBuffer(); + const int idx = (i * numChannels + ch) * 3; + int32_t sample = (src[idx + 2] << 24) | (src[idx + 1] << 16) | (src[idx] << 8); + sample >>= 8; // sign-extend the 24-bit value + planarPCM[ch][i] = sample / 8388608.0f; } - sampleIdx += chunk; } - p_pBuff->UnlockBuffer(); - } else { - if (m_ringBufferFill > 0) { - PadAndFlushRingBuffer(); - AVFrame* tempFrame = av_frame_alloc(); - tempFrame->format = m_ffmpegCtx->codecCtx->sample_fmt; - tempFrame->nb_samples = frame_size; - av_channel_layout_copy(&tempFrame->ch_layout, &m_ffmpegCtx->codecCtx->ch_layout); - av_frame_get_buffer(tempFrame, 0); - float** dst = (float**)tempFrame->extended_data; - GetFrameFromRingBuffer(dst, frame_size); - tempFrame->pts = m_ffmpegCtx->pts; - m_ffmpegCtx->pts += frame_size; - int ret = avcodec_send_frame(m_ffmpegCtx->codecCtx, tempFrame); - av_frame_free(&tempFrame); - if (ret < 0) { - return errFail; - } - do { - ret = avcodec_receive_packet(m_ffmpegCtx->codecCtx, m_ffmpegCtx->pkt); - if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) - break; - else if (ret < 0) { - break; - } - HostBufferRef outBuf; - outBuf.Resize(m_ffmpegCtx->pkt->size); - char* outData = nullptr; - size_t outSize = 0; - if (outBuf.LockBuffer(&outData, &outSize) && outSize >= (size_t)m_ffmpegCtx->pkt->size) { - memcpy(outData, m_ffmpegCtx->pkt->data, m_ffmpegCtx->pkt->size); - outBuf.UnlockBuffer(); - outBuf.SetProperty(pIOPropBitDepth, propTypeUInt32, &m_outputBitDepth_, 1); - outBuf.SetProperty(pIOPropSamplingRate, propTypeUInt32, &m_ffmpegCtx->codecCtx->sample_rate, 1); - uint32_t numChannels_ = m_ffmpegCtx->codecCtx->ch_layout.nb_channels; - outBuf.SetProperty(pIOPropNumChannels, propTypeUInt32, &numChannels_, 1); - uint8_t isKey = 0; - outBuf.SetProperty(pIOPropIsKeyFrame, propTypeUInt8, &isKey, 1); - int64_t pkt_pts = m_ffmpegCtx->pkt->pts; - outBuf.SetProperty(pIOPropPTS, propTypeInt64, &pkt_pts, 1); - int64_t pkt_dur = m_ffmpegCtx->pkt->duration; - outBuf.SetProperty(pIOPropDuration, propTypeInt64, &pkt_dur, 1); - IPluginCodecRef::DoProcess(&outBuf); - } - av_packet_unref(m_ffmpegCtx->pkt); - } while (ret >= 0); + } + + StatusCode sts = errNone; + int sampleIdx = 0; + while (sampleIdx < totalSamples) { + const int chunk = std::min(static_cast(frame_size - m_ringBufferFill), totalSamples - sampleIdx); + std::vector chunkPtrs(numChannels); + for (int ch = 0; ch < numChannels; ++ch) { + chunkPtrs[ch] = &planarPCM[ch][sampleIdx]; + } + AddPCMToRingBuffer(chunkPtrs.data(), chunk); + if (IsRingBufferFull()) { + sts = EncodeRingBufferFrame(); ResetRingBuffer(); + if (sts != errNone) { + break; + } } - // --- FLUSH --- - int ret = avcodec_send_frame(m_ffmpegCtx->codecCtx, nullptr); - if (ret < 0 && ret != AVERROR_EOF) { + sampleIdx += chunk; + } + + p_pBuff->UnlockBuffer(); + return sts; +} + +// Encodes exactly one frame worth of samples currently held in the ring buffer +// and forwards the resulting packets to the host. +StatusCode AudioEncoder::EncodeRingBufferFrame() +{ + AVFrame* frame = av_frame_alloc(); + if (!frame) { + g_Log(logLevelError, "AAC Audio Plugin :: av_frame_alloc failed"); + return errFail; + } + frame->format = m_ffmpegCtx->codecCtx->sample_fmt; + frame->nb_samples = static_cast(m_frameSize); + av_channel_layout_copy(&frame->ch_layout, &m_ffmpegCtx->codecCtx->ch_layout); + if (av_frame_get_buffer(frame, 0) < 0) { + g_Log(logLevelError, "AAC Audio Plugin :: av_frame_get_buffer failed"); + av_frame_free(&frame); + return errFail; + } + + GetFrameFromRingBuffer(reinterpret_cast(frame->extended_data), m_frameSize); + frame->pts = m_ffmpegCtx->pts; + m_ffmpegCtx->pts += m_frameSize; + + const int ret = avcodec_send_frame(m_ffmpegCtx->codecCtx, frame); + av_frame_free(&frame); + if (ret < 0) { + g_Log(logLevelError, "AAC Audio Plugin :: avcodec_send_frame failed (%d)", ret); + return errFail; + } + + return DrainEncoder(); +} + +// Pulls every packet currently available from the encoder and forwards each to +// the host. +StatusCode AudioEncoder::DrainEncoder() +{ + for (;;) { + const int ret = avcodec_receive_packet(m_ffmpegCtx->codecCtx, m_ffmpegCtx->pkt); + if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) { + return errNone; + } + if (ret < 0) { + g_Log(logLevelError, "AAC Audio Plugin :: avcodec_receive_packet failed (%d)", ret); return errFail; } - do { - ret = avcodec_receive_packet(m_ffmpegCtx->codecCtx, m_ffmpegCtx->pkt); - if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) - break; - else if (ret < 0) { - break; - } - HostBufferRef outBuf; - outBuf.Resize(m_ffmpegCtx->pkt->size); - char* outData = nullptr; - size_t outSize = 0; - if (outBuf.LockBuffer(&outData, &outSize) && outSize >= (size_t)m_ffmpegCtx->pkt->size) { - memcpy(outData, m_ffmpegCtx->pkt->data, m_ffmpegCtx->pkt->size); - outBuf.UnlockBuffer(); - outBuf.SetProperty(pIOPropBitDepth, propTypeUInt32, &m_outputBitDepth_, 1); - outBuf.SetProperty(pIOPropSamplingRate, propTypeUInt32, &m_ffmpegCtx->codecCtx->sample_rate, 1); - uint32_t numChannels_ = m_ffmpegCtx->codecCtx->ch_layout.nb_channels; - outBuf.SetProperty(pIOPropNumChannels, propTypeUInt32, &numChannels_, 1); - uint8_t isKey = 0; - outBuf.SetProperty(pIOPropIsKeyFrame, propTypeUInt8, &isKey, 1); - int64_t pkt_pts = m_ffmpegCtx->pkt->pts; - outBuf.SetProperty(pIOPropPTS, propTypeInt64, &pkt_pts, 1); - int64_t pkt_dur = m_ffmpegCtx->pkt->duration; - outBuf.SetProperty(pIOPropDuration, propTypeInt64, &pkt_dur, 1); - IPluginCodecRef::DoProcess(&outBuf); - } - av_packet_unref(m_ffmpegCtx->pkt); - } while (ret >= 0); + EmitPacket(); + av_packet_unref(m_ffmpegCtx->pkt); } - return errNone; +} + +// Copies the current encoder packet into a host buffer and hands it to the +// container, tagging it with the metadata the muxer needs. +void AudioEncoder::EmitPacket() +{ + const AVPacket* pkt = m_ffmpegCtx->pkt; + HostBufferRef outBuf; + if (!outBuf.Resize(pkt->size)) { + g_Log(logLevelError, "AAC Audio Plugin :: failed to allocate output buffer (%d bytes)", pkt->size); + return; + } + char* outData = nullptr; + size_t outSize = 0; + if (!outBuf.LockBuffer(&outData, &outSize) || outSize < static_cast(pkt->size)) { + return; + } + memcpy(outData, pkt->data, pkt->size); + outBuf.UnlockBuffer(); + + outBuf.SetProperty(pIOPropBitDepth, propTypeUInt32, &m_outputBitDepth_, 1); + const uint32_t sampleRate = m_ffmpegCtx->codecCtx->sample_rate; + outBuf.SetProperty(pIOPropSamplingRate, propTypeUInt32, &sampleRate, 1); + const uint32_t numChannels = m_ffmpegCtx->codecCtx->ch_layout.nb_channels; + outBuf.SetProperty(pIOPropNumChannels, propTypeUInt32, &numChannels, 1); + // Every AAC frame is independently decodable, so it is a sync sample. The + // native encoder sets AV_PKT_FLAG_KEY on every packet; mirror that so the + // muxer marks audio sync samples correctly (matters for seeking). + const uint8_t isKey = (pkt->flags & AV_PKT_FLAG_KEY) ? 1 : 0; + outBuf.SetProperty(pIOPropIsKeyFrame, propTypeUInt8, &isKey, 1); + const int64_t pkt_pts = pkt->pts; + outBuf.SetProperty(pIOPropPTS, propTypeInt64, &pkt_pts, 1); + const int64_t pkt_dur = pkt->duration; + outBuf.SetProperty(pIOPropDuration, propTypeInt64, &pkt_dur, 1); + IPluginCodecRef::DoProcess(&outBuf); } // --- RINGBUFFER HELPERS --- diff --git a/audio_encoder.h b/audio_encoder.h index fda9202..5340638 100755 --- a/audio_encoder.h +++ b/audio_encoder.h @@ -32,6 +32,26 @@ class AudioEncoder : public IPluginCodecRef std::unique_ptr m_ffmpegCtx; uint32_t m_outputBitDepth_ = 16; // Желаемый bitDepth (выбранный пользователем) + // Encoder parameters cached from DoInit/DoOpen. The encoder is created lazily + // in DoOpen() once both the media format (DoInit) and the user settings + // (DoOpen) are known, so the chosen bit rate is actually applied. + uint32_t m_samplingRate = 0; + uint32_t m_numChannels = 0; + int m_bitRate = 128000; + + // AAC AudioSpecificConfig (codec extradata), handed to the container as the + // magic cookie so the MP4/MOV gets a proper esds box (mp4a.40.2 / AAC-LC). + std::vector m_magicCookie; + + StatusCode InitEncoder(); + + // Encode helpers. EncodeRingBufferFrame() pushes one full frame from the + // ring buffer into the encoder; DrainEncoder() forwards every packet the + // encoder produces to the host; EmitPacket() forwards the current packet. + StatusCode EncodeRingBufferFrame(); + StatusCode DrainEncoder(); + void EmitPacket(); + // Ringbuffer для PCM (float planar) std::vector> m_pcmRingBuffer; // [channel][sample] size_t m_ringBufferFill = 0;