torchcodec 0.7.0__cp312-cp312-win_amd64.whl

torchcodec/_core/DeviceInterface.cpp

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+// Copyright (c) Meta Platforms, Inc. and affiliates.
+// All rights reserved.
+//
+// This source code is licensed under the BSD-style license found in the
+// LICENSE file in the root directory of this source tree.
+
+#include "src/torchcodec/_core/DeviceInterface.h"
+#include <map>
+#include <mutex>
+
+namespace facebook::torchcodec {
+
+namespace {
+using DeviceInterfaceMap = std::map<torch::DeviceType, CreateDeviceInterfaceFn>;
+static std::mutex g_interface_mutex;
+
+DeviceInterfaceMap& getDeviceMap() {
+  static DeviceInterfaceMap deviceMap;
+  return deviceMap;
+}
+
+std::string getDeviceType(const std::string& device) {
+  size_t pos = device.find(':');
+  if (pos == std::string::npos) {
+    return device;
+  }
+  return device.substr(0, pos);
+}
+
+} // namespace
+
+bool registerDeviceInterface(
+    torch::DeviceType deviceType,
+    CreateDeviceInterfaceFn createInterface) {
+  std::scoped_lock lock(g_interface_mutex);
+  DeviceInterfaceMap& deviceMap = getDeviceMap();
+
+  TORCH_CHECK(
+      deviceMap.find(deviceType) == deviceMap.end(),
+      "Device interface already registered for ",
+      deviceType);
+  deviceMap.insert({deviceType, createInterface});
+
+  return true;
+}
+
+torch::Device createTorchDevice(const std::string device) {
+  std::scoped_lock lock(g_interface_mutex);
+  std::string deviceType = getDeviceType(device);
+  DeviceInterfaceMap& deviceMap = getDeviceMap();
+
+  auto deviceInterface = std::find_if(
+      deviceMap.begin(),
+      deviceMap.end(),
+      [&](const std::pair<torch::DeviceType, CreateDeviceInterfaceFn>& arg) {
+        return device.rfind(
+                   torch::DeviceTypeName(arg.first, /*lcase*/ true), 0) == 0;
+      });
+  TORCH_CHECK(
+      deviceInterface != deviceMap.end(), "Unsupported device: ", device);
+
+  return torch::Device(device);
+}
+
+std::unique_ptr<DeviceInterface> createDeviceInterface(
+    const torch::Device& device) {
+  auto deviceType = device.type();
+  std::scoped_lock lock(g_interface_mutex);
+  DeviceInterfaceMap& deviceMap = getDeviceMap();
+
+  TORCH_CHECK(
+      deviceMap.find(deviceType) != deviceMap.end(),
+      "Unsupported device: ",
+      device);
+
+  return std::unique_ptr<DeviceInterface>(deviceMap[deviceType](device));
+}
+
+} // namespace facebook::torchcodec

torchcodec/_core/DeviceInterface.h

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+// Copyright (c) Meta Platforms, Inc. and affiliates.
+// All rights reserved.
+//
+// This source code is licensed under the BSD-style license found in the
+// LICENSE file in the root directory of this source tree.
+
+#pragma once
+
+#include <torch/types.h>
+#include <functional>
+#include <memory>
+#include <stdexcept>
+#include <string>
+#include "FFMPEGCommon.h"
+#include "src/torchcodec/_core/Frame.h"
+#include "src/torchcodec/_core/StreamOptions.h"
+
+namespace facebook::torchcodec {
+
+// Note that all these device functions should only be called if the device is
+// not a CPU device. CPU device functions are already implemented in the
+// SingleStreamDecoder implementation.
+// These functions should only be called from within an if block like this:
+// if (device.type() != torch::kCPU) {
+//   deviceFunction(device, ...);
+// }
+
+class DeviceInterface {
+ public:
+  DeviceInterface(const torch::Device& device) : device_(device) {}
+
+  virtual ~DeviceInterface(){};
+
+  torch::Device& device() {
+    return device_;
+  };
+
+  virtual std::optional<const AVCodec*> findCodec(const AVCodecID& codecId) = 0;
+
+  // Initialize the hardware device that is specified in `device`. Some builds
+  // support CUDA and others only support CPU.
+  virtual void initializeContext(AVCodecContext* codecContext) = 0;
+
+  virtual void convertAVFrameToFrameOutput(
+      const VideoStreamOptions& videoStreamOptions,
+      const AVRational& timeBase,
+      UniqueAVFrame& avFrame,
+      FrameOutput& frameOutput,
+      std::optional<torch::Tensor> preAllocatedOutputTensor = std::nullopt) = 0;
+
+ protected:
+  torch::Device device_;
+};
+
+using CreateDeviceInterfaceFn =
+    std::function<DeviceInterface*(const torch::Device& device)>;
+
+bool registerDeviceInterface(
+    torch::DeviceType deviceType,
+    const CreateDeviceInterfaceFn createInterface);
+
+torch::Device createTorchDevice(const std::string device);
+
+std::unique_ptr<DeviceInterface> createDeviceInterface(
+    const torch::Device& device);
+
+} // namespace facebook::torchcodec

torchcodec/_core/Encoder.cpp

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+#include <sstream>
+
+#include "src/torchcodec/_core/AVIOTensorContext.h"
+#include "src/torchcodec/_core/Encoder.h"
+#include "torch/types.h"
+
+namespace facebook::torchcodec {
+
+namespace {
+
+torch::Tensor validateSamples(const torch::Tensor& samples) {
+  TORCH_CHECK(
+      samples.dtype() == torch::kFloat32,
+      "samples must have float32 dtype, got ",
+      samples.dtype());
+  TORCH_CHECK(
+      samples.dim() == 2,
+      "samples must have 2 dimensions, got ",
+      samples.dim());
+
+  // We enforce this, but if we get user reports we should investigate whether
+  // that's actually needed.
+  int numChannels = static_cast<int>(samples.sizes()[0]);
+  TORCH_CHECK(
+      numChannels <= AV_NUM_DATA_POINTERS,
+      "Trying to encode ",
+      numChannels,
+      " channels, but FFmpeg only supports ",
+      AV_NUM_DATA_POINTERS,
+      " channels per frame.");
+
+  return samples.contiguous();
+}
+
+void validateSampleRate(const AVCodec& avCodec, int sampleRate) {
+  if (avCodec.supported_samplerates == nullptr) {
+    return;
+  }
+
+  for (auto i = 0; avCodec.supported_samplerates[i] != 0; ++i) {
+    if (sampleRate == avCodec.supported_samplerates[i]) {
+      return;
+    }
+  }
+  std::stringstream supportedRates;
+  for (auto i = 0; avCodec.supported_samplerates[i] != 0; ++i) {
+    if (i > 0) {
+      supportedRates << ", ";
+    }
+    supportedRates << avCodec.supported_samplerates[i];
+  }
+
+  TORCH_CHECK(
+      false,
+      "invalid sample rate=",
+      sampleRate,
+      ". Supported sample rate values are: ",
+      supportedRates.str());
+}
+
+static const std::vector<AVSampleFormat> preferredFormatsOrder = {
+    AV_SAMPLE_FMT_FLTP,
+    AV_SAMPLE_FMT_FLT,
+    AV_SAMPLE_FMT_DBLP,
+    AV_SAMPLE_FMT_DBL,
+    AV_SAMPLE_FMT_S64P,
+    AV_SAMPLE_FMT_S64,
+    AV_SAMPLE_FMT_S32P,
+    AV_SAMPLE_FMT_S32,
+    AV_SAMPLE_FMT_S16P,
+    AV_SAMPLE_FMT_S16,
+    AV_SAMPLE_FMT_U8P,
+    AV_SAMPLE_FMT_U8};
+
+AVSampleFormat findBestOutputSampleFormat(const AVCodec& avCodec) {
+  // Find a sample format that the encoder supports. We prefer using FLT[P],
+  // since this is the format of the input samples. If FLTP isn't supported
+  // then we'll need to convert the AVFrame's format. Our heuristic is to encode
+  // into the format with the highest resolution.
+  if (avCodec.sample_fmts == nullptr) {
+    // Can't really validate anything in this case, best we can do is hope that
+    // FLTP is supported by the encoder. If not, FFmpeg will raise.
+    return AV_SAMPLE_FMT_FLTP;
+  }
+
+  for (AVSampleFormat preferredFormat : preferredFormatsOrder) {
+    for (int i = 0; avCodec.sample_fmts[i] != -1; ++i) {
+      if (avCodec.sample_fmts[i] == preferredFormat) {
+        return preferredFormat;
+      }
+    }
+  }
+  // We should always find a match in preferredFormatsOrder, so we should always
+  // return earlier. But in the event that a future FFmpeg version defines an
+  // additional sample format that isn't in preferredFormatsOrder, we fallback:
+  return avCodec.sample_fmts[0];
+}
+
+} // namespace
+
+AudioEncoder::~AudioEncoder() {
+  close_avio();
+}
+
+void AudioEncoder::close_avio() {
+  if (avFormatContext_ && avFormatContext_->pb) {
+    if (avFormatContext_->pb->error == 0) {
+      avio_flush(avFormatContext_->pb);
+    }
+
+    if (!avioContextHolder_) {
+      if (avFormatContext_->pb->error == 0) {
+        avio_close(avFormatContext_->pb);
+      }
+      // avoids closing again in destructor, which would segfault.
+      avFormatContext_->pb = nullptr;
+    }
+  }
+}
+
+AudioEncoder::AudioEncoder(
+    const torch::Tensor& samples,
+    int sampleRate,
+    std::string_view fileName,
+    const AudioStreamOptions& audioStreamOptions)
+    : samples_(validateSamples(samples)), inSampleRate_(sampleRate) {
+  setFFmpegLogLevel();
+  AVFormatContext* avFormatContext = nullptr;
+  int status = avformat_alloc_output_context2(
+      &avFormatContext, nullptr, nullptr, fileName.data());
+
+  TORCH_CHECK(
+      avFormatContext != nullptr,
+      "Couldn't allocate AVFormatContext. ",
+      "The destination file is ",
+      fileName,
+      ", check the desired extension? ",
+      getFFMPEGErrorStringFromErrorCode(status));
+  avFormatContext_.reset(avFormatContext);
+
+  status = avio_open(&avFormatContext_->pb, fileName.data(), AVIO_FLAG_WRITE);
+  TORCH_CHECK(
+      status >= 0,
+      "avio_open failed. The destination file is ",
+      fileName,
+      ", make sure it's a valid path? ",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  initializeEncoder(audioStreamOptions);
+}
+
+AudioEncoder::AudioEncoder(
+    const torch::Tensor& samples,
+    int sampleRate,
+    std::string_view formatName,
+    std::unique_ptr<AVIOContextHolder> avioContextHolder,
+    const AudioStreamOptions& audioStreamOptions)
+    : samples_(validateSamples(samples)),
+      inSampleRate_(sampleRate),
+      avioContextHolder_(std::move(avioContextHolder)) {
+  setFFmpegLogLevel();
+  AVFormatContext* avFormatContext = nullptr;
+  int status = avformat_alloc_output_context2(
+      &avFormatContext, nullptr, formatName.data(), nullptr);
+
+  TORCH_CHECK(
+      avFormatContext != nullptr,
+      "Couldn't allocate AVFormatContext. ",
+      "Check the desired format? Got format=",
+      formatName,
+      ". ",
+      getFFMPEGErrorStringFromErrorCode(status));
+  avFormatContext_.reset(avFormatContext);
+
+  avFormatContext_->pb = avioContextHolder_->getAVIOContext();
+
+  initializeEncoder(audioStreamOptions);
+}
+
+void AudioEncoder::initializeEncoder(
+    const AudioStreamOptions& audioStreamOptions) {
+  // We use the AVFormatContext's default codec for that
+  // specific format/container.
+  const AVCodec* avCodec =
+      avcodec_find_encoder(avFormatContext_->oformat->audio_codec);
+  TORCH_CHECK(avCodec != nullptr, "Codec not found");
+
+  AVCodecContext* avCodecContext = avcodec_alloc_context3(avCodec);
+  TORCH_CHECK(avCodecContext != nullptr, "Couldn't allocate codec context.");
+  avCodecContext_.reset(avCodecContext);
+
+  auto desiredBitRate = audioStreamOptions.bitRate;
+  if (desiredBitRate.has_value()) {
+    TORCH_CHECK(
+        *desiredBitRate >= 0, "bit_rate=", *desiredBitRate, " must be >= 0.");
+  }
+  // bit_rate=None defaults to 0, which is what the FFmpeg CLI seems to use as
+  // well when "-b:a" isn't specified.
+  avCodecContext_->bit_rate = desiredBitRate.value_or(0);
+
+  outNumChannels_ = static_cast<int>(
+      audioStreamOptions.numChannels.value_or(samples_.sizes()[0]));
+  validateNumChannels(*avCodec, outNumChannels_);
+  // The avCodecContext layout defines the layout of the encoded output, it's
+  // not related to the input sampes.
+  setDefaultChannelLayout(avCodecContext_, outNumChannels_);
+
+  outSampleRate_ = audioStreamOptions.sampleRate.value_or(inSampleRate_);
+  validateSampleRate(*avCodec, outSampleRate_);
+  avCodecContext_->sample_rate = outSampleRate_;
+
+  // Input samples are expected to be FLTP. Not all encoders support FLTP, so we
+  // may need to convert the samples into a supported output sample format,
+  // which is what the `.sample_fmt` defines.
+  avCodecContext_->sample_fmt = findBestOutputSampleFormat(*avCodec);
+
+  int status = avcodec_open2(avCodecContext_.get(), avCodec, nullptr);
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "avcodec_open2 failed: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  // We're allocating the stream here. Streams are meant to be freed by
+  // avformat_free_context(avFormatContext), which we call in the
+  // avFormatContext_'s destructor.
+  AVStream* avStream = avformat_new_stream(avFormatContext_.get(), nullptr);
+  TORCH_CHECK(avStream != nullptr, "Couldn't create new stream.");
+  status = avcodec_parameters_from_context(
+      avStream->codecpar, avCodecContext_.get());
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "avcodec_parameters_from_context failed: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+  streamIndex_ = avStream->index;
+
+  // If sample rate conversion is needed and the encoder doesn't support
+  // variable frame size, we need to create an intermediate FIFO. See
+  // [Encoding loop, sample rate conversion and FIFO].
+  if (((avCodec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE) == 0) &&
+      (inSampleRate_ != outSampleRate_)) {
+    // frame_size * 2 is a decent default size. FFmpeg automatically
+    // re-allocates the fifo if more space is needed.
+    auto avAudioFifo = av_audio_fifo_alloc(
+        avCodecContext_->sample_fmt,
+        outNumChannels_,
+        avCodecContext_->frame_size * 2);
+    TORCH_CHECK(avAudioFifo != nullptr, "Couldn't create AVAudioFifo.");
+    avAudioFifo_.reset(avAudioFifo);
+  }
+}
+
+torch::Tensor AudioEncoder::encodeToTensor() {
+  TORCH_CHECK(
+      avioContextHolder_ != nullptr,
+      "Cannot encode to tensor, avio tensor context doesn't exist.");
+  encode();
+  auto avioToTensorContext =
+      dynamic_cast<AVIOToTensorContext*>(avioContextHolder_.get());
+  TORCH_CHECK(avioToTensorContext != nullptr, "Invalid AVIO context holder.");
+  return avioToTensorContext->getOutputTensor();
+}
+
+void AudioEncoder::encode() {
+  // To be on the safe side we enforce that encode() can only be called once on
+  // an encoder object. Whether this is actually necessary is unknown, so this
+  // may be relaxed if needed.
+  TORCH_CHECK(!encodeWasCalled_, "Cannot call encode() twice.");
+  encodeWasCalled_ = true;
+
+  //  Default to 256 like in torchaudio
+  int numSamplesAllocatedPerFrame =
+      avCodecContext_->frame_size > 0 ? avCodecContext_->frame_size : 256;
+  UniqueAVFrame avFrame = allocateAVFrame(
+      numSamplesAllocatedPerFrame,
+      inSampleRate_,
+      static_cast<int>(samples_.sizes()[0]),
+      AV_SAMPLE_FMT_FLTP);
+  avFrame->pts = 0;
+
+  AutoAVPacket autoAVPacket;
+
+  uint8_t* psamples = static_cast<uint8_t*>(samples_.data_ptr());
+  int numSamples = static_cast<int>(samples_.sizes()[1]); // per channel
+  int numEncodedSamples = 0; // per channel
+  int numBytesPerSample = static_cast<int>(samples_.element_size());
+  int numBytesPerChannel = numSamples * numBytesPerSample;
+
+  auto status = avformat_write_header(avFormatContext_.get(), nullptr);
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Error in avformat_write_header: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  while (numEncodedSamples < numSamples) {
+    int numSamplesToEncode =
+        std::min(numSamplesAllocatedPerFrame, numSamples - numEncodedSamples);
+    int numBytesToEncode = numSamplesToEncode * numBytesPerSample;
+
+    for (int ch = 0; ch < samples_.sizes()[0]; ch++) {
+      std::memcpy(
+          avFrame->data[ch],
+          psamples + ch * numBytesPerChannel,
+          numBytesToEncode);
+    }
+    psamples += numBytesToEncode;
+
+    // Above, we set the AVFrame's .nb_samples to AVCodecContext.frame_size so
+    // that the frame buffers are allocated to a big enough size. Here, we reset
+    // it to the exact number of samples that need to be encoded, otherwise the
+    // encoded frame would contain more samples than necessary and our results
+    // wouldn't match the ffmpeg CLI.
+    avFrame->nb_samples = numSamplesToEncode;
+
+    UniqueAVFrame convertedAVFrame = maybeConvertAVFrame(avFrame);
+    encodeFrameThroughFifo(autoAVPacket, convertedAVFrame);
+
+    numEncodedSamples += numSamplesToEncode;
+  }
+  TORCH_CHECK(numEncodedSamples == numSamples, "Hmmmmmm something went wrong.");
+
+  flushBuffers();
+
+  status = av_write_trailer(avFormatContext_.get());
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Error in: av_write_trailer",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  close_avio();
+}
+
+UniqueAVFrame AudioEncoder::maybeConvertAVFrame(const UniqueAVFrame& avFrame) {
+  if (static_cast<AVSampleFormat>(avFrame->format) ==
+          avCodecContext_->sample_fmt &&
+      getNumChannels(avFrame) == outNumChannels_ &&
+      avFrame->sample_rate == outSampleRate_) {
+    // Note: the clone references the same underlying data, it's a cheap copy.
+    return UniqueAVFrame(av_frame_clone(avFrame.get()));
+  }
+
+  if (!swrContext_) {
+    swrContext_.reset(createSwrContext(
+        static_cast<AVSampleFormat>(avFrame->format),
+        avCodecContext_->sample_fmt,
+        avFrame->sample_rate,
+        outSampleRate_,
+        avFrame,
+        outNumChannels_));
+  }
+  UniqueAVFrame convertedAVFrame = convertAudioAVFrameSamples(
+      swrContext_,
+      avFrame,
+      avCodecContext_->sample_fmt,
+      outSampleRate_,
+      outNumChannels_);
+
+  if (avFrame->sample_rate == outSampleRate_) {
+    TORCH_CHECK(
+        convertedAVFrame->nb_samples == avFrame->nb_samples,
+        "convertedAVFrame->nb_samples=",
+        convertedAVFrame->nb_samples,
+        " differs from ",
+        "avFrame->nb_samples=",
+        avFrame->nb_samples,
+        "This is unexpected, please report on the TorchCodec bug tracker.");
+  }
+  return convertedAVFrame;
+}
+
+void AudioEncoder::encodeFrameThroughFifo(
+    AutoAVPacket& autoAVPacket,
+    const UniqueAVFrame& avFrame,
+    // flushFifo is only set to true in maybeFlushSwrBuffers(), i.e. at the very
+    // end of the encoding process when we're flushing buffers. We also want to
+    // flush the FIFO so as to not leave any remaining samples in it.
+    bool flushFifo) {
+  if (avAudioFifo_ == nullptr) {
+    encodeFrame(autoAVPacket, avFrame);
+    return;
+  }
+  int numSamplesWritten = av_audio_fifo_write(
+      avAudioFifo_.get(),
+      reinterpret_cast<void**>(avFrame->data),
+      avFrame->nb_samples);
+  TORCH_CHECK(
+      numSamplesWritten == avFrame->nb_samples,
+      "Tried to write ",
+      avFrame->nb_samples,
+      " samples, but only wrote ",
+      numSamplesWritten);
+
+  UniqueAVFrame newavFrame = allocateAVFrame(
+      avCodecContext_->frame_size,
+      outSampleRate_,
+      outNumChannels_,
+      avCodecContext_->sample_fmt);
+
+  // Explaining the while bound:
+  // - if we're not flushing the FIFO, i.e. in most cases, we want to pull
+  //   exactly `frame_size` samples from the FIFO, so we have to stop before it
+  //   contains less than `frame_size` samples.
+  // - if we're flushing the FIFO, we want to read from the FIFO until the very
+  //   last sample it contains.
+  //
+  // In both cases, for as long as we can, we're trying to pull exatly
+  // `frame_size` samples from the FIFO and send each `frame_size`-sized avFrame
+  // to encodeFrame(). Only the very last avFrame of the encoding process is
+  // allowed to contained less than frame_size samples. That only happens when
+  // flushFifo is true.
+  while (av_audio_fifo_size(avAudioFifo_.get()) >=
+         (flushFifo ? 1 : avCodecContext_->frame_size)) {
+    int samplesToRead = std::min(
+        av_audio_fifo_size(avAudioFifo_.get()), newavFrame->nb_samples);
+    int numSamplesRead = av_audio_fifo_read(
+        avAudioFifo_.get(),
+        reinterpret_cast<void**>(newavFrame->data),
+        samplesToRead);
+    TORCH_CHECK(
+        numSamplesRead == samplesToRead,
+        "Tried to read ",
+        samplesToRead,
+        " samples, but only read ",
+        numSamplesRead);
+
+    newavFrame->nb_samples = numSamplesRead;
+    encodeFrame(autoAVPacket, newavFrame);
+  }
+}
+
+void AudioEncoder::encodeFrame(
+    AutoAVPacket& autoAVPacket,
+    const UniqueAVFrame& avFrame) {
+  if (avFrame != nullptr) {
+    avFrame->pts = lastEncodedAVFramePts_;
+    lastEncodedAVFramePts_ += avFrame->nb_samples;
+  }
+
+  auto status = avcodec_send_frame(avCodecContext_.get(), avFrame.get());
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Error while sending frame: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  while (status >= 0) {
+    ReferenceAVPacket packet(autoAVPacket);
+    status = avcodec_receive_packet(avCodecContext_.get(), packet.get());
+    if (status == AVERROR(EAGAIN) || status == AVERROR_EOF) {
+      if (status == AVERROR_EOF) {
+        // Flush the packets that were potentially buffered by
+        // av_interleaved_write_frame(). See corresponding block in
+        // TorchAudio:
+        // https://github.com/pytorch/audio/blob/d60ce09e2c532d5bf2e05619e700ab520543465e/src/libtorio/ffmpeg/stream_writer/encoder.cpp#L21
+        status = av_interleaved_write_frame(avFormatContext_.get(), nullptr);
+        TORCH_CHECK(
+            status == AVSUCCESS,
+            "Failed to flush packet: ",
+            getFFMPEGErrorStringFromErrorCode(status));
+      }
+      return;
+    }
+    TORCH_CHECK(
+        status >= 0,
+        "Error receiving packet: ",
+        getFFMPEGErrorStringFromErrorCode(status));
+
+    packet->stream_index = streamIndex_;
+
+    status = av_interleaved_write_frame(avFormatContext_.get(), packet.get());
+    TORCH_CHECK(
+        status == AVSUCCESS,
+        "Error in av_interleaved_write_frame: ",
+        getFFMPEGErrorStringFromErrorCode(status));
+  }
+}
+
+void AudioEncoder::maybeFlushSwrBuffers(AutoAVPacket& autoAVPacket) {
+  // Similar to the decoder's method with the same name, but for encoding this
+  // time. That is, when sample conversion is involved, libswresample may have
+  // buffered some samples that we now need to flush and send to the encoder.
+  if (swrContext_ == nullptr && inSampleRate_ == outSampleRate_) {
+    return;
+  }
+  TORCH_CHECK(
+      swrContext_ != nullptr,
+      "swrContext is null, but sample rate conversion is needed. ",
+      "This is unexpected, please report on the TorchCodec bug tracker.");
+
+  int numRemainingSamples = // this is an upper bound
+      swr_get_out_samples(swrContext_.get(), 0);
+  if (numRemainingSamples == 0) {
+    return;
+  }
+
+  UniqueAVFrame avFrame = allocateAVFrame(
+      numRemainingSamples,
+      outSampleRate_,
+      outNumChannels_,
+      avCodecContext_->sample_fmt);
+  int actualNumRemainingSamples = swr_convert(
+      swrContext_.get(), avFrame->data, avFrame->nb_samples, nullptr, 0);
+  avFrame->nb_samples = actualNumRemainingSamples;
+
+  // We're potentially sending avFrame through the FIFO (if it exists), in which
+  // case we also want to flush the FIFO itself.
+  encodeFrameThroughFifo(autoAVPacket, avFrame, /*flushFifo=*/true);
+}
+
+void AudioEncoder::flushBuffers() {
+  AutoAVPacket autoAVPacket;
+  maybeFlushSwrBuffers(autoAVPacket);
+
+  encodeFrame(autoAVPacket, UniqueAVFrame(nullptr));
+}
+} // namespace facebook::torchcodec