torchcodec 0.3.0__cp313-cp313t-macosx_11_0_arm64.whl

torchcodec/_core/Encoder.cpp

@@ -0,0 +1,319 @@
+#include <sstream>
+
+#include "src/torchcodec/_core/Encoder.h"
+#include "torch/types.h"
+
+namespace facebook::torchcodec {
+
+namespace {
+
+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 waveform. 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() {}
+
+AudioEncoder::AudioEncoder(
+    const torch::Tensor wf,
+    int sampleRate,
+    std::string_view fileName,
+    std::optional<int64_t> bitRate)
+    : wf_(wf) {
+  TORCH_CHECK(
+      wf_.dtype() == torch::kFloat32,
+      "waveform must have float32 dtype, got ",
+      wf_.dtype());
+  // TODO-ENCODING check contiguity of the input wf to ensure that it is indeed
+  // planar (fltp).
+  TORCH_CHECK(
+      wf_.dim() == 2, "waveform must have 2 dimensions, got ", wf_.dim());
+
+  setFFmpegLogLevel();
+  AVFormatContext* avFormatContext = nullptr;
+  auto status = avformat_alloc_output_context2(
+      &avFormatContext, nullptr, nullptr, fileName.data());
+  TORCH_CHECK(
+      avFormatContext != nullptr,
+      "Couldn't allocate AVFormatContext. ",
+      "Check the desired extension? ",
+      getFFMPEGErrorStringFromErrorCode(status));
+  avFormatContext_.reset(avFormatContext);
+
+  // TODO-ENCODING: Should also support encoding into bytes (use
+  // AVIOBytesContext)
+  TORCH_CHECK(
+      !(avFormatContext->oformat->flags & AVFMT_NOFILE),
+      "AVFMT_NOFILE is set. We only support writing to a file.");
+  status = avio_open(&avFormatContext_->pb, fileName.data(), AVIO_FLAG_WRITE);
+  TORCH_CHECK(
+      status >= 0,
+      "avio_open failed: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  // 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);
+
+  if (bitRate.has_value()) {
+    TORCH_CHECK(*bitRate >= 0, "bit_rate=", *bitRate, " 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 = bitRate.value_or(0);
+
+  validateSampleRate(*avCodec, sampleRate);
+  avCodecContext_->sample_rate = sampleRate;
+
+  // Input waveform is expected to be FLTP. Not all encoders support FLTP, so we
+  // may need to convert the wf into a supported output sample format, which is
+  // what the `.sample_fmt` defines.
+  avCodecContext_->sample_fmt = findBestOutputSampleFormat(*avCodec);
+
+  int numChannels = static_cast<int>(wf_.sizes()[0]);
+  TORCH_CHECK(
+      // TODO-ENCODING is this even true / needed? We can probably support more
+      // with non-planar data?
+      numChannels <= AV_NUM_DATA_POINTERS,
+      "Trying to encode ",
+      numChannels,
+      " channels, but FFmpeg only supports ",
+      AV_NUM_DATA_POINTERS,
+      " channels per frame.");
+
+  setDefaultChannelLayout(avCodecContext_, numChannels);
+
+  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;
+}
+
+void AudioEncoder::encode() {
+  UniqueAVFrame avFrame(av_frame_alloc());
+  TORCH_CHECK(avFrame != nullptr, "Couldn't allocate AVFrame.");
+  //  Default to 256 like in torchaudio
+  int numSamplesAllocatedPerFrame =
+      avCodecContext_->frame_size > 0 ? avCodecContext_->frame_size : 256;
+  avFrame->nb_samples = numSamplesAllocatedPerFrame;
+  avFrame->format = AV_SAMPLE_FMT_FLTP;
+  avFrame->sample_rate = avCodecContext_->sample_rate;
+  avFrame->pts = 0;
+  setChannelLayout(avFrame, avCodecContext_);
+
+  auto status = av_frame_get_buffer(avFrame.get(), 0);
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Couldn't allocate avFrame's buffers: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  AutoAVPacket autoAVPacket;
+
+  uint8_t* pwf = static_cast<uint8_t*>(wf_.data_ptr());
+  int numSamples = static_cast<int>(wf_.sizes()[1]); // per channel
+  int numEncodedSamples = 0; // per channel
+  int numBytesPerSample = static_cast<int>(wf_.element_size());
+  int numBytesPerChannel = numSamples * numBytesPerSample;
+
+  status = avformat_write_header(avFormatContext_.get(), nullptr);
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Error in avformat_write_header: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  while (numEncodedSamples < numSamples) {
+    status = av_frame_make_writable(avFrame.get());
+    TORCH_CHECK(
+        status == AVSUCCESS,
+        "Couldn't make AVFrame writable: ",
+        getFFMPEGErrorStringFromErrorCode(status));
+
+    int numSamplesToEncode =
+        std::min(numSamplesAllocatedPerFrame, numSamples - numEncodedSamples);
+    int numBytesToEncode = numSamplesToEncode * numBytesPerSample;
+
+    for (int ch = 0; ch < wf_.sizes()[0]; ch++) {
+      std::memcpy(
+          avFrame->data[ch], pwf + ch * numBytesPerChannel, numBytesToEncode);
+    }
+    pwf += 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;
+    encodeInnerLoop(autoAVPacket, avFrame);
+
+    avFrame->pts += static_cast<int64_t>(numSamplesToEncode);
+    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));
+}
+
+void AudioEncoder::encodeInnerLoop(
+    AutoAVPacket& autoAVPacket,
+    const UniqueAVFrame& srcAVFrame) {
+  bool mustConvert =
+      (avCodecContext_->sample_fmt != AV_SAMPLE_FMT_FLTP &&
+       srcAVFrame != nullptr);
+  UniqueAVFrame convertedAVFrame;
+  if (mustConvert) {
+    if (!swrContext_) {
+      swrContext_.reset(createSwrContext(
+          avCodecContext_,
+          AV_SAMPLE_FMT_FLTP,
+          avCodecContext_->sample_fmt,
+          srcAVFrame->sample_rate, // No sample rate conversion
+          srcAVFrame->sample_rate));
+    }
+    convertedAVFrame = convertAudioAVFrameSampleFormatAndSampleRate(
+        swrContext_,
+        srcAVFrame,
+        avCodecContext_->sample_fmt,
+        srcAVFrame->sample_rate, // No sample rate conversion
+        srcAVFrame->sample_rate);
+    TORCH_CHECK(
+        convertedAVFrame->nb_samples == srcAVFrame->nb_samples,
+        "convertedAVFrame->nb_samples=",
+        convertedAVFrame->nb_samples,
+        " differs from ",
+        "srcAVFrame->nb_samples=",
+        srcAVFrame->nb_samples,
+        "This is unexpected, please report on the TorchCodec bug tracker.");
+  }
+  const UniqueAVFrame& avFrame = mustConvert ? convertedAVFrame : srcAVFrame;
+
+  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) {
+      // TODO-ENCODING this is from TorchAudio, probably needed, but not sure.
+      //   if (status == AVERROR_EOF) {
+      //     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::flushBuffers() {
+  // We flush the main FFmpeg buffers, but not swresample buffers. Flushing
+  // swresample is only necessary when converting sample rates, which we don't
+  // do for encoding.
+  AutoAVPacket autoAVPacket;
+  encodeInnerLoop(autoAVPacket, UniqueAVFrame(nullptr));
+}
+} // namespace facebook::torchcodec

torchcodec/_core/Encoder.h

@@ -0,0 +1,39 @@
+#pragma once
+#include <torch/types.h>
+#include "src/torchcodec/_core/FFMPEGCommon.h"
+
+namespace facebook::torchcodec {
+class AudioEncoder {
+ public:
+  ~AudioEncoder();
+
+  // TODO-ENCODING: document in public docs that bit_rate value is only
+  // best-effort, matching to the closest supported bit_rate. I.e. passing 1 is
+  // like passing 0, which results in choosing the minimum supported bit rate.
+  // Passing 44_100 could result in output being 44000 if only 44000 is
+  // supported.
+  AudioEncoder(
+      const torch::Tensor wf,
+      // The *output* sample rate. We can't really decide for the user what it
+      // should be. Particularly, the sample rate of the input waveform should
+      // match this, and that's up to the user. If sample rates don't match,
+      // encoding will still work but audio will be distorted.
+      int sampleRate,
+      std::string_view fileName,
+      std::optional<int64_t> bitRate = std::nullopt);
+  void encode();
+
+ private:
+  void encodeInnerLoop(
+      AutoAVPacket& autoAVPacket,
+      const UniqueAVFrame& srcAVFrame);
+  void flushBuffers();
+
+  UniqueEncodingAVFormatContext avFormatContext_;
+  UniqueAVCodecContext avCodecContext_;
+  int streamIndex_;
+  UniqueSwrContext swrContext_;
+
+  const torch::Tensor wf_;
+};
+} // namespace facebook::torchcodec

torchcodec/_core/FFMPEGCommon.cpp

@@ -0,0 +1,264 @@
+// 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/FFMPEGCommon.h"
+
+#include <c10/util/Exception.h>
+
+namespace facebook::torchcodec {
+
+AutoAVPacket::AutoAVPacket() : avPacket_(av_packet_alloc()) {
+  TORCH_CHECK(avPacket_ != nullptr, "Couldn't allocate avPacket.");
+}
+
+AutoAVPacket::~AutoAVPacket() {
+  av_packet_free(&avPacket_);
+}
+
+ReferenceAVPacket::ReferenceAVPacket(AutoAVPacket& shared)
+    : avPacket_(shared.avPacket_) {}
+
+ReferenceAVPacket::~ReferenceAVPacket() {
+  av_packet_unref(avPacket_);
+}
+
+AVPacket* ReferenceAVPacket::get() {
+  return avPacket_;
+}
+
+AVPacket* ReferenceAVPacket::operator->() {
+  return avPacket_;
+}
+
+AVCodecOnlyUseForCallingAVFindBestStream
+makeAVCodecOnlyUseForCallingAVFindBestStream(const AVCodec* codec) {
+#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(59, 18, 100)
+  return const_cast<AVCodec*>(codec);
+#else
+  return codec;
+#endif
+}
+
+std::string getFFMPEGErrorStringFromErrorCode(int errorCode) {
+  char errorBuffer[AV_ERROR_MAX_STRING_SIZE] = {0};
+  av_strerror(errorCode, errorBuffer, AV_ERROR_MAX_STRING_SIZE);
+  return std::string(errorBuffer);
+}
+
+int64_t getDuration(const UniqueAVFrame& avFrame) {
+#if LIBAVUTIL_VERSION_MAJOR < 58
+  return avFrame->pkt_duration;
+#else
+  return avFrame->duration;
+#endif
+}
+
+int getNumChannels(const UniqueAVFrame& avFrame) {
+#if LIBAVFILTER_VERSION_MAJOR > 8 || \
+    (LIBAVFILTER_VERSION_MAJOR == 8 && LIBAVFILTER_VERSION_MINOR >= 44)
+  return avFrame->ch_layout.nb_channels;
+#else
+  return av_get_channel_layout_nb_channels(avFrame->channel_layout);
+#endif
+}
+
+int getNumChannels(const UniqueAVCodecContext& avCodecContext) {
+#if LIBAVFILTER_VERSION_MAJOR > 8 || \
+    (LIBAVFILTER_VERSION_MAJOR == 8 && LIBAVFILTER_VERSION_MINOR >= 44)
+  return avCodecContext->ch_layout.nb_channels;
+#else
+  return avCodecContext->channels;
+#endif
+}
+
+void setDefaultChannelLayout(
+    UniqueAVCodecContext& avCodecContext,
+    int numChannels) {
+#if LIBAVFILTER_VERSION_MAJOR > 7 // FFmpeg > 4
+  AVChannelLayout channel_layout;
+  av_channel_layout_default(&channel_layout, numChannels);
+  avCodecContext->ch_layout = channel_layout;
+
+#else
+  uint64_t channel_layout = av_get_default_channel_layout(numChannels);
+  avCodecContext->channel_layout = channel_layout;
+  avCodecContext->channels = numChannels;
+#endif
+}
+
+void setChannelLayout(
+    UniqueAVFrame& dstAVFrame,
+    const UniqueAVCodecContext& avCodecContext) {
+#if LIBAVFILTER_VERSION_MAJOR > 7 // FFmpeg > 4
+  auto status = av_channel_layout_copy(
+      &dstAVFrame->ch_layout, &avCodecContext->ch_layout);
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Couldn't copy channel layout to avFrame: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+#else
+  dstAVFrame->channel_layout = avCodecContext->channel_layout;
+  dstAVFrame->channels = avCodecContext->channels;
+
+#endif
+}
+
+void setChannelLayout(
+    UniqueAVFrame& dstAVFrame,
+    const UniqueAVFrame& srcAVFrame) {
+#if LIBAVFILTER_VERSION_MAJOR > 7 // FFmpeg > 4
+  dstAVFrame->ch_layout = srcAVFrame->ch_layout;
+#else
+  dstAVFrame->channel_layout = srcAVFrame->channel_layout;
+#endif
+}
+
+SwrContext* createSwrContext(
+    UniqueAVCodecContext& avCodecContext,
+    AVSampleFormat sourceSampleFormat,
+    AVSampleFormat desiredSampleFormat,
+    int sourceSampleRate,
+    int desiredSampleRate) {
+  SwrContext* swrContext = nullptr;
+  int status = AVSUCCESS;
+#if LIBAVFILTER_VERSION_MAJOR > 7 // FFmpeg > 4
+  AVChannelLayout layout = avCodecContext->ch_layout;
+  status = swr_alloc_set_opts2(
+      &swrContext,
+      &layout,
+      desiredSampleFormat,
+      desiredSampleRate,
+      &layout,
+      sourceSampleFormat,
+      sourceSampleRate,
+      0,
+      nullptr);
+
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Couldn't create SwrContext: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+#else
+  int64_t layout = static_cast<int64_t>(avCodecContext->channel_layout);
+  swrContext = swr_alloc_set_opts(
+      nullptr,
+      layout,
+      desiredSampleFormat,
+      desiredSampleRate,
+      layout,
+      sourceSampleFormat,
+      sourceSampleRate,
+      0,
+      nullptr);
+#endif
+
+  TORCH_CHECK(swrContext != nullptr, "Couldn't create swrContext");
+  status = swr_init(swrContext);
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Couldn't initialize SwrContext: ",
+      getFFMPEGErrorStringFromErrorCode(status),
+      ". If the error says 'Invalid argument', it's likely that you are using "
+      "a buggy FFmpeg version. FFmpeg4 is known to fail here in some "
+      "valid scenarios. Try to upgrade FFmpeg?");
+  return swrContext;
+}
+
+UniqueAVFrame convertAudioAVFrameSampleFormatAndSampleRate(
+    const UniqueSwrContext& swrContext,
+    const UniqueAVFrame& srcAVFrame,
+    AVSampleFormat desiredSampleFormat,
+    int sourceSampleRate,
+    int desiredSampleRate) {
+  UniqueAVFrame convertedAVFrame(av_frame_alloc());
+  TORCH_CHECK(
+      convertedAVFrame,
+      "Could not allocate frame for sample format conversion.");
+
+  setChannelLayout(convertedAVFrame, srcAVFrame);
+  convertedAVFrame->format = static_cast<int>(desiredSampleFormat);
+  convertedAVFrame->sample_rate = desiredSampleRate;
+  if (sourceSampleRate != desiredSampleRate) {
+    // Note that this is an upper bound on the number of output samples.
+    // `swr_convert()` will likely not fill convertedAVFrame with that many
+    // samples if sample rate conversion is needed. It will buffer the last few
+    // ones because those require future samples. That's also why we reset
+    // nb_samples after the call to `swr_convert()`.
+    // We could also use `swr_get_out_samples()` to determine the number of
+    // output samples, but empirically `av_rescale_rnd()` seems to provide a
+    // tighter bound.
+    convertedAVFrame->nb_samples = av_rescale_rnd(
+        swr_get_delay(swrContext.get(), sourceSampleRate) +
+            srcAVFrame->nb_samples,
+        desiredSampleRate,
+        sourceSampleRate,
+        AV_ROUND_UP);
+  } else {
+    convertedAVFrame->nb_samples = srcAVFrame->nb_samples;
+  }
+
+  auto status = av_frame_get_buffer(convertedAVFrame.get(), 0);
+  TORCH_CHECK(
+      status == AVSUCCESS,
+      "Could not allocate frame buffers for sample format conversion: ",
+      getFFMPEGErrorStringFromErrorCode(status));
+
+  auto numConvertedSamples = swr_convert(
+      swrContext.get(),
+      convertedAVFrame->data,
+      convertedAVFrame->nb_samples,
+      static_cast<const uint8_t**>(
+          const_cast<const uint8_t**>(srcAVFrame->data)),
+      srcAVFrame->nb_samples);
+  // numConvertedSamples can be 0 if we're downsampling by a great factor and
+  // the first frame doesn't contain a lot of samples. It should be handled
+  // properly by the caller.
+  TORCH_CHECK(
+      numConvertedSamples >= 0,
+      "Error in swr_convert: ",
+      getFFMPEGErrorStringFromErrorCode(numConvertedSamples));
+
+  // See comment above about nb_samples
+  convertedAVFrame->nb_samples = numConvertedSamples;
+
+  return convertedAVFrame;
+}
+
+void setFFmpegLogLevel() {
+  auto logLevel = AV_LOG_QUIET;
+  const char* logLevelEnvPtr = std::getenv("TORCHCODEC_FFMPEG_LOG_LEVEL");
+  if (logLevelEnvPtr != nullptr) {
+    std::string logLevelEnv(logLevelEnvPtr);
+    if (logLevelEnv == "QUIET") {
+      logLevel = AV_LOG_QUIET;
+    } else if (logLevelEnv == "PANIC") {
+      logLevel = AV_LOG_PANIC;
+    } else if (logLevelEnv == "FATAL") {
+      logLevel = AV_LOG_FATAL;
+    } else if (logLevelEnv == "ERROR") {
+      logLevel = AV_LOG_ERROR;
+    } else if (logLevelEnv == "WARNING") {
+      logLevel = AV_LOG_WARNING;
+    } else if (logLevelEnv == "INFO") {
+      logLevel = AV_LOG_INFO;
+    } else if (logLevelEnv == "VERBOSE") {
+      logLevel = AV_LOG_VERBOSE;
+    } else if (logLevelEnv == "DEBUG") {
+      logLevel = AV_LOG_DEBUG;
+    } else if (logLevelEnv == "TRACE") {
+      logLevel = AV_LOG_TRACE;
+    } else {
+      TORCH_CHECK(
+          false,
+          "Invalid TORCHCODEC_FFMPEG_LOG_LEVEL: ",
+          logLevelEnv,
+          ". Use e.g. 'QUIET', 'PANIC', 'VERBOSE', etc.");
+    }
+  }
+  av_log_set_level(logLevel);
+}
+
+} // namespace facebook::torchcodec