Refactor _compat.save function

When dealing with vorbis format, FFmpeg expects "ogg" container/extension
with "vorbis" encoder. It does not recognize "vorbis" container/extension.

libsox-based torchaudio I/O used to handle vorbis extension.

This commit refactors the internal of save argument and adds support to vorbis
as extension for those cases with FFmpeg backend.

This also fixes the case of mp3

#3385

test/torchaudio_unittest/backend/dispatcher/ffmpeg/save_test.py

@@ -8,7 +8,7 @@
 import torch
 from parameterized import parameterized
 from torchaudio._backend.utils import get_save_func
-from torchaudio.io._compat import _get_encoder, _get_encoder_format
+from torchaudio.io._compat import _parse_save_args
 
 from torchaudio_unittest.backend.dispatcher.sox.common import get_enc_params, name_func
 from torchaudio_unittest.common_utils import (
@@ -24,12 +24,14 @@
 )
 
 
-def _convert_audio_file(src_path, dst_path, format=None, acodec=None):
-    command = ["ffmpeg", "-y", "-i", src_path, "-strict", "-2"]
-    if format:
-        command += ["-sample_fmt", format]
-    if acodec:
-        command += ["-acodec", acodec]
+def _convert_audio_file(src_path, dst_path, muxer=None, encoder=None, sample_fmt=None):
+    command = ["ffmpeg", "-hide_banner", "-y", "-i", src_path, "-strict", "-2"]
+    if muxer:
+        command += ["-f", muxer]
+    if encoder:
+        command += ["-acodec", encoder]
+    if sample_fmt:
+        command += ["-sample_fmt", sample_fmt]
     command += [dst_path]
     print(" ".join(command), file=sys.stderr)
     subprocess.run(command, check=True)
@@ -97,11 +99,15 @@ def assert_save_consistency(
         data = get_wav_data(src_dtype, num_channels, normalize=False, num_frames=num_frames)
         save_wav(src_path, data, sample_rate)
 
+        import torchaudio
+        torchaudio.utils.ffmpeg_utils.set_log_level(32)
         # 2.1. Convert the original wav to target format with torchaudio
         data = load_wav(src_path, normalize=False)[0]
         if test_mode == "path":
-            self._save(tgt_path, data, sample_rate, encoding=encoding, bits_per_sample=bits_per_sample)
+            ext = format
+            self._save(tgt_path, data, sample_rate, format=format, encoding=encoding, bits_per_sample=bits_per_sample)
         elif test_mode == "fileobj":
+            ext = None
             with open(tgt_path, "bw") as file_:
                 self._save(
                     file_,
@@ -113,6 +119,7 @@ def assert_save_consistency(
                 )
         elif test_mode == "bytesio":
             file_ = io.BytesIO()
+            ext = None
             self._save(
                 file_,
                 data,
@@ -127,16 +134,15 @@ def assert_save_consistency(
         else:
             raise ValueError(f"Unexpected test mode: {test_mode}")
         # 2.2. Convert the target format to wav with ffmpeg
-        _convert_audio_file(tgt_path, tst_path, acodec="pcm_f32le")
+        _convert_audio_file(tgt_path, tst_path, encoder="pcm_f32le")
         # 2.3. Load with SciPy
         found = load_wav(tst_path, normalize=False)[0]
 
         # 3.1. Convert the original wav to target format with ffmpeg
-        acodec = _get_encoder(data.dtype, format, encoding, bits_per_sample)
-        sample_fmt = _get_encoder_format(format, bits_per_sample)
-        _convert_audio_file(src_path, sox_path, acodec=acodec, format=sample_fmt)
+        muxer, encoder, sample_fmt = _parse_save_args(ext, format, encoding, bits_per_sample)
+        _convert_audio_file(src_path, sox_path, muxer=muxer, encoder=encoder, sample_fmt=sample_fmt)
         # 3.2. Convert the target format to wav with ffmpeg
-        _convert_audio_file(sox_path, ref_path, acodec="pcm_f32le")
+        _convert_audio_file(sox_path, ref_path, encoder="pcm_f32le")
         # 3.3. Load with SciPy
         expected = load_wav(ref_path, normalize=False)[0]
 

torchaudio/io/_compat.py

@@ -102,7 +102,8 @@ def load_audio_fileobj(
     format: Optional[str] = None,
     buffer_size: int = 4096,
 ) -> Tuple[torch.Tensor, int]:
-    s = torchaudio.lib._torchaudio_ffmpeg.StreamReaderFileObj(src, format, None, buffer_size)
+    file_format = "ogg" if format == "vorbis" else format
+    s = torchaudio.lib._torchaudio_ffmpeg.StreamReaderFileObj(src, file_format, None, buffer_size)
     sample_rate = int(s.get_src_stream_info(s.find_best_audio_stream()).sample_rate)
     filter = _get_load_filter(frame_offset, num_frames, convert)
     waveform = _load_audio_fileobj(s, filter, channels_first)
@@ -131,7 +132,7 @@ def _native_endianness() -> str:
         return "be"
 
 
-def _get_encoder_for_wav(dtype: torch.dtype, encoding: str, bits_per_sample: int) -> str:
+def _get_encoder_for_wav(encoding: str, bits_per_sample: int) -> str:
     if bits_per_sample not in {None, 8, 16, 24, 32, 64}:
         raise ValueError(f"Invalid bits_per_sample {bits_per_sample} for WAV encoding.")
     endianness = _native_endianness()
@@ -148,49 +149,80 @@ def _get_encoder_for_wav(dtype: torch.dtype, encoding: str, bits_per_sample: int
         if bits_per_sample == 8:
             raise ValueError("For WAV signed PCM, 8-bit encoding is not supported.")
         return f"pcm_s{bits_per_sample}{endianness}"
-    elif encoding == "PCM_U":
+    if encoding == "PCM_U":
         if bits_per_sample in (None, 8):
             return "pcm_u8"
         raise ValueError("For WAV unsigned PCM, only 8-bit encoding is supported.")
-    elif encoding == "PCM_F":
+    if encoding == "PCM_F":
         if not bits_per_sample:
             bits_per_sample = 32
         if bits_per_sample in (32, 64):
             return f"pcm_f{bits_per_sample}{endianness}"
         raise ValueError("For WAV float PCM, only 32- and 64-bit encodings are supported.")
-    elif encoding == "ULAW":
+    if encoding == "ULAW":
         if bits_per_sample in (None, 8):
             return "pcm_mulaw"
         raise ValueError("For WAV PCM mu-law, only 8-bit encoding is supported.")
-    elif encoding == "ALAW":
+    if encoding == "ALAW":
         if bits_per_sample in (None, 8):
             return "pcm_alaw"
         raise ValueError("For WAV PCM A-law, only 8-bit encoding is supported.")
     raise ValueError(f"WAV encoding {encoding} is not supported.")
 
 
-def _get_encoder(dtype: torch.dtype, format: str, encoding: str, bits_per_sample: int) -> str:
-    if format == "wav":
-        return _get_encoder_for_wav(dtype, encoding, bits_per_sample)
-    if format == "flac":
-        return "flac"
-    if format in ("ogg", "vorbis"):
-        if encoding or bits_per_sample:
-            raise ValueError("ogg/vorbis does not support encoding/bits_per_sample.")
-        return "vorbis"
-    return format
+def _get_flac_sample_fmt(bps):
+    if bps is None or bps == 16:
+        return "s16"
+    if bps == 24:
+        return "s32"
+    raise ValueError(f"FLAC only supports bits_per_sample values of 16 and 24 ({bps} specified).")
 
 
-def _get_encoder_format(format: str, bits_per_sample: Optional[int]) -> str:
-    if format == "flac":
-        if not bits_per_sample:
-            return "s16"
-        if bits_per_sample == 24:
-            return "s32"
-        if bits_per_sample == 16:
-            return "s16"
-        raise ValueError(f"FLAC only supports bits_per_sample values of 16 and 24 ({bits_per_sample} specified).")
-    return None
+def _parse_save_args(
+    ext: Optional[str],
+    format: Optional[str],
+    encoding: Optional[str],
+    bps: Optional[int],
+):
+    # torchaudio's save function accepts the followings, which do not 1to1 map
+    # to FFmpeg.
+    #
+    # - format: audio format
+    # - bits_per_sample: encoder sample format
+    # - encoding: such as PCM_U8.
+    #
+    # In FFmpeg, format is specified with the following three (and more)
+    #
+    # - muxer: could be audio format or container format.
+    # the one we passed to the constructor of StreamWriter
+    # - encoder: the audio encoder used to encode audio
+    # - encoder sample format: the format used by encoder to encode audio.
+    #
+    # If encoder sample format is different from source sample format, StreamWriter
+    # will insert a filter automatically.
+    #
+    if format == "wav" or (format is None and ext == "wav"):
+        # wav is special because it supports different encoding through encoders
+        # each encoder only supports one encoder format
+        muxer = "wav"
+        encoder = _get_encoder_for_wav(encoding, bps)
+        sample_fmt = None
+    elif format == "vorbis" or (format is None and ext == "vorbis"):
+        # FFpmeg does not recognize vorbis extension, while libsox used to do.
+        # For the sake of bakward compatibility, (and the simplicity),
+        # we support the case where users want to do save("foo.vorbis")
+        muxer = "ogg"
+        encoder = "vorbis"
+        sample_fmt = None
+    else:
+        muxer = format
+        encoder = None
+        sample_fmt = None
+        if format == "flac" or format is None and ext == "flac":
+            sample_fmt = _get_flac_sample_fmt(bps)
+        if format == "ogg" or format is None and ext == "ogg":
+            sample_fmt = _get_flac_sample_fmt(bps)
+    return muxer, encoder, sample_fmt
 
 
 # NOTE: in contrast to load_audio* and info_audio*, this function is NOT compatible with TorchScript.
@@ -204,25 +236,27 @@ def save_audio(
     bits_per_sample: Optional[int] = None,
     buffer_size: int = 4096,
 ) -> None:
+    ext = None
     if hasattr(uri, "write"):
         if format is None:
             raise RuntimeError("'format' is required when saving to file object.")
     else:
         uri = os.path.normpath(uri)
-    s = StreamWriter(uri, format=format, buffer_size=buffer_size)
-    if format is None:
-        tokens = str(uri).split(".")
-        if len(tokens) > 1:
-            format = tokens[-1].lower()
+        if tokens := str(uri).split(".")[1:]:
+            ext = tokens[-1].lower()
+
+    muxer, encoder, enc_fmt = _parse_save_args(ext, format, encoding, bits_per_sample)
 
     if channels_first:
         src = src.T
+
+    s = StreamWriter(uri, format=muxer, buffer_size=buffer_size)
     s.add_audio_stream(
         sample_rate,
         num_channels=src.size(-1),
         format=_get_sample_format(src.dtype),
-        encoder=_get_encoder(src.dtype, format, encoding, bits_per_sample),
-        encoder_format=_get_encoder_format(format, bits_per_sample),
+        encoder=encoder,
+        encoder_format=enc_fmt,
     )
     with s.open():
         s.write_audio_chunk(0, src)