Merge branch 'preserve-encoding'

* preserve-encoding:
  README.adoc: minor wording fix.
  Move (not really portable) builtin bswap function into header.
  README.adoc: document float / double / 32 bit raw stream format
  raw-format-test.sh: also validate 8-bit raw input/output
  RawConverter: fix sign flip bug for big endian encoding.
  raw-format-test.sh: enable more test cases for unsigned encoding.
  Make testrawconverter fail if int16/uint16 values don't match.
  Extend testrawconverter to check 16-bit conversion values.
  TESTS: add raw-format-test (using ffmpeg)
  Make HLS work again with the encoding related changes.
  SFInputStream: handle more soundfile subformat types
	This ensures that libsndfile input .ogg and .mp3 files are treated as 24-bit
	again (like it used to be in previous audiowmark versions).
  Add wave subformat test.
  wav-pipe-test.sh: update test to handle 24:24 and 32:32 in:out bit checks.
  For raw streams, use float / double as encoding name.
	Changing raw bit depth is no longer supported for float / double encoding.
  Support 8 bit format in WavPipeInputStream.
  StdoutWavOutputStream: preserve encoding rather than optimizing for speed.
  Support float/double input for WavPipeInputStream.
  Float encoding fixes.
  Write floating point output in SFOutputStream for Encoding::FLOAT.
  Implement Encoding::FLOAT conversions in RawConverter.
  Fix testwavformat utility.
  Add Encoding accessor for streams/formats to represent float/double input.
  Add utility to test sndfile encoding.

Signed-off-by: Stefan Westerfeld <[email protected]>

Author: swesterfeld

README.adoc

@@ -512,9 +512,9 @@ so this parameter must always be specified for raw streams.
 --raw-output-bits <bits>::
 --raw-bits <bits>::
 
-The options can be used to set the input number of bits, the output number
-of bits or both. The number of bits can either be `16` or `24`. The default
-number of bits is `16`.
+The options can be used to set the input number of bits, the output number of
+bits or both. The number of bits can be `16`, `24` or `32`. The default number
+of bits is `16`.
 
 --raw-input-endian <endian>::
 --raw-output-endian <endian>::
@@ -528,9 +528,10 @@ endianness is `little`.
 --raw-output-encoding <encoding>::
 --raw-encoding <encoding>::
 
-These options can be used to set the input/output encoding or both.
-The <encoding> parameter can either be `signed` or `unsigned`. The
-default encoding is `signed`.
+These options can be used to set the input/output encoding or both.  The
+<encoding> parameter can be `signed`, `unsigned`, `float` or `double`. The
+default encoding is `signed`. Using `float` (or `double`) encoding
+automatically sets the number of bits to `32` (or `64`).
 
 --raw-channels <channels>::
 

src/Makefile.am

@@ -16,7 +16,7 @@ audiowmark_SOURCES = audiowmark.cc $(COMMON_SRC)
 audiowmark_LDFLAGS = $(COMMON_LIBS)
 
 noinst_PROGRAMS = testconvcode testrandom testmp3 teststream testlimiter testshortcode testmpegts testthreadpool \
-		  testrawconverter
+		  testrawconverter testwavformat
 
 testconvcode_SOURCES = testconvcode.cc $(COMMON_SRC)
 testconvcode_LDFLAGS = $(COMMON_LIBS)
@@ -45,6 +45,9 @@ testthreadpool_LDFLAGS = $(COMMON_LIBS)
 testrawconverter_SOURCES = testrawconverter.cc $(COMMON_SRC)
 testrawconverter_LDFLAGS = $(COMMON_LIBS)
 
+testwavformat_SOURCES = testwavformat.cc $(COMMON_SRC)
+testwavformat_LDFLAGS = $(COMMON_LIBS)
+
 if COND_WITH_FFMPEG
 COMMON_SRC += hlsoutputstream.cc hlsoutputstream.hh
 

src/audiostream.cc

@@ -80,7 +80,7 @@ AudioInputStream::create (const string& filename, Error& err)
 }
 
 std::unique_ptr<AudioOutputStream>
-AudioOutputStream::create (const string& filename, int n_channels, int sample_rate, int bit_depth, size_t n_frames, Error& err)
+AudioOutputStream::create (const string& filename, int n_channels, int sample_rate, int bit_depth, Encoding encoding, size_t n_frames, Error& err)
 {
   std::unique_ptr<AudioOutputStream> out_stream;
 
@@ -98,7 +98,7 @@ AudioOutputStream::create (const string& filename, int n_channels, int sample_ra
 
       StdoutWavOutputStream *swstream = new StdoutWavOutputStream();
       out_stream.reset (swstream);
-      err = swstream->open (n_channels, sample_rate, bit_depth, n_frames, wav_pipe);
+      err = swstream->open (n_channels, sample_rate, bit_depth, encoding, n_frames, wav_pipe);
       if (err)
         return nullptr;
     }
@@ -113,7 +113,7 @@ AudioOutputStream::create (const string& filename, int n_channels, int sample_ra
 
       SFOutputStream *sfostream = new SFOutputStream();
       out_stream.reset (sfostream);
-      err = sfostream->open (filename, n_channels, sample_rate, bit_depth, out_format);
+      err = sfostream->open (filename, n_channels, sample_rate, bit_depth, encoding, out_format);
       if (err)
         return nullptr;
     }

src/audiostream.hh

@@ -22,6 +22,12 @@
 #include <memory>
 #include "utils.hh"
 
+enum class Encoding {
+  SIGNED,
+  UNSIGNED,
+  FLOAT
+};
+
 class AudioStream
 {
 public:
@@ -40,6 +46,7 @@ public:
   // for streams that do not know the number of frames in advance (i.e. raw input stream)
   static constexpr size_t N_FRAMES_UNKNOWN = ~size_t (0);
   virtual size_t n_frames() const = 0;
+  virtual Encoding encoding() const = 0;
 
   virtual Error read_frames (std::vector<float>& samples, size_t count) = 0;
 };
@@ -48,7 +55,7 @@ class AudioOutputStream : public AudioStream
 {
 public:
   static std::unique_ptr<AudioOutputStream> create (const std::string& filename,
-    int n_channels, int sample_rate, int bit_depth, size_t n_frames, Error& err);
+    int n_channels, int sample_rate, int bit_depth, Encoding encoding, size_t n_frames, Error& err);
 
   virtual Error write_frames (const std::vector<float>& frames) = 0;
   virtual Error close() = 0;

src/audiowmark.cc

@@ -137,15 +137,39 @@ parse_endian (const string& str)
   exit (1);
 }
 
-RawFormat::Encoding
-parse_encoding (const string& str)
+void
+parse_encoding (const string& str, RawFormat& fmt)
 {
   if (str == "signed")
-    return RawFormat::Encoding::SIGNED;
-  if (str == "unsigned")
-    return RawFormat::Encoding::UNSIGNED;
-  error ("audiowmark: unsupported encoding '%s'\n", str.c_str());
-  exit (1);
+    fmt.set_encoding (Encoding::SIGNED);
+  else if (str == "unsigned")
+    fmt.set_encoding (Encoding::UNSIGNED);
+  else if (str == "float")
+    {
+      fmt.set_encoding (Encoding::FLOAT);
+      fmt.set_bit_depth (32);
+    }
+  else if (str == "double")
+    {
+      fmt.set_encoding (Encoding::FLOAT);
+      fmt.set_bit_depth (64);
+    }
+  else
+    {
+      error ("audiowmark: unsupported encoding '%s'\n", str.c_str());
+      exit (1);
+    }
+}
+
+void
+update_raw_bits (RawFormat& fmt, int bits)
+{
+  if (fmt.encoding() == Encoding::FLOAT)
+    {
+      error ("audiowmark: bit depth can not be changed for float / double encoding\n");
+      exit (1);
+    }
+  fmt.set_bit_depth (bits);
 }
 
 int
@@ -722,19 +746,6 @@ parse_add_options (ArgParser& ap)
     {
       Params::input_format = Params::output_format = parse_format (s);
     }
-  if (ap.parse_opt ("--raw-input-bits", i))
-    {
-      Params::raw_input_format.set_bit_depth (i);
-    }
-  if (ap.parse_opt ("--raw-output-bits", i))
-    {
-      Params::raw_output_format.set_bit_depth (i);
-    }
-  if (ap.parse_opt ("--raw-bits", i))
-    {
-      Params::raw_input_format.set_bit_depth (i);
-      Params::raw_output_format.set_bit_depth (i);
-    }
   if (ap.parse_opt ( "--raw-input-endian", s))
     {
       auto e = parse_endian (s);
@@ -753,19 +764,29 @@ parse_add_options (ArgParser& ap)
     }
   if (ap.parse_opt ("--raw-input-encoding", s))
     {
-      auto e = parse_encoding (s);
-      Params::raw_input_format.set_encoding (e);
+      parse_encoding (s, Params::raw_input_format);
     }
   if (ap.parse_opt ("--raw-output-encoding", s))
     {
-      auto e = parse_encoding (s);
-      Params::raw_output_format.set_encoding (e);
+      parse_encoding (s, Params::raw_output_format);
     }
   if (ap.parse_opt ("--raw-encoding", s))
     {
-      auto e = parse_encoding (s);
-      Params::raw_input_format.set_encoding (e);
-      Params::raw_output_format.set_encoding (e);
+      parse_encoding (s, Params::raw_input_format);
+      parse_encoding (s, Params::raw_output_format);
+    }
+  if (ap.parse_opt ("--raw-input-bits", i))
+    {
+      update_raw_bits (Params::raw_input_format, i);
+    }
+  if (ap.parse_opt ("--raw-output-bits", i))
+    {
+      update_raw_bits (Params::raw_output_format, i);
+    }
+  if (ap.parse_opt ("--raw-bits", i))
+    {
+      update_raw_bits (Params::raw_input_format, i);
+      update_raw_bits (Params::raw_output_format, i);
     }
   if (ap.parse_opt ("--raw-channels", i))
     {

src/hls.cc

@@ -568,6 +568,7 @@ hls_prepare (const string& in_dir, const string& out_dir, const string& filename
       SFOutputStream out_stream;
       err = out_stream.open (&full_flac_mem,
                              audio_master_data.n_channels(), audio_master_data.sample_rate(), audio_master_data.bit_depth(),
+                             Encoding::SIGNED, /* flac does not support floating point audio */
                              SFOutputStream::OutFormat::FLAC);
       if (err)
         {

src/mp3inputstream.cc

@@ -181,6 +181,12 @@ MP3InputStream::bit_depth() const
   return 24; /* mp3 decoder is running on floats */
 }
 
+Encoding
+MP3InputStream::encoding() const
+{
+  return Encoding::SIGNED; /* should use 24-bit pcm output */
+}
+
 int
 MP3InputStream::sample_rate() const
 {

src/mp3inputstream.hh

@@ -51,6 +51,7 @@ public:
   int     sample_rate() const override;
   int     n_channels()  const override;
   size_t  n_frames() const override;
+  Encoding encoding() const override;
 
   static bool detect (const std::string& filename);
 };