Sync opus codec to upstream git

Sync opus codec to upstream commit
02fed471a4568852d6618e041c4f2af0d7730ee2 (August 30 2013)

This brings in a lot of optimizations but also makes the diff
between our codec and the upstream much smaller as most of our
optimizations have been upstreamed or supeceded.

Speedups across the board for CELT mode files:

        64kbps      128kbps
H300    9.82MHz     15.48MHz
c200	4.86MHz     9.63MHz
fuze v1 10.32MHz    15.92MHz

For the silk mode test file (16kbps) arm targets get a speedup
of about 2MHz while the H300 is 7.8MHz slower, likely because it's
now using the pseudostack more rather than the real stack which
is in iram. Patches to get around that are upcomming.

Change-Id: Ifecf963e461c51ac42e09dac1e91bc4bc3b12fa3

1 files changed, 52 insertions, 14 deletions

diff --git a/lib/rbcodec/codecs/libopus/opus.h b/lib/rbcodec/codecs/libopus/opus.h
index c242fec0e7..38817b405a 100644
--- a/lib/rbcodec/codecs/libopus/opus.h
+++ b/lib/rbcodec/codecs/libopus/opus.h
@@ -136,10 +136,11 @@ extern "C" {
   * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
   * <li>frame_size is the duration of the frame in samples (per channel)</li>
   * <li>packet is the byte array to which the compressed data is written</li>
-  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended)</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
   * </ul>
   *
-  * opus_encode() and opus_encode_frame() return the number of bytes actually written to the packet.
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
   * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
   * is 1 byte, then the packet does not need to be transmitted (DTX).
   *
@@ -252,9 +253,10 @@ OPUS_EXPORT int opus_encoder_init(
   *                                                 memory for the output
   *                                                 payload. This may be
   *                                                 used to impose an upper limit on
-  *                                                 the variable bitrate, but should
+  *                                                 the instant bitrate, but should
   *                                                 not be used as the only bitrate
-  *                                                 control.
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
   * @returns The length of the encoded packet (in bytes) on success or a
   *          negative error code (see @ref opus_errorcodes) on failure.
   */
@@ -292,9 +294,10 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
   *                                                 memory for the output
   *                                                 payload. This may be
   *                                                 used to impose an upper limit on
-  *                                                 the variable bitrate, but should
+  *                                                 the instant bitrate, but should
   *                                                 not be used as the only bitrate
-  *                                                 control.
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
   * @returns The length of the encoded packet (in bytes) on success or a
   *          negative error code (see @ref opus_errorcodes) on failure.
   */
@@ -340,7 +343,7 @@ OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NON
   * where
   * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
   * @li channels is the number of channels (1 or 2)
-  * @li error will hold the error code in case or failure (or #OPUS_OK on success)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
   * @li the return value is a newly created decoder state to be used for decoding
   *
   * While opus_decoder_create() allocates memory for the state, it's also possible
@@ -371,7 +374,7 @@ OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NON
   * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
   *
   * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
-  * If that value is negative, then an error has occured. This can occur if the packet is corrupted or if the audio
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
   * buffer is too small to hold the decoded audio.
   *
   * Opus is a stateful codec with overlapping blocks and as a result Opus
@@ -447,8 +450,11 @@ OPUS_EXPORT int opus_decoder_init(
   * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
   *  is frame_size*channels*sizeof(opus_int16)
   * @param [in] frame_size Number of samples per channel of available space in \a pcm.
-  *  If this is less than the maximum frame size (120 ms), this function will
-  *  not be capable of decoding some packets.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
   * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
   *  decoded. If no such data is available, the frame is decoded as if it were lost.
   * @returns Number of decoded samples or @ref opus_errorcodes
@@ -468,8 +474,12 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
   * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
   * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
   *  is frame_size*channels*sizeof(float)
-  * @param [in] frame_size Number of samples per channel of available space in *pcm,
-  *  if less than the maximum frame size (120ms) some frames can not be decoded
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
   * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
   *  decoded. If no such data is available the frame is decoded as if it were lost.
   * @returns Number of decoded samples or @ref opus_errorcodes
@@ -510,7 +520,7 @@ OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
   * @param [in] len <tt>opus_int32</tt>: size of data
   * @param [out] out_toc <tt>char*</tt>: TOC pointer
   * @param [out] frames <tt>char*[48]</tt> encapsulated frames
-  * @param [out] size <tt>short[48]</tt> sizes of the encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
   * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
   * @returns number of frames
   */
@@ -519,7 +529,7 @@ OPUS_EXPORT int opus_packet_parse(
    opus_int32 len,
    unsigned char *out_toc,
    const unsigned char *frames[48],
-   short size[48],
+   opus_int16 size[48],
    int *payload_offset
 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
 
@@ -556,18 +566,46 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsign
   * @param [in] packet <tt>char*</tt>: Opus packet
   * @param [in] len <tt>opus_int32</tt>: Length of packet
   * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
   * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
   */
 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
 
 /** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
   * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
   * @param [in] packet <tt>char*</tt>: Opus packet
   * @param [in] len <tt>opus_int32</tt>: Length of packet
   * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
   * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
   */
 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
 /**@}*/
 
 /** @defgroup opus_repacketizer Repacketizer