Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12218 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 42 insertions, 87 deletions

diff --git a/apps/playback.c b/apps/playback.c
index ff681bb..9396438 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -1022,81 +1022,60 @@ static const char * get_codec_filename(int cod_spec)
 
 #ifdef PLAYBACK_VOICE
 
-static bool voice_pcmbuf_insert_split_callback(
-        const void *ch1, const void *ch2, size_t length)
+static bool voice_pcmbuf_insert_callback(
+        const void *ch1, const void *ch2, int count)
 {
-    const char* src[2];
-    char *dest;
-    long input_size;
-    size_t output_size;
+    const char *src[2] = { ch1, ch2 };
 
-    src[0] = ch1;
-    src[1] = ch2;
-
-    if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
-        length *= 2;    /* Length is per channel */
-
-    while (length)
+    while (count > 0)
     {
-        long est_output_size = dsp_output_size(length);
-        
-        while ((dest = pcmbuf_request_voice_buffer(est_output_size,
-                        &output_size, playing)) == NULL)
+        int out_count = dsp_output_count(count);
+        int inp_count;
+        char *dest;
+
+        while ((dest = pcmbuf_request_voice_buffer(
+                        &out_count, playing)) == NULL)
         {
             if (playing && audio_codec_loaded)
                 swap_codec();
             else
                 yield();
         }
-        
+
         /* Get the real input_size for output_size bytes, guarding
          * against resampling buffer overflows. */
-        input_size = dsp_input_size(output_size);
+        inp_count = dsp_input_count(out_count);
 
-        if (input_size <= 0) 
+        if (inp_count <= 0) 
         {
-            DEBUGF("Error: dsp_input_size(%ld=dsp_output_size(%ld))=%ld<=0\n",
-                    output_size, length, input_size);
+            DEBUGF("Error: dsp_input_count(%ld=dsp_output_count(%ld))=%ld<=0\n",
+                    out_count, count, inp_count);
             /* If this happens, there are samples of codec data that don't
              * become a number of pcm samples, and something is broken */
             return false;
         }
 
         /* Input size has grown, no error, just don't write more than length */
-        if ((size_t)input_size > length)
-            input_size = length;
+        if (inp_count > count)
+            inp_count = count;
 
-        output_size = dsp_process(dest, src, input_size);
+        out_count = dsp_process(dest, src, inp_count);
 
         if (playing)
         {
-            pcmbuf_mix_voice(output_size);
+            pcmbuf_mix_voice(out_count);
             if ((pcmbuf_usage() < 10 || pcmbuf_mix_free() < 30) &&
                     audio_codec_loaded)
                 swap_codec();
         }
         else
-            pcmbuf_write_complete(output_size);
+            pcmbuf_write_complete(out_count);
 
-        length -= input_size;
+        count -= inp_count;
     }
 
     return true;
-} /* voice_pcmbuf_insert_split_callback */
-
-static bool voice_pcmbuf_insert_callback(const char *buf, size_t length)
-{
-    /* TODO: The audiobuffer API should probably be updated, and be based on
-     *       pcmbuf_insert_split().  */
-    long real_length = length;
-
-    if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
-        length /= 2;    /* Length is per channel */
-
-    /* Second channel is only used for non-interleaved stereo. */
-    return voice_pcmbuf_insert_split_callback(buf, buf + (real_length / 2),
-        length);
-}
+} /* voice_pcmbuf_insert_callback */
 
 static void* voice_get_memory_callback(size_t *size)
 {
@@ -1321,30 +1300,22 @@ static void voice_thread(void)
 #endif /* PLAYBACK_VOICE */
 
 /* --- Codec thread --- */
-
-static bool codec_pcmbuf_insert_split_callback(
-        const void *ch1, const void *ch2, size_t length)
+static bool codec_pcmbuf_insert_callback(
+        const void *ch1, const void *ch2, int count)
 {
-    const char* src[2];
-    char *dest;
-    long input_size;
-    size_t output_size;
-
-    src[0] = ch1;
-    src[1] = ch2;
-
-    if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
-        length *= 2;    /* Length is per channel */
+    const char *src[2] = { ch1, ch2 };
 
-    while (length)
+    while (count > 0)
     {
-        long est_output_size = dsp_output_size(length);
+        int out_count = dsp_output_count(count);
+        int inp_count;
+        char *dest;
+
         /* Prevent audio from a previous track from playing */
         if (ci.new_track || ci.stop_codec)
             return true;
 
-        while ((dest = pcmbuf_request_buffer(est_output_size,
-                        &output_size)) == NULL) 
+        while ((dest = pcmbuf_request_buffer(&out_count)) == NULL) 
         {
             sleep(1);
             if (ci.seek_time || ci.new_track || ci.stop_codec)
@@ -1353,24 +1324,24 @@ static bool codec_pcmbuf_insert_split_callback(
 
         /* Get the real input_size for output_size bytes, guarding
          * against resampling buffer overflows. */
-        input_size = dsp_input_size(output_size);
+        inp_count = dsp_input_count(out_count);
 
-        if (input_size <= 0) 
+        if (inp_count <= 0) 
         {
-            DEBUGF("Error: dsp_input_size(%ld=dsp_output_size(%ld))=%ld<=0\n",
-                    output_size, length, input_size);
+            DEBUGF("Error: dsp_input_count(%ld=dsp_output_count(%ld))=%ld<=0\n",
+                    out_count, count, inp_count);
             /* If this happens, there are samples of codec data that don't
              * become a number of pcm samples, and something is broken */
             return false;
         }
 
         /* Input size has grown, no error, just don't write more than length */
-        if ((size_t)input_size > length)
-            input_size = length;
+        if (inp_count > count)
+            inp_count = count;
 
-        output_size = dsp_process(dest, src, input_size);
+        out_count = dsp_process(dest, src, inp_count);
 
-        pcmbuf_write_complete(output_size);
+        pcmbuf_write_complete(out_count);
 
 #ifdef PLAYBACK_VOICE
         if ((voice_is_playing || voice_thread_start) 
@@ -1381,26 +1352,12 @@ static bool codec_pcmbuf_insert_split_callback(
             swap_codec();
         }
 #endif
-        
-        length -= input_size;
+
+        count -= inp_count;
     }
 
     return true;
-} /* codec_pcmbuf_insert_split_callback */
-
-static bool codec_pcmbuf_insert_callback(const char *buf, size_t length)
-{
-    /* TODO: The audiobuffer API should probably be updated, and be based on
-     *       pcmbuf_insert_split().  */
-    long real_length = length;
-
-    if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
-        length /= 2;    /* Length is per channel */
-
-    /* Second channel is only used for non-interleaved stereo. */
-    return codec_pcmbuf_insert_split_callback(buf, buf + (real_length / 2),
-        length);
-}
+} /* codec_pcmbuf_insert_callback */
 
 static void* codec_get_memory_callback(size_t *size)
 {
@@ -3523,7 +3480,6 @@ static void audio_playback_init(void)
     /* Initialize codec api. */
     ci.read_filebuf = codec_filebuf_callback;
     ci.pcmbuf_insert = codec_pcmbuf_insert_callback;
-    ci.pcmbuf_insert_split = codec_pcmbuf_insert_split_callback;
     ci.get_codec_memory = codec_get_memory_callback;
     ci.request_buffer = codec_request_buffer_callback;
     ci.advance_buffer = codec_advance_buffer_callback;
@@ -3543,7 +3499,6 @@ static void audio_playback_init(void)
     memset(&id3_voice, 0, sizeof(struct mp3entry));
     ci_voice.read_filebuf = voice_filebuf_callback;
     ci_voice.pcmbuf_insert = voice_pcmbuf_insert_callback;
-    ci_voice.pcmbuf_insert_split = voice_pcmbuf_insert_split_callback;
     ci_voice.get_codec_memory = voice_get_memory_callback;
     ci_voice.request_buffer = voice_request_buffer_callback;
     ci_voice.advance_buffer = voice_advance_buffer_callback;